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[alsa-devel] [sound:for-linus 5/7] sound/pci/hda/patch_ca0132.c:8416:3: error: implicit declaration of function 'pci_iounmap'; did you mean 'pcim_iounmap'?
by kbuild test robot 20 Nov '18

20 Nov '18

Hi Takashi, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: a6b0961b39896a9f9f1350d26d202f078a7d9dbc commit: d99501b8575dc1248bacf1b58d2241cb4b265d49 [5/7] ALSA: hda/ca0132 - Call pci_iounmap() instead of iounmap() config: sh-allmodconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout d99501b8575dc1248bacf1b58d2241cb4b265d49 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): sound/pci/hda/patch_ca0132.c: In function 'ca0132_free': >> sound/pci/hda/patch_ca0132.c:8416:3: error: implicit declaration of function 'pci_iounmap'; did you mean 'pcim_iounmap'? [-Werror=implicit-function-declaration] pci_iounmap(codec->bus->pci, spec->mem_base); ^~~~~~~~~~~ pcim_iounmap sound/pci/hda/patch_ca0132.c: In function 'patch_ca0132': sound/pci/hda/patch_ca0132.c:8799:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'? [-Werror=implicit-function-declaration] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^~~~~~~~~ pcim_iomap sound/pci/hda/patch_ca0132.c:8799:18: warning: assignment makes pointer from integer without a cast [-Wint-conversion] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^ cc1: some warnings being treated as errors vim +8416 sound/pci/hda/patch_ca0132.c 8386 8387 static void ca0132_free(struct hda_codec *codec) 8388 { 8389 struct ca0132_spec *spec = codec->spec; 8390 8391 cancel_delayed_work_sync(&spec->unsol_hp_work); 8392 snd_hda_power_up(codec); 8393 switch (spec->quirk) { 8394 case QUIRK_SBZ: 8395 sbz_exit_chip(codec); 8396 break; 8397 case QUIRK_ZXR: 8398 zxr_exit_chip(codec); 8399 break; 8400 case QUIRK_R3D: 8401 r3d_exit_chip(codec); 8402 break; 8403 case QUIRK_AE5: 8404 ae5_exit_chip(codec); 8405 break; 8406 case QUIRK_R3DI: 8407 r3di_gpio_shutdown(codec); 8408 break; 8409 } 8410 8411 snd_hda_sequence_write(codec, spec->base_exit_verbs); 8412 ca0132_exit_chip(codec); 8413 8414 snd_hda_power_down(codec); 8415 if (spec->mem_base) > 8416 pci_iounmap(codec->bus->pci, spec->mem_base); 8417 kfree(spec->spec_init_verbs); 8418 kfree(codec->spec); 8419 } 8420 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ALSA: aoa: Use device_type helpers to access the node type
by Rob Herring 19 Nov '18

19 Nov '18

Remove directly accessing device_node.type pointer and use the accessors instead. This will eventually allow removing the type pointer. Replace the open coded iterating over child nodes with for_each_child_of_node() while we're here. Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- sound/aoa/fabrics/layout.c | 4 ++-- sound/aoa/soundbus/core.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 1eddf8fa188f..3a23fd032611 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -1008,8 +1008,8 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) return -ENODEV; /* by breaking out we keep a reference */ - while ((sound = of_get_next_child(sdev->ofdev.dev.of_node, sound))) { - if (sound->type && strcasecmp(sound->type, "soundchip") == 0) + for_each_child_of_node(sdev->ofdev.dev.of_node, sound) { + if (of_node_is_type(sound, "soundchip")) break; } if (!sound) diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c index 70bcaa7f93dd..f3076a3f7f2f 100644 --- a/sound/aoa/soundbus/core.c +++ b/sound/aoa/soundbus/core.c @@ -78,7 +78,7 @@ static int soundbus_uevent(struct device *dev, struct kobj_uevent_env *env) if (retval) return retval; - retval = add_uevent_var(env, "OF_TYPE=%s", of->dev.of_node->type); + retval = add_uevent_var(env, "OF_TYPE=%s", of_node_get_device_type(of->dev.of_node)); if (retval) return retval; -- 2.19.1

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda/ca0132 - Add new ZxR quirk
by Connor McAdams 19 Nov '18

19 Nov '18

This patch adds a new PCI subsys ID for the ZxR, as found and tested by other users. Without a way to know if any Z's use it as well, it keeps the quirk of QUIRK_SBZ and goes through the HDA subsys test function. Signed-off-by: Connor McAdams <conmanx360(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 9c0b94ba786e..4c21bd24bcbd 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1189,6 +1189,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ), SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1102, 0x0033, "Sound Blaster ZxR", QUIRK_SBZ), SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI), -- 2.18.1

2 3

[alsa-devel] [PATCH 1/5] ALSA: asihpi: clean up indentation, replace spaces with tab
by Colin King 19 Nov '18

19 Nov '18

From: Colin Ian King <colin.king(a)canonical.com> The struct declaration is not indented correctly. Fix this by replacing spaces with a tab. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/asihpi/asihpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index a31fe1550903..aad74e809797 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1183,7 +1183,7 @@ static int snd_card_asihpi_capture_prepare(struct snd_pcm_substream *substream) static u64 snd_card_asihpi_capture_formats(struct snd_card_asihpi *asihpi, u32 h_stream) { - struct hpi_format hpi_format; + struct hpi_format hpi_format; u16 format; u16 err; u32 h_control; -- 2.19.1

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[alsa-devel] [PATCH v2 1/2] ASoC: ak4118: Add support for AK4118 S/PDIF transceiver
by Clément Péron 19 Nov '18

19 Nov '18

From: Adrien Charruel <adrien.charruel(a)devialet.com> The AK4118A is a digital audio transceiver supporting 8 input channels at 192kHz and with 24bits resolution. It converts the S/PDIF signal to I2S format and is configurable over I2C. This driver introduce a minimal support of the AK4118, like selecting the input channel, reading input frequency and detecting some errors. Datasheet is available here: https://www.akm.com/akm/en/file/datasheet/AK4118AEQ.pdf Signed-off-by: Adrien Charruel <adrien.charruel(a)devialet.com> Signed-off-by: Clément Péron <peron.clem(a)gmail.com> --- v2: - request threaded_irq in driver probe - change fs_enum from VALUE_SINGLE to VALUE_ENUM_SINGLE sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4118.c | 438 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 446 insertions(+) create mode 100644 sound/soc/codecs/ak4118.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9cc4f1848c9b..62bdb7e333b8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -35,6 +35,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADAU7002 select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER + select SND_SOC_AK4118 if I2C select SND_SOC_AK4458 if I2C select SND_SOC_AK4535 if I2C select SND_SOC_AK4554 @@ -392,6 +393,11 @@ config SND_SOC_AK4104 tristate "AKM AK4104 CODEC" depends on SPI_MASTER +config SND_SOC_AK4118 + tristate "AKM AK4118 CODEC" + depends on I2C + select REGMAP_I2C + config SND_SOC_AK4458 tristate "AKM AK4458 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8ffab8c8dbfa..66f55d185620 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -27,6 +27,7 @@ snd-soc-adav801-objs := adav801.o snd-soc-adav803-objs := adav803.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o +snd-soc-ak4118-objs := ak4118.o snd-soc-ak4458-objs := ak4458.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4554-objs := ak4554.o @@ -290,6 +291,7 @@ obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o +obj-$(CONFIG_SND_SOC_AK4118) += snd-soc-ak4118.o obj-$(CONFIG_SND_SOC_AK4458) += snd-soc-ak4458.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o diff --git a/sound/soc/codecs/ak4118.c b/sound/soc/codecs/ak4118.c new file mode 100644 index 000000000000..238ab29f2bf4 --- /dev/null +++ b/sound/soc/codecs/ak4118.c @@ -0,0 +1,438 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * ak4118.c -- Asahi Kasei ALSA Soc Audio driver + * + * Copyright 2018 DEVIALET + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <linux/slab.h> + +#include <sound/asoundef.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#define AK4118_REG_CLK_PWR_CTL 0x00 +#define AK4118_REG_FORMAT_CTL 0x01 +#define AK4118_REG_IO_CTL0 0x02 +#define AK4118_REG_IO_CTL1 0x03 +#define AK4118_REG_INT0_MASK 0x04 +#define AK4118_REG_INT1_MASK 0x05 +#define AK4118_REG_RCV_STATUS0 0x06 +#define AK4118_REG_RCV_STATUS1 0x07 +#define AK4118_REG_RXCHAN_STATUS0 0x08 +#define AK4118_REG_RXCHAN_STATUS1 0x09 +#define AK4118_REG_RXCHAN_STATUS2 0x0a +#define AK4118_REG_RXCHAN_STATUS3 0x0b +#define AK4118_REG_RXCHAN_STATUS4 0x0c +#define AK4118_REG_TXCHAN_STATUS0 0x0d +#define AK4118_REG_TXCHAN_STATUS1 0x0e +#define AK4118_REG_TXCHAN_STATUS2 0x0f +#define AK4118_REG_TXCHAN_STATUS3 0x10 +#define AK4118_REG_TXCHAN_STATUS4 0x11 +#define AK4118_REG_BURST_PREAMB_PC0 0x12 +#define AK4118_REG_BURST_PREAMB_PC1 0x13 +#define AK4118_REG_BURST_PREAMB_PD0 0x14 +#define AK4118_REG_BURST_PREAMB_PD1 0x15 +#define AK4118_REG_QSUB_CTL 0x16 +#define AK4118_REG_QSUB_TRACK 0x17 +#define AK4118_REG_QSUB_INDEX 0x18 +#define AK4118_REG_QSUB_MIN 0x19 +#define AK4118_REG_QSUB_SEC 0x1a +#define AK4118_REG_QSUB_FRAME 0x1b +#define AK4118_REG_QSUB_ZERO 0x1c +#define AK4118_REG_QSUB_ABS_MIN 0x1d +#define AK4118_REG_QSUB_ABS_SEC 0x1e +#define AK4118_REG_QSUB_ABS_FRAME 0x1f +#define AK4118_REG_GPE 0x20 +#define AK4118_REG_GPDR 0x21 +#define AK4118_REG_GPSCR 0x22 +#define AK4118_REG_GPLR 0x23 +#define AK4118_REG_DAT_MASK_DTS 0x24 +#define AK4118_REG_RX_DETECT 0x25 +#define AK4118_REG_STC_DAT_DETECT 0x26 +#define AK4118_REG_RXCHAN_STATUS5 0x27 +#define AK4118_REG_TXCHAN_STATUS5 0x28 +#define AK4118_REG_MAX 0x29 + +#define AK4118_REG_FORMAT_CTL_DIF0 (1 << 4) +#define AK4118_REG_FORMAT_CTL_DIF1 (1 << 5) +#define AK4118_REG_FORMAT_CTL_DIF2 (1 << 6) + +struct ak4118_priv { + struct regmap *regmap; + struct gpio_desc *reset; + struct gpio_desc *irq; + struct snd_soc_component *component; +}; + +static const struct reg_default ak4118_reg_defaults[] = { + {AK4118_REG_CLK_PWR_CTL, 0x43}, + {AK4118_REG_FORMAT_CTL, 0x6a}, + {AK4118_REG_IO_CTL0, 0x88}, + {AK4118_REG_IO_CTL1, 0x48}, + {AK4118_REG_INT0_MASK, 0xee}, + {AK4118_REG_INT1_MASK, 0xb5}, + {AK4118_REG_RCV_STATUS0, 0x00}, + {AK4118_REG_RCV_STATUS1, 0x10}, + {AK4118_REG_TXCHAN_STATUS0, 0x00}, + {AK4118_REG_TXCHAN_STATUS1, 0x00}, + {AK4118_REG_TXCHAN_STATUS2, 0x00}, + {AK4118_REG_TXCHAN_STATUS3, 0x00}, + {AK4118_REG_TXCHAN_STATUS4, 0x00}, + {AK4118_REG_GPE, 0x77}, + {AK4118_REG_GPDR, 0x00}, + {AK4118_REG_GPSCR, 0x00}, + {AK4118_REG_GPLR, 0x00}, + {AK4118_REG_DAT_MASK_DTS, 0x3f}, + {AK4118_REG_RX_DETECT, 0x00}, + {AK4118_REG_STC_DAT_DETECT, 0x00}, + {AK4118_REG_TXCHAN_STATUS5, 0x00}, +}; + +static const char * const ak4118_input_select_txt[] = { + "RX0", "RX1", "RX2", "RX3", "RX4", "RX5", "RX6", "RX7", +}; +static SOC_ENUM_SINGLE_DECL(ak4118_insel_enum, AK4118_REG_IO_CTL1, 0x0, + ak4118_input_select_txt); + +static const struct snd_kcontrol_new ak4118_input_mux_controls = + SOC_DAPM_ENUM("Input Select", ak4118_insel_enum); + +static const char * const ak4118_iec958_fs_txt[] = { + "44100", "48000", "32000", "22050", "11025", "24000", "16000", "88200", + "8000", "96000", "64000", "176400", "192000", +}; + +static const int ak4118_iec958_fs_val[] = { + 0x0, 0x2, 0x3, 0x4, 0x5, 0x6, 0x7, 0x8, 0x9, 0xA, 0xB, 0xC, 0xE, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(ak4118_iec958_fs_enum, AK4118_REG_RCV_STATUS1, + 0x4, 0x4, ak4118_iec958_fs_txt, + ak4118_iec958_fs_val); + +static struct snd_kcontrol_new ak4118_iec958_controls[] = { + SOC_SINGLE("IEC958 Parity Errors", AK4118_REG_RCV_STATUS0, 0, 1, 0), + SOC_SINGLE("IEC958 No Audio", AK4118_REG_RCV_STATUS0, 1, 1, 0), + SOC_SINGLE("IEC958 PLL Lock", AK4118_REG_RCV_STATUS0, 4, 1, 1), + SOC_SINGLE("IEC958 Non PCM", AK4118_REG_RCV_STATUS0, 6, 1, 0), + SOC_ENUM("IEC958 Sampling Freq", ak4118_iec958_fs_enum), +}; + +static const struct snd_soc_dapm_widget ak4118_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("INRX0"), + SND_SOC_DAPM_INPUT("INRX1"), + SND_SOC_DAPM_INPUT("INRX2"), + SND_SOC_DAPM_INPUT("INRX3"), + SND_SOC_DAPM_INPUT("INRX4"), + SND_SOC_DAPM_INPUT("INRX5"), + SND_SOC_DAPM_INPUT("INRX6"), + SND_SOC_DAPM_INPUT("INRX7"), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ak4118_input_mux_controls), +}; + +static const struct snd_soc_dapm_route ak4118_dapm_routes[] = { + {"Input Mux", "RX0", "INRX0"}, + {"Input Mux", "RX1", "INRX1"}, + {"Input Mux", "RX2", "INRX2"}, + {"Input Mux", "RX3", "INRX3"}, + {"Input Mux", "RX4", "INRX4"}, + {"Input Mux", "RX5", "INRX5"}, + {"Input Mux", "RX6", "INRX6"}, + {"Input Mux", "RX7", "INRX7"}, +}; + + +static int ak4118_set_dai_fmt_master(struct ak4118_priv *ak4118, + unsigned int format) +{ + int dif; + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dif = AK4118_REG_FORMAT_CTL_DIF0 | AK4118_REG_FORMAT_CTL_DIF2; + break; + case SND_SOC_DAIFMT_RIGHT_J: + dif = AK4118_REG_FORMAT_CTL_DIF0 | AK4118_REG_FORMAT_CTL_DIF1; + break; + case SND_SOC_DAIFMT_LEFT_J: + dif = AK4118_REG_FORMAT_CTL_DIF2; + break; + default: + return -ENOTSUPP; + } + + return dif; +} + +static int ak4118_set_dai_fmt_slave(struct ak4118_priv *ak4118, + unsigned int format) +{ + int dif; + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dif = AK4118_REG_FORMAT_CTL_DIF0 | AK4118_REG_FORMAT_CTL_DIF1 | + AK4118_REG_FORMAT_CTL_DIF2; + break; + case SND_SOC_DAIFMT_LEFT_J: + dif = AK4118_REG_FORMAT_CTL_DIF1 | AK4118_REG_FORMAT_CTL_DIF2; + break; + default: + return -ENOTSUPP; + } + + return dif; +} + +static int ak4118_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int format) +{ + struct snd_soc_component *component = dai->component; + struct ak4118_priv *ak4118 = snd_soc_component_get_drvdata(component); + int dif; + int ret = 0; + + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* component is master */ + dif = ak4118_set_dai_fmt_master(ak4118, format); + break; + case SND_SOC_DAIFMT_CBS_CFS: + /*component is slave */ + dif = ak4118_set_dai_fmt_slave(ak4118, format); + break; + default: + ret = -ENOTSUPP; + goto exit; + } + + /* format not supported */ + if (dif < 0) { + ret = dif; + goto exit; + } + + ret = regmap_update_bits(ak4118->regmap, AK4118_REG_FORMAT_CTL, + AK4118_REG_FORMAT_CTL_DIF0 | + AK4118_REG_FORMAT_CTL_DIF1 | + AK4118_REG_FORMAT_CTL_DIF2, dif); + if (ret < 0) + goto exit; + +exit: + return ret; +} + +static int ak4118_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + return 0; +} + +static const struct snd_soc_dai_ops ak4118_dai_ops = { + .hw_params = ak4118_hw_params, + .set_fmt = ak4118_set_dai_fmt, +}; + +static struct snd_soc_dai_driver ak4118_dai = { + .name = "ak4118-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_LE + }, + .ops = &ak4118_dai_ops, +}; + +static irqreturn_t ak4118_irq_handler(int irq, void *data) +{ + struct ak4118_priv *ak4118 = data; + struct snd_soc_component *component = ak4118->component; + struct snd_kcontrol_new *kctl_new; + struct snd_kcontrol *kctl; + struct snd_ctl_elem_id *id; + unsigned int i; + + if (!component) + return IRQ_NONE; + + for (i = 0; i < ARRAY_SIZE(ak4118_iec958_controls); i++) { + kctl_new = &ak4118_iec958_controls[i]; + kctl = snd_soc_card_get_kcontrol(component->card, + kctl_new->name); + if (!kctl) + continue; + id = &kctl->id; + snd_ctl_notify(component->card->snd_card, + SNDRV_CTL_EVENT_MASK_VALUE, id); + } + + return IRQ_HANDLED; +} + +static int ak4118_probe(struct snd_soc_component *component) +{ + struct ak4118_priv *ak4118 = snd_soc_component_get_drvdata(component); + int ret = 0; + + ak4118->component = component; + + /* release reset */ + gpiod_set_value(ak4118->reset, 0); + + /* unmask all int1 sources */ + ret = regmap_write(ak4118->regmap, AK4118_REG_INT1_MASK, 0x00); + if (ret < 0) { + dev_err(component->dev, + "failed to write regmap 0x%x 0x%x: %d\n", + AK4118_REG_INT1_MASK, 0x00, ret); + return ret; + } + + /* rx detect enable on all channels */ + ret = regmap_write(ak4118->regmap, AK4118_REG_RX_DETECT, 0xff); + if (ret < 0) { + dev_err(component->dev, + "failed to write regmap 0x%x 0x%x: %d\n", + AK4118_REG_RX_DETECT, 0xff, ret); + return ret; + } + + ret = snd_soc_add_component_controls(component, ak4118_iec958_controls, + ARRAY_SIZE(ak4118_iec958_controls)); + if (ret) { + dev_err(component->dev, + "failed to add component kcontrols: %d\n", ret); + return ret; + } + + return 0; +} + +static void ak4118_remove(struct snd_soc_component *component) +{ + struct ak4118_priv *ak4118 = snd_soc_component_get_drvdata(component); + + /* hold reset */ + gpiod_set_value(ak4118->reset, 1); +} + +static const struct snd_soc_component_driver soc_component_drv_ak4118 = { + .probe = ak4118_probe, + .remove = ak4118_remove, + .dapm_widgets = ak4118_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4118_dapm_widgets), + .dapm_routes = ak4118_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak4118_dapm_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config ak4118_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = ak4118_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ak4118_reg_defaults), + + .cache_type = REGCACHE_NONE, + .max_register = AK4118_REG_MAX - 1, +}; + +static int ak4118_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ak4118_priv *ak4118; + int ret; + + ak4118 = devm_kzalloc(&i2c->dev, sizeof(struct ak4118_priv), + GFP_KERNEL); + if (ak4118 == NULL) + return -ENOMEM; + + ak4118->regmap = devm_regmap_init_i2c(i2c, &ak4118_regmap); + if (IS_ERR(ak4118->regmap)) + return PTR_ERR(ak4118->regmap); + + i2c_set_clientdata(i2c, ak4118); + + ak4118->reset = devm_gpiod_get(&i2c->dev, "reset", GPIOD_OUT_HIGH); + if (IS_ERR(ak4118->reset)) { + ret = PTR_ERR(ak4118->reset); + if (ret != -EPROBE_DEFER) + dev_err(&i2c->dev, "Failed to get reset: %d\n", ret); + return ret; + } + + ak4118->irq = devm_gpiod_get(&i2c->dev, "irq", GPIOD_IN); + if (IS_ERR(ak4118->irq)) { + ret = PTR_ERR(ak4118->irq); + if (ret != -EPROBE_DEFER) + dev_err(&i2c->dev, "Failed to get IRQ: %d\n", ret); + return ret; + } + + ret = devm_request_threaded_irq(&i2c->dev, gpiod_to_irq(ak4118->irq), + NULL, ak4118_irq_handler, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, + "ak4118-irq", ak4118); + if (ret < 0) { + dev_err(&i2c->dev, "Fail to request_irq: %d\n", ret); + return ret; + } + + return snd_soc_register_component(&i2c->dev, &soc_component_drv_ak4118, + &ak4118_dai, 1); +} + +static int ak4118_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_component(&i2c->dev); + return 0; +} + +static const struct of_device_id ak4118_of_match[] = { + { .compatible = "asahi-kasei,ak4118", }, + {} +}; +MODULE_DEVICE_TABLE(of, ak4118_of_match); + +static const struct i2c_device_id ak4118_id_table[] = { + { "ak4118", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, ak4118_id_table); + +static struct i2c_driver ak4118_i2c_driver = { + .driver = { + .name = "ak4118", + .of_match_table = of_match_ptr(ak4118_of_match), + }, + .id_table = ak4118_id_table, + .probe = ak4118_i2c_probe, + .remove = ak4118_i2c_remove, +}; + +module_i2c_driver(ak4118_i2c_driver); + +MODULE_DESCRIPTION("Asahi Kasei AK4118 ALSA SoC driver"); +MODULE_AUTHOR("Adrien Charruel <adrien.charruel(a)devialet.com>"); +MODULE_LICENSE("GPL"); -- 2.19.1

3 6

[alsa-devel] [PATCH v2] ASoC: rt5663: Add documentation for power supply support
by Cheng-Yi Chiang 17 Nov '18

17 Nov '18

rt5663 codec driver will support setting CPVDD and AVDD power supply from device tree. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Moved power supply properties to required properties. Documentation/devicetree/bindings/sound/rt5663.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt index 23386446c63d6..2a55e91334082 100644 --- a/Documentation/devicetree/bindings/sound/rt5663.txt +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -10,6 +10,10 @@ Required properties: - interrupts : The CODEC's interrupt output. +- avdd-supply: Power supply for AVDD, providing 1.8V. + +- cpvdd-supply: Power supply for CPVDD, providing 3.5V. + Optional properties: - "realtek,dc_offset_l_manual" @@ -51,4 +55,6 @@ rt5663: codec@12 { compatible = "realtek,rt5663"; reg = <0x12>; interrupts = <7 IRQ_TYPE_EDGE_FALLING>; + avdd-supply = <&pp1800_a_alc5662>; + cpvdd-supply = <&pp3500_a_alc5662>; }; -- 2.19.1.1215.g8438c0b245-goog

2 1

[alsa-devel] Applied "ASoC: Convert a few more users to using %pOFn instead of device_node.name" to the asoc tree
by Mark Brown 17 Nov '18

17 Nov '18

The patch ASoC: Convert a few more users to using %pOFn instead of device_node.name 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 dc43d3aa045c6588f163f4c0f5084e09cb5c0295 Mon Sep 17 00:00:00 2001 From: Rob Herring <robh(a)kernel.org> Date: Fri, 16 Nov 2018 15:43:49 -0600 Subject: [PATCH] ASoC: Convert a few more users to using %pOFn instead of device_node.name In preparation to remove the node name pointer from struct device_node, convert printf users to use the %pOFn format specifier. 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: Olivier Moysan <olivier.moysan(a)st.com> Cc: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Cc: Maxime Coquelin <mcoquelin.stm32(a)gmail.com> Cc: Alexandre Torgue <alexandre.torgue(a)st.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-stm32(a)st-md-mailman.stormreply.com Cc: linux-arm-kernel(a)lists.infradead.org Signed-off-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/stm/stm32_sai.c | 8 ++++---- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index 192f4d7b37b6..bff7d71d0742 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -828,7 +828,7 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) /* request irq */ irq_id = platform_get_irq(pdev, 0); if (!irq_id) { - dev_err(dev, "%s no irq found\n", dev->of_node->name); + dev_err(dev, "%pOFn no irq found\n", dev->of_node); return -ENXIO; } ret = devm_request_irq(dev, irq_id, mt6797_afe_irq_handler, diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index c0b6697503fd..166aed28330d 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -1092,7 +1092,7 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev) irq_id = platform_get_irq(pdev, 0); if (irq_id <= 0) { - dev_err(afe->dev, "np %s no irq\n", afe->dev->of_node->name); + dev_err(afe->dev, "np %pOFn no irq\n", afe->dev->of_node); return irq_id < 0 ? irq_id : -ENXIO; } ret = devm_request_irq(afe->dev, irq_id, mt8173_afe_irq_handler, diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index d597eba61992..bcb35cae2a2c 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -74,14 +74,14 @@ static int stm32_sai_sync_conf_provider(struct stm32_sai_data *sai, int synco) return ret; } - dev_dbg(&sai->pdev->dev, "Set %s%s as synchro provider\n", - sai->pdev->dev.of_node->name, + dev_dbg(&sai->pdev->dev, "Set %pOFn%s as synchro provider\n", + sai->pdev->dev.of_node, synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); prev_synco = FIELD_GET(SAI_GCR_SYNCOUT_MASK, readl_relaxed(sai->base)); if (prev_synco != STM_SAI_SYNC_OUT_NONE && synco != prev_synco) { - dev_err(&sai->pdev->dev, "%s%s already set as sync provider\n", - sai->pdev->dev.of_node->name, + dev_err(&sai->pdev->dev, "%pOFn%s already set as sync provider\n", + sai->pdev->dev.of_node, prev_synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); clk_disable_unprepare(sai->pclk); return -EINVAL; -- 2.19.1

1 0

[alsa-devel] [PATCH][next] ASoC: amd: fix spelling mistake "Inavlid" -> "Invalid"
by Colin King 17 Nov '18

17 Nov '18

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_err message. Fix this. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/raven/pci-acp3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index ef805e71a98f..c28457fd9b81 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -105,7 +105,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, } break; default: - dev_err(&pci->dev, "Inavlid ACP audio mode : %d\n", val); + dev_err(&pci->dev, "Invalid ACP audio mode : %d\n", val); ret = -ENODEV; goto unmap_mmio; } -- 2.19.1

2 1

[alsa-devel] [PATCH] ASoC: rt5663: Fix error handling of regulator_set_load
by Cheng-Yi Chiang 17 Nov '18

17 Nov '18

The default implementation of regulator_set_load returns REGULATOR_MODE_NORMAL, which is positive. rt5663_i2c_probe should only do error handling when return value of regulator_set_load is negative. In this case, rt5663_i2c_probe should return error. Also, consolidate err_irq into err_enable. Fix the missing goto for temporary regmap and rt5663->regmap. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- This patch is the fixup of the patch e81a2a6d12e85 ASoC: rt5663: Add regulator support sound/soc/codecs/rt5663.c | 21 ++++++++++++--------- 1 file changed, 12 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index bb2c1ee4a836f..473a54d1ae2c5 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3522,10 +3522,11 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, for (i = 0; i < ARRAY_SIZE(rt5663->supplies); i++) { ret = regulator_set_load(rt5663->supplies[i].consumer, RT5663_SUPPLY_CURRENT_UA); - if (ret) { + if (ret < 0) { dev_err(&i2c->dev, - "Failed to set regulator %s, ret: %d\n", + "Failed to set regulator load on %s, ret: %d\n", rt5663->supplies[i].supply, ret); + return ret; } } @@ -3543,7 +3544,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, ret = PTR_ERR(regmap); dev_err(&i2c->dev, "Failed to allocate temp register map: %d\n", ret); - return ret; + goto err_enable; } ret = regmap_read(regmap, RT5663_VENDOR_ID_2, &val); @@ -3576,7 +3577,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, ret = PTR_ERR(rt5663->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); - return ret; + goto err_enable; } /* reset and calibrate */ @@ -3686,17 +3687,19 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, rt5663_dai, ARRAY_SIZE(rt5663_dai)); if (ret) - goto err_irq; + goto err_enable; return 0; -err_irq: + + /* + * Error after enabling regulators should goto err_enable + * to disable regulators. + */ +err_enable: if (i2c->irq) free_irq(i2c->irq, rt5663); -err_enable: - dev_err(&i2c->dev, - "%s: Disable regulator after probe error\n", __func__); regulator_bulk_disable(ARRAY_SIZE(rt5663->supplies), rt5663->supplies); return ret; } -- 2.19.1.1215.g8438c0b245-goog

2 2

[alsa-devel] [PATCH v3 0/2] Introduce dmic mode switch delay parameter
by Jenny TC 17 Nov '18

17 Nov '18

Some DMICs need clock to be running for a specified duration as per the DMIC spec to complete the mode transition like sleep to normal, off to normal etc. This series add a mode-switch delay which will prevent the clock from being turned off without complying with manufacturer timing specifications. Also introduces a module parameter for the wakeup_delay parameter. v1: First patch set v2: 1). Removed ACPI device entry since a) The DMIC vendor doesn't have an ACPI vendor ID b) The ACPI enumeration creates one more device and conflicts with the existing skl.c dmic platform device registration. 2). Updated commit message v3: Updated commit message and introduced module param for wakeup_delay Jenny TC (2): ASoC: dmic: introduce mode switch delay ASoC: dmic: introduce module_param wakeup_delay sound/soc/codecs/dmic.c | 36 ++++++++++++++++++++++++++++++++++++ 1 file changed, 36 insertions(+) -- 1.9.1

4 5

[alsa-devel] [PATCH] ASoC: Convert a few more users to using %pOFn instead of device_node.name
by Rob Herring 16 Nov '18

16 Nov '18

In preparation to remove the node name pointer from struct device_node, convert printf users to use the %pOFn format specifier. 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: Olivier Moysan <olivier.moysan(a)st.com> Cc: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Cc: Maxime Coquelin <mcoquelin.stm32(a)gmail.com> Cc: Alexandre Torgue <alexandre.torgue(a)st.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-stm32(a)st-md-mailman.stormreply.com Cc: linux-arm-kernel(a)lists.infradead.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/stm/stm32_sai.c | 8 ++++---- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index 192f4d7b37b6..bff7d71d0742 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -828,7 +828,7 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) /* request irq */ irq_id = platform_get_irq(pdev, 0); if (!irq_id) { - dev_err(dev, "%s no irq found\n", dev->of_node->name); + dev_err(dev, "%pOFn no irq found\n", dev->of_node); return -ENXIO; } ret = devm_request_irq(dev, irq_id, mt6797_afe_irq_handler, diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index c0b6697503fd..166aed28330d 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -1092,7 +1092,7 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev) irq_id = platform_get_irq(pdev, 0); if (irq_id <= 0) { - dev_err(afe->dev, "np %s no irq\n", afe->dev->of_node->name); + dev_err(afe->dev, "np %pOFn no irq\n", afe->dev->of_node); return irq_id < 0 ? irq_id : -ENXIO; } ret = devm_request_irq(afe->dev, irq_id, mt8173_afe_irq_handler, diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index d597eba61992..bcb35cae2a2c 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -74,14 +74,14 @@ static int stm32_sai_sync_conf_provider(struct stm32_sai_data *sai, int synco) return ret; } - dev_dbg(&sai->pdev->dev, "Set %s%s as synchro provider\n", - sai->pdev->dev.of_node->name, + dev_dbg(&sai->pdev->dev, "Set %pOFn%s as synchro provider\n", + sai->pdev->dev.of_node, synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); prev_synco = FIELD_GET(SAI_GCR_SYNCOUT_MASK, readl_relaxed(sai->base)); if (prev_synco != STM_SAI_SYNC_OUT_NONE && synco != prev_synco) { - dev_err(&sai->pdev->dev, "%s%s already set as sync provider\n", - sai->pdev->dev.of_node->name, + dev_err(&sai->pdev->dev, "%pOFn%s already set as sync provider\n", + sai->pdev->dev.of_node, prev_synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); clk_disable_unprepare(sai->pclk); return -EINVAL; -- 2.19.1

1 0

Re: [alsa-devel] Controlling the Tascam FW-1884
by Takashi Sakamoto 16 Nov '18

16 Nov '18

Hi Scott, (C.Ced to alsa-devel, ffado-devel) On Sep 7 2018 23:01, Scott Bahling wrote: > I recently got my hands on a FW-1884 and would like to get it working in > Linux. I'm mostly interested in using the control interface which should > support several midi control interface protocols like Mackie Control or > Mackie HUI. From the user manual it appears that the host computer needs to > set the registers on the FW-1884 to switch modes (they reference Windows > applications). > > I see that you have recently been working on user space support asynchronous > commands to such devices. The hinawa_utils code has some beginnings of > support for setting options on the Tascam devices and I have just started > playing with that. Thanks for your suggestion. At present, the most of features on control surface of FW-1884 are not available, especially for fader, knob and start/stop buttons. In data transmission between system and devices on IEEE 1394 bus, both of asynchronous transactions and isochronous packet streams are available. In a case of TASCAM FireWire series, messages via MIDI physical interface is bypassed to asynchronous transaction, while messages generated on control surface are bypassed to isochronous packets, multiplexed with PCM frames. When enabling multiplexing messages to isochronous packet, demultiplexer should be implemented to ALSA firewire-tascam driver, but not yet. You can see my initial RFC in alsa-devel[1]. This patch enables userspace applications to perform the demultiplexing, however this idea easily brings cache-coherency problems due to page mapping between kernel/userspace and impossible to be applied. We need to investigate another idea to achieve the demultiplexing in kernel land. As another option, according to page 26 in 'TASCAM FW-1884 Owner's Manual', when programming Keys to port 1-4, system could receive MIDI messages from control surface of FW-1884. However, I've not investigated it. > How are you determining the registers and flags? Do you have documentation, > or are you reverse engineering? Do you know the registers and flags for > setting the control protocol? > > If you have any hints on how figure out the proper registers and flags I'm > happy to contribute to the work you have started and help support more > functions of the FW-1884. User manuals are documents to which we can refer. The most vendors licenses their drivers/software with EULA and in most cases it can't allow users to binary-based reverse engineering. All of my work are done with external protocol analyzer unit on IEEE 1394 bus. When working to implement drivers, I read/classified/parsed content of many packets sniffed by the analyzer. An initial idea of libhinawa/hinawa-utils is to assist this my work. I heard FFADO developers uses some ways for this type of work: - Usage of host controller of TI PCILynx and driver module (nosy.ko) - Windows application; bushound[2] However, this type of work takes you long time, so I don't necessarily recommend you. It's 2018 and I think you can find alternative control surfaces on USB. If you take long time to make devices available on GNU/Linux, it's better to select devices on 'live' buses instead of 'legacy' buses such as IEEE 1394 bus, IMO. In a case of USB, wireshark with usbcap is the easiest way for us to sniff packets on Windows. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/094817.html [2] http://perisoft.net/bushound/ Regards Takashi Sakamoto

2 21

[alsa-devel] [PATCH] ASoC: acpi: fix: continue searching when machine is ignored
by Keyon Jie 16 Nov '18

16 Nov '18

The machine_quirk may return NULL which means the acpi entries should be skipped and search for next matched entry is needed, here add return check here and continue for NULL case. Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> --- sound/soc/soc-acpi.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c index b8e72b52db30..faf8941363e4 100644 --- a/sound/soc/soc-acpi.c +++ b/sound/soc/soc-acpi.c @@ -15,7 +15,10 @@ snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines) if (acpi_dev_present(mach->id, NULL, -1)) { if (mach->machine_quirk) mach = mach->machine_quirk(mach); - return mach; + + /* return matched machine, continue otherwise */ + if (mach) + return mach; } } return NULL; -- 2.17.1

2 4

[alsa-devel] How to propagate parameters from FE to BE with compressed audio interfaces
by Daniel Baluta 16 Nov '18

16 Nov '18

Hello, soc_compr_set_params_fe calls dpcm_be_dai_hw_params in order to setup BE hw_params. Is there a way to transfer parameters like for example the rate from FE to BE? In soc_compr_set_params_fe there is this code: * * Create an empty hw_params for the BE as the machine driver must * fix this up to match DSP decoder and ASRC configuration. * I.e. machine driver fixup for compressed BE is mandatory. */ memset(&fe->dpcm[fe_substream->stream].hw_params, 0, sizeof(struct snd_pcm_hw_params)); So, information about parameters is just ignored. The only way we can setup hw_params for BE is via be_hw_params_fixup() but this function doesn't have any information from FE. My use case is something like this: FE: - cpu: FSL SAI DAI interface - platform: DSP - codec: dummy codec BE: - cpu: dummy - platform: dummy - codec: cs42888. codec could handle 44100Hz and 48000Hz streams, but the BE doesn't have any way to know what is the rate cplay sets to FE. thanks, Daniel.

3 2

[alsa-devel] Applied "ASoC: Add documentation for pcm3060 property out-single-ended" to the asoc tree
by Mark Brown 16 Nov '18

16 Nov '18

The patch ASoC: Add documentation for pcm3060 property out-single-ended 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 fd7de6370cb62b147185834d60069568a21acacd Mon Sep 17 00:00:00 2001 From: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Date: Thu, 15 Nov 2018 08:12:53 +0100 Subject: [PATCH] ASoC: Add documentation for pcm3060 property out-single-ended Output of pcm3060 codec may be configured as single-ended or differential Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/pcm3060.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/pcm3060.txt b/Documentation/devicetree/bindings/sound/pcm3060.txt index 90fcb8523099..97de66932d44 100644 --- a/Documentation/devicetree/bindings/sound/pcm3060.txt +++ b/Documentation/devicetree/bindings/sound/pcm3060.txt @@ -9,9 +9,15 @@ Required properties: - reg : the I2C address of the device for I2C, the chip select number for SPI. +Optional properties: + +- ti,out-single-ended: "true" if output is single-ended; + "false" or not specified if output is differential. + Examples: pcm3060: pcm3060@46 { compatible = "ti,pcm3060"; reg = <0x46>; + ti,out-single-ended = "true"; }; -- 2.19.1

1 0

[alsa-devel] [PATCH] ASoC: nau8822: convert to SPDX identifiers
by David Lin 15 Nov '18

15 Nov '18

This patch updates license to SPDX-License-Identifier instead of verbose license text. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- sound/soc/codecs/nau8822.c | 26 +++++++++++--------------- sound/soc/codecs/nau8822.h | 11 +++++------ 2 files changed, 16 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 622ce947f134..c6152a044416 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -1,18 +1,14 @@ -/* - * nau8822.c -- NAU8822 ALSA Soc Audio Codec driver - * - * Copyright 2017 Nuvoton Technology Corp. - * - * Author: David Lin <ctlin0(a)nuvoton.com> - * Co-author: John Hsu <kchsu0(a)nuvoton.com> - * Co-author: Seven Li <wtli(a)nuvoton.com> - * - * Based on WM8974.c - * - * 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 +// +// nau8822.c -- NAU8822 ALSA Soc Audio driver +// +// Copyright 2017 Nuvoton Technology Crop. +// +// Author: David Lin <ctlin0(a)nuvoton.com> +// Co-author: John Hsu <kchsu0(a)nuvoton.com> +// Co-author: Seven Li <wtli(a)nuvoton.com> +// +// Based on WM8974.c #include <linux/module.h> #include <linux/moduleparam.h> diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index aa79c969cd44..d15afd782815 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -1,13 +1,12 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* - * nau8822.h -- NAU8822 Soc Audio Codec driver + * nau8822.h -- NAU8822 ALSA SoC Audio driver + * + * Copyright 2017 Nuvoton Technology Crop. * * Author: David Lin <ctlin0(a)nuvoton.com> - * Co-author: John Hsu <kchsu0(a)nuvoton.com> + * Co-author: John Hsu <kchsu(a)nuvoton.com> * Co-author: Seven Li <wtli(a)nuvoton.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 __NAU8822_H__ -- 2.18.0

2 3

[alsa-devel] [PATCH 1/2] ASoC: amd: Fixed build errors
by Vijendar Mukunda 15 Nov '18

15 Nov '18

Fixed build errors. - Implicit declaration of pci_enable_msi() & pci_disable_msi() api's for openrisc architecture. - type defaults to 'int' in declaration of 'module_pci_driver' Enabled build for x86 architecture. Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 5f3b560..33ebec9 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -22,5 +22,6 @@ config SND_SOC_AMD_CZ_RT5645_MACH config SND_SOC_AMD_ACP3x tristate "AMD Audio Coprocessor-v3.x support" + depends on X86 && PCI help This option enables ACP v3.x I2S support on AMD platform -- 2.7.4

2 3

[alsa-devel] Applied "ASoC: amd: remove set but not used variable 'dma_buffer'" to the asoc tree
by Mark Brown 15 Nov '18

15 Nov '18

The patch ASoC: amd: remove set but not used variable 'dma_buffer' 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 424d5b56d020b750803134cc15bbd4d0addaaee7 Mon Sep 17 00:00:00 2001 From: YueHaibing <yuehaibing(a)huawei.com> Date: Thu, 15 Nov 2018 01:56:35 +0000 Subject: [PATCH] ASoC: amd: remove set but not used variable 'dma_buffer' Fixes gcc '-Wunused-but-set-variable' warning: sound/soc/amd/raven/acp3x-pcm-dma.c: In function 'acp3x_dma_hw_params': sound/soc/amd/raven/acp3x-pcm-dma.c:333:25: warning: variable 'dma_buffer' set but not used [-Wunused-but-set-variable] It never used since introduction in commit 8de1b5ed0337 ("ASoC: amd: add acp3x system resume pm op") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 2aa34208ae92..022a8912c8a2 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -322,7 +322,6 @@ static int acp3x_dma_hw_params(struct snd_pcm_substream *substream, { int status; u64 size; - struct snd_dma_buffer *dma_buffer; struct page *pg; struct snd_pcm_runtime *runtime = substream->runtime; struct i2s_stream_instance *rtd = runtime->private_data; @@ -330,7 +329,6 @@ static int acp3x_dma_hw_params(struct snd_pcm_substream *substream, if (!rtd) return -EINVAL; - dma_buffer = &substream->dma_buffer; size = params_buffer_bytes(params); status = snd_pcm_lib_malloc_pages(substream, size); if (status < 0) -- 2.19.1

1 0

[alsa-devel] [PATCH] ASoC: pcm186x: Fix device reset-registers trigger value
by Andreas Dannenberg 15 Nov '18

15 Nov '18

According to the current device datasheet (TI Lit # SLAS831D, revised March 2018) the value written to the device's PAGE register to trigger a complete register reset should be 0xfe, not 0xff. So go ahead and update to the correct value. Reported-by: Stephane Le Provost <stephane.leprovost(a)mediatek.com> Tested-by: Stephane Le Provost <stephane.leprovost(a)mediatek.com> Signed-off-by: Andreas Dannenberg <dannenberg(a)ti.com> --- sound/soc/codecs/pcm186x.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm186x.h b/sound/soc/codecs/pcm186x.h index 2c6ba55bf394..bb3f0c42a1cd 100644 --- a/sound/soc/codecs/pcm186x.h +++ b/sound/soc/codecs/pcm186x.h @@ -139,7 +139,7 @@ enum pcm186x_type { #define PCM186X_MAX_REGISTER PCM186X_CURR_TRIM_CTRL /* PCM186X_PAGE */ -#define PCM186X_RESET 0xff +#define PCM186X_RESET 0xfe /* PCM186X_ADCX_INPUT_SEL_X */ #define PCM186X_ADC_INPUT_SEL_POL BIT(7) -- 2.15.1

3 2

[alsa-devel] Applied "ASoC: rt5663: Add regulator support" to the asoc tree
by Mark Brown 15 Nov '18

15 Nov '18

The patch ASoC: rt5663: Add regulator support 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 e81a2a6d12e85fbd7b19e96ad72b82a3cc8a6b2d Mon Sep 17 00:00:00 2001 From: Cheng-Yi Chiang <cychiang(a)chromium.org> Date: Thu, 15 Nov 2018 12:13:34 +0800 Subject: [PATCH] ASoC: rt5663: Add regulator support Add regulator support to turn on cpvdd and avdd in probe. If a regulator is not given from device tree, a dummy regulator will be used. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5663.c | 68 +++++++++++++++++++++++++++++++++++---- 1 file changed, 61 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 7eb2cbd39d6e..29c059ed0682 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -17,6 +17,7 @@ #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/acpi.h> +#include <linux/regulator/consumer.h> #include <linux/workqueue.h> #include <sound/core.h> #include <sound/pcm.h> @@ -33,6 +34,9 @@ #define RT5663_DEVICE_ID_2 0x6451 #define RT5663_DEVICE_ID_1 0x6406 +#define RT5663_POWER_ON_DELAY_MS 300 +#define RT5663_SUPPLY_CURRENT_UA 500000 + enum { CODEC_VER_1, CODEC_VER_0, @@ -48,6 +52,11 @@ struct impedance_mapping_table { unsigned int dc_offset_r_manual_mic; }; +static const char *const rt5663_supply_names[] = { + "avdd", + "cpvdd", +}; + struct rt5663_priv { struct snd_soc_component *component; struct rt5663_platform_data pdata; @@ -56,6 +65,7 @@ struct rt5663_priv { struct snd_soc_jack *hs_jack; struct timer_list btn_check_timer; struct impedance_mapping_table *imp_table; + struct regulator_bulk_data supplies[ARRAY_SIZE(rt5663_supply_names)]; int codec_ver; int sysclk; @@ -3483,7 +3493,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, { struct rt5663_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt5663_priv *rt5663; - int ret; + int ret, i; unsigned int val; struct regmap *regmap; @@ -3500,6 +3510,37 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, else rt5663_parse_dp(rt5663, &i2c->dev); + for (i = 0; i < ARRAY_SIZE(rt5663->supplies); i++) + rt5663->supplies[i].supply = rt5663_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, + ARRAY_SIZE(rt5663->supplies), + rt5663->supplies); + if (ret) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + /* Set load for regulator. */ + for (i = 0; i < ARRAY_SIZE(rt5663->supplies); i++) { + ret = regulator_set_load(rt5663->supplies[i].consumer, + RT5663_SUPPLY_CURRENT_UA); + if (ret) { + dev_err(&i2c->dev, + "Failed to set regulator %s, ret: %d\n", + rt5663->supplies[i].supply, ret); + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(rt5663->supplies), + rt5663->supplies); + + if (ret) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + msleep(RT5663_POWER_ON_DELAY_MS); + regmap = devm_regmap_init_i2c(i2c, &temp_regmap); if (IS_ERR(regmap)) { ret = PTR_ERR(regmap); @@ -3530,7 +3571,8 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Device with ID register %#x is not rt5663\n", val); - return -ENODEV; + ret = -ENODEV; + goto err_enable; } if (IS_ERR(rt5663->regmap)) { @@ -3635,20 +3677,30 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, ret = request_irq(i2c->irq, rt5663_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5663", rt5663); - if (ret) + if (ret) { dev_err(&i2c->dev, "%s Failed to reguest IRQ: %d\n", __func__, ret); + goto err_enable; + } } ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5663, rt5663_dai, ARRAY_SIZE(rt5663_dai)); - if (ret) { - if (i2c->irq) - free_irq(i2c->irq, rt5663); - } + if (ret) + goto err_irq; + return 0; + +err_irq: + if (i2c->irq) + free_irq(i2c->irq, rt5663); + +err_enable: + dev_err(&i2c->dev, + "%s: Disable regulator after probe error\n", __func__); + regulator_bulk_disable(ARRAY_SIZE(rt5663->supplies), rt5663->supplies); return ret; } @@ -3659,6 +3711,8 @@ static int rt5663_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5663); + regulator_bulk_disable(ARRAY_SIZE(rt5663->supplies), rt5663->supplies); + return 0; } -- 2.19.1

1 0

[alsa-devel] [PATCH 1/2] ASoC: rt5663: Add documentation for power supply support
by Cheng-Yi Chiang 15 Nov '18

15 Nov '18

rt5663 codec driver will support setting CPVDD and AVDD power supply from device tree. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Fixed the commit title of v1. Documentation/devicetree/bindings/sound/rt5663.txt | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt index 23386446c63d6..d4058dfde0392 100644 --- a/Documentation/devicetree/bindings/sound/rt5663.txt +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -36,6 +36,9 @@ Optional properties: "realtek,impedance_sensing_num" is 2. It means that there are 2 ranges of impedance in the impedance sensing function. +- avdd-supply: Power supply for AVDD, providing 1.8V. +- cpvdd-supply: Power supply for CPVDD, providing 3.5V. + Pins on the device (for linking into audio routes) for RT5663: * IN1P @@ -51,4 +54,6 @@ rt5663: codec@12 { compatible = "realtek,rt5663"; reg = <0x12>; interrupts = <7 IRQ_TYPE_EDGE_FALLING>; + avdd-supply = <&pp1800_a_alc5662>; + cpvdd-supply = <&pp3500_a_alc5662>; }; -- 2.19.1.930.g4563a0d9d0-goog

2 2

[alsa-devel] [PATCH alsa-lib 1/2] pcm_share: Fix compiler warnings -Wunused-result
by Kirill Marinushkin 15 Nov '18

15 Nov '18

Before this commit, compilation of `pcm_share` causes warnings "ignoring return value" for several `read` and `write` operations: ~~~~ pcm_share.c: In function '_snd_pcm_share_missing': pcm_share.c:293:5: warning: ignoring return value of 'read', declared with attribute warn_unused_result [-Wunused-result] read(share->slave_socket, buf, 1); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ <...> ~~~~ However, ignoring return values in these use-cases is safe and optimal, because `read` and `write` operations are executed upon descriptors, created with `socketpair()` and `pipe()`. This commit fixes the warnings by introducing the return value with the attribute `unused` (for details, see [1]). The macro, used in this commit, is defined at `global.h`: ~~~~ define ATTRIBUTE_UNUSED __attribute__ ((__unused__)) ~~~~ I explicitly checked the objdump: this commit doesn't change the machine code, related to the modified C code lines. The execution performance is not affected. [1] https://gcc.gnu.org/onlinedocs/gcc-8.2.0/gcc/Common-Variable-Attributes.html Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> --- src/pcm/pcm_share.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/src/pcm/pcm_share.c b/src/pcm/pcm_share.c index 5a540c4f..9592e003 100644 --- a/src/pcm/pcm_share.c +++ b/src/pcm/pcm_share.c @@ -287,17 +287,18 @@ static snd_pcm_uframes_t _snd_pcm_share_missing(snd_pcm_t *pcm) update_poll: if (ready != share->ready) { + int un ATTRIBUTE_UNUSED; char buf[1]; if (pcm->stream == SND_PCM_STREAM_PLAYBACK) { if (ready) - read(share->slave_socket, buf, 1); + un = read(share->slave_socket, buf, 1); else - write(share->client_socket, buf, 1); + un = write(share->client_socket, buf, 1); } else { if (ready) - write(share->slave_socket, buf, 1); + un = write(share->slave_socket, buf, 1); else - read(share->client_socket, buf, 1); + un = read(share->client_socket, buf, 1); } share->ready = ready; } @@ -404,8 +405,9 @@ static void *snd_pcm_share_thread(void *data) err = poll(pfd, 2, -1); Pthread_mutex_lock(&slave->mutex); if (pfd[0].revents & POLLIN) { + int un ATTRIBUTE_UNUSED; char buf[1]; - read(pfd[0].fd, buf, 1); + un = read(pfd[0].fd, buf, 1); } } else { slave->polling = 0; -- 2.13.6

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[alsa-devel] [PATCH 1/2] ASoC: ak4118: Add support for AK4118 S/PDIF transceiver
by Clément Péron 15 Nov '18

15 Nov '18

From: Adrien Charruel <adrien.charruel(a)devialet.com> The AK4118A is a digital audio transceiver supporting 8 input channels at 192kHz and with 24bits resolution. It converts the S/PDIF signal to I2S format and is configurable over I2C. This driver introduce a minimal support of the AK4118, like selecting the input channel, reading input frequency and detecting some errors. Datasheet is available here: https://www.akm.com/akm/en/file/datasheet/AK4118AEQ.pdf Signed-off-by: Adrien Charruel <adrien.charruel(a)devialet.com> Signed-off-by: Clément Péron <peron.clem(a)gmail.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4118.c | 449 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 457 insertions(+) create mode 100644 sound/soc/codecs/ak4118.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9cc4f1848c9b..62bdb7e333b8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -35,6 +35,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADAU7002 select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER + select SND_SOC_AK4118 if I2C select SND_SOC_AK4458 if I2C select SND_SOC_AK4535 if I2C select SND_SOC_AK4554 @@ -392,6 +393,11 @@ config SND_SOC_AK4104 tristate "AKM AK4104 CODEC" depends on SPI_MASTER +config SND_SOC_AK4118 + tristate "AKM AK4118 CODEC" + depends on I2C + select REGMAP_I2C + config SND_SOC_AK4458 tristate "AKM AK4458 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8ffab8c8dbfa..66f55d185620 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -27,6 +27,7 @@ snd-soc-adav801-objs := adav801.o snd-soc-adav803-objs := adav803.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o +snd-soc-ak4118-objs := ak4118.o snd-soc-ak4458-objs := ak4458.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4554-objs := ak4554.o @@ -290,6 +291,7 @@ obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o +obj-$(CONFIG_SND_SOC_AK4118) += snd-soc-ak4118.o obj-$(CONFIG_SND_SOC_AK4458) += snd-soc-ak4458.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o diff --git a/sound/soc/codecs/ak4118.c b/sound/soc/codecs/ak4118.c new file mode 100644 index 000000000000..a88e7d6a65a3 --- /dev/null +++ b/sound/soc/codecs/ak4118.c @@ -0,0 +1,449 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * ak4118.c -- Asahi Kasei ALSA Soc Audio driver + * + * Copyright 2018 DEVIALET + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <linux/slab.h> + +#include <sound/asoundef.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#define AK4118_REG_CLK_PWR_CTL 0x00 +#define AK4118_REG_FORMAT_CTL 0x01 +#define AK4118_REG_IO_CTL0 0x02 +#define AK4118_REG_IO_CTL1 0x03 +#define AK4118_REG_INT0_MASK 0x04 +#define AK4118_REG_INT1_MASK 0x05 +#define AK4118_REG_RCV_STATUS0 0x06 +#define AK4118_REG_RCV_STATUS1 0x07 +#define AK4118_REG_RXCHAN_STATUS0 0x08 +#define AK4118_REG_RXCHAN_STATUS1 0x09 +#define AK4118_REG_RXCHAN_STATUS2 0x0a +#define AK4118_REG_RXCHAN_STATUS3 0x0b +#define AK4118_REG_RXCHAN_STATUS4 0x0c +#define AK4118_REG_TXCHAN_STATUS0 0x0d +#define AK4118_REG_TXCHAN_STATUS1 0x0e +#define AK4118_REG_TXCHAN_STATUS2 0x0f +#define AK4118_REG_TXCHAN_STATUS3 0x10 +#define AK4118_REG_TXCHAN_STATUS4 0x11 +#define AK4118_REG_BURST_PREAMB_PC0 0x12 +#define AK4118_REG_BURST_PREAMB_PC1 0x13 +#define AK4118_REG_BURST_PREAMB_PD0 0x14 +#define AK4118_REG_BURST_PREAMB_PD1 0x15 +#define AK4118_REG_QSUB_CTL 0x16 +#define AK4118_REG_QSUB_TRACK 0x17 +#define AK4118_REG_QSUB_INDEX 0x18 +#define AK4118_REG_QSUB_MIN 0x19 +#define AK4118_REG_QSUB_SEC 0x1a +#define AK4118_REG_QSUB_FRAME 0x1b +#define AK4118_REG_QSUB_ZERO 0x1c +#define AK4118_REG_QSUB_ABS_MIN 0x1d +#define AK4118_REG_QSUB_ABS_SEC 0x1e +#define AK4118_REG_QSUB_ABS_FRAME 0x1f +#define AK4118_REG_GPE 0x20 +#define AK4118_REG_GPDR 0x21 +#define AK4118_REG_GPSCR 0x22 +#define AK4118_REG_GPLR 0x23 +#define AK4118_REG_DAT_MASK_DTS 0x24 +#define AK4118_REG_RX_DETECT 0x25 +#define AK4118_REG_STC_DAT_DETECT 0x26 +#define AK4118_REG_RXCHAN_STATUS5 0x27 +#define AK4118_REG_TXCHAN_STATUS5 0x28 +#define AK4118_REG_MAX 0x29 + +#define AK4118_REG_FORMAT_CTL_DIF0 (1 << 4) +#define AK4118_REG_FORMAT_CTL_DIF1 (1 << 5) +#define AK4118_REG_FORMAT_CTL_DIF2 (1 << 6) + +struct ak4118_priv { + struct regmap *regmap; + struct gpio_desc *reset; + struct gpio_desc *irq; + struct snd_soc_component *component; +}; + +static const struct reg_default ak4118_reg_defaults[] = { + {AK4118_REG_CLK_PWR_CTL, 0x43}, + {AK4118_REG_FORMAT_CTL, 0x6a}, + {AK4118_REG_IO_CTL0, 0x88}, + {AK4118_REG_IO_CTL1, 0x48}, + {AK4118_REG_INT0_MASK, 0xee}, + {AK4118_REG_INT1_MASK, 0xb5}, + {AK4118_REG_RCV_STATUS0, 0x00}, + {AK4118_REG_RCV_STATUS1, 0x10}, + {AK4118_REG_TXCHAN_STATUS0, 0x00}, + {AK4118_REG_TXCHAN_STATUS1, 0x00}, + {AK4118_REG_TXCHAN_STATUS2, 0x00}, + {AK4118_REG_TXCHAN_STATUS3, 0x00}, + {AK4118_REG_TXCHAN_STATUS4, 0x00}, + {AK4118_REG_GPE, 0x77}, + {AK4118_REG_GPDR, 0x00}, + {AK4118_REG_GPSCR, 0x00}, + {AK4118_REG_GPLR, 0x00}, + {AK4118_REG_DAT_MASK_DTS, 0x3f}, + {AK4118_REG_RX_DETECT, 0x00}, + {AK4118_REG_STC_DAT_DETECT, 0x00}, + {AK4118_REG_TXCHAN_STATUS5, 0x00}, +}; + +static const char * const ak4118_input_select_txt[] = { + "RX0", "RX1", "RX2", "RX3", "RX4", "RX5", "RX6", "RX7", +}; +static SOC_ENUM_SINGLE_DECL(ak4118_insel_enum, AK4118_REG_IO_CTL1, 0, + ak4118_input_select_txt); + +static const struct snd_kcontrol_new ak4118_input_mux_controls = + SOC_DAPM_ENUM("Input Select", ak4118_insel_enum); + +static const char * const ak4118_iec958_fs_texts[] = { + "44100", + "Reserved", + "48000", + "32000", + "22050", + "11025", + "24000", + "16000", + "88200", + "8000", + "96000", + "64000", + "176400", + "Unused", + "192000", +}; + +static const struct soc_enum ak4118_iec958_fs_enum = + SOC_ENUM_SINGLE(AK4118_REG_RCV_STATUS1, 4, + ARRAY_SIZE(ak4118_iec958_fs_texts), + ak4118_iec958_fs_texts); + +static struct snd_kcontrol_new ak4118_iec958_controls[] = { + SOC_SINGLE("IEC958 Parity Errors", AK4118_REG_RCV_STATUS0, 0, 1, 0), + SOC_SINGLE("IEC958 No Audio", AK4118_REG_RCV_STATUS0, 1, 1, 0), + SOC_SINGLE("IEC958 PLL Lock", AK4118_REG_RCV_STATUS0, 4, 1, 1), + SOC_SINGLE("IEC958 Non PCM", AK4118_REG_RCV_STATUS0, 6, 1, 0), + SOC_ENUM("IEC958 Sampling Freq", ak4118_iec958_fs_enum), +}; + +static const struct snd_soc_dapm_widget ak4118_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("INRX0"), + SND_SOC_DAPM_INPUT("INRX1"), + SND_SOC_DAPM_INPUT("INRX2"), + SND_SOC_DAPM_INPUT("INRX3"), + SND_SOC_DAPM_INPUT("INRX4"), + SND_SOC_DAPM_INPUT("INRX5"), + SND_SOC_DAPM_INPUT("INRX6"), + SND_SOC_DAPM_INPUT("INRX7"), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ak4118_input_mux_controls), +}; + +static const struct snd_soc_dapm_route ak4118_dapm_routes[] = { + {"Input Mux", "RX0", "INRX0"}, + {"Input Mux", "RX1", "INRX1"}, + {"Input Mux", "RX2", "INRX2"}, + {"Input Mux", "RX3", "INRX3"}, + {"Input Mux", "RX4", "INRX4"}, + {"Input Mux", "RX5", "INRX5"}, + {"Input Mux", "RX6", "INRX6"}, + {"Input Mux", "RX7", "INRX7"}, +}; + + +static int ak4118_set_dai_fmt_master(struct ak4118_priv *ak4118, + unsigned int format) +{ + int dif; + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dif = AK4118_REG_FORMAT_CTL_DIF0 | AK4118_REG_FORMAT_CTL_DIF2; + break; + case SND_SOC_DAIFMT_RIGHT_J: + dif = AK4118_REG_FORMAT_CTL_DIF0 | AK4118_REG_FORMAT_CTL_DIF1; + break; + case SND_SOC_DAIFMT_LEFT_J: + dif = AK4118_REG_FORMAT_CTL_DIF2; + break; + default: + return -ENOTSUPP; + } + + return dif; +} + +static int ak4118_set_dai_fmt_slave(struct ak4118_priv *ak4118, + unsigned int format) +{ + int dif; + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dif = AK4118_REG_FORMAT_CTL_DIF0 | AK4118_REG_FORMAT_CTL_DIF1 | + AK4118_REG_FORMAT_CTL_DIF2; + break; + case SND_SOC_DAIFMT_LEFT_J: + dif = AK4118_REG_FORMAT_CTL_DIF1 | AK4118_REG_FORMAT_CTL_DIF2; + break; + default: + return -ENOTSUPP; + } + + return dif; +} + +static int ak4118_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int format) +{ + struct snd_soc_component *component = dai->component; + struct ak4118_priv *ak4118 = snd_soc_component_get_drvdata(component); + int dif; + int ret = 0; + + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* component is master */ + dif = ak4118_set_dai_fmt_master(ak4118, format); + break; + case SND_SOC_DAIFMT_CBS_CFS: + /*component is slave */ + dif = ak4118_set_dai_fmt_slave(ak4118, format); + break; + default: + ret = -ENOTSUPP; + goto exit; + } + + /* format not supported */ + if (dif < 0) { + ret = dif; + goto exit; + } + + ret = regmap_update_bits(ak4118->regmap, AK4118_REG_FORMAT_CTL, + AK4118_REG_FORMAT_CTL_DIF0 | + AK4118_REG_FORMAT_CTL_DIF1 | + AK4118_REG_FORMAT_CTL_DIF2, dif); + if (ret < 0) + goto exit; + +exit: + return ret; +} + +static int ak4118_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + return 0; +} + +static const struct snd_soc_dai_ops ak4118_dai_ops = { + .hw_params = ak4118_hw_params, + .set_fmt = ak4118_set_dai_fmt, +}; + +static struct snd_soc_dai_driver ak4118_dai = { + .name = "ak4118-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_LE + }, + .ops = &ak4118_dai_ops, +}; + +static irqreturn_t ak4118_irq_handler(int irq, void *data) +{ + struct ak4118_priv *ak4118 = data; + struct snd_soc_component *component = ak4118->component; + struct snd_kcontrol_new *kctl_new; + struct snd_kcontrol *kctl; + struct snd_ctl_elem_id *id; + unsigned int i; + + if (!component) + return IRQ_NONE; + + for (i = 0; i < ARRAY_SIZE(ak4118_iec958_controls); i++) { + kctl_new = &ak4118_iec958_controls[i]; + kctl = snd_soc_card_get_kcontrol(component->card, + kctl_new->name); + if (!kctl) + continue; + id = &kctl->id; + snd_ctl_notify(component->card->snd_card, + SNDRV_CTL_EVENT_MASK_VALUE, id); + } + + return IRQ_HANDLED; +} + +static int ak4118_probe(struct snd_soc_component *component) +{ + struct ak4118_priv *ak4118 = snd_soc_component_get_drvdata(component); + int ret = 0; + + ak4118->component = component; + + /* release reset */ + gpiod_set_value(ak4118->reset, 0); + + /* unmask all int1 sources */ + ret = regmap_write(ak4118->regmap, AK4118_REG_INT1_MASK, 0x00); + if (ret < 0) { + dev_err(component->dev, + "failed to write regmap 0x%x 0x%x: %d\n", + AK4118_REG_INT1_MASK, 0x00, ret); + return ret; + } + + /* rx detect enable on all channels */ + ret = regmap_write(ak4118->regmap, AK4118_REG_RX_DETECT, 0xff); + if (ret < 0) { + dev_err(component->dev, + "failed to write regmap 0x%x 0x%x: %d\n", + AK4118_REG_RX_DETECT, 0xff, ret); + return ret; + } + + ret = snd_soc_add_component_controls(component, ak4118_iec958_controls, + ARRAY_SIZE(ak4118_iec958_controls)); + if (ret) { + dev_err(component->dev, + "failed to add component kcontrols: %d\n", ret); + return ret; + } + + ret = request_threaded_irq(gpiod_to_irq(ak4118->irq), NULL, + ak4118_irq_handler, IRQF_TRIGGER_RISING | + IRQF_ONESHOT, "ak4118-irq", ak4118); + if (ret < 0) { + dev_err(component->dev, "Fail to request_irq: %d\n", ret); + return ret; + } + + return 0; +} + +static void ak4118_remove(struct snd_soc_component *component) +{ + struct ak4118_priv *ak4118 = snd_soc_component_get_drvdata(component); + + /* hold reset */ + gpiod_set_value(ak4118->reset, 1); + + free_irq(gpiod_to_irq(ak4118->irq), ak4118); +} + +static const struct snd_soc_component_driver soc_component_drv_ak4118 = { + .probe = ak4118_probe, + .remove = ak4118_remove, + .dapm_widgets = ak4118_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4118_dapm_widgets), + .dapm_routes = ak4118_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak4118_dapm_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config ak4118_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = ak4118_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ak4118_reg_defaults), + + .cache_type = REGCACHE_NONE, + .max_register = AK4118_REG_MAX - 1, +}; + +static int ak4118_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ak4118_priv *ak4118; + int ret; + + ak4118 = devm_kzalloc(&i2c->dev, sizeof(struct ak4118_priv), + GFP_KERNEL); + if (ak4118 == NULL) + return -ENOMEM; + + ak4118->regmap = devm_regmap_init_i2c(i2c, &ak4118_regmap); + if (IS_ERR(ak4118->regmap)) + return PTR_ERR(ak4118->regmap); + + i2c_set_clientdata(i2c, ak4118); + + ak4118->reset = devm_gpiod_get(&i2c->dev, "reset", GPIOD_OUT_HIGH); + if (IS_ERR(ak4118->reset)) { + ret = PTR_ERR(ak4118->reset); + if (ret != -EPROBE_DEFER) + dev_err(&i2c->dev, "Failed to get reset: %d\n", ret); + return ret; + } + + ak4118->irq = devm_gpiod_get(&i2c->dev, "irq", GPIOD_IN); + if (IS_ERR(ak4118->irq)) { + ret = PTR_ERR(ak4118->irq); + if (ret != -EPROBE_DEFER) + dev_err(&i2c->dev, "Failed to get IRQ: %d\n", ret); + return ret; + } + + return snd_soc_register_component(&i2c->dev, &soc_component_drv_ak4118, + &ak4118_dai, 1); +} + +static int ak4118_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_component(&i2c->dev); + return 0; +} + +static const struct of_device_id ak4118_of_match[] = { + { .compatible = "asahi-kasei,ak4118", }, + {} +}; +MODULE_DEVICE_TABLE(of, ak4118_of_match); + +static const struct i2c_device_id ak4118_id_table[] = { + { "ak4118", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, ak4118_id_table); + +static struct i2c_driver ak4118_i2c_driver = { + .driver = { + .name = "ak4118", + .of_match_table = of_match_ptr(ak4118_of_match), + }, + .id_table = ak4118_id_table, + .probe = ak4118_i2c_probe, + .remove = ak4118_i2c_remove, +}; + +module_i2c_driver(ak4118_i2c_driver); + +MODULE_DESCRIPTION("Asahi Kasei AK4118 ALSA SoC driver"); +MODULE_AUTHOR("Adrien Charruel <adrien.charruel(a)devialet.com>"); +MODULE_LICENSE("GPL"); -- 2.19.1

2 5

[alsa-devel] [RFC][PATCH] ASoC: rsnd: add SSIU BUSIF support
by Kuninori Morimoto 15 Nov '18

15 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Gen2 has BUSIF0-3, Gen3 has BUSIF0-7 on some SSIU. Current driver is assuming it is using BUSIF0 as default. Thus, SSI is attaching SSIU (with BUSIF0) by using rsnd_ssiu_attach(). But, TDM split mode also needs other BUSIF to use it. This patch adds missing SSIU BUSIFx support. BUSIF is handled by SSIU instead of SSI anymore. Thus, its settings no longer needed on SSI node on DT. This patch removes its settings from Document, but driver is still keeping compatibility. Thus, old DT style is still working. But, to avoid confusing, it doesn't indicate old compatiblity things on Document. New SoC should have SSIU on DT from this patch. 1) old style DT is still supported (= no rcar_sound,ssiu node on DT) 2) If ssiu is not indicated on playback/capture, BUSIF0 will be used as default playback = <&ssi3>; /* ssiu30 will be selected */ 3) you can select own ssiu playback = <&ssi32 &ssi3>; /* ssiu32 will be selected */ Not-yet-Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- > Jiada This patch supports BUSIF, you can select it on DT. Is this style OK for you ? For example playback = <&src0 &dvc0 &ssiu02 &ssi0>; ~~~~~~~ It is keeping compatiblity. no <&ssiu> is still OK playback = <&src0 &dvc0 &ssi0>; .../devicetree/bindings/sound/renesas,rsnd.txt | 78 ++++----- sound/soc/sh/rcar/core.c | 1 + sound/soc/sh/rcar/dma.c | 37 ++++- sound/soc/sh/rcar/rsnd.h | 6 +- sound/soc/sh/rcar/ssi.c | 22 +-- sound/soc/sh/rcar/ssiu.c | 178 +++++++++++++++++++-- 6 files changed, 255 insertions(+), 67 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index d92b705..8dfbe5b 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -363,6 +363,9 @@ Required properties: - rcar_sound,ssi : Should contain SSI feature. The number of SSI subnode should be same as HW. see below for detail. +- rcar_sound,ssiu : Should contain SSIU feature. + The number of SSIU subnode should be same as HW. + see below for detail. - rcar_sound,src : Should contain SRC feature. The number of SRC subnode should be same as HW. see below for detail. @@ -532,56 +535,55 @@ rcar_sound: sound@ec500000 { }; }; + rcar_sound,ssiu { + ssiu00: ssiu-0 { + dmas = <&audma0 0x15>, <&audma1 0x16>; + dma-names = "rx", "tx"; + }; + ssiu01: ssiu-1 { + dmas = <&audma0 0x35>, <&audma1 0x36>; + dma-names = "rx", "tx"; + }; + + ... + + ssiu95: ssiu-49 { + dmas = <&audma0 0xA5>, <&audma1 0xA6>; + dma-names = "rx", "tx"; + }; + ssiu96: ssiu-50 { + dmas = <&audma0 0xA7>, <&audma1 0xA8>; + dma-names = "rx", "tx"; + }; + ssiu97: ssiu-51 { + dmas = <&audma0 0xA9>, <&audma1 0xAA>; + dma-names = "rx", "tx"; + }; + }; + rcar_sound,ssi { ssi0: ssi-0 { interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x01>, <&audma1 0x02>, <&audma0 0x15>, <&audma1 0x16>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x01>, <&audma1 0x02>; + dma-names = "rx", "tx"; }; ssi1: ssi-1 { interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x03>, <&audma1 0x04>, <&audma0 0x49>, <&audma1 0x4a>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi2: ssi-2 { - interrupts = <0 372 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x05>, <&audma1 0x06>, <&audma0 0x63>, <&audma1 0x64>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi3: ssi-3 { - interrupts = <0 373 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x07>, <&audma1 0x08>, <&audma0 0x6f>, <&audma1 0x70>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi4: ssi-4 { - interrupts = <0 374 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x09>, <&audma1 0x0a>, <&audma0 0x71>, <&audma1 0x72>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi5: ssi-5 { - interrupts = <0 375 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x0b>, <&audma1 0x0c>, <&audma0 0x73>, <&audma1 0x74>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi6: ssi-6 { - interrupts = <0 376 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x0d>, <&audma1 0x0e>, <&audma0 0x75>, <&audma1 0x76>; - dma-names = "rx", "tx", "rxu", "txu"; - }; - ssi7: ssi-7 { - interrupts = <0 377 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x0f>, <&audma1 0x10>, <&audma0 0x79>, <&audma1 0x7a>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x03>, <&audma1 0x04>; + dma-names = "rx", "tx"; }; + + ... + ssi8: ssi-8 { interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x11>, <&audma1 0x12>, <&audma0 0x7b>, <&audma1 0x7c>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x11>, <&audma1 0x12>; + dma-names = "rx", "tx"; }; ssi9: ssi-9 { interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x13>, <&audma1 0x14>, <&audma0 0x7d>, <&audma1 0x7e>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x13>, <&audma1 0x14>; + dma-names = "rx", "tx"; }; }; diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 5e10012..fff0a3a 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1121,6 +1121,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, break; rsnd_parse_connect_ssi(rdai, playback, capture); + rsnd_parse_connect_ssiu(rdai, playback, capture); rsnd_parse_connect_src(rdai, playback, capture); rsnd_parse_connect_ctu(rdai, playback, capture); rsnd_parse_connect_mix(rdai, playback, capture); diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 5daa6c9..0324a5c 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -218,7 +218,7 @@ struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, int i = 0; for_each_child_of_node(of_node, np) { - if (i == rsnd_mod_id(mod) && (!chan)) + if (i == rsnd_mod_id_raw(mod) && (!chan)) chan = of_dma_request_slave_channel(np, name); i++; } @@ -344,14 +344,16 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io, struct rsnd_mod *mod) { struct rsnd_mod *ssi = rsnd_io_to_mod_ssi(io); + struct rsnd_mod *ssiu = rsnd_io_to_mod_ssiu(io); struct rsnd_mod *src = rsnd_io_to_mod_src(io); struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io); const u8 *entry = NULL; int id = 255; int size = 0; - if (mod == ssi) { - int busif = rsnd_ssi_get_busif(io); + if ((mod == ssi) || + (mod == ssiu)) { + int busif = rsnd_mod_id_sub(ssiu); entry = gen2_id_table_ssiu; size = ARRAY_SIZE(gen2_id_table_ssiu); @@ -530,13 +532,14 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io, struct device *dev = rsnd_priv_to_dev(priv); phys_addr_t ssi_reg = rsnd_gen_get_phy_addr(priv, RSND_GEN2_SSI); phys_addr_t src_reg = rsnd_gen_get_phy_addr(priv, RSND_GEN2_SCU); - int is_ssi = !!(rsnd_io_to_mod_ssi(io) == mod); + int is_ssi = !!(rsnd_io_to_mod_ssi(io) == mod) || + !!(rsnd_io_to_mod_ssiu(io) == mod); int use_src = !!rsnd_io_to_mod_src(io); int use_cmd = !!rsnd_io_to_mod_dvc(io) || !!rsnd_io_to_mod_mix(io) || !!rsnd_io_to_mod_ctu(io); int id = rsnd_mod_id(mod); - int busif = rsnd_ssi_get_busif(io); + int busif = rsnd_mod_id_sub(rsnd_io_to_mod_ssiu(io)); struct dma_addr { dma_addr_t out_addr; dma_addr_t in_addr; @@ -620,7 +623,7 @@ static void rsnd_dma_of_path(struct rsnd_mod *this, struct rsnd_mod **mod_from, struct rsnd_mod **mod_to) { - struct rsnd_mod *ssi = rsnd_io_to_mod_ssi(io); + struct rsnd_mod *ssi; struct rsnd_mod *src = rsnd_io_to_mod_src(io); struct rsnd_mod *ctu = rsnd_io_to_mod_ctu(io); struct rsnd_mod *mix = rsnd_io_to_mod_mix(io); @@ -631,6 +634,28 @@ static void rsnd_dma_of_path(struct rsnd_mod *this, struct device *dev = rsnd_priv_to_dev(priv); int nr, i, idx; + /* + * It should use "rcar_sound,ssiu" on DT. + * But, we need to keep compatibility for old version. + * + * If it has "rcar_sound.ssiu", it will be used. + * If not, "rcar_sound.ssi" will be used. + * see + * rsnd_ssiu_dma_req() + * rsnd_ssi_dma_req() + */ + if (rsnd_ssiu_of_node(priv)) { + struct rsnd_mod *ssiu = rsnd_io_to_mod_ssiu(io); + + /* use SSIU */ + ssi = ssiu; + if (this == rsnd_io_to_mod_ssi(io)) + this = ssiu; + } else { + /* keep compatible, use SSI */ + ssi = rsnd_io_to_mod_ssi(io); + } + if (!ssi) return; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 28bd90a..1875fd9 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -441,6 +441,7 @@ int rsnd_runtime_is_ssi_tdm(struct rsnd_dai_stream *io); of_get_child_by_name(rsnd_priv_to_dev(priv)->of_node, node) #define RSND_NODE_DAI "rcar_sound,dai" #define RSND_NODE_SSI "rcar_sound,ssi" +#define RSND_NODE_SSIU "rcar_sound,ssiu" #define RSND_NODE_SRC "rcar_sound,src" #define RSND_NODE_CTU "rcar_sound,ctu" #define RSND_NODE_MIX "rcar_sound,mix" @@ -715,7 +716,6 @@ void rsnd_ssi_remove(struct rsnd_priv *priv); struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id); int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod); int rsnd_ssi_use_busif(struct rsnd_dai_stream *io); -int rsnd_ssi_get_busif(struct rsnd_dai_stream *io); u32 rsnd_ssi_multi_slaves_runtime(struct rsnd_dai_stream *io); #define RSND_SSI_HDMI_PORT0 0xf0 @@ -743,6 +743,10 @@ int rsnd_ssiu_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod); int rsnd_ssiu_probe(struct rsnd_priv *priv); void rsnd_ssiu_remove(struct rsnd_priv *priv); +void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, + struct device_node *playback, + struct device_node *capture); +#define rsnd_ssiu_of_node(priv) rsnd_parse_of_node(priv, RSND_NODE_SSIU) /* * R-Car SRC diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index a032819..95acc8b 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -150,11 +150,6 @@ int rsnd_ssi_use_busif(struct rsnd_dai_stream *io) return use_busif; } -int rsnd_ssi_get_busif(struct rsnd_dai_stream *io) -{ - return 0; /* BUSIF0 only for now */ -} - static void rsnd_ssi_status_clear(struct rsnd_mod *mod) { rsnd_mod_write(mod, SSISR, 0); @@ -761,7 +756,7 @@ static int rsnd_ssi_common_probe(struct rsnd_mod *mod, { struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - int ret; + int ret = 0; /* * SSIP/SSIU/IRQ are not needed on @@ -775,10 +770,6 @@ static int rsnd_ssi_common_probe(struct rsnd_mod *mod, * see rsnd_ssi_pcm_new() */ - ret = rsnd_ssiu_attach(io, mod); - if (ret < 0) - return ret; - /* * SSI might be called again as PIO fallback * It is easy to manual handling for IRQ request/free @@ -972,6 +963,17 @@ static struct dma_chan *rsnd_ssi_dma_req(struct rsnd_dai_stream *io, int is_play = rsnd_io_is_play(io); char *name; + /* + * It should use "rcar_sound,ssiu" on DT. + * But, we need to keep compatibility for old version. + * + * If it has "rcar_sound.ssiu", it will be used. + * If not, "rcar_sound.ssi" will be used. + * see + * rsnd_ssiu_dma_req() + * rsnd_dma_of_path() + */ + if (rsnd_ssi_use_busif(io)) name = is_play ? "rxu" : "txu"; else diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index ebcb826..20f7b04 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -12,6 +12,8 @@ struct rsnd_ssiu { struct rsnd_mod mod; u32 busif_status[8]; /* for BUSIF0 - BUSIF7 */ unsigned int usrcnt; + int id; + int id_sub; }; #define rsnd_ssiu_nr(priv) ((priv)->ssiu_nr) @@ -22,12 +24,29 @@ struct rsnd_ssiu { ((pos) = ((struct rsnd_ssiu *)(priv)->ssiu + i)); \ i++) +/* + * SSI Gen2 Gen3 + * 0 BUSIF0-3 BUSIF0-7 + * 1 BUSIF0-3 BUSIF0-7 + * 2 BUSIF0-3 BUSIF0-7 + * 3 BUSIF0 BUSIF0-7 + * 4 BUSIF0 BUSIF0-7 + * 5 BUSIF0 BUSIF0 + * 6 BUSIF0 BUSIF0 + * 7 BUSIF0 BUSIF0 + * 8 BUSIF0 BUSIF0 + * 9 BUSIF0-3 BUSIF0-7 + * total 22 52 + */ +static const int gen2_id[] = { 0, 4, 8, 12, 13, 14, 15, 16, 17, 18 }; +static const int gen3_id[] = { 0, 8, 16, 24, 32, 40, 41, 42, 43, 44 }; + static u32 *rsnd_ssiu_get_status(struct rsnd_mod *mod, struct rsnd_dai_stream *io, enum rsnd_mod_type type) { struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); - int busif = rsnd_ssi_get_busif(io); + int busif = rsnd_mod_id_sub(mod); return &ssiu->busif_status[busif]; } @@ -158,7 +177,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, if (rsnd_ssi_use_busif(io)) { int id = rsnd_mod_id(mod); - int busif = rsnd_ssi_get_busif(io); + int busif = rsnd_mod_id_sub(mod); /* * FIXME @@ -255,7 +274,7 @@ static int rsnd_ssiu_start_gen2(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - int busif = rsnd_ssi_get_busif(io); + int busif = rsnd_mod_id_sub(mod); if (!rsnd_ssi_use_busif(io)) return 0; @@ -273,7 +292,7 @@ static int rsnd_ssiu_stop_gen2(struct rsnd_mod *mod, struct rsnd_priv *priv) { struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); - int busif = rsnd_ssi_get_busif(io); + int busif = rsnd_mod_id_sub(mod); if (!rsnd_ssi_use_busif(io)) return 0; @@ -289,8 +308,49 @@ static int rsnd_ssiu_stop_gen2(struct rsnd_mod *mod, return 0; } +static int rsnd_ssiu_id(struct rsnd_mod *mod) +{ + struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); + + /* see rsnd_ssiu_probe() */ + return ssiu->id; +} + +static int rsnd_ssiu_id_sub(struct rsnd_mod *mod) +{ + struct rsnd_ssiu *ssiu = rsnd_mod_to_ssiu(mod); + + /* see rsnd_ssiu_probe() */ + return ssiu->id_sub; +} + +static struct dma_chan *rsnd_ssiu_dma_req(struct rsnd_dai_stream *io, + struct rsnd_mod *mod) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + int is_play = rsnd_io_is_play(io); + char *name; + + /* + * It should use "rcar_sound,ssiu" on DT. + * But, we need to keep compatibility for old version. + * + * If it has "rcar_sound.ssiu", it will be used. + * If not, "rcar_sound.ssi" will be used. + * see + * rsnd_ssi_dma_req() + * rsnd_dma_of_path() + */ + + name = is_play ? "rx" : "tx"; + + return rsnd_dma_request_channel(rsnd_ssiu_of_node(priv), + mod, name); +} + static struct rsnd_mod_ops rsnd_ssiu_ops_gen2 = { .name = SSIU_NAME, + .dma_req = rsnd_ssiu_dma_req, .init = rsnd_ssiu_init_gen2, .start = rsnd_ssiu_start_gen2, .stop = rsnd_ssiu_stop_gen2, @@ -305,26 +365,83 @@ static struct rsnd_mod *rsnd_ssiu_mod_get(struct rsnd_priv *priv, int id) return rsnd_mod_get((struct rsnd_ssiu *)(priv->ssiu) + id); } -int rsnd_ssiu_attach(struct rsnd_dai_stream *io, - struct rsnd_mod *ssi_mod) +static void rsnd_parse_connect_ssiu_compatible(struct rsnd_priv *priv, + struct rsnd_dai_stream *io) +{ + struct rsnd_mod *ssi_mod = rsnd_io_to_mod_ssi(io); + struct rsnd_mod *mod; + struct rsnd_ssiu *ssiu; + int i; + + if (!ssi_mod) + return; + + /* select BUSIF0 */ + for_each_rsnd_ssiu(ssiu, priv, i) { + mod = rsnd_mod_get(ssiu); + + if ((rsnd_mod_id(ssi_mod) == rsnd_ssiu_id(mod)) && + (rsnd_mod_id_sub(mod) == 0)) + rsnd_dai_connect(mod, io, mod->type); + } +} + +void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, + struct device_node *playback, + struct device_node *capture) { - struct rsnd_priv *priv = rsnd_io_to_priv(io); - struct rsnd_mod *mod = rsnd_ssiu_mod_get(priv, rsnd_mod_id(ssi_mod)); + struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device_node *node = rsnd_ssiu_of_node(priv); + struct device_node *np; + struct rsnd_mod *mod; + struct rsnd_dai_stream *io_p = &rdai->playback; + struct rsnd_dai_stream *io_c = &rdai->capture; + int i; - rsnd_mod_confirm_ssi(ssi_mod); + /* use rcar_sound,ssiu if exist */ + if (node) { + i = 0; + for_each_child_of_node(node, np) { + mod = rsnd_ssiu_mod_get(priv, i); + if (np == playback) + rsnd_dai_connect(mod, io_p, mod->type); + if (np == capture) + rsnd_dai_connect(mod, io_c, mod->type); + i++; + } - return rsnd_dai_connect(mod, io, mod->type); + of_node_put(node); + } + + /* Keep DT compatibility */ + if (!rsnd_io_to_mod_ssiu(io_p)) + rsnd_parse_connect_ssiu_compatible(priv, io_p); + if (!rsnd_io_to_mod_ssiu(io_c)) + rsnd_parse_connect_ssiu_compatible(priv, io_c); } int rsnd_ssiu_probe(struct rsnd_priv *priv) { struct device *dev = rsnd_priv_to_dev(priv); + struct device_node *node; struct rsnd_ssiu *ssiu; struct rsnd_mod_ops *ops; + const int *list = NULL; int i, nr, ret; - /* same number to SSI */ - nr = priv->ssi_nr; + /* + * Keep DT compatibility. + * if it has "rcar_sound,ssiu", use it. + * if not, use "rcar_sound,ssi" + * see + * rsnd_ssiu_bufsif_to_id() + */ + node = rsnd_ssiu_of_node(priv); + if (node) + nr = of_get_child_count(node); + else + nr = priv->ssi_nr; + ssiu = devm_kcalloc(dev, nr, sizeof(*ssiu), GFP_KERNEL); if (!ssiu) return -ENOMEM; @@ -337,7 +454,44 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv) else ops = &rsnd_ssiu_ops_gen2; + /* Keep compatibility */ + nr = 0; + if ((node) && + (ops == &rsnd_ssiu_ops_gen2)) { + ops->id = rsnd_ssiu_id; + ops->id_sub = rsnd_ssiu_id_sub; + + if (rsnd_is_gen2(priv)) { + list = gen2_id; + nr = ARRAY_SIZE(gen2_id); + } else if (rsnd_is_gen3(priv)) { + list = gen3_id; + nr = ARRAY_SIZE(gen3_id); + } else { + dev_err(dev, "unknown SSIU\n"); + return -ENODEV; + } + } + for_each_rsnd_ssiu(ssiu, priv, i) { + if (node) { + int j; + + /* + * see + * rsnd_ssiu_get_id() + * rsnd_ssiu_get_id_sub() + */ + for (j = 0; j < nr; j++) { + if (list[j] > i) + break; + ssiu->id = j; + ssiu->id_sub = i - list[ssiu->id]; + } + } else { + ssiu->id = i; + } + ret = rsnd_mod_init(priv, rsnd_mod_get(ssiu), ops, NULL, RSND_MOD_SSIU, i); if (ret) -- 2.7.4

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Re: [alsa-devel] [EXTERNAL] Re: [PATCH 1/2] ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier
by Schulman, James 14 Nov '18

14 Nov '18

On Tue, 13 Nov 2018, Mark Brown wrote: > On Tue, Nov 13, 2018 at 12:49:16PM -0600, James Schulman wrote: > >> +++ b/sound/soc/codecs/cs35l36-tables.c >> @@ -0,0 +1,315 @@ >> +/* >> + * cs35l36-tables.c -- CS35L36 ALSA SoC audio driver >> + * >> + * Copyright 2018 Cirrus Logic, Inc. > > Please use SPDX headers. > Will do. >> + SOC_SINGLE_TLV("AMP PCM Gain", CS35L36_AMP_GAIN_CTRL, 5, 0x13, 0, >> + amp_gain_tlv), > > All volume controls should end in Volume, see control-names.rst. > >> + regmap_read(cs35l36->regmap, CS35L36_INT4_RAW_STATUS, &reg); >> + if (reg & CS35L36_PLL_UNLOCK_MASK) >> + dev_crit(cs35l36->dev, "PLL Unlocked\n"); > > WARN_ON_ONCE? > Will do. >> + } >> + return 0; >> +} >> +static const char * const cs35l36_chan_text[] = { > > Missing blank line. > Will do. >> +static const char * const cs35l36_boost_text[] = { >> + "On", >> + "Off", >> +}; >> + >> +static SOC_ENUM_SINGLE_DECL(boost_enum, SND_SOC_NOPM, 0, >> + cs35l36_boost_text); >> + >> +static const struct snd_kcontrol_new cs35l36_boost_mux[] = { >> + SOC_DAPM_ENUM("Boost Enable", boost_enum), >> +}; > > Simple on/off conttrols should be _SINGLE controls with Switch at the > end of their name. It's not super clear why this is in DAPM as a widget > at all, the routing around it is: > > + {"BOOST Mux", "On", "Channel Mux"}, > + {"CLASS H", NULL, "BOOST Mux"}, > > which is a bit perplexing. Possibly this should be handled in the event > for the main amplifier? > We used _ENUM instead of _SINGLE because we wanted the default value to be "On" instead of "Off". We don't want the default assumption to have boost turned off. However, we're happy to change this if you would rather have consitency with the _SINGLE controls. >> + if (cs35l36->sclk > 6000000) { >> + fs1_val = 3 * 4 + 4; >> + fs2_val = 8 * 4 + 4; >> + } > > These are just constants, why write out the operations? > Will do. >> + >> + if (cs35l36->sclk <= 6000000) { > > if/else? > Will do. >> + /* >> + * Rev B0 has 2 versions >> + * L36 is 10V >> + * L37 is 12V > > \o/ > Yes we had 2 silicon spins for this revision >_< is there anything about this comment you want us to change? >> + } >> + >> + return IRQ_HANDLED; > > This unconditionally reports that it handled the interrupt, even if it > didn't. This will stop the interrupt core handling for faulty interrupt > lines working and will disrupt any support that gets added for handling > shared threaded interrupts. > Earlier in the handler, we have a check to see if any unmasked bits are set. If not we return IRQ_NONE. I figured this would be enough to ensure we don't unconditionally report IRQ_HANDLED. >> + if (pdata) >> + cs35l36->pdata = *pdata; >> + else { > > Use { } on both sides of the if (see coding-style.rst)? > Will do. >> + cs35l36->reset_gpio = devm_gpiod_get_optional(dev, "reset", >> + GPIOD_OUT_LOW); >> + if (IS_ERR(cs35l36->reset_gpio)) { >> + ret = PTR_ERR(cs35l36->reset_gpio); >> + cs35l36->reset_gpio = NULL; >> + if (ret == -EBUSY) >> + dev_info(dev, >> + "Reset line busy, assuming shared reset\n"); >> + else { >> + dev_err(dev, "Failed to get reset GPIO: %d\n", ret); >> + goto err; >> + } >> + } > > This doesn't handle -EPROBE_DEFER. > I think we will add an err_disable_regs jump statement to ensure we don't try to use the uninitialized reset_gpio handle. Once we do that, I think this will handle -EPROBE_DEFER cleanly? Thanks for the comments Mark. - James

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[alsa-devel] [PATCH] ASoC: dapm: Recalculate audio map forcely when card instantiated
by Tzung-Bi Shih 14 Nov '18

14 Nov '18

Audio map are possible in wrong state before card->instantiated has been set to true. Imaging the following examples: time 1: at the beginning in:-1 in:-1 in:-1 in:-1 out:-1 out:-1 out:-1 out:-1 SIGGEN A B Spk time 2: after someone called snd_soc_dapm_new_widgets() (e.g. create_fill_widget_route_map() in sound/soc/codecs/hdac_hdmi.c) in:1 in:0 in:0 in:0 out:0 out:0 out:0 out:1 SIGGEN A B Spk time 3: routes added in:1 in:0 in:0 in:0 out:0 out:0 out:0 out:1 SIGGEN -----> A -----> B ---> Spk In the end, the path should be powered on but it did not. At time 3, "in" of SIGGEN and "out" of Spk did not propagate to their neighbors because snd_soc_dapm_add_path() will not invalidate the paths if the card has not instantiated (i.e. card->instantiated is false). To correct the state of audio map, recalculate the whole map forcely. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6ddcf12bc030..b29d0f65611e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2131,6 +2131,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } card->instantiated = 1; + dapm_mark_endpoints_dirty(card); snd_soc_dapm_sync(&card->dapm); mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); -- 2.19.1.930.g4563a0d9d0-goog

2 1

[alsa-devel] Applied "ASoC: pcm3060: Add DT property for single-ended output" to the asoc tree
by Mark Brown 14 Nov '18

14 Nov '18

The patch ASoC: pcm3060: Add DT property for single-ended output 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 9d4b17efa51f3bb0918021090c09b0013039df00 Mon Sep 17 00:00:00 2001 From: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Date: Mon, 12 Nov 2018 08:08:33 +0100 Subject: [PATCH] ASoC: pcm3060: Add DT property for single-ended output DAC output may be differential (default) or single-ended. Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/pcm3060.c | 14 ++++++++++++++ sound/soc/codecs/pcm3060.h | 1 + 2 files changed, 15 insertions(+) diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index 494d9d662be8..5f479aa61097 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -274,9 +274,23 @@ EXPORT_SYMBOL(pcm3060_regmap); /* device */ +static void pcm3060_parse_dt(const struct device_node *np, + struct pcm3060_priv *priv) +{ + priv->out_se = of_property_read_bool(np, "ti,out-single-ended"); +} + int pcm3060_probe(struct device *dev) { int rc; + struct pcm3060_priv *priv = dev_get_drvdata(dev); + + if (dev->of_node) + pcm3060_parse_dt(dev->of_node, priv); + + if (priv->out_se) + regmap_update_bits(priv->regmap, PCM3060_REG64, + PCM3060_REG_SE, PCM3060_REG_SE); rc = devm_snd_soc_register_component(dev, &pcm3060_soc_comp_driver, pcm3060_dai, diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h index fd89a68aa8a7..c895cf40ee10 100644 --- a/sound/soc/codecs/pcm3060.h +++ b/sound/soc/codecs/pcm3060.h @@ -25,6 +25,7 @@ struct pcm3060_priv_dai { struct pcm3060_priv { struct regmap *regmap; struct pcm3060_priv_dai dai[PCM3060_DAI_IDS_NUM]; + u8 out_se: 1; }; int pcm3060_probe(struct device *dev); -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: amd: fix memory leak of i2s_data on error return" to the asoc tree
by Mark Brown 14 Nov '18

14 Nov '18

The patch ASoC: amd: fix memory leak of i2s_data on error return 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 46dce4042659755a2b9753369d0c2d8ca33e35cf Mon Sep 17 00:00:00 2001 From: Colin Ian King <colin.king(a)canonical.com> Date: Wed, 14 Nov 2018 21:31:48 +0000 Subject: [PATCH] ASoC: amd: fix memory leak of i2s_data on error return Currently when snd_pcm_hw_constraint_integer fails there is a memory leak of i2s_data on the error return path. Fix this by kfree'ing i2s_data before returning. Detected by CoverityScan, CID#1475479 ("Resource leak") Fixes: 0b87d6bcd648 ("ASoC: amd: add acp3x pcm driver dma ops") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 2e61cef00f41..b16cdd187139 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -309,6 +309,7 @@ static int acp3x_dma_open(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) { dev_err(component->dev, "set integer constraint failed\n"); + kfree(i2s_data); return ret; } -- 2.19.1

1 0

[alsa-devel] [PATCH] ASoC: omap-abe-twl6040: Fix missing audio card caused by deferred probing
by Peter Ujfalusi 14 Nov '18

14 Nov '18

The machine driver fails to probe in next-20181113 with: [ 2.539093] omap-abe-twl6040 sound: ASoC: CODEC DAI twl6040-legacy not registered [ 2.546630] omap-abe-twl6040 sound: devm_snd_soc_register_card() failed: -517 ... [ 3.693206] omap-abe-twl6040 sound: ASoC: Both platform name/of_node are set for TWL6040 [ 3.701446] omap-abe-twl6040 sound: ASoC: failed to init link TWL6040 [ 3.708007] omap-abe-twl6040 sound: devm_snd_soc_register_card() failed: -22 [ 3.715148] omap-abe-twl6040: probe of sound failed with error -22 Bisect pointed to a merge commit: first bad commit: [0f688ab20a540aafa984c5dbd68a71debebf4d7f] Merge remote-tracking branch 'net-next/master' and a diff between a working kernel does not reveal anything which would explain the change in behavior. Further investigation showed that on the second try of loading fails because the dai_link->platform is no longer NULL and it might be pointing to uninitialized memory. The fix is to move the snd_soc_dai_link and snd_soc_card inside of the abe_twl6040 struct, which is dynamically allocated every time the driver probes. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-abe-twl6040.c | 67 +++++++++++++------------------ 1 file changed, 29 insertions(+), 38 deletions(-) diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index d5ae9eb8c756..fed45b41f9d3 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -36,6 +36,8 @@ #include "../codecs/twl6040.h" struct abe_twl6040 { + struct snd_soc_card card; + struct snd_soc_dai_link dai_links[2]; int jack_detection; /* board can detect jack events */ int mclk_freq; /* MCLK frequency speed for twl6040 */ }; @@ -208,40 +210,10 @@ static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(dmic_audio_map)); } -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link abe_twl6040_dai_links[] = { - { - .name = "TWL6040", - .stream_name = "TWL6040", - .codec_dai_name = "twl6040-legacy", - .codec_name = "twl6040-codec", - .init = omap_abe_twl6040_init, - .ops = &omap_abe_ops, - }, - { - .name = "DMIC", - .stream_name = "DMIC Capture", - .codec_dai_name = "dmic-hifi", - .codec_name = "dmic-codec", - .init = omap_abe_dmic_init, - .ops = &omap_abe_dmic_ops, - }, -}; - -/* Audio machine driver */ -static struct snd_soc_card omap_abe_card = { - .owner = THIS_MODULE, - - .dapm_widgets = twl6040_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), -}; - static int omap_abe_probe(struct platform_device *pdev) { struct device_node *node = pdev->dev.of_node; - struct snd_soc_card *card = &omap_abe_card; + struct snd_soc_card *card; struct device_node *dai_node; struct abe_twl6040 *priv; int num_links = 0; @@ -252,12 +224,18 @@ static int omap_abe_probe(struct platform_device *pdev) return -ENODEV; } - card->dev = &pdev->dev; - priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL); if (priv == NULL) return -ENOMEM; + card = &priv->card; + card->dev = &pdev->dev; + card->owner = THIS_MODULE; + card->dapm_widgets = twl6040_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets); + card->dapm_routes = audio_map; + card->num_dapm_routes = ARRAY_SIZE(audio_map); + if (snd_soc_of_parse_card_name(card, "ti,model")) { dev_err(&pdev->dev, "Card name is not provided\n"); return -ENODEV; @@ -274,14 +252,27 @@ static int omap_abe_probe(struct platform_device *pdev) dev_err(&pdev->dev, "McPDM node is not provided\n"); return -EINVAL; } - abe_twl6040_dai_links[0].cpu_of_node = dai_node; - abe_twl6040_dai_links[0].platform_of_node = dai_node; + + priv->dai_links[0].name = "DMIC"; + priv->dai_links[0].stream_name = "TWL6040"; + priv->dai_links[0].cpu_of_node = dai_node; + priv->dai_links[0].platform_of_node = dai_node; + priv->dai_links[0].codec_dai_name = "twl6040-legacy"; + priv->dai_links[0].codec_name = "twl6040-codec"; + priv->dai_links[0].init = omap_abe_twl6040_init; + priv->dai_links[0].ops = &omap_abe_ops; dai_node = of_parse_phandle(node, "ti,dmic", 0); if (dai_node) { num_links = 2; - abe_twl6040_dai_links[1].cpu_of_node = dai_node; - abe_twl6040_dai_links[1].platform_of_node = dai_node; + priv->dai_links[1].name = "TWL6040"; + priv->dai_links[1].stream_name = "DMIC Capture"; + priv->dai_links[1].cpu_of_node = dai_node; + priv->dai_links[1].platform_of_node = dai_node; + priv->dai_links[1].codec_dai_name = "dmic-hifi"; + priv->dai_links[1].codec_name = "dmic-codec"; + priv->dai_links[1].init = omap_abe_dmic_init; + priv->dai_links[1].ops = &omap_abe_dmic_ops; } else { num_links = 1; } @@ -300,7 +291,7 @@ static int omap_abe_probe(struct platform_device *pdev) return -ENODEV; } - card->dai_link = abe_twl6040_dai_links; + card->dai_link = priv->dai_links; card->num_links = num_links; snd_soc_card_set_drvdata(card, priv); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH][next] ASoC: amd: fix memory leak of i2s_data on error return
by Colin King 14 Nov '18

14 Nov '18

From: Colin Ian King <colin.king(a)canonical.com> Currently when snd_pcm_hw_constraint_integer fails there is a memory leak of i2s_data on the error return path. Fix this by kfree'ing i2s_data before returning. Detected by CoverityScan, CID#1475479 ("Resource leak") Fixes: 0b87d6bcd648 ("ASoC: amd: add acp3x pcm driver dma ops") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 2e61cef00f41..b16cdd187139 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -309,6 +309,7 @@ static int acp3x_dma_open(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) { dev_err(component->dev, "set integer constraint failed\n"); + kfree(i2s_data); return ret; } -- 2.19.1

1 0

Re: [alsa-devel] linux-next: Tree for Nov 14 (sound/soc/amd/raven/)
by Randy Dunlap 14 Nov '18

14 Nov '18

On 11/13/18 9:26 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20181113: > on i386 or x86_64: when CONFIG_PCI is not enabled. ../sound/soc/amd/raven/pci-acp3x.c: In function 'snd_acp3x_probe': ../sound/soc/amd/raven/pci-acp3x.c:59:2: error: implicit declaration of function 'pci_enable_msi' [-Werror=implicit-function-declaration] ret = pci_enable_msi(pci); ^ ../sound/soc/amd/raven/pci-acp3x.c:124:2: error: implicit declaration of function 'pci_disable_msi' [-Werror=implicit-function-declaration] pci_disable_msi(pci); ^ ../sound/soc/amd/raven/pci-acp3x.c: At top level: ../sound/soc/amd/raven/pci-acp3x.c:161:1: warning: data definition has no type or storage class [enabled by default] module_pci_driver(acp3x_driver); ^ ../sound/soc/amd/raven/pci-acp3x.c:161:1: error: type defaults to 'int' in declaration of 'module_pci_driver' [-Werror=implicit-int] ../sound/soc/amd/raven/pci-acp3x.c:161:1: warning: parameter names (without types) in function declaration [enabled by default] ../sound/soc/amd/raven/pci-acp3x.c:154:26: warning: 'acp3x_driver' defined but not used [-Wunused-variable] static struct pci_driver acp3x_driver = { ^ config SND_SOC_AMD_ACP3x tristate "AMD Audio Coprocessor-v3.x support" help This option enables ACP v3.x I2S support on AMD platform Needs "depends on PCI"?? -- ~Randy

1 0

[alsa-devel] alsa-utils: axfer: rewrite aplay, adding 'timer-based scheduling' option
by GitHub pull_request - opened 14 Nov '18

14 Nov '18

alsa-project/alsa-utils pull request #3 was opened from takaswie: Hi, This patchset is an updated version of my previous RFCv3, and for merging to upstream. [alsa-devel] [RFCv3][PATCH 00/39] axfer: rewrite aplay, adding 'timer-based scheduling' option http://mailman.alsa-project.org/pipermail/alsa-devel/2017-October/125961.ht… As I noted in RFCv3, some patches are dropped in this patchset, due to some technical issues. In detail, please refer to RFCv3. Changes from RFCv3: - rename scheduling option to 'sched-model' - improve waiter implementation and add support for select(2) - categorize supported options as common and backend-dependent - release alsa-lib configuration space at the end of runtime - valgrind test and fix some memory leaks - add 'frame-cache' internal interface to share implementation by several backends. - add 'libffado' backend just for technical preview (linkage of this library brings problems. Please refet to patch comment.) I apologize but in this time I don't prepare any help document and manuals as well as RFCv3. Let me put them to a part of task for this development period. Regards Takashi Sakamoto (35): axfer: add an entry point for this command axfer: add a sub-command to print list of PCMs/devices axfer: add a common interface to handle a file with audio-specific data format axfer: add support for a container of Microsoft/IBM RIFF/Wave format axfer: add support for a container of Sparc AU format axfer: add support for a container of Creative Tech. voice format axfer: add support for a container of raw data axfer: add unit test for container interface axfer: add a common interface to align data frames on different layout axfer: add support for a mapper for single target axfer: add support for a mapper for multiple target axfer: add a unit test for mapper interface axfer: add a common interface to transfer data frames axfer: add a parser for command-line options axfer: add support to transfer data frames by alsa-lib PCM APIs axfer: add support for blocking data transmission operation of alsa-lib PCM API axfer: add a sub-command to transfer data frames axfer: add informative output and an option to suppress it axfer: add an option to dump available hardware parameters axfer: add options related to duration and obsolete '--max-file-size' option axfer: add an option to finish transmission at XRUN axfer: add support for non-blocking operation axfer: add support for MMAP PCM operation axfer: add an option to suppress event waiting axfer: add options for buffer arrangement axfer: add options for software parameters of PCM substream axfer: add options for plugins in alsa-lib axfer: add a common interface of waiter for I/O event notification axfer: add an option for waiter type axfer: add an implementation of waiter for poll(2) axfer: add an implementation of waiter for select(2) axfer: add an implementation of waiter for epoll(7) axfer: add support for timer-based scheduling model with MMAP operation axfer: obsolete some unimplemented options axfer: add support for libffado transmission backend ``` Makefile.am | 2 +- axfer/Makefile.am | 62 +++ axfer/container-au.c | 203 +++++++ axfer/container-raw.c | 66 +++ axfer/container-riff-wave.c | 575 +++++++++++++++++++ axfer/container-voc.c | 833 +++++++++++++++++++++++++++ axfer/container.c | 457 +++++++++++++++ axfer/container.h | 126 +++++ axfer/frame-cache.c | 111 ++++ axfer/frame-cache.h | 47 ++ axfer/main.c | 229 ++++++++ axfer/mapper-multiple.c | 259 +++++++++ axfer/mapper-single.c | 191 +++++++ axfer/mapper.c | 150 +++++ axfer/mapper.h | 87 +++ axfer/misc.h | 19 + axfer/subcmd-list.c | 234 ++++++++ axfer/subcmd-transfer.c | 467 ++++++++++++++++ axfer/subcmd.h | 18 + axfer/test/Makefile.am | 31 ++ axfer/test/container-test.c | 299 ++++++++++ axfer/test/generator.c | 260 +++++++++ axfer/test/generator.h | 47 ++ axfer/test/mapper-test.c | 491 ++++++++++++++++ axfer/waiter-epoll.c | 107 ++++ axfer/waiter-poll.c | 45 ++ axfer/waiter-select.c | 100 ++++ axfer/waiter.c | 104 ++++ axfer/waiter.h | 61 ++ axfer/xfer-libasound-irq-mmap.c | 282 ++++++++++ axfer/xfer-libasound-irq-rw.c | 427 ++++++++++++++ axfer/xfer-libasound-timer-mmap.c | 455 +++++++++++++++ axfer/xfer-libasound.c | 898 ++++++++++++++++++++++++++++++ axfer/xfer-libasound.h | 92 +++ axfer/xfer-libffado.c | 555 ++++++++++++++++++ axfer/xfer-options.c | 589 ++++++++++++++++++++ axfer/xfer.c | 254 +++++++++ axfer/xfer.h | 115 ++++ configure.ac | 15 +- 39 files changed, 9361 insertions(+), 2 deletions(-) create mode 100644 axfer/Makefile.am create mode 100644 axfer/container-au.c create mode 100644 axfer/container-raw.c create mode 100644 axfer/container-riff-wave.c create mode 100644 axfer/container-voc.c create mode 100644 axfer/container.c create mode 100644 axfer/container.h create mode 100644 axfer/frame-cache.c create mode 100644 axfer/frame-cache.h create mode 100644 axfer/main.c create mode 100644 axfer/mapper-multiple.c create mode 100644 axfer/mapper-single.c create mode 100644 axfer/mapper.c create mode 100644 axfer/mapper.h create mode 100644 axfer/misc.h create mode 100644 axfer/subcmd-list.c create mode 100644 axfer/subcmd-transfer.c create mode 100644 axfer/subcmd.h create mode 100644 axfer/test/Makefile.am create mode 100644 axfer/test/container-test.c create mode 100644 axfer/test/generator.c create mode 100644 axfer/test/generator.h create mode 100644 axfer/test/mapper-test.c create mode 100644 axfer/waiter-epoll.c create mode 100644 axfer/waiter-poll.c create mode 100644 axfer/waiter-select.c create mode 100644 axfer/waiter.c create mode 100644 axfer/waiter.h create mode 100644 axfer/xfer-libasound-irq-mmap.c create mode 100644 axfer/xfer-libasound-irq-rw.c create mode 100644 axfer/xfer-libasound-timer-mmap.c create mode 100644 axfer/xfer-libasound.c create mode 100644 axfer/xfer-libasound.h create mode 100644 axfer/xfer-libffado.c create mode 100644 axfer/xfer-options.c create mode 100644 axfer/xfer.c create mode 100644 axfer/xfer.h ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/3 Patch URL : https://github.com/alsa-project/alsa-utils/pull/3.patch Repository URL: https://github.com/alsa-project/alsa-utils

3 38

[alsa-devel] GitHub - alsa-project - repositories
by Jaroslav Kysela 14 Nov '18

14 Nov '18

Hi all, URL: https://github.com/alsa-project I finally finished the first phase of the integration with GitHub. All repositories are now on GitHub and all repositories are mirrored to alsa-project.org. The github repositories are master (developers should push changes only to those repositories). If you refer the repository as the code source (for packaging or so), please, keep to use the repositories on alsa-project.org (in case when we migrate to other service in future like GitLab or so). The mirroring is realtime, so the changes should be visible on git.alsa-project.org in few seconds after the push. I invited few people to the GitHub team and actually Takashi has full access to all repos, Vinod Koul should have the write access to tinycompress and Takashi Sakamoto should have the write access to alsa-gi. If I omitted someone (or someone is not on github), please, let me know (and register before). Because github adds possibility for the pull requests and issue tracking, I added notifications for them to this (alsa-devel) mailing list. The short notification should be sent when an pull request or an issue is opened or changed. I think that it would be best to handle this per request, so the developer can ask to resend the patch to the mailing list for the wider review or just push the change with the signing. I activated Travis CI for alsa-lib and alsa-utils and I will add other repos soon, too. URL: https://travis-ci.org/alsa-project If you have some ideas which other github applications can be used to improve the code maintenance, let me know. I will probably play with the coverity checker (http://scan.coverity.com), too. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

5 10

[alsa-devel] [PATCH 01/11] ASoC: AMD: add ACP 3.x IP register header
by Vijendar Mukunda 14 Nov '18

14 Nov '18

From: Maruthi Srinivas Bayyavarapu <Maruthi.Bayyavarapu(a)amd.com> ACP 3.x is a new audio block in raven. Added register header of the same. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/chip_offset_byte.h | 655 +++++++++++++++++++++++++++++++++ 1 file changed, 655 insertions(+) create mode 100644 sound/soc/amd/raven/chip_offset_byte.h diff --git a/sound/soc/amd/raven/chip_offset_byte.h b/sound/soc/amd/raven/chip_offset_byte.h new file mode 100644 index 0000000..fceb7bb --- /dev/null +++ b/sound/soc/amd/raven/chip_offset_byte.h @@ -0,0 +1,655 @@ +/* + * ACP 3.0 Register documentation + * + * Copyright (C) 2016 Advanced Micro Devices, Inc. + * + * Permission is hereby granted, free of charge, to any person obtaining a + * copy of this software and associated documentation files (the "Software"), + * to deal in the Software without restriction, including without limitation + * the rights to use, copy, modify, merge, publish, distribute, sublicense, + * and/or sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included + * in all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS + * OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE COPYRIGHT HOLDER(S) BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR + * IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. + */ + +#ifndef _acp_ip_OFFSET_HEADER +#define _acp_ip_OFFSET_HEADER +// Registers from ACP_DMA block + +#define mmACP_DMA_CNTL_0 0x1240000 +#define mmACP_DMA_CNTL_1 0x1240004 +#define mmACP_DMA_CNTL_2 0x1240008 +#define mmACP_DMA_CNTL_3 0x124000C +#define mmACP_DMA_CNTL_4 0x1240010 +#define mmACP_DMA_CNTL_5 0x1240014 +#define mmACP_DMA_CNTL_6 0x1240018 +#define mmACP_DMA_CNTL_7 0x124001C +#define mmACP_DMA_DSCR_STRT_IDX_0 0x1240020 +#define mmACP_DMA_DSCR_STRT_IDX_1 0x1240024 +#define mmACP_DMA_DSCR_STRT_IDX_2 0x1240028 +#define mmACP_DMA_DSCR_STRT_IDX_3 0x124002C +#define mmACP_DMA_DSCR_STRT_IDX_4 0x1240030 +#define mmACP_DMA_DSCR_STRT_IDX_5 0x1240034 +#define mmACP_DMA_DSCR_STRT_IDX_6 0x1240038 +#define mmACP_DMA_DSCR_STRT_IDX_7 0x124003C +#define mmACP_DMA_DSCR_CNT_0 0x1240040 +#define mmACP_DMA_DSCR_CNT_1 0x1240044 +#define mmACP_DMA_DSCR_CNT_2 0x1240048 +#define mmACP_DMA_DSCR_CNT_3 0x124004C +#define mmACP_DMA_DSCR_CNT_4 0x1240050 +#define mmACP_DMA_DSCR_CNT_5 0x1240054 +#define mmACP_DMA_DSCR_CNT_6 0x1240058 +#define mmACP_DMA_DSCR_CNT_7 0x124005C +#define mmACP_DMA_PRIO_0 0x1240060 +#define mmACP_DMA_PRIO_1 0x1240064 +#define mmACP_DMA_PRIO_2 0x1240068 +#define mmACP_DMA_PRIO_3 0x124006C +#define mmACP_DMA_PRIO_4 0x1240070 +#define mmACP_DMA_PRIO_5 0x1240074 +#define mmACP_DMA_PRIO_6 0x1240078 +#define mmACP_DMA_PRIO_7 0x124007C +#define mmACP_DMA_CUR_DSCR_0 0x1240080 +#define mmACP_DMA_CUR_DSCR_1 0x1240084 +#define mmACP_DMA_CUR_DSCR_2 0x1240088 +#define mmACP_DMA_CUR_DSCR_3 0x124008C +#define mmACP_DMA_CUR_DSCR_4 0x1240090 +#define mmACP_DMA_CUR_DSCR_5 0x1240094 +#define mmACP_DMA_CUR_DSCR_6 0x1240098 +#define mmACP_DMA_CUR_DSCR_7 0x124009C +#define mmACP_DMA_CUR_TRANS_CNT_0 0x12400A0 +#define mmACP_DMA_CUR_TRANS_CNT_1 0x12400A4 +#define mmACP_DMA_CUR_TRANS_CNT_2 0x12400A8 +#define mmACP_DMA_CUR_TRANS_CNT_3 0x12400AC +#define mmACP_DMA_CUR_TRANS_CNT_4 0x12400B0 +#define mmACP_DMA_CUR_TRANS_CNT_5 0x12400B4 +#define mmACP_DMA_CUR_TRANS_CNT_6 0x12400B8 +#define mmACP_DMA_CUR_TRANS_CNT_7 0x12400BC +#define mmACP_DMA_ERR_STS_0 0x12400C0 +#define mmACP_DMA_ERR_STS_1 0x12400C4 +#define mmACP_DMA_ERR_STS_2 0x12400C8 +#define mmACP_DMA_ERR_STS_3 0x12400CC +#define mmACP_DMA_ERR_STS_4 0x12400D0 +#define mmACP_DMA_ERR_STS_5 0x12400D4 +#define mmACP_DMA_ERR_STS_6 0x12400D8 +#define mmACP_DMA_ERR_STS_7 0x12400DC +#define mmACP_DMA_DESC_BASE_ADDR 0x12400E0 +#define mmACP_DMA_DESC_MAX_NUM_DSCR 0x12400E4 +#define mmACP_DMA_CH_STS 0x12400E8 +#define mmACP_DMA_CH_GROUP 0x12400EC +#define mmACP_DMA_CH_RST_STS 0x12400F0 + + +// Registers from ACP_AXI2AXIATU block + +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_1 0x1240C00 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_1 0x1240C04 +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_2 0x1240C08 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_2 0x1240C0C +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_3 0x1240C10 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_3 0x1240C14 +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_4 0x1240C18 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_4 0x1240C1C +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_5 0x1240C20 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_5 0x1240C24 +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_6 0x1240C28 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_6 0x1240C2C +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_7 0x1240C30 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_7 0x1240C34 +#define mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_8 0x1240C38 +#define mmACPAXI2AXI_ATU_BASE_ADDR_GRP_8 0x1240C3C +#define mmACPAXI2AXI_ATU_CTRL 0x1240C40 + + +// Registers from ACP_CLKRST block + +#define mmACP_SOFT_RESET 0x1241000 +#define mmACP_CONTROL 0x1241004 +#define mmACP_STATUS 0x1241008 +#define mmACP_DSP0_OCD_HALT_ON_RST 0x124100C +#define mmACP_DYNAMIC_CG_MASTER_CONTROL 0x1241010 + + +// Registers from ACP_MISC block + +#define mmACP_EXTERNAL_INTR_ENB 0x1241800 +#define mmACP_EXTERNAL_INTR_CNTL 0x1241804 +#define mmACP_EXTERNAL_INTR_STAT 0x1241808 +#define mmACP_DSP0_INTR_CNTL 0x124180C +#define mmACP_DSP0_INTR_STAT 0x1241810 +#define mmACP_DSP_SW_INTR_CNTL 0x1241814 +#define mmACP_DSP_SW_INTR_STAT 0x1241818 +#define mmACP_SW_INTR_TRIG 0x124181C +#define mmACP_SMU_MAILBOX 0x1241820 +#define mmDSP_INTERRUPT_ROUTING_CTRL 0x1241824 +#define mmACP_DSP0_WATCHDOG_TIMER_CNTL 0x1241828 +#define mmACP_DSP0_EXT_TIMER1_CNTL 0x124182C +#define mmACP_DSP0_EXT_TIMER2_CNTL 0x1241830 +#define mmACP_DSP0_EXT_TIMER3_CNTL 0x1241834 +#define mmACP_DSP0_EXT_TIMER4_CNTL 0x1241838 +#define mmACP_DSP0_EXT_TIMER5_CNTL 0x124183C +#define mmACP_DSP0_EXT_TIMER6_CNTL 0x1241840 +#define mmACP_DSP0_EXT_TIMER1_CURR_VALUE 0x1241844 +#define mmACP_DSP0_EXT_TIMER2_CURR_VALUE 0x1241848 +#define mmACP_DSP0_EXT_TIMER3_CURR_VALUE 0x124184C +#define mmACP_DSP0_EXT_TIMER4_CURR_VALUE 0x1241850 +#define mmACP_DSP0_EXT_TIMER5_CURR_VALUE 0x1241854 +#define mmACP_DSP0_EXT_TIMER6_CURR_VALUE 0x1241858 +#define mmACP_FW_STATUS 0x124185C +#define mmACP_TIMER 0x1241874 +#define mmACP_TIMER_CNTL 0x1241878 +#define mmACP_PGMEM_CTRL 0x12418C0 +#define mmACP_ERROR_STATUS 0x12418C4 +#define mmACP_SW_I2S_ERROR_REASON 0x12418C8 +#define mmACP_MEM_PG_STS 0x12418CC + + +// Registers from ACP_PGFSM block + +#define mmACP_I2S_PIN_CONFIG 0x1241400 +#define mmACP_PAD_PULLUP_PULLDOWN_CTRL 0x1241404 +#define mmACP_PAD_DRIVE_STRENGTH_CTRL 0x1241408 +#define mmACP_SW_PAD_KEEPER_EN 0x124140C +#define mmACP_SW_WAKE_EN 0x1241410 +#define mmACP_I2S_WAKE_EN 0x1241414 +#define mmACP_PME_EN 0x1241418 +#define mmACP_PGFSM_CONTROL 0x124141C +#define mmACP_PGFSM_STATUS 0x1241420 + + +// Registers from ACP_SCRATCH block + +#define mmACP_SCRATCH_REG_0 0x1250000 +#define mmACP_SCRATCH_REG_1 0x1250004 +#define mmACP_SCRATCH_REG_2 0x1250008 +#define mmACP_SCRATCH_REG_3 0x125000C +#define mmACP_SCRATCH_REG_4 0x1250010 +#define mmACP_SCRATCH_REG_5 0x1250014 +#define mmACP_SCRATCH_REG_6 0x1250018 +#define mmACP_SCRATCH_REG_7 0x125001C +#define mmACP_SCRATCH_REG_8 0x1250020 +#define mmACP_SCRATCH_REG_9 0x1250024 +#define mmACP_SCRATCH_REG_10 0x1250028 +#define mmACP_SCRATCH_REG_11 0x125002C +#define mmACP_SCRATCH_REG_12 0x1250030 +#define mmACP_SCRATCH_REG_13 0x1250034 +#define mmACP_SCRATCH_REG_14 0x1250038 +#define mmACP_SCRATCH_REG_15 0x125003C +#define mmACP_SCRATCH_REG_16 0x1250040 +#define mmACP_SCRATCH_REG_17 0x1250044 +#define mmACP_SCRATCH_REG_18 0x1250048 +#define mmACP_SCRATCH_REG_19 0x125004C +#define mmACP_SCRATCH_REG_20 0x1250050 +#define mmACP_SCRATCH_REG_21 0x1250054 +#define mmACP_SCRATCH_REG_22 0x1250058 +#define mmACP_SCRATCH_REG_23 0x125005C +#define mmACP_SCRATCH_REG_24 0x1250060 +#define mmACP_SCRATCH_REG_25 0x1250064 +#define mmACP_SCRATCH_REG_26 0x1250068 +#define mmACP_SCRATCH_REG_27 0x125006C +#define mmACP_SCRATCH_REG_28 0x1250070 +#define mmACP_SCRATCH_REG_29 0x1250074 +#define mmACP_SCRATCH_REG_30 0x1250078 +#define mmACP_SCRATCH_REG_31 0x125007C +#define mmACP_SCRATCH_REG_32 0x1250080 +#define mmACP_SCRATCH_REG_33 0x1250084 +#define mmACP_SCRATCH_REG_34 0x1250088 +#define mmACP_SCRATCH_REG_35 0x125008C +#define mmACP_SCRATCH_REG_36 0x1250090 +#define mmACP_SCRATCH_REG_37 0x1250094 +#define mmACP_SCRATCH_REG_38 0x1250098 +#define mmACP_SCRATCH_REG_39 0x125009C +#define mmACP_SCRATCH_REG_40 0x12500A0 +#define mmACP_SCRATCH_REG_41 0x12500A4 +#define mmACP_SCRATCH_REG_42 0x12500A8 +#define mmACP_SCRATCH_REG_43 0x12500AC +#define mmACP_SCRATCH_REG_44 0x12500B0 +#define mmACP_SCRATCH_REG_45 0x12500B4 +#define mmACP_SCRATCH_REG_46 0x12500B8 +#define mmACP_SCRATCH_REG_47 0x12500BC +#define mmACP_SCRATCH_REG_48 0x12500C0 +#define mmACP_SCRATCH_REG_49 0x12500C4 +#define mmACP_SCRATCH_REG_50 0x12500C8 +#define mmACP_SCRATCH_REG_51 0x12500CC +#define mmACP_SCRATCH_REG_52 0x12500D0 +#define mmACP_SCRATCH_REG_53 0x12500D4 +#define mmACP_SCRATCH_REG_54 0x12500D8 +#define mmACP_SCRATCH_REG_55 0x12500DC +#define mmACP_SCRATCH_REG_56 0x12500E0 +#define mmACP_SCRATCH_REG_57 0x12500E4 +#define mmACP_SCRATCH_REG_58 0x12500E8 +#define mmACP_SCRATCH_REG_59 0x12500EC +#define mmACP_SCRATCH_REG_60 0x12500F0 +#define mmACP_SCRATCH_REG_61 0x12500F4 +#define mmACP_SCRATCH_REG_62 0x12500F8 +#define mmACP_SCRATCH_REG_63 0x12500FC +#define mmACP_SCRATCH_REG_64 0x1250100 +#define mmACP_SCRATCH_REG_65 0x1250104 +#define mmACP_SCRATCH_REG_66 0x1250108 +#define mmACP_SCRATCH_REG_67 0x125010C +#define mmACP_SCRATCH_REG_68 0x1250110 +#define mmACP_SCRATCH_REG_69 0x1250114 +#define mmACP_SCRATCH_REG_70 0x1250118 +#define mmACP_SCRATCH_REG_71 0x125011C +#define mmACP_SCRATCH_REG_72 0x1250120 +#define mmACP_SCRATCH_REG_73 0x1250124 +#define mmACP_SCRATCH_REG_74 0x1250128 +#define mmACP_SCRATCH_REG_75 0x125012C +#define mmACP_SCRATCH_REG_76 0x1250130 +#define mmACP_SCRATCH_REG_77 0x1250134 +#define mmACP_SCRATCH_REG_78 0x1250138 +#define mmACP_SCRATCH_REG_79 0x125013C +#define mmACP_SCRATCH_REG_80 0x1250140 +#define mmACP_SCRATCH_REG_81 0x1250144 +#define mmACP_SCRATCH_REG_82 0x1250148 +#define mmACP_SCRATCH_REG_83 0x125014C +#define mmACP_SCRATCH_REG_84 0x1250150 +#define mmACP_SCRATCH_REG_85 0x1250154 +#define mmACP_SCRATCH_REG_86 0x1250158 +#define mmACP_SCRATCH_REG_87 0x125015C +#define mmACP_SCRATCH_REG_88 0x1250160 +#define mmACP_SCRATCH_REG_89 0x1250164 +#define mmACP_SCRATCH_REG_90 0x1250168 +#define mmACP_SCRATCH_REG_91 0x125016C +#define mmACP_SCRATCH_REG_92 0x1250170 +#define mmACP_SCRATCH_REG_93 0x1250174 +#define mmACP_SCRATCH_REG_94 0x1250178 +#define mmACP_SCRATCH_REG_95 0x125017C +#define mmACP_SCRATCH_REG_96 0x1250180 +#define mmACP_SCRATCH_REG_97 0x1250184 +#define mmACP_SCRATCH_REG_98 0x1250188 +#define mmACP_SCRATCH_REG_99 0x125018C +#define mmACP_SCRATCH_REG_100 0x1250190 +#define mmACP_SCRATCH_REG_101 0x1250194 +#define mmACP_SCRATCH_REG_102 0x1250198 +#define mmACP_SCRATCH_REG_103 0x125019C +#define mmACP_SCRATCH_REG_104 0x12501A0 +#define mmACP_SCRATCH_REG_105 0x12501A4 +#define mmACP_SCRATCH_REG_106 0x12501A8 +#define mmACP_SCRATCH_REG_107 0x12501AC +#define mmACP_SCRATCH_REG_108 0x12501B0 +#define mmACP_SCRATCH_REG_109 0x12501B4 +#define mmACP_SCRATCH_REG_110 0x12501B8 +#define mmACP_SCRATCH_REG_111 0x12501BC +#define mmACP_SCRATCH_REG_112 0x12501C0 +#define mmACP_SCRATCH_REG_113 0x12501C4 +#define mmACP_SCRATCH_REG_114 0x12501C8 +#define mmACP_SCRATCH_REG_115 0x12501CC +#define mmACP_SCRATCH_REG_116 0x12501D0 +#define mmACP_SCRATCH_REG_117 0x12501D4 +#define mmACP_SCRATCH_REG_118 0x12501D8 +#define mmACP_SCRATCH_REG_119 0x12501DC +#define mmACP_SCRATCH_REG_120 0x12501E0 +#define mmACP_SCRATCH_REG_121 0x12501E4 +#define mmACP_SCRATCH_REG_122 0x12501E8 +#define mmACP_SCRATCH_REG_123 0x12501EC +#define mmACP_SCRATCH_REG_124 0x12501F0 +#define mmACP_SCRATCH_REG_125 0x12501F4 +#define mmACP_SCRATCH_REG_126 0x12501F8 +#define mmACP_SCRATCH_REG_127 0x12501FC +#define mmACP_SCRATCH_REG_128 0x1250200 + + +// Registers from ACP_SW_ACLK block + +#define mmSW_CORB_Base_Address 0x1243200 +#define mmSW_CORB_Write_Pointer 0x1243204 +#define mmSW_CORB_Read_Pointer 0x1243208 +#define mmSW_CORB_Control 0x124320C +#define mmSW_CORB_Size 0x1243214 +#define mmSW_RIRB_Base_Address 0x1243218 +#define mmSW_RIRB_Write_Pointer 0x124321C +#define mmSW_RIRB_Response_Interrupt_Count 0x1243220 +#define mmSW_RIRB_Control 0x1243224 +#define mmSW_RIRB_Size 0x1243228 +#define mmSW_RIRB_FIFO_MIN_THDL 0x124322C +#define mmSW_imm_cmd_UPPER_WORD 0x1243230 +#define mmSW_imm_cmd_LOWER_QWORD 0x1243234 +#define mmSW_imm_resp_UPPER_WORD 0x1243238 +#define mmSW_imm_resp_LOWER_QWORD 0x124323C +#define mmSW_imm_cmd_sts 0x1243240 +#define mmSW_BRA_BASE_ADDRESS 0x1243244 +#define mmSW_BRA_TRANSFER_SIZE 0x1243248 +#define mmSW_BRA_DMA_BUSY 0x124324C +#define mmSW_BRA_RESP 0x1243250 +#define mmSW_BRA_RESP_FRAME_ADDR 0x1243254 +#define mmSW_BRA_CURRENT_TRANSFER_SIZE 0x1243258 +#define mmSW_STATE_CHANGE_STATUS_0TO7 0x124325C +#define mmSW_STATE_CHANGE_STATUS_8TO11 0x1243260 +#define mmSW_STATE_CHANGE_STATUS_MASK_0to7 0x1243264 +#define mmSW_STATE_CHANGE_STATUS_MASK_8to11 0x1243268 +#define mmSW_CLK_FREQUENCY_CTRL 0x124326C +#define mmSW_ERROR_INTR_MASK 0x1243270 +#define mmSW_PHY_TEST_MODE_DATA_OFF 0x1243274 + + +// Registers from ACP_SW_SWCLK block + +#define mmACP_SW_EN 0x1243000 +#define mmACP_SW_EN_STATUS 0x1243004 +#define mmACP_SW_FRAMESIZE 0x1243008 +#define mmACP_SW_SSP_Counter 0x124300C +#define mmACP_SW_Audio_TX_EN 0x1243010 +#define mmACP_SW_Audio_TX_EN_STATUS 0x1243014 +#define mmACP_SW_Audio_TX_Frame_Format 0x1243018 +#define mmACP_SW_Audio_TX_SampleInterval 0x124301C +#define mmACP_SW_Audio_TX_Hctrl_DP0 0x1243020 +#define mmACP_SW_Audio_TX_Hctrl_DP1 0x1243024 +#define mmACP_SW_Audio_TX_Hctrl_DP2 0x1243028 +#define mmACP_SW_Audio_TX_Hctrl_DP3 0x124302C +#define mmACP_SW_Audio_TX_offset_DP0 0x1243030 +#define mmACP_SW_Audio_TX_offset_DP1 0x1243034 +#define mmACP_SW_Audio_TX_offset_DP2 0x1243038 +#define mmACP_SW_Audio_TX_offset_DP3 0x124303C +#define mmACP_SW_Audio_TX_Channel_Enable_DP0 0x1243040 +#define mmACP_SW_Audio_TX_Channel_Enable_DP1 0x1243044 +#define mmACP_SW_Audio_TX_Channel_Enable_DP2 0x1243048 +#define mmACP_SW_Audio_TX_Channel_Enable_DP3 0x124304C +#define mmACP_SW_BT_TX_EN 0x1243050 +#define mmACP_SW_BT_TX_EN_STATUS 0x1243054 +#define mmACP_SW_BT_TX_Frame_Format 0x1243058 +#define mmACP_SW_BT_TX_SampleInterval 0x124305C +#define mmACP_SW_BT_TX_Hctrl 0x1243060 +#define mmACP_SW_BT_TX_offset 0x1243064 +#define mmACP_SW_BT_TX_Channel_Enable_DP0 0x1243068 +#define mmACP_SW_Headset_TX_EN 0x124306C +#define mmACP_SW_Headset_TX_EN_STATUS 0x1243070 +#define mmACP_SW_Headset_TX_Frame_Format 0x1243074 +#define mmACP_SW_Headset_TX_SampleInterval 0x1243078 +#define mmACP_SW_Headset_TX_Hctrl 0x124307C +#define mmACP_SW_Headset_TX_offset 0x1243080 +#define mmACP_SW_Headset_TX_Channel_Enable_DP0 0x1243084 +#define mmACP_SW_Audio_RX_EN 0x1243088 +#define mmACP_SW_Audio_RX_EN_STATUS 0x124308C +#define mmACP_SW_Audio_RX_Frame_Format 0x1243090 +#define mmACP_SW_Audio_RX_SampleInterval 0x1243094 +#define mmACP_SW_Audio_RX_Hctrl_DP0 0x1243098 +#define mmACP_SW_Audio_RX_Hctrl_DP1 0x124309C +#define mmACP_SW_Audio_RX_Hctrl_DP2 0x1243100 +#define mmACP_SW_Audio_RX_Hctrl_DP3 0x1243104 +#define mmACP_SW_Audio_RX_offset_DP0 0x1243108 +#define mmACP_SW_Audio_RX_offset_DP1 0x124310C +#define mmACP_SW_Audio_RX_offset_DP2 0x1243110 +#define mmACP_SW_Audio_RX_offset_DP3 0x1243114 +#define mmACP_SW_Audio_RX_Channel_Enable_DP0 0x1243118 +#define mmACP_SW_Audio_RX_Channel_Enable_DP1 0x124311C +#define mmACP_SW_Audio_RX_Channel_Enable_DP2 0x1243120 +#define mmACP_SW_Audio_RX_Channel_Enable_DP3 0x1243124 +#define mmACP_SW_BT_RX_EN 0x1243128 +#define mmACP_SW_BT_RX_EN_STATUS 0x124312C +#define mmACP_SW_BT_RX_Frame_Format 0x1243130 +#define mmACP_SW_BT_RX_SampleInterval 0x1243134 +#define mmACP_SW_BT_RX_Hctrl 0x1243138 +#define mmACP_SW_BT_RX_offset 0x124313C +#define mmACP_SW_BT_RX_Channel_Enable_DP0 0x1243140 +#define mmACP_SW_Headset_RX_EN 0x1243144 +#define mmACP_SW_Headset_RX_EN_STATUS 0x1243148 +#define mmACP_SW_Headset_RX_Frame_Format 0x124314C +#define mmACP_SW_Headset_RX_SampleInterval 0x1243150 +#define mmACP_SW_Headset_RX_Hctrl 0x1243154 +#define mmACP_SW_Headset_RX_offset 0x1243158 +#define mmACP_SW_Headset_RX_Channel_Enable_DP0 0x124315C +#define mmACP_SW_BPT_PORT_EN 0x1243160 +#define mmACP_SW_BPT_PORT_EN_STATUS 0x1243164 +#define mmACP_SW_BPT_PORT_Frame_Format 0x1243168 +#define mmACP_SW_BPT_PORT_SampleInterval 0x124316C +#define mmACP_SW_BPT_PORT_Hctrl 0x1243170 +#define mmACP_SW_BPT_PORT_offset 0x1243174 +#define mmACP_SW_BPT_PORT_Channel_Enable 0x1243178 +#define mmACP_SW_BPT_PORT_First_byte_addr 0x124317C +#define mmACP_SW_CLK_RESUME_CTRL 0x1243180 +#define mmACP_SW_CLK_RESUME_Delay_Cntr 0x1243184 +#define mmACP_SW_BUS_RESET_CTRL 0x1243188 +#define mmACP_SW_PRBS_ERR_STATUS 0x124318C + + +// Registers from ACP_AUDIO_BUFFERS block + +#define mmACP_I2S_RX_RINGBUFADDR 0x1242000 +#define mmACP_I2S_RX_RINGBUFSIZE 0x1242004 +#define mmACP_I2S_RX_LINKPOSITIONCNTR 0x1242008 +#define mmACP_I2S_RX_FIFOADDR 0x124200C +#define mmACP_I2S_RX_FIFOSIZE 0x1242010 +#define mmACP_I2S_RX_DMA_SIZE 0x1242014 +#define mmACP_I2S_RX_LINEARPOSITIONCNTR_HIGH 0x1242018 +#define mmACP_I2S_RX_LINEARPOSITIONCNTR_LOW 0x124201C +#define mmACP_I2S_RX_INTR_WATERMARK_SIZE 0x1242020 +#define mmACP_I2S_TX_RINGBUFADDR 0x1242024 +#define mmACP_I2S_TX_RINGBUFSIZE 0x1242028 +#define mmACP_I2S_TX_LINKPOSITIONCNTR 0x124202C +#define mmACP_I2S_TX_FIFOADDR 0x1242030 +#define mmACP_I2S_TX_FIFOSIZE 0x1242034 +#define mmACP_I2S_TX_DMA_SIZE 0x1242038 +#define mmACP_I2S_TX_LINEARPOSITIONCNTR_HIGH 0x124203C +#define mmACP_I2S_TX_LINEARPOSITIONCNTR_LOW 0x1242040 +#define mmACP_I2S_TX_INTR_WATERMARK_SIZE 0x1242044 +#define mmACP_BT_RX_RINGBUFADDR 0x1242048 +#define mmACP_BT_RX_RINGBUFSIZE 0x124204C +#define mmACP_BT_RX_LINKPOSITIONCNTR 0x1242050 +#define mmACP_BT_RX_FIFOADDR 0x1242054 +#define mmACP_BT_RX_FIFOSIZE 0x1242058 +#define mmACP_BT_RX_DMA_SIZE 0x124205C +#define mmACP_BT_RX_LINEARPOSITIONCNTR_HIGH 0x1242060 +#define mmACP_BT_RX_LINEARPOSITIONCNTR_LOW 0x1242064 +#define mmACP_BT_RX_INTR_WATERMARK_SIZE 0x1242068 +#define mmACP_BT_TX_RINGBUFADDR 0x124206C +#define mmACP_BT_TX_RINGBUFSIZE 0x1242070 +#define mmACP_BT_TX_LINKPOSITIONCNTR 0x1242074 +#define mmACP_BT_TX_FIFOADDR 0x1242078 +#define mmACP_BT_TX_FIFOSIZE 0x124207C +#define mmACP_BT_TX_DMA_SIZE 0x1242080 +#define mmACP_BT_TX_LINEARPOSITIONCNTR_HIGH 0x1242084 +#define mmACP_BT_TX_LINEARPOSITIONCNTR_LOW 0x1242088 +#define mmACP_BT_TX_INTR_WATERMARK_SIZE 0x124208C +#define mmACP_HS_RX_RINGBUFADDR 0x1242090 +#define mmACP_HS_RX_RINGBUFSIZE 0x1242094 +#define mmACP_HS_RX_LINKPOSITIONCNTR 0x1242098 +#define mmACP_HS_RX_FIFOADDR 0x124209C +#define mmACP_HS_RX_FIFOSIZE 0x12420A0 +#define mmACP_HS_RX_DMA_SIZE 0x12420A4 +#define mmACP_HS_RX_LINEARPOSITIONCNTR_HIGH 0x12420A8 +#define mmACP_HS_RX_LINEARPOSITIONCNTR_LOW 0x12420AC +#define mmACP_HS_RX_INTR_WATERMARK_SIZE 0x12420B0 +#define mmACP_HS_TX_RINGBUFADDR 0x12420B4 +#define mmACP_HS_TX_RINGBUFSIZE 0x12420B8 +#define mmACP_HS_TX_LINKPOSITIONCNTR 0x12420BC +#define mmACP_HS_TX_FIFOADDR 0x12420C0 +#define mmACP_HS_TX_FIFOSIZE 0x12420C4 +#define mmACP_HS_TX_DMA_SIZE 0x12420C8 +#define mmACP_HS_TX_LINEARPOSITIONCNTR_HIGH 0x12420CC +#define mmACP_HS_TX_LINEARPOSITIONCNTR_LOW 0x12420D0 +#define mmACP_HS_TX_INTR_WATERMARK_SIZE 0x12420D4 + + +// Registers from ACP_I2S_TDM block + +#define mmACP_I2STDM_IER 0x1242400 +#define mmACP_I2STDM_IRER 0x1242404 +#define mmACP_I2STDM_RXFRMT 0x1242408 +#define mmACP_I2STDM_ITER 0x124240C +#define mmACP_I2STDM_TXFRMT 0x1242410 + + +// Registers from ACP_BT_TDM block + +#define mmACP_BTTDM_IER 0x1242800 +#define mmACP_BTTDM_IRER 0x1242804 +#define mmACP_BTTDM_RXFRMT 0x1242808 +#define mmACP_BTTDM_ITER 0x124280C +#define mmACP_BTTDM_TXFRMT 0x1242810 + + +// Registers from AZALIA_IP block + +#define mmAudio_Az_Global_Capabilities 0x1200000 +#define mmAudio_Az_Minor_Version 0x1200002 +#define mmAudio_Az_Major_Version 0x1200003 +#define mmAudio_Az_Output_Payload_Capability 0x1200004 +#define mmAudio_Az_Input_Payload_Capability 0x1200006 +#define mmAudio_Az_Global_Control 0x1200008 +#define mmAudio_Az_Wake_Enable 0x120000C +#define mmAudio_Az_State_Change_Status 0x120000E +#define mmAudio_Az_Global_Status 0x1200010 +#define mmAudio_Az_Linked_List_Capability_Header 0x1200014 +#define mmAudio_Az_Output_Stream_Payload_Capability 0x1200018 +#define mmAudio_Az_Input_Stream_Payload_Capability 0x120001A +#define mmAudio_Az_Interrupt_Control 0x1200020 +#define mmAudio_Az_Interrupt_Status 0x1200024 +#define mmAudio_Az_Wall_Clock_Counter 0x1200030 +#define mmAudio_Az_Stream_Synchronization 0x1200038 +#define mmAudio_Az_CORB_Lower_Base_Address 0x1200040 +#define mmAudio_Az_CORB_Upper_Base_Address 0x1200044 +#define mmAudio_Az_CORB_Write_Pointer 0x1200048 +#define mmAudio_Az_CORB_Read_Pointer 0x120004A +#define mmAudio_Az_CORB_Control 0x120004C +#define mmAudio_Az_CORB_Status 0x120004D +#define mmAudio_Az_CORB_Size 0x120004E +#define mmAudio_Az_RIRB_Lower_Base_Address 0x1200050 +#define mmAudio_Az_RIRB_Upper_Base_Address 0x1200054 +#define mmAudio_Az_RIRB_Write_Pointer 0x1200058 +#define mmAudio_Az_RIRB_Response_Interrupt_Count 0x120005A +#define mmAudio_Az_RIRB_Control 0x120005C +#define mmAudio_Az_RIRB_Status 0x120005D +#define mmAudio_Az_RIRB_Size 0x120005E +#define mmAudio_Az_Immediate_Command_Output_Interface 0x1200060 +#define mmAudio_Az_Immediate_Response_Input_Interface 0x1200064 +#define mmAudio_Az_Immediate_Command_Status 0x1200068 +#define mmAudio_Az_DPLBASE 0x1200070 +#define mmAudio_Az_DPUBASE 0x1200074 +#define mmAudio_Az_Input_SD0CTL_and_STS 0x1200080 +#define mmAudio_Az_Input_SD0LPIB 0x1200084 +#define mmAudio_Az_Input_SD0CBL 0x1200088 +#define mmAudio_Az_Input_SD0LVI 0x120008C +#define mmAudio_Az_Input_SD0FIFOS 0x1200090 +#define mmAudio_Az_Input_SD0FMT 0x1200092 +#define mmAudio_Az_Input_SD0BDPL 0x1200098 +#define mmAudio_Az_Input_SD0BDPU 0x120009C +#define mmAudio_Az_Input_SD1CTL_and_STS 0x12000A0 +#define mmAudio_Az_Input_SD1LPIB 0x12000A4 +#define mmAudio_Az_Input_SD1CBL 0x12000A8 +#define mmAudio_Az_Input_SD1LVI 0x12000AC +#define mmAudio_Az_Input_SD1FIFOS 0x12000B0 +#define mmAudio_Az_Input_SD1FMT 0x12000B2 +#define mmAudio_Az_Input_SD1BDPL 0x12000B8 +#define mmAudio_Az_Input_SD1BDPU 0x12000BC +#define mmAudio_Az_Input_SD2CTL_and_STS 0x12000C0 +#define mmAudio_Az_Input_SD2LPIB 0x12000C4 +#define mmAudio_Az_Input_SD2CBL 0x12000C8 +#define mmAudio_Az_Input_SD2LVI 0x12000CC +#define mmAudio_Az_Input_SD2FIFOS 0x12000D0 +#define mmAudio_Az_Input_SD2FMT 0x12000D2 +#define mmAudio_Az_Input_SD2BDPL 0x12000D8 +#define mmAudio_Az_Input_SD2BDPU 0x12000DC +#define mmAudio_Az_Input_SD3CTL_and_STS 0x12000E0 +#define mmAudio_Az_Input_SD3LPIB 0x12000E4 +#define mmAudio_Az_Input_SD3CBL 0x12000E8 +#define mmAudio_Az_Input_SD3LVI 0x12000EC +#define mmAudio_Az_Input_SD3FIFOS 0x12000F0 +#define mmAudio_Az_Input_SD3FMT 0x12000F2 +#define mmAudio_Az_Input_SD3BDPL 0x12000F8 +#define mmAudio_Az_Input_SD3BDPU 0x12000FC +#define mmAudio_Az_Output_SD0CTL_and_STS 0x1200100 +#define mmAudio_Az_Output_SD0LPIB 0x1200104 +#define mmAudio_Az_Output_SD0CBL 0x1200108 +#define mmAudio_Az_Output_SD0LVI 0x120010C +#define mmAudio_Az_Output_SD0FIFOS 0x1200110 +#define mmAudio_Az_Output_SD0FMT 0x1200112 +#define mmAudio_Az_Output_SD0BDPL 0x1200118 +#define mmAudio_Az_Output_SD0BDPU 0x120011C +#define mmAudio_Az_Output_SD1CTL_and_STS 0x1200120 +#define mmAudio_Az_Output_SD1LPIB 0x1200124 +#define mmAudio_Az_Output_SD1CBL 0x1200128 +#define mmAudio_Az_Output_SD1LVI 0x120012C +#define mmAudio_Az_Output_SD1FIFOS 0x1200130 +#define mmAudio_Az_Output_SD1FMT 0x1200132 +#define mmAudio_Az_Output_SD1BDPL 0x1200138 +#define mmAudio_Az_Output_SD1BDPU 0x120013C +#define mmAudio_Az_Output_SD2CTL_and_STS 0x1200140 +#define mmAudio_Az_Output_SD2LPIB 0x1200144 +#define mmAudio_Az_Output_SD2CBL 0x1200148 +#define mmAudio_Az_Output_SD2LVI 0x120014C +#define mmAudio_Az_Output_SD2FIFOS 0x1200150 +#define mmAudio_Az_Output_SD2FMT 0x1200152 +#define mmAudio_Az_Output_SD2BDPL 0x1200158 +#define mmAudio_Az_Output_SD2BDPU 0x120015C +#define mmAudio_Az_Output_SD3CTL_and_STS 0x1200160 +#define mmAudio_Az_Output_SD3LPIB 0x1200164 +#define mmAudio_Az_Output_SD3CBL 0x1200168 +#define mmAudio_Az_Output_SD3LVI 0x120016C +#define mmAudio_Az_Output_SD3FIFOS 0x1200170 +#define mmAudio_Az_Output_SD3FMT 0x1200172 +#define mmAudio_Az_Output_SD3BDPL 0x1200178 +#define mmAudio_Az_Output_SD3BDPU 0x120017C +#define mmAudioAZ_Misc_Control_Register_1 0x1200180 +#define mmAudioAZ_Misc_Control_Register_2 0x1200182 +#define mmAudioAZ_Misc_Control_Register_3 0x1200183 +#define mmAudio_AZ_Multiple_Links_Capability_Header 0x1200200 +#define mmAudio_AZ_Multiple_Links_Capability_Declaration 0x1200204 +#define mmAudio_AZ_Link0_Capabilities 0x1200240 +#define mmAudio_AZ_Link0_Control 0x1200244 +#define mmAudio_AZ_Link0_Output_Stream_ID 0x1200248 +#define mmAudio_AZ_Link0_SDI_Identifier 0x120024C +#define mmAudio_AZ_Link0_Per_Stream_Overhead 0x1200250 +#define mmAudio_AZ_Link0_Wall_Frame_Counter 0x1200258 +#define mmAudio_AZ_Link0_Output_Payload_Capability_L 0x1200260 +#define mmAudio_AZ_Link0_Output_Payload_Capability_U 0x1200264 +#define mmAudio_AZ_Link0_Input_Payload_Capability_L 0x1200270 +#define mmAudio_AZ_Link0_Input_Payload_Capability_U 0x1200274 +#define mmAudio_Az_Input_SD0LICBA 0x1202084 +#define mmAudio_Az_Input_SD1LICBA 0x12020A4 +#define mmAudio_Az_Input_SD2LICBA 0x12020C4 +#define mmAudio_Az_Input_SD3LICBA 0x12020E4 +#define mmAudio_Az_Output_SD0LICBA 0x1202104 +#define mmAudio_Az_Output_SD1LICBA 0x1202124 +#define mmAudio_Az_Output_SD2LICBA 0x1202144 +#define mmAudio_Az_Output_SD3LICBA 0x1202164 +#define mmAUDIO_AZ_POWER_MANAGEMENT_CONTROL 0x1204000 +#define mmAUDIO_AZ_IOC_SOFTRST_CONTROL 0x1204004 +#define mmAUDIO_AZ_IOC_CLKGATE_CONTROL 0x1204008 + + +// Registers from ACP_AZALIA block + +#define mmACP_AZ_PAGE0_LBASE_ADDR 0x1243800 +#define mmACP_AZ_PAGE0_UBASE_ADDR 0x1243804 +#define mmACP_AZ_PAGE0_PGEN_SIZE 0x1243808 +#define mmACP_AZ_PAGE0_OFFSET 0x124380C +#define mmACP_AZ_PAGE1_LBASE_ADDR 0x1243810 +#define mmACP_AZ_PAGE1_UBASE_ADDR 0x1243814 +#define mmACP_AZ_PAGE1_PGEN_SIZE 0x1243818 +#define mmACP_AZ_PAGE1_OFFSET 0x124381C +#define mmACP_AZ_PAGE2_LBASE_ADDR 0x1243820 +#define mmACP_AZ_PAGE2_UBASE_ADDR 0x1243824 +#define mmACP_AZ_PAGE2_PGEN_SIZE 0x1243828 +#define mmACP_AZ_PAGE2_OFFSET 0x124382C +#define mmACP_AZ_PAGE3_LBASE_ADDR 0x1243830 +#define mmACP_AZ_PAGE3_UBASE_ADDR 0x1243834 +#define mmACP_AZ_PAGE3_PGEN_SIZE 0x1243838 +#define mmACP_AZ_PAGE3_OFFSET 0x124383C +#define mmACP_AZ_PAGE4_LBASE_ADDR 0x1243840 +#define mmACP_AZ_PAGE4_UBASE_ADDR 0x1243844 +#define mmACP_AZ_PAGE4_PGEN_SIZE 0x1243848 +#define mmACP_AZ_PAGE4_OFFSET 0x124384C +#define mmACP_AZ_PAGE5_LBASE_ADDR 0x1243850 +#define mmACP_AZ_PAGE5_UBASE_ADDR 0x1243854 +#define mmACP_AZ_PAGE5_PGEN_SIZE 0x1243858 +#define mmACP_AZ_PAGE5_OFFSET 0x124385C +#define mmACP_AZ_PAGE6_LBASE_ADDR 0x1243860 +#define mmACP_AZ_PAGE6_UBASE_ADDR 0x1243864 +#define mmACP_AZ_PAGE6_PGEN_SIZE 0x1243868 +#define mmACP_AZ_PAGE6_OFFSET 0x124386C +#define mmACP_AZ_PAGE7_LBASE_ADDR 0x1243870 +#define mmACP_AZ_PAGE7_UBASE_ADDR 0x1243874 +#define mmACP_AZ_PAGE7_PGEN_SIZE 0x1243878 +#define mmACP_AZ_PAGE7_OFFSET 0x124387C + + +#endif -- 2.7.4

3 3

[alsa-devel] [PATCH v6 3/3] ALSA: hda: add support for Huawei WMI micmute LED
by Ayman Bagabas 14 Nov '18

14 Nov '18

Some of Huawei laptops come with a LED in the micmute key. This patch enables the use of micmute LED for these devices: 1. Matebook X (19e5:3200), (19e5:3201) 2. Matebook X Pro (19e5:3204) Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/huawei_wmi_helper.c | 47 +++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 14 ++++++++- 2 files changed, 60 insertions(+), 1 deletion(-) create mode 100644 sound/pci/hda/huawei_wmi_helper.c diff --git a/sound/pci/hda/huawei_wmi_helper.c b/sound/pci/hda/huawei_wmi_helper.c new file mode 100644 index 000000000000..7c91f914ffba --- /dev/null +++ b/sound/pci/hda/huawei_wmi_helper.c @@ -0,0 +1,47 @@ +// SPDX-License-Identifier: GPL-2.0 +/* Helper functions for Huawei WMI micmute LED; + * to be included from codec driver + */ + +#if IS_ENABLED(CONFIG_HUAWEI_WMI) +#include <linux/platform_data/x86/huawei-wmi.h> + +static int (*huawei_wmi_micmute_led_set_func)(bool); + +static void update_huawei_wmi_micmute_led(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + huawei_wmi_micmute_led_set_func(spec->micmute_led.led_value); +} + +static void alc_fixup_huawei_wmi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + bool removefunc = false; + + if (action == HDA_FIXUP_ACT_PROBE) { + if (!huawei_wmi_micmute_led_set_func) + huawei_wmi_micmute_led_set_func = symbol_request(huawei_wmi_micmute_led_set); + if (!huawei_wmi_micmute_led_set_func) { + codec_warn(codec, "Failed to find huawei_wmi symbol huawei_wmi_micmute_led_set\n"); + return; + } + + removefunc = (huawei_wmi_micmute_led_set_func(false) < 0) + || (snd_hda_gen_add_micmute_led(codec, update_huawei_wmi_micmute_led) < 0); + } + + if (huawei_wmi_micmute_led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { + symbol_put(huawei_wmi_micmute_led_set); + huawei_wmi_micmute_led_set_func = NULL; + } +} + +#else + +static void alc_fixup_huawei_wmi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ +} + +#endif diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 135c0826859b..5431dee40f4f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5374,6 +5374,9 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" +/* for alc_fixup_huawei_wmi */ +#include "huawei_wmi_helper.c" + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -5494,6 +5497,7 @@ enum { ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, ALC256_FIXUP_HUAWEI_MBXP_PINS, + ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, }; @@ -5765,6 +5769,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_huawei_wmi + }, [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5779,7 +5787,9 @@ static const struct hda_fixup alc269_fixups[] = { {0x1e, 0x411111f0}, {0x21, 0x04211020}, { } - } + }, + .chained = true, + .chain_id = ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, @@ -6611,6 +6621,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x19e5, 0x3200, "Huawei MBX", ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED), + SND_PCI_QUIRK(0x19e5, 0x3201, "Huawei MBX", ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- 2.17.2

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[alsa-devel] [PATCH v6 2/3] ALSA: hda: fix front speakers on Huawei MBXP.
by Ayman Bagabas 14 Nov '18

14 Nov '18

This patch solves bug 200501 'Only 2 of 4 speakers playing sound.' https://bugzilla.kernel.org/show_bug.cgi?id=200501 It enables the front speakers on Huawei Matebook X Pro laptops. These laptops come with Dolby Atmos sound system and these pins configuration enables the front speakers. Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fa61674a5605..135c0826859b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5493,6 +5493,7 @@ enum { ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, + ALC256_FIXUP_HUAWEI_MBXP_PINS, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, }; @@ -5764,6 +5765,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x12, 0x90a60130}, + {0x13, 0x40000000}, + {0x14, 0x90170110}, + {0x18, 0x411111f0}, + {0x19, 0x04a11040}, + {0x1a, 0x411111f0}, + {0x1b, 0x90170112}, + {0x1d, 0x40759a05}, + {0x1e, 0x411111f0}, + {0x21, 0x04211020}, + { } + } + }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_x101_headset_mic, @@ -6594,6 +6611,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ #if 0 -- 2.17.2

1 0

[alsa-devel] [asoc:for-4.21 60/62] sound/soc/amd/raven/pci-acp3x.c:59:8: error: implicit declaration of function 'pci_enable_msi'
by kbuild test robot 14 Nov '18

14 Nov '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.21 head: 9d4b17efa51f3bb0918021090c09b0013039df00 commit: 818ba93b0e9ed4a08cba575dd49c1391ef4b7776 [60/62] ASoC: amd: enable acp3x drivers build config: openrisc-allmodconfig (attached as .config) compiler: or1k-linux-gcc (GCC) 6.0.0 20160327 (experimental) reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 818ba93b0e9ed4a08cba575dd49c1391ef4b7776 # save the attached .config to linux build tree make.cross ARCH=openrisc All errors (new ones prefixed by >>): sound/soc/amd/raven/pci-acp3x.c: In function 'snd_acp3x_probe': >> sound/soc/amd/raven/pci-acp3x.c:59:8: error: implicit declaration of function 'pci_enable_msi' [-Werror=implicit-function-declaration] ret = pci_enable_msi(pci); ^~~~~~~~~~~~~~ >> sound/soc/amd/raven/pci-acp3x.c:124:2: error: implicit declaration of function 'pci_disable_msi' [-Werror=implicit-function-declaration] pci_disable_msi(pci); ^~~~~~~~~~~~~~~ sound/soc/amd/raven/pci-acp3x.c: At top level: sound/soc/amd/raven/pci-acp3x.c:161:1: warning: data definition has no type or storage class module_pci_driver(acp3x_driver); ^~~~~~~~~~~~~~~~~ sound/soc/amd/raven/pci-acp3x.c:161:1: error: type defaults to 'int' in declaration of 'module_pci_driver' [-Werror=implicit-int] sound/soc/amd/raven/pci-acp3x.c:161:1: warning: parameter names (without types) in function declaration sound/soc/amd/raven/pci-acp3x.c:154:26: warning: 'acp3x_driver' defined but not used [-Wunused-variable] static struct pci_driver acp3x_driver = { ^~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/pci_enable_msi +59 sound/soc/amd/raven/pci-acp3x.c e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 30 e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 31 static int snd_acp3x_probe(struct pci_dev *pci, e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 32 const struct pci_device_id *pci_id) e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 33 { e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 34 int ret; 7894a7e7 Vijendar Mukunda 2018-11-12 35 u32 addr, val; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 36 struct acp3x_dev_data *adata; 7894a7e7 Vijendar Mukunda 2018-11-12 37 struct platform_device_info pdevinfo; 7894a7e7 Vijendar Mukunda 2018-11-12 38 unsigned int irqflags; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 39 e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 40 if (pci_enable_device(pci)) { e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 41 dev_err(&pci->dev, "pci_enable_device failed\n"); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 42 return -ENODEV; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 43 } e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 44 e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 45 ret = pci_request_regions(pci, "AMD ACP3x audio"); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 46 if (ret < 0) { e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 47 dev_err(&pci->dev, "pci_request_regions failed\n"); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 48 goto disable_pci; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 49 } e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 50 e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 51 adata = devm_kzalloc(&pci->dev, sizeof(struct acp3x_dev_data), e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 52 GFP_KERNEL); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 53 if (!adata) { e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 54 ret = -ENOMEM; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 55 goto release_regions; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 56 } e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 57 7894a7e7 Vijendar Mukunda 2018-11-12 58 /* check for msi interrupt support */ 7894a7e7 Vijendar Mukunda 2018-11-12 @59 ret = pci_enable_msi(pci); 7894a7e7 Vijendar Mukunda 2018-11-12 60 if (ret) 7894a7e7 Vijendar Mukunda 2018-11-12 61 /* msi is not enabled */ 7894a7e7 Vijendar Mukunda 2018-11-12 62 irqflags = IRQF_SHARED; 7894a7e7 Vijendar Mukunda 2018-11-12 63 else 7894a7e7 Vijendar Mukunda 2018-11-12 64 /* msi is enabled */ 7894a7e7 Vijendar Mukunda 2018-11-12 65 irqflags = 0; 7894a7e7 Vijendar Mukunda 2018-11-12 66 e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 67 addr = pci_resource_start(pci, 0); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 68 adata->acp3x_base = ioremap(addr, pci_resource_len(pci, 0)); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 69 if (!adata->acp3x_base) { e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 70 ret = -ENOMEM; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 71 goto release_regions; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 72 } e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 73 pci_set_master(pci); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 74 pci_set_drvdata(pci, adata); 7894a7e7 Vijendar Mukunda 2018-11-12 75 7894a7e7 Vijendar Mukunda 2018-11-12 76 val = rv_readl(adata->acp3x_base + mmACP_I2S_PIN_CONFIG); 7894a7e7 Vijendar Mukunda 2018-11-12 77 switch (val) { 7894a7e7 Vijendar Mukunda 2018-11-12 78 case I2S_MODE: 7894a7e7 Vijendar Mukunda 2018-11-12 79 adata->res = devm_kzalloc(&pci->dev, 7894a7e7 Vijendar Mukunda 2018-11-12 80 sizeof(struct resource) * 2, 7894a7e7 Vijendar Mukunda 2018-11-12 81 GFP_KERNEL); 7894a7e7 Vijendar Mukunda 2018-11-12 82 if (!adata->res) { 7894a7e7 Vijendar Mukunda 2018-11-12 83 ret = -ENOMEM; 7894a7e7 Vijendar Mukunda 2018-11-12 84 goto unmap_mmio; 7894a7e7 Vijendar Mukunda 2018-11-12 85 } 7894a7e7 Vijendar Mukunda 2018-11-12 86 7894a7e7 Vijendar Mukunda 2018-11-12 87 adata->res[0].name = "acp3x_i2s_iomem"; 7894a7e7 Vijendar Mukunda 2018-11-12 88 adata->res[0].flags = IORESOURCE_MEM; 7894a7e7 Vijendar Mukunda 2018-11-12 89 adata->res[0].start = addr; 7894a7e7 Vijendar Mukunda 2018-11-12 90 adata->res[0].end = addr + (ACP3x_REG_END - ACP3x_REG_START); 7894a7e7 Vijendar Mukunda 2018-11-12 91 7894a7e7 Vijendar Mukunda 2018-11-12 92 adata->res[1].name = "acp3x_i2s_irq"; 7894a7e7 Vijendar Mukunda 2018-11-12 93 adata->res[1].flags = IORESOURCE_IRQ; 7894a7e7 Vijendar Mukunda 2018-11-12 94 adata->res[1].start = pci->irq; 7894a7e7 Vijendar Mukunda 2018-11-12 95 adata->res[1].end = pci->irq; 7894a7e7 Vijendar Mukunda 2018-11-12 96 7894a7e7 Vijendar Mukunda 2018-11-12 97 adata->acp3x_audio_mode = ACP3x_I2S_MODE; 7894a7e7 Vijendar Mukunda 2018-11-12 98 7894a7e7 Vijendar Mukunda 2018-11-12 99 memset(&pdevinfo, 0, sizeof(pdevinfo)); 7894a7e7 Vijendar Mukunda 2018-11-12 100 pdevinfo.name = "acp3x_rv_i2s"; 7894a7e7 Vijendar Mukunda 2018-11-12 101 pdevinfo.id = 0; 7894a7e7 Vijendar Mukunda 2018-11-12 102 pdevinfo.parent = &pci->dev; 7894a7e7 Vijendar Mukunda 2018-11-12 103 pdevinfo.num_res = 2; 7894a7e7 Vijendar Mukunda 2018-11-12 104 pdevinfo.res = adata->res; 7894a7e7 Vijendar Mukunda 2018-11-12 105 pdevinfo.data = &irqflags; 7894a7e7 Vijendar Mukunda 2018-11-12 106 pdevinfo.size_data = sizeof(irqflags); 7894a7e7 Vijendar Mukunda 2018-11-12 107 7894a7e7 Vijendar Mukunda 2018-11-12 108 adata->pdev = platform_device_register_full(&pdevinfo); 7894a7e7 Vijendar Mukunda 2018-11-12 109 if (!adata->pdev) { 7894a7e7 Vijendar Mukunda 2018-11-12 110 dev_err(&pci->dev, "cannot register %s device\n", 7894a7e7 Vijendar Mukunda 2018-11-12 111 pdevinfo.name); 7894a7e7 Vijendar Mukunda 2018-11-12 112 ret = -ENODEV; 7894a7e7 Vijendar Mukunda 2018-11-12 113 goto unmap_mmio; 7894a7e7 Vijendar Mukunda 2018-11-12 114 } 7894a7e7 Vijendar Mukunda 2018-11-12 115 break; 7894a7e7 Vijendar Mukunda 2018-11-12 116 default: 7894a7e7 Vijendar Mukunda 2018-11-12 117 dev_err(&pci->dev, "Inavlid ACP audio mode : %d\n", val); 7894a7e7 Vijendar Mukunda 2018-11-12 118 ret = -ENODEV; 7894a7e7 Vijendar Mukunda 2018-11-12 119 goto unmap_mmio; 7894a7e7 Vijendar Mukunda 2018-11-12 120 } e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 121 return 0; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 122 7894a7e7 Vijendar Mukunda 2018-11-12 123 unmap_mmio: 7894a7e7 Vijendar Mukunda 2018-11-12 @124 pci_disable_msi(pci); 7894a7e7 Vijendar Mukunda 2018-11-12 125 iounmap(adata->acp3x_base); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 126 release_regions: e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 127 pci_release_regions(pci); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 128 disable_pci: e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 129 pci_disable_device(pci); e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 130 e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 131 return ret; e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 132 } e30d9128 Maruthi Srinivas Bayyavarapu 2018-11-12 133 :::::: The code at line 59 was first introduced by commit :::::: 7894a7e7ea3de6c1fd5b2b43f3a73106c94ef584 ASoC: amd: create ACP3x PCM platform device :::::: TO: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH v5 3/3] ALSA: hda: add support for Huawei WMI micmute LED
by Ayman Bagabas 14 Nov '18

14 Nov '18

Some of Huawei laptops come with a LED in the micmute key. This patch enables and disable this LED accordingly. This enables micmute LED for: 1. Matebook X (19e5:3200), (19e5:3201) 2. Matebook X Pro (19e5:3204) Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> --- sound/pci/hda/huawei_wmi_helper.c | 47 +++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 14 ++++++++- 2 files changed, 60 insertions(+), 1 deletion(-) create mode 100644 sound/pci/hda/huawei_wmi_helper.c diff --git a/sound/pci/hda/huawei_wmi_helper.c b/sound/pci/hda/huawei_wmi_helper.c new file mode 100644 index 000000000000..7c91f914ffba --- /dev/null +++ b/sound/pci/hda/huawei_wmi_helper.c @@ -0,0 +1,47 @@ +// SPDX-License-Identifier: GPL-2.0 +/* Helper functions for Huawei WMI micmute LED; + * to be included from codec driver + */ + +#if IS_ENABLED(CONFIG_HUAWEI_WMI) +#include <linux/platform_data/x86/huawei-wmi.h> + +static int (*huawei_wmi_micmute_led_set_func)(bool); + +static void update_huawei_wmi_micmute_led(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + huawei_wmi_micmute_led_set_func(spec->micmute_led.led_value); +} + +static void alc_fixup_huawei_wmi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + bool removefunc = false; + + if (action == HDA_FIXUP_ACT_PROBE) { + if (!huawei_wmi_micmute_led_set_func) + huawei_wmi_micmute_led_set_func = symbol_request(huawei_wmi_micmute_led_set); + if (!huawei_wmi_micmute_led_set_func) { + codec_warn(codec, "Failed to find huawei_wmi symbol huawei_wmi_micmute_led_set\n"); + return; + } + + removefunc = (huawei_wmi_micmute_led_set_func(false) < 0) + || (snd_hda_gen_add_micmute_led(codec, update_huawei_wmi_micmute_led) < 0); + } + + if (huawei_wmi_micmute_led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { + symbol_put(huawei_wmi_micmute_led_set); + huawei_wmi_micmute_led_set_func = NULL; + } +} + +#else + +static void alc_fixup_huawei_wmi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ +} + +#endif diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 135c0826859b..5431dee40f4f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5374,6 +5374,9 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" +/* for alc_fixup_huawei_micmute_led */ +#include "huawei_wmi_helper.c" + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -5494,6 +5497,7 @@ enum { ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, ALC256_FIXUP_HUAWEI_MBXP_PINS, + ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, }; @@ -5765,6 +5769,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_huawei_wmi + }, [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5779,7 +5787,9 @@ static const struct hda_fixup alc269_fixups[] = { {0x1e, 0x411111f0}, {0x21, 0x04211020}, { } - } + }, + .chained = true, + .chain_id = ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, @@ -6611,6 +6621,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x19e5, 0x3200, "Huawei MBX", ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED), + SND_PCI_QUIRK(0x19e5, 0x3201, "Huawei MBX", ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- 2.17.2

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[alsa-devel] HTTP GIT access to alsa-lib seems to be misaligned
by Timo Wischer 14 Nov '18

14 Nov '18

Hi all, The master branch of $ git clone http://git.alsa-project.org/http/alsa-lib.git is a different one as of $ git clone git://git.alsa-project.org/alsa-lib.git alsa-lib $ git log --graph --oneline http/master * dd2f6f8 update README * 9021f6c samples: make them work well * eec7186 tests: add a test for client instance of ALSATimer * c46b9e6 tests: add a test for port instance of ALSASeq * d6467d9 tests: add a test for client instance of ALSASeq * 03a8b8b tests: add test for public API of client class of ALSACtl * 2f94c2d rawmidi; add GObject enumerator for information flags * ed49bff rawmidi: add GObject enums for stream direction * 25e37bc rawmidi: add support for GObject enumerations * 709c4ce timer: add GObject enums for event type of timer instance * 10c9f2e timer: add GObject flags for paramter flags of timer instance * 3118597 timer: add GObject flags for flag of global timer instance * b28e1d7 timer: add GObject enums for type of global timer instance * 48f904b timer: add GObject enums for slave class of timer instance * 3bc28ec timer: add GObject enums for device class of timer instance * 480b358 timer: add support for GObject enumerations versus $ git log --graph --oneline origin/master * 178fe66 utils/alsa.m4: conditionally enable libdl in AM_PATH_ALSA m4 macro * 9a8d0b0 README.md: add link to www.alsa-project.org * fa72edc initial version of README.md for github * 6fbee19 initial version of .travis.yml file * b420056 pcm: interval: Interpret (x x+1] correctly and return x+1 * 92172ad test/audio_time: remove unused variables * 956bc15 test: rename code to more approriate mixtest * 82ae026 test/code: make it work again * 6852f88 test/latecy: fix typo in tstamp compare * 6eced6b conf/ucm/Dell-WD15-Dock: Fix incorrect device names * 3ec6dce Release v1.1.7 * fd13267 ioplug: Fix comment in struct snd_pcm_ioplug * 60d74ce timer: fix wrong comment to refer to 'SNDRV_TIMER_PSFLG_*' * 7c5c050 seq: Fix signedness in MIDI encoder/decoder * a849163 control_hw: Fix issue when applying seccomp policy * 040cfea conf/ucm: bytcr-rt5640: Improve human readable input/output names @Takashi The latest commit of http/master is yours. May be you have an idea how to solve this issue. -- Best regards *Timo Wischer* Engineering Software Base (ADITG/ESB) Tel. +49 5121 49 6938

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[alsa-devel] [PATCH 1/2] ASoC: pcm3060: Add control for differential output
by Kirill Marinushkin 14 Nov '18

14 Nov '18

DAC may be switched between differential and single-ended output. Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> --- sound/soc/codecs/pcm3060.c | 9 +++++++++ sound/soc/codecs/pcm3060.h | 1 + 2 files changed, 10 insertions(+) diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index 494d9d662be8..f792ad637ae4 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -179,6 +179,13 @@ static struct snd_soc_dai_driver pcm3060_dai[] = { /* dapm */ +static const char * const pcm3060_dapm_out_dif_txt[] = { + "Differential", "Single-Ended" +}; + +static SOC_ENUM_SINGLE_DECL(pcm3060_dapm_out_dif_enum, PCM3060_REG64, + PCM3060_REG_SHIFT_SE, pcm3060_dapm_out_dif_txt); + static DECLARE_TLV_DB_SCALE(pcm3060_dapm_tlv, -10050, 50, 1); static const struct snd_kcontrol_new pcm3060_dapm_controls[] = { @@ -195,6 +202,8 @@ static const struct snd_kcontrol_new pcm3060_dapm_controls[] = { 0, pcm3060_dapm_tlv), SOC_DOUBLE("Master Capture Switch", PCM3060_REG73, PCM3060_REG_SHIFT_MUT11, PCM3060_REG_SHIFT_MUT12, 1, 1), + + SOC_ENUM("DAC Out Differential", pcm3060_dapm_out_dif_enum), }; static const struct snd_soc_dapm_widget pcm3060_dapm_widgets[] = { diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h index fd89a68aa8a7..32080ff00166 100644 --- a/sound/soc/codecs/pcm3060.h +++ b/sound/soc/codecs/pcm3060.h @@ -38,6 +38,7 @@ int pcm3060_remove(struct device *dev); #define PCM3060_REG_ADPSV 0x20 #define PCM3060_REG_DAPSV 0x10 #define PCM3060_REG_SE 0x01 +#define PCM3060_REG_SHIFT_SE 0x00 #define PCM3060_REG65 0x41 #define PCM3060_REG66 0x42 -- 2.13.6

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[alsa-devel] [RFC PATCH] cplay: Always write frag * fragment_size
by Daniel Baluta 14 Nov '18

14 Nov '18

cplay first writes frag * fragment_size and then it only writes one fragment at a time. This means for example than if the user supplied a buffer_size it will only be used for the first write. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- I noticed this while investigating why cplay prints buffer_size as 0 when not specified as command line argument to cplay. I also noticed that cred always reads frag * frament_size, so I think this patch should be OK, but marking it as RFC to get your thoughts. src/utils/cplay.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/src/utils/cplay.c b/src/utils/cplay.c index 2a52b52..b016f52 100644 --- a/src/utils/cplay.c +++ b/src/utils/cplay.c @@ -435,15 +435,15 @@ void play_samples(char *name, unsigned int card, unsigned int device, }; if (verbose) printf("%s: Opened compress device\n", __func__); - size = config.fragment_size; - buffer = malloc(size * config.fragments); + size = config.fragments * config.fragment_size; + buffer = malloc(size); if (!buffer) { fprintf(stderr, "Unable to allocate %d bytes\n", size); goto COMP_EXIT; } /* we will write frag fragment_size and then start */ - num_read = fread(buffer, 1, size * config.fragments, file); + num_read = fread(buffer, 1, size, file); if (num_read > 0) { if (verbose) printf("%s: Doing first buffer write of %d\n", __func__, num_read); @@ -459,7 +459,7 @@ void play_samples(char *name, unsigned int card, unsigned int device, } } printf("Playing file %s On Card %u device %u, with buffer of %lu bytes\n", - name, card, device, buffer_size); + name, card, device, size); printf("Format %u Channels %u, %u Hz, Bit Rate %d\n", codec.id, codec.ch_in, codec.sample_rate, codec.bit_rate); -- 2.7.4

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[alsa-devel] Applied "ASoC: stm32: sai: fix invalid use of sizeof in stm32_sai_add_mclk_provider()" to the asoc tree
by Mark Brown 14 Nov '18

14 Nov '18

The patch ASoC: stm32: sai: fix invalid use of sizeof in stm32_sai_add_mclk_provider() 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 496fa3ba0a66c0192be86a45f44f94827ab8bcb6 Mon Sep 17 00:00:00 2001 From: Wei Yongjun <weiyongjun1(a)huawei.com> Date: Sat, 27 Oct 2018 02:19:59 +0000 Subject: [PATCH] ASoC: stm32: sai: fix invalid use of sizeof in stm32_sai_add_mclk_provider() sizeof() when applied to a pointer typed expression gives the size of the pointer, not that of the pointed data. Fixes: 8307b2afd386 ("ASoC: stm32: sai: set sai as mclk clock provider") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/stm/stm32_sai_sub.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 506423d5861a..d4825700b63f 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -389,7 +389,7 @@ static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai) char *mclk_name, *p, *s = (char *)pname; int ret, i = 0; - mclk = devm_kzalloc(dev, sizeof(mclk), GFP_KERNEL); + mclk = devm_kzalloc(dev, sizeof(*mclk), GFP_KERNEL); if (!mclk) return -ENOMEM; -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: pcm3060: Rename output widgets" to the asoc tree
by Mark Brown 14 Nov '18

14 Nov '18

The patch ASoC: pcm3060: Rename output widgets 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 efcf8f9ee39dd4d63fc95d05e460a78605a6cbf2 Mon Sep 17 00:00:00 2001 From: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Date: Mon, 12 Nov 2018 08:08:34 +0100 Subject: [PATCH] ASoC: pcm3060: Rename output widgets In the initial commit [1], I added differential output of the codec as separate `+` and `-` widgets: OUTL+ OUTR+ OUTL- OUTR- Later, in the commit [2], I added a device tree property to configure the output as single-ended or differential. Having this property, the `+` and `-` separation in widgets seems for me confusing. There are no functional benefits in such separation, so I find reasonable to get rid of it: OUTL OUTR The new naming is more friendly for sound cards, and is better aligned with other codec drivers in kernel. Renaming the output widgets now should not be a problem from the backwards- compatibility perspective, as the driver for PCM3060 is added into the mainline very recently, and did not yet appear in any releases. [1] commit 6ee47d4a8dac ("ASoC: pcm3060: Add codec driver") [2] commit a78c62de00d5 ("ASoC: pcm3060: Add DT property for single-ended output") Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/pcm3060.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index 494d9d662be8..771b46e1974b 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -198,20 +198,16 @@ static const struct snd_kcontrol_new pcm3060_dapm_controls[] = { }; static const struct snd_soc_dapm_widget pcm3060_dapm_widgets[] = { - SND_SOC_DAPM_OUTPUT("OUTL+"), - SND_SOC_DAPM_OUTPUT("OUTR+"), - SND_SOC_DAPM_OUTPUT("OUTL-"), - SND_SOC_DAPM_OUTPUT("OUTR-"), + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), SND_SOC_DAPM_INPUT("INL"), SND_SOC_DAPM_INPUT("INR"), }; static const struct snd_soc_dapm_route pcm3060_dapm_map[] = { - { "OUTL+", NULL, "Playback" }, - { "OUTR+", NULL, "Playback" }, - { "OUTL-", NULL, "Playback" }, - { "OUTR-", NULL, "Playback" }, + { "OUTL", NULL, "Playback" }, + { "OUTR", NULL, "Playback" }, { "Capture", NULL, "INL" }, { "Capture", NULL, "INR" }, -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: pcm3060: Rename output widgets" to the asoc tree
by Mark Brown 14 Nov '18

14 Nov '18

The patch ASoC: pcm3060: Rename output widgets 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 933a95496e48bde40e366a79e6dd9f9740905573 Mon Sep 17 00:00:00 2001 From: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Date: Mon, 12 Nov 2018 08:08:34 +0100 Subject: [PATCH] ASoC: pcm3060: Rename output widgets In the initial commit [1], I added differential output of the codec as separate `+` and `-` widgets: OUTL+ OUTR+ OUTL- OUTR- Later, in the commit [2], I added a device tree property to configure the output as single-ended or differential. Having this property, the `+` and `-` separation in widgets seems for me confusing. There are no functional benefits in such separation, so I find reasonable to get rid of it: OUTL OUTR The new naming is more friendly for sound cards, and is better aligned with other codec drivers in kernel. Renaming the output widgets now should not be a problem from the backwards- compatibility perspective, as the driver for PCM3060 is added into the mainline very recently, and did not yet appear in any releases. [1] commit 6ee47d4a8dac ("ASoC: pcm3060: Add codec driver") [2] commit a78c62de00d5 ("ASoC: pcm3060: Add DT property for single-ended output") Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/pcm3060.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index 494d9d662be8..771b46e1974b 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -198,20 +198,16 @@ static const struct snd_kcontrol_new pcm3060_dapm_controls[] = { }; static const struct snd_soc_dapm_widget pcm3060_dapm_widgets[] = { - SND_SOC_DAPM_OUTPUT("OUTL+"), - SND_SOC_DAPM_OUTPUT("OUTR+"), - SND_SOC_DAPM_OUTPUT("OUTL-"), - SND_SOC_DAPM_OUTPUT("OUTR-"), + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), SND_SOC_DAPM_INPUT("INL"), SND_SOC_DAPM_INPUT("INR"), }; static const struct snd_soc_dapm_route pcm3060_dapm_map[] = { - { "OUTL+", NULL, "Playback" }, - { "OUTR+", NULL, "Playback" }, - { "OUTL-", NULL, "Playback" }, - { "OUTR-", NULL, "Playback" }, + { "OUTL", NULL, "Playback" }, + { "OUTR", NULL, "Playback" }, { "Capture", NULL, "INL" }, { "Capture", NULL, "INR" }, -- 2.19.1

1 0

[alsa-devel] [PATCH] ASoC: dapm: Use less aggressive caching to ensure correctness
by Tzung-Bi Shih 14 Nov '18

14 Nov '18

Commit 92a99ea439c4 ("ASoC: dapm: Use more aggressive caching") only invalidates necessary caches to improve the cache hit rate. However, in snd_soc_dapm_add_path(), the cache is too aggressive which results in wrong state of the audio map. E.g. imaging the following examples during snd_soc_instantiate_card(): time 1: at the beginning in:-1 in:-1 in:-1 out:-1 out:-1 out:-1 A -----> B Spk time 2: after someone called snd_soc_dapm_new_widgets() (e.g. create_fill_widget_route_map() in sound/soc/codecs/hdac_hdmi.c) in:0 in:0 in:0 out:0 out:0 out:1 A ----> B Spk time 3: route added to Spk in:0 in:0 in:0 out:0 out:0 out:1 A ----> B ----> Spk time N: route added from SIGGEN in:1 in:0 in:0 in:0 out:0 out:0 out:0 out:1 SIGGEN ----> A ----> B ----> Spk In the end, the path will not power on but it should be. At time 2, the caches have been polluted. At time 3, it will not get the correct state if snd_soc_dapm_add_path() does not invalidate "out" of B and A. time 3 (expected result): in:0 in:0 in:0 out:1 out:1 out:1 A ----> B ----> Spk Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a5178845065b..2914f3adb333 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2722,7 +2722,7 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, dapm_mark_dirty(widgets[dir], "Route added"); } - if (dapm->card->instantiated && path->connect) + if (path->connect) dapm_path_invalidate(path); return 0; -- 2.19.1.930.g4563a0d9d0-goog

2 3

[alsa-devel] [PATCH v7 0/4] Add support for audiocodec in Allwinner A64
by Vasily Khoruzhick 14 Nov '18

14 Nov '18

v2: - Use simple-amplifier for speaker amp on Pinebook - Rename sun50i-a64-i2s to sun50i-a64-codec-i2s to preserve compatible string for other 3 I2S modules in A64 in case if there's any incompatibility with H3 v3: - renamed sunxi-adda-pr-regmap to sun8i-adda-pr-regmap - use ilog2() to calculate reg value for LRCK div instead of using a table v4: - dts: don't use 'Mic' and 'Headset Mic' widgets from sun8i-codec, define our board-level widgets instead. v5: - collect all the tags v6: - driver patches has been merged through ASoC tree - rebase onto 4.20-rc1 - Drop 'Speaker' from routes on sopine and pine64, they don't have speaker. v7: - drop 'Mic' and 'Headset Mic' from dtsi - move AIF1 Slot 0 ADC to ADC route to sun8i-codec - drop BIAS routes for Pine64 and SoPine Vasily Khoruzhick (4): ASoC: sun8i-codec: add missing route for ADC arm64: dts: allwinner: a64: add nodes necessary for analog sound support arm64: dts: allwinner: a64: enable sound on Pine64 and SoPine arm64: dts: allwinner: a64: enable sound on Pinebook .../boot/dts/allwinner/sun50i-a64-pine64.dts | 25 +++++++++ .../dts/allwinner/sun50i-a64-pinebook.dts | 40 ++++++++++++++ .../allwinner/sun50i-a64-sopine-baseboard.dts | 25 +++++++++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 54 +++++++++++++++++++ sound/soc/sunxi/sun8i-codec.c | 6 ++- 5 files changed, 149 insertions(+), 1 deletion(-) -- 2.19.1

2 5

[alsa-devel] [PATCH] ASoC: qcom: Set dai_link id to each dai_link
by Rohit kumar 14 Nov '18

14 Nov '18

Frontend dai_link id is used for closing ADM sessions. During concurrent usecase when one session is closed, it closes other ADM session associated with other usecase too. Dai_link->id should always point to Frontend dai id. Set cpu_dai id as dai_link id to fix the issue. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- sound/soc/qcom/common.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index eb1b9da..4715527 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -13,6 +13,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) struct device_node *cpu = NULL; struct device *dev = card->dev; struct snd_soc_dai_link *link; + struct of_phandle_args args; int ret, num_links; ret = snd_soc_of_parse_card_name(card, "model"); @@ -47,12 +48,14 @@ int qcom_snd_parse_of(struct snd_soc_card *card) goto err; } - link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); - if (!link->cpu_of_node) { + ret = of_parse_phandle_with_args(cpu, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { dev_err(card->dev, "error getting cpu phandle\n"); - ret = -EINVAL; goto err; } + link->cpu_of_node = args.np; + link->id = args.args[0]; ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); if (ret) { -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 2

[alsa-devel] [PATCH v2] ASoC: Intel: Power down links before turning off display audio power
by Pierre-Louis Bossart 13 Nov '18

13 Nov '18

On certain platforms, Display HDMI HDA codec was not going to sleep state after the use when links are powered down after turning off the display power. As per the HW recommendation, links are powered down before turning off the display power to ensure that the codec goes to sleep state. This patch was updated from an earlier version submitted upstream [1] which conflicted with the changes merged for HDaudio codec support with the Intel DSP. [1] https://patchwork.kernel.org/patch/10540213/ Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v2: fix copy/paste in error handling. sound/soc/codecs/hdac_hdmi.c | 11 +++++------ sound/soc/intel/skylake/skl.c | 12 ++++++------ 2 files changed, 11 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 4e9854889a95..e63d6e33df48 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -2187,11 +2187,6 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) */ snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - err = snd_hdac_display_power(bus, false); - if (err < 0) { - dev_err(dev, "Cannot turn on display power on i915\n"); - return err; - } hlink = snd_hdac_ext_bus_get_link(bus, dev_name(dev)); if (!hlink) { @@ -2201,7 +2196,11 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) snd_hdac_ext_bus_link_put(bus, hlink); - return 0; + err = snd_hdac_display_power(bus, false); + if (err < 0) + dev_err(dev, "Cannot turn off display power on i915\n"); + + return err; } static int hdac_hdmi_runtime_resume(struct device *dev) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index ebab04b51bc1..ac0b4ff21acc 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -811,6 +811,12 @@ static void skl_probe_work(struct work_struct *work) } } + /* + * we are done probing so decrement link counts + */ + list_for_each_entry(hlink, &bus->hlink_list, list) + snd_hdac_ext_bus_link_put(bus, hlink); + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { err = snd_hdac_display_power(bus, false); if (err < 0) { @@ -820,12 +826,6 @@ static void skl_probe_work(struct work_struct *work) } } - /* - * we are done probing so decrement link counts - */ - list_for_each_entry(hlink, &bus->hlink_list, list) - snd_hdac_ext_bus_link_put(bus, hlink); - /* configure PM */ pm_runtime_put_noidle(bus->dev); pm_runtime_allow(bus->dev); -- 2.17.1

2 1

[alsa-devel] [PATCH 02/11] ASoC: AMD: add ACP3.0 PCI driver
by Vijendar Mukunda 13 Nov '18

13 Nov '18

From: Maruthi Srinivas Bayyavarapu <Maruthi.Bayyavarapu(a)amd.com> ACP 3.0 is a PCI audio device. This patch adds PCI driver to bind to this device and get PCI resources. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Signed-off-by: Sanju R Mehta <sanju.mehta(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x.h | 13 ++++++ sound/soc/amd/raven/pci-acp3x.c | 97 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 110 insertions(+) create mode 100644 sound/soc/amd/raven/acp3x.h create mode 100644 sound/soc/amd/raven/pci-acp3x.c diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h new file mode 100644 index 0000000..e9b4df0 --- /dev/null +++ b/sound/soc/amd/raven/acp3x.h @@ -0,0 +1,13 @@ +#include "chip_offset_byte.h" + +#define ACP3x_PHY_BASE_ADDRESS 0x1240000 + +static inline u32 rv_readl(void __iomem *base_addr) +{ + return readl(base_addr - ACP3x_PHY_BASE_ADDRESS); +} + +static inline void rv_writel(u32 val, void __iomem *base_addr) +{ + writel(val, base_addr - ACP3x_PHY_BASE_ADDRESS); +} diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c new file mode 100644 index 0000000..27588ed --- /dev/null +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -0,0 +1,97 @@ +/* + * AMD ALSA SoC PCM Driver + * + * Copyright 2016 Advanced Micro Devices, Inc. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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. + */ + +#include <linux/pci.h> +#include <linux/module.h> +#include <linux/io.h> + +#include "acp3x.h" + +struct acp3x_dev_data { + void __iomem *acp3x_base; +}; + +static int snd_acp3x_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + int ret; + u32 addr; + struct acp3x_dev_data *adata; + + if (pci_enable_device(pci)) { + dev_err(&pci->dev, "pci_enable_device failed\n"); + return -ENODEV; + } + + ret = pci_request_regions(pci, "AMD ACP3x audio"); + if (ret < 0) { + dev_err(&pci->dev, "pci_request_regions failed\n"); + goto disable_pci; + } + + adata = devm_kzalloc(&pci->dev, sizeof(struct acp3x_dev_data), + GFP_KERNEL); + if (!adata) { + ret = -ENOMEM; + goto release_regions; + } + + addr = pci_resource_start(pci, 0); + adata->acp3x_base = ioremap(addr, pci_resource_len(pci, 0)); + if (!adata->acp3x_base) { + ret = -ENOMEM; + goto release_regions; + } + pci_set_master(pci); + pci_set_drvdata(pci, adata); + return 0; + +release_regions: + pci_release_regions(pci); +disable_pci: + pci_disable_device(pci); + + return ret; +} + +static void snd_acp3x_remove(struct pci_dev *pci) +{ + struct acp3x_dev_data *adata = pci_get_drvdata(pci); + + iounmap(adata->acp3x_base); + pci_release_regions(pci); + pci_disable_device(pci); +} + +static const struct pci_device_id snd_acp3x_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_AMD, 0x15e2), + .class = PCI_CLASS_MULTIMEDIA_OTHER << 8, + .class_mask = 0xffffff }, + { 0, }, +}; +MODULE_DEVICE_TABLE(pci, snd_acp3x_ids); + +static struct pci_driver acp3x_driver = { + .name = KBUILD_MODNAME, + .id_table = snd_acp3x_ids, + .probe = snd_acp3x_probe, + .remove = snd_acp3x_remove, +}; + +module_pci_driver(acp3x_driver); + +MODULE_AUTHOR("Maruthi.Bayyavarapu(a)amd.com"); +MODULE_DESCRIPTION("AMD ACP3x PCI driver"); +MODULE_LICENSE("GPL v2"); -- 2.7.4

2 1

[alsa-devel] [PATCH 03/11] ASoC: amd: create ACP3x PCM platform device
by Vijendar Mukunda 13 Nov '18

13 Nov '18

ACP 3x IP has I2S controller device as one of IP blocks. Create a platform device for it, so that the PCM platform driver can be bound to this device. Pass PCI resources like MMIO, irq to the platform device. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <vishnuvardhanrao.ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x.h | 4 +++ sound/soc/amd/raven/pci-acp3x.c | 70 ++++++++++++++++++++++++++++++++++++++++- 2 files changed, 73 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index e9b4df0..83b1ed8 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -1,6 +1,10 @@ #include "chip_offset_byte.h" #define ACP3x_PHY_BASE_ADDRESS 0x1240000 +#define ACP3x_I2S_MODE 0 +#define ACP3x_REG_START 0x1240000 +#define ACP3x_REG_END 0x1250200 +#define I2S_MODE 0x04 static inline u32 rv_readl(void __iomem *base_addr) { diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index 27588ed..0700162 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -16,19 +16,26 @@ #include <linux/pci.h> #include <linux/module.h> #include <linux/io.h> +#include <linux/platform_device.h> +#include <linux/interrupt.h> #include "acp3x.h" struct acp3x_dev_data { void __iomem *acp3x_base; + bool acp3x_audio_mode; + struct resource *res; + struct platform_device *pdev; }; static int snd_acp3x_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { int ret; - u32 addr; + u32 addr, val; struct acp3x_dev_data *adata; + struct platform_device_info pdevinfo; + unsigned int irqflags; if (pci_enable_device(pci)) { dev_err(&pci->dev, "pci_enable_device failed\n"); @@ -48,6 +55,15 @@ static int snd_acp3x_probe(struct pci_dev *pci, goto release_regions; } + /* check for msi interrupt support */ + ret = pci_enable_msi(pci); + if (ret) + /* msi is not enabled */ + irqflags = IRQF_SHARED; + else + /* msi is enabled */ + irqflags = 0; + addr = pci_resource_start(pci, 0); adata->acp3x_base = ioremap(addr, pci_resource_len(pci, 0)); if (!adata->acp3x_base) { @@ -56,8 +72,57 @@ static int snd_acp3x_probe(struct pci_dev *pci, } pci_set_master(pci); pci_set_drvdata(pci, adata); + + val = rv_readl(adata->acp3x_base + mmACP_I2S_PIN_CONFIG); + switch (val) { + case I2S_MODE: + adata->res = devm_kzalloc(&pci->dev, + sizeof(struct resource) * 2, + GFP_KERNEL); + if (!adata->res) { + ret = -ENOMEM; + goto unmap_mmio; + } + + adata->res[0].name = "acp3x_i2s_iomem"; + adata->res[0].flags = IORESOURCE_MEM; + adata->res[0].start = addr; + adata->res[0].end = addr + (ACP3x_REG_END - ACP3x_REG_START); + + adata->res[1].name = "acp3x_i2s_irq"; + adata->res[1].flags = IORESOURCE_IRQ; + adata->res[1].start = pci->irq; + adata->res[1].end = pci->irq; + + adata->acp3x_audio_mode = ACP3x_I2S_MODE; + + memset(&pdevinfo, 0, sizeof(pdevinfo)); + pdevinfo.name = "acp3x_rv_i2s"; + pdevinfo.id = 0; + pdevinfo.parent = &pci->dev; + pdevinfo.num_res = 2; + pdevinfo.res = adata->res; + pdevinfo.data = &irqflags; + pdevinfo.size_data = sizeof(irqflags); + + adata->pdev = platform_device_register_full(&pdevinfo); + if (!adata->pdev) { + dev_err(&pci->dev, "cannot register %s device\n", + pdevinfo.name); + ret = -ENODEV; + goto unmap_mmio; + } + break; + default: + dev_err(&pci->dev, "Inavlid ACP audio mode : %d\n", val); + ret = -ENODEV; + goto unmap_mmio; + } return 0; +unmap_mmio: + pci_disable_msi(pci); + iounmap(adata->acp3x_base); release_regions: pci_release_regions(pci); disable_pci: @@ -70,7 +135,10 @@ static void snd_acp3x_remove(struct pci_dev *pci) { struct acp3x_dev_data *adata = pci_get_drvdata(pci); + platform_device_unregister(adata->pdev); iounmap(adata->acp3x_base); + + pci_disable_msi(pci); pci_release_regions(pci); pci_disable_device(pci); } -- 2.7.4

2 1

[alsa-devel] [PATCH 04/11] ASoC: amd: add ACP3x PCM platform driver
by Vijendar Mukunda 13 Nov '18

13 Nov '18

PCM platform driver binds to the platform device created by ACP3x PCI device. PCM driver registers ALSA DMA and CPU DAI components with ASoC framework. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 263 ++++++++++++++++++++++++++++++++++++ sound/soc/amd/raven/acp3x.h | 7 + 2 files changed, 270 insertions(+) create mode 100644 sound/soc/amd/raven/acp3x-pcm-dma.c diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c new file mode 100644 index 0000000..3e806f7 --- /dev/null +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -0,0 +1,263 @@ +/* + * AMD ALSA SoC PCM Driver + * + * Copyright 2016 Advanced Micro Devices, Inc. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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. + */ + +#include <linux/platform_device.h> +#include <linux/module.h> +#include <linux/err.h> +#include <linux/io.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "acp3x.h" + +#define DRV_NAME "acp3x-i2s-audio" + +struct i2s_dev_data { + void __iomem *acp3x_base; + struct snd_pcm_substream *play_stream; + struct snd_pcm_substream *capture_stream; +}; + +static int acp3x_power_on(void __iomem *acp3x_base, bool on) +{ + u16 val, mask; + u32 timeout; + + if (on == true) { + val = 1; + mask = ACP3x_POWER_ON; + } else { + val = 0; + mask = ACP3x_POWER_OFF; + } + + rv_writel(val, acp3x_base + mmACP_PGFSM_CONTROL); + timeout = 0; + while (true) { + val = rv_readl(acp3x_base + mmACP_PGFSM_STATUS); + if ((val & ACP3x_POWER_OFF_IN_PROGRESS) == mask) + break; + if (timeout > 100) { + pr_err("ACP3x power state change failure\n"); + return -ENODEV; + } + timeout++; + cpu_relax(); + } + return 0; +} + +static int acp3x_reset(void __iomem *acp3x_base) +{ + u32 val, timeout; + + rv_writel(1, acp3x_base + mmACP_SOFT_RESET); + timeout = 0; + while (true) { + val = rv_readl(acp3x_base + mmACP_SOFT_RESET); + if ((val & ACP3x_SOFT_RESET__SoftResetAudDone_MASK) || + timeout > 100) { + if (val & ACP3x_SOFT_RESET__SoftResetAudDone_MASK) + break; + return -ENODEV; + } + timeout++; + cpu_relax(); + } + + rv_writel(0, acp3x_base + mmACP_SOFT_RESET); + timeout = 0; + while (true) { + val = rv_readl(acp3x_base + mmACP_SOFT_RESET); + if (!val || timeout > 100) { + if (!val) + break; + return -ENODEV; + } + timeout++; + cpu_relax(); + } + return 0; +} + +static int acp3x_init(void __iomem *acp3x_base) +{ + int ret; + + /* power on */ + ret = acp3x_power_on(acp3x_base, true); + if (ret) { + pr_err("ACP3x power on failed\n"); + return ret; + } + /* Reset */ + ret = acp3x_reset(acp3x_base); + if (ret) { + pr_err("ACP3x reset failed\n"); + return ret; + } + return 0; +} + +static int acp3x_deinit(void __iomem *acp3x_base) +{ + int ret; + + /* Reset */ + ret = acp3x_reset(acp3x_base); + if (ret) { + pr_err("ACP3x reset failed\n"); + return ret; + } + /* power off */ + ret = acp3x_power_on(acp3x_base, false); + if (ret) { + pr_err("ACP3x power off failed\n"); + return ret; + } + return 0; +} + +static struct snd_pcm_ops acp3x_dma_ops = { + .open = NULL, + .close = NULL, + .ioctl = NULL, + .hw_params = NULL, + .hw_free = NULL, + .pointer = NULL, + .mmap = NULL, +}; + +struct snd_soc_dai_ops acp3x_dai_i2s_ops = { + .hw_params = NULL, + .trigger = NULL, + .set_fmt = NULL, +}; + +static struct snd_soc_dai_driver acp3x_i2s_dai_driver = { + .playback = { + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + }, + .ops = &acp3x_dai_i2s_ops, +}; + +static const struct snd_soc_component_driver acp3x_i2s_component = { + .name = DRV_NAME, + .ops = &acp3x_dma_ops, + .pcm_new = acp3x_dma_new, +}; + +static int acp3x_audio_probe(struct platform_device *pdev) +{ + int status; + struct resource *res; + struct i2s_dev_data *adata; + unsigned int irqflags; + + if (!pdev->dev.platform_data) { + dev_err(&pdev->dev, "platform_data not retrieved\n"); + return -ENODEV; + } + irqflags = *((unsigned int *)(pdev->dev.platform_data)); + + adata = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dev_data), + GFP_KERNEL); + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + return -ENODEV; + } + + adata->acp3x_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + + adata->play_stream = NULL; + adata->capture_stream = NULL; + + dev_set_drvdata(&pdev->dev, adata); + /* Initialize ACP */ + status = acp3x_init(adata->acp3x_base); + if (status) + return -ENODEV; + status = devm_snd_soc_register_component(&pdev->dev, + &acp3x_i2s_component, + &acp3x_i2s_dai_driver, 1); + if (status) { + dev_err(&pdev->dev, "Fail to register acp i2s dai\n"); + goto dev_err; + } + + return 0; +dev_err: + status = acp3x_deinit(adata->acp3x_base); + if (status) + dev_err(&pdev->dev, "ACP de-init failed\n"); + else + dev_info(&pdev->dev, "ACP de-initialized\n"); + /*ignore device status and return driver probe error*/ + return -ENODEV; +} + +static int acp3x_audio_remove(struct platform_device *pdev) +{ + int ret; + struct i2s_dev_data *adata = dev_get_drvdata(&pdev->dev); + + ret = acp3x_deinit(adata->acp3x_base); + if (ret) + dev_err(&pdev->dev, "ACP de-init failed\n"); + else + dev_info(&pdev->dev, "ACP de-initialized\n"); + + return 0; +} + +static struct platform_driver acp3x_dma_driver = { + .probe = acp3x_audio_probe, + .remove = acp3x_audio_remove, + .driver = { + .name = "acp3x_rv_i2s", + }, +}; + +module_platform_driver(acp3x_dma_driver); + +MODULE_AUTHOR("Maruthi.Bayyavarapu(a)amd.com"); +MODULE_AUTHOR("Vijendar.Mukunda(a)amd.com"); +MODULE_DESCRIPTION("AMD ACP 3.x PCM Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index 83b1ed8..f3d3e51 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -5,6 +5,13 @@ #define ACP3x_REG_START 0x1240000 #define ACP3x_REG_END 0x1250200 #define I2S_MODE 0x04 +#define BT_TX_THRESHOLD 26 +#define BT_RX_THRESHOLD 25 +#define ACP3x_POWER_ON 0x00 +#define ACP3x_POWER_ON_IN_PROGRESS 0x01 +#define ACP3x_POWER_OFF 0x02 +#define ACP3x_POWER_OFF_IN_PROGRESS 0x03 +#define ACP3x_SOFT_RESET__SoftResetAudDone_MASK 0x00010001 static inline u32 rv_readl(void __iomem *base_addr) { -- 2.7.4

2 2

[alsa-devel] [PATCH 05/11] ASoC: amd: Interrupt handler changes for ACP3x DMA driver
by Vijendar Mukunda 13 Nov '18

13 Nov '18

Whenever audio data equal to the I2S FIFO watermark level are produced/consumed, interrupt is generated. Acknowledge the interrupt. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 46 +++++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 3e806f7..94f915a 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -27,6 +27,7 @@ #define DRV_NAME "acp3x-i2s-audio" struct i2s_dev_data { + unsigned int i2s_irq; void __iomem *acp3x_base; struct snd_pcm_substream *play_stream; struct snd_pcm_substream *capture_stream; @@ -132,6 +133,38 @@ static int acp3x_deinit(void __iomem *acp3x_base) return 0; } +static irqreturn_t i2s_irq_handler(int irq, void *dev_id) +{ + u16 play_flag, cap_flag; + u32 val; + struct i2s_dev_data *rv_i2s_data = dev_id; + + if (!rv_i2s_data) + return IRQ_NONE; + + play_flag = 0; + cap_flag = 0; + val = rv_readl(rv_i2s_data->acp3x_base + mmACP_EXTERNAL_INTR_STAT); + if ((val & BIT(BT_TX_THRESHOLD)) && rv_i2s_data->play_stream) { + rv_writel(BIT(BT_TX_THRESHOLD), rv_i2s_data->acp3x_base + + mmACP_EXTERNAL_INTR_STAT); + snd_pcm_period_elapsed(rv_i2s_data->play_stream); + play_flag = 1; + } + + if ((val & BIT(BT_RX_THRESHOLD)) && rv_i2s_data->capture_stream) { + rv_writel(BIT(BT_RX_THRESHOLD), rv_i2s_data->acp3x_base + + mmACP_EXTERNAL_INTR_STAT); + snd_pcm_period_elapsed(rv_i2s_data->capture_stream); + cap_flag = 1; + } + + if (play_flag | cap_flag) + return IRQ_HANDLED; + else + return IRQ_NONE; +} + static struct snd_pcm_ops acp3x_dma_ops = { .open = NULL, .close = NULL, @@ -205,6 +238,13 @@ static int acp3x_audio_probe(struct platform_device *pdev) adata->acp3x_base = devm_ioremap(&pdev->dev, res->start, resource_size(res)); + res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!res) { + dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + return -ENODEV; + } + + adata->i2s_irq = res->start; adata->play_stream = NULL; adata->capture_stream = NULL; @@ -220,6 +260,12 @@ static int acp3x_audio_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Fail to register acp i2s dai\n"); goto dev_err; } + status = devm_request_irq(&pdev->dev, adata->i2s_irq, i2s_irq_handler, + irqflags, "ACP3x_I2S_IRQ", adata); + if (status) { + dev_err(&pdev->dev, "ACP3x I2S IRQ request failed\n"); + goto dev_err; + } return 0; dev_err: -- 2.7.4

2 1

[alsa-devel] [PATCH 06/11] ASoC: amd: add acp3x pcm driver dma ops
by Vijendar Mukunda 13 Nov '18

13 Nov '18

ACP3x has a DMA controller to access system memory. This controller transfers data from/to system memory to/from the ACP internal FIFO. The patch adds PCM driver DMA operations. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 262 +++++++++++++++++++++++++++++++++++- sound/soc/amd/raven/acp3x.h | 20 +++ 2 files changed, 275 insertions(+), 7 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 94f915a..06c1985 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -33,6 +33,54 @@ struct i2s_dev_data { struct snd_pcm_substream *capture_stream; }; +struct i2s_stream_instance { + u16 num_pages; + u16 channels; + u32 xfer_resolution; + struct page *pg; + void __iomem *acp3x_base; +}; + +static const struct snd_pcm_hardware acp3x_pcm_hardware_playback = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_96000, + .rate_min = 8000, + .rate_max = 96000, + .buffer_bytes_max = PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE, + .period_bytes_min = PLAYBACK_MIN_PERIOD_SIZE, + .period_bytes_max = PLAYBACK_MAX_PERIOD_SIZE, + .periods_min = PLAYBACK_MIN_NUM_PERIODS, + .periods_max = PLAYBACK_MAX_NUM_PERIODS, +}; + +static const struct snd_pcm_hardware acp3x_pcm_hardware_capture = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .buffer_bytes_max = CAPTURE_MAX_NUM_PERIODS * CAPTURE_MAX_PERIOD_SIZE, + .period_bytes_min = CAPTURE_MIN_PERIOD_SIZE, + .period_bytes_max = CAPTURE_MAX_PERIOD_SIZE, + .periods_min = CAPTURE_MIN_NUM_PERIODS, + .periods_max = CAPTURE_MAX_NUM_PERIODS, +}; + static int acp3x_power_on(void __iomem *acp3x_base, bool on) { u16 val, mask; @@ -165,14 +213,214 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) return IRQ_NONE; } +static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction) +{ + u16 page_idx; + u64 addr; + u32 low, high, val, acp_fifo_addr; + struct page *pg = rtd->pg; + + /* 8 scratch registers used to map one 64 bit address */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + val = 0; + else + val = rtd->num_pages * 8; + + /* Group Enable */ + rv_writel(ACP_SRAM_PTE_OFFSET | BIT(31), rtd->acp3x_base + + mmACPAXI2AXI_ATU_BASE_ADDR_GRP_1); + rv_writel(PAGE_SIZE_4K_ENABLE, rtd->acp3x_base + + mmACPAXI2AXI_ATU_PAGE_SIZE_GRP_1); + + for (page_idx = 0; page_idx < rtd->num_pages; page_idx++) { + /* Load the low address of page int ACP SRAM through SRBM */ + addr = page_to_phys(pg); + low = lower_32_bits(addr); + high = upper_32_bits(addr); + + rv_writel(low, rtd->acp3x_base + mmACP_SCRATCH_REG_0 + val); + high |= BIT(31); + rv_writel(high, rtd->acp3x_base + mmACP_SCRATCH_REG_0 + val + + 4); + /* Move to next physically contiguos page */ + val += 8; + pg++; + } + + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + /* Config ringbuffer */ + rv_writel(MEM_WINDOW_START, rtd->acp3x_base + + mmACP_BT_TX_RINGBUFADDR); + rv_writel(MAX_BUFFER, rtd->acp3x_base + + mmACP_BT_TX_RINGBUFSIZE); + rv_writel(DMA_SIZE, rtd->acp3x_base + mmACP_BT_TX_DMA_SIZE); + + /* Config audio fifo */ + acp_fifo_addr = ACP_SRAM_PTE_OFFSET + (rtd->num_pages * 8) + + PLAYBACK_FIFO_ADDR_OFFSET; + rv_writel(acp_fifo_addr, rtd->acp3x_base + + mmACP_BT_TX_FIFOADDR); + rv_writel(FIFO_SIZE, rtd->acp3x_base + mmACP_BT_TX_FIFOSIZE); + } else { + /* Config ringbuffer */ + rv_writel(MEM_WINDOW_START + MAX_BUFFER, rtd->acp3x_base + + mmACP_BT_RX_RINGBUFADDR); + rv_writel(MAX_BUFFER, rtd->acp3x_base + + mmACP_BT_RX_RINGBUFSIZE); + rv_writel(DMA_SIZE, rtd->acp3x_base + mmACP_BT_RX_DMA_SIZE); + + /* Config audio fifo */ + acp_fifo_addr = ACP_SRAM_PTE_OFFSET + + (rtd->num_pages * 8) + CAPTURE_FIFO_ADDR_OFFSET; + rv_writel(acp_fifo_addr, rtd->acp3x_base + + mmACP_BT_RX_FIFOADDR); + rv_writel(FIFO_SIZE, rtd->acp3x_base + mmACP_BT_RX_FIFOSIZE); + } + + /* Enable watermark/period interrupt to host */ + rv_writel(BIT(BT_TX_THRESHOLD) | BIT(BT_RX_THRESHOLD), + rtd->acp3x_base + mmACP_EXTERNAL_INTR_CNTL); +} + +static int acp3x_dma_open(struct snd_pcm_substream *substream) +{ + int ret = 0; + + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, + DRV_NAME); + struct i2s_dev_data *adata = dev_get_drvdata(component->dev); + + struct i2s_stream_instance *i2s_data = kzalloc(sizeof(struct i2s_stream_instance), + GFP_KERNEL); + if (!i2s_data) + return -EINVAL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + runtime->hw = acp3x_pcm_hardware_playback; + else + runtime->hw = acp3x_pcm_hardware_capture; + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(component->dev, "set integer constraint failed\n"); + return ret; + } + + if (!adata->play_stream && !adata->capture_stream) + rv_writel(1, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + adata->play_stream = substream; + else + adata->capture_stream = substream; + + i2s_data->acp3x_base = adata->acp3x_base; + runtime->private_data = i2s_data; + return 0; +} + +static int acp3x_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int status; + u64 size; + struct snd_dma_buffer *dma_buffer; + struct page *pg; + struct snd_pcm_runtime *runtime = substream->runtime; + struct i2s_stream_instance *rtd = runtime->private_data; + + if (!rtd) + return -EINVAL; + + dma_buffer = &substream->dma_buffer; + size = params_buffer_bytes(params); + status = snd_pcm_lib_malloc_pages(substream, size); + if (status < 0) + return status; + + memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); + pg = virt_to_page(substream->dma_buffer.area); + if (pg) { + rtd->pg = pg; + rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT); + config_acp3x_dma(rtd, substream->stream); + status = 0; + } else { + status = -ENOMEM; + } + return status; +} + +static snd_pcm_uframes_t acp3x_dma_pointer(struct snd_pcm_substream *substream) +{ + u32 pos = 0; + struct i2s_stream_instance *rtd = substream->runtime->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + pos = rv_readl(rtd->acp3x_base + + mmACP_BT_TX_LINKPOSITIONCNTR); + else + pos = rv_readl(rtd->acp3x_base + + mmACP_BT_RX_LINKPOSITIONCNTR); + + if (pos >= MAX_BUFFER) + pos = 0; + + return bytes_to_frames(substream->runtime, pos); +} + +static int acp3x_dma_new(struct snd_soc_pcm_runtime *rtd) +{ + return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + SNDRV_DMA_TYPE_DEV, + NULL, MIN_BUFFER, + MAX_BUFFER); +} + +static int acp3x_dma_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int acp3x_dma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return snd_pcm_lib_default_mmap(substream, vma); +} + +static int acp3x_dma_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct i2s_stream_instance *rtd = substream->runtime->private_data; + struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, + DRV_NAME); + struct i2s_dev_data *adata = dev_get_drvdata(component->dev); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + adata->play_stream = NULL; + else + adata->capture_stream = NULL; + + /* Disable ACP irq, when the current stream is being closed and + * another stream is also not active. + */ + if (!adata->play_stream && !adata->capture_stream) + rv_writel(0, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); + kfree(rtd); + return 0; +} + static struct snd_pcm_ops acp3x_dma_ops = { - .open = NULL, - .close = NULL, - .ioctl = NULL, - .hw_params = NULL, - .hw_free = NULL, - .pointer = NULL, - .mmap = NULL, + .open = acp3x_dma_open, + .close = acp3x_dma_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = acp3x_dma_hw_params, + .hw_free = acp3x_dma_hw_free, + .pointer = acp3x_dma_pointer, + .mmap = acp3x_dma_mmap, }; struct snd_soc_dai_ops acp3x_dai_i2s_ops = { diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index f3d3e51..9f33e34 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -13,6 +13,26 @@ #define ACP3x_POWER_OFF_IN_PROGRESS 0x03 #define ACP3x_SOFT_RESET__SoftResetAudDone_MASK 0x00010001 +#define ACP_SRAM_PTE_OFFSET 0x02050000 +#define PAGE_SIZE_4K_ENABLE 0x2 +#define MEM_WINDOW_START 0x4000000 +#define PLAYBACK_FIFO_ADDR_OFFSET 0x400 +#define CAPTURE_FIFO_ADDR_OFFSET 0x500 + +#define PLAYBACK_MIN_NUM_PERIODS 2 +#define PLAYBACK_MAX_NUM_PERIODS 8 +#define PLAYBACK_MAX_PERIOD_SIZE 16384 +#define PLAYBACK_MIN_PERIOD_SIZE 4096 +#define CAPTURE_MIN_NUM_PERIODS 2 +#define CAPTURE_MAX_NUM_PERIODS 8 +#define CAPTURE_MAX_PERIOD_SIZE 16384 +#define CAPTURE_MIN_PERIOD_SIZE 4096 + +#define MAX_BUFFER (PLAYBACK_MAX_PERIOD_SIZE * PLAYBACK_MAX_NUM_PERIODS) +#define MIN_BUFFER MAX_BUFFER +#define FIFO_SIZE 0x100 +#define DMA_SIZE 0x40 + static inline u32 rv_readl(void __iomem *base_addr) { return readl(base_addr - ACP3x_PHY_BASE_ADDRESS); -- 2.7.4

2 1

[alsa-devel] [PATCH 07/11] ASoC: amd: add acp3x i2s ops
by Vijendar Mukunda 13 Nov '18

13 Nov '18

ACP3x has a i2s controller block for playback and capture. This patch adds ACP3x i2s DAI operations. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 89 +++++++++++++++++++++++++++++++++++-- 1 file changed, 86 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 06c1985..ed2c163 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -423,10 +423,93 @@ static struct snd_pcm_ops acp3x_dma_ops = { .mmap = acp3x_dma_mmap, }; +static int acp3x_dai_i2s_hwparams(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + u32 val = 0; + struct i2s_stream_instance *rtd = substream->runtime->private_data; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + rtd->xfer_resolution = 0x0; + break; + case SNDRV_PCM_FORMAT_S16_LE: + rtd->xfer_resolution = 0x02; + break; + case SNDRV_PCM_FORMAT_S24_LE: + rtd->xfer_resolution = 0x04; + break; + case SNDRV_PCM_FORMAT_S32_LE: + rtd->xfer_resolution = 0x05; + break; + default: + return -EINVAL; + } + val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_ITER); + val = val | (rtd->xfer_resolution << 3); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_ITER); + else + rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_IRER); + + return 0; +} + +static int acp3x_dai_i2s_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + int ret = 0; + struct i2s_stream_instance *rtd = substream->runtime->private_data; + u32 val, period_bytes; + + period_bytes = frames_to_bytes(substream->runtime, + substream->runtime->period_size); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + rv_writel(period_bytes, rtd->acp3x_base + + mmACP_BT_TX_INTR_WATERMARK_SIZE); + val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_ITER); + val = val | BIT(0); + rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_ITER); + } else { + rv_writel(period_bytes, rtd->acp3x_base + + mmACP_BT_RX_INTR_WATERMARK_SIZE); + val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_IRER); + val = val | BIT(0); + rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_IRER); + } + rv_writel(1, rtd->acp3x_base + mmACP_BTTDM_IER); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_ITER); + val = val & ~BIT(0); + rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_ITER); + } else { + val = rv_readl(rtd->acp3x_base + mmACP_BTTDM_IRER); + val = val & ~BIT(0); + rv_writel(val, rtd->acp3x_base + mmACP_BTTDM_IRER); + } + rv_writel(0, rtd->acp3x_base + mmACP_BTTDM_IER); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + struct snd_soc_dai_ops acp3x_dai_i2s_ops = { - .hw_params = NULL, - .trigger = NULL, - .set_fmt = NULL, + .hw_params = acp3x_dai_i2s_hwparams, + .trigger = acp3x_dai_i2s_trigger, }; static struct snd_soc_dai_driver acp3x_i2s_dai_driver = { -- 2.7.4

2 1

[alsa-devel] [PATCH 08/11] ASoC: amd: add acp3x tdm mode support
by Vijendar Mukunda 13 Nov '18

13 Nov '18

ACP3x I2S (CPU DAI) can act in normal I2S and TDM modes. Added support for TDM mode. Desired mode can be selected from ASoC machine driver. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 62 +++++++++++++++++++++++++++++++++++++ sound/soc/amd/raven/acp3x.h | 7 +++++ 2 files changed, 69 insertions(+) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index ed2c163..15a5f3b 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -27,7 +27,9 @@ #define DRV_NAME "acp3x-i2s-audio" struct i2s_dev_data { + bool tdm_mode; unsigned int i2s_irq; + u32 tdm_fmt; void __iomem *acp3x_base; struct snd_pcm_substream *play_stream; struct snd_pcm_substream *capture_stream; @@ -423,6 +425,64 @@ static struct snd_pcm_ops acp3x_dma_ops = { .mmap = acp3x_dma_mmap, }; + +static int acp3x_dai_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + + struct i2s_dev_data *adata = snd_soc_dai_get_drvdata(cpu_dai); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + adata->tdm_mode = false; + break; + case SND_SOC_DAIFMT_DSP_A: + adata->tdm_mode = true; + break; + default: + return -EINVAL; + } + + return 0; +} + +static int acp3x_dai_set_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, + u32 rx_mask, int slots, int slot_width) +{ + u32 val = 0; + u16 slot_len; + + struct i2s_dev_data *adata = snd_soc_dai_get_drvdata(cpu_dai); + + switch (slot_width) { + case SLOT_WIDTH_8: + slot_len = 8; + break; + case SLOT_WIDTH_16: + slot_len = 16; + break; + case SLOT_WIDTH_24: + slot_len = 24; + break; + case SLOT_WIDTH_32: + slot_len = 0; + break; + default: + return -EINVAL; + } + + val = rv_readl(adata->acp3x_base + mmACP_BTTDM_ITER); + rv_writel((val | 0x2), adata->acp3x_base + mmACP_BTTDM_ITER); + val = rv_readl(adata->acp3x_base + mmACP_BTTDM_IRER); + rv_writel((val | 0x2), adata->acp3x_base + mmACP_BTTDM_IRER); + + val = (FRM_LEN | (slots << 15) | (slot_len << 18)); + rv_writel(val, adata->acp3x_base + mmACP_BTTDM_TXFRMT); + rv_writel(val, adata->acp3x_base + mmACP_BTTDM_RXFRMT); + + adata->tdm_fmt = val; + return 0; +} + static int acp3x_dai_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -510,6 +570,8 @@ static int acp3x_dai_i2s_trigger(struct snd_pcm_substream *substream, struct snd_soc_dai_ops acp3x_dai_i2s_ops = { .hw_params = acp3x_dai_i2s_hwparams, .trigger = acp3x_dai_i2s_trigger, + .set_fmt = acp3x_dai_i2s_set_fmt, + .set_tdm_slot = acp3x_dai_set_tdm_slot, }; static struct snd_soc_dai_driver acp3x_i2s_dai_driver = { diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index 9f33e34..95fef85 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -32,6 +32,13 @@ #define MIN_BUFFER MAX_BUFFER #define FIFO_SIZE 0x100 #define DMA_SIZE 0x40 +#define FRM_LEN 0x100 + +#define SLOT_WIDTH_8 0x08 +#define SLOT_WIDTH_16 0x10 +#define SLOT_WIDTH_24 0x18 +#define SLOT_WIDTH_32 0x20 + static inline u32 rv_readl(void __iomem *base_addr) { -- 2.7.4

2 1

[alsa-devel] [PATCH 09/11] ASoC: amd: add acp3x runtime pm ops
by Vijendar Mukunda 13 Nov '18

13 Nov '18

Added runtime PM operations for ACP3x PCM platform device. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 39 +++++++++++++++++++++++++++++++++++++ 1 file changed, 39 insertions(+) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 15a5f3b..710c5c5 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -17,6 +17,7 @@ #include <linux/module.h> #include <linux/err.h> #include <linux/io.h> +#include <linux/pm_runtime.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -660,6 +661,9 @@ static int acp3x_audio_probe(struct platform_device *pdev) goto dev_err; } + pm_runtime_set_autosuspend_delay(&pdev->dev, 10000); + pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_enable(&pdev->dev); return 0; dev_err: status = acp3x_deinit(adata->acp3x_base); @@ -682,14 +686,49 @@ static int acp3x_audio_remove(struct platform_device *pdev) else dev_info(&pdev->dev, "ACP de-initialized\n"); + pm_runtime_disable(&pdev->dev); return 0; } +static int acp3x_pcm_runtime_suspend(struct device *dev) +{ + int status; + struct i2s_dev_data *adata = dev_get_drvdata(dev); + + status = acp3x_deinit(adata->acp3x_base); + if (status) + dev_err(dev, "ACP de-init failed\n"); + else + dev_info(dev, "ACP de-initialized\n"); + + rv_writel(0, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); + + return 0; +} + +static int acp3x_pcm_runtime_resume(struct device *dev) +{ + int status; + struct i2s_dev_data *adata = dev_get_drvdata(dev); + + status = acp3x_init(adata->acp3x_base); + if (status) + return -ENODEV; + rv_writel(1, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); + return 0; +} + +static const struct dev_pm_ops acp3x_pm_ops = { + .runtime_suspend = acp3x_pcm_runtime_suspend, + .runtime_resume = acp3x_pcm_runtime_resume, +}; + static struct platform_driver acp3x_dma_driver = { .probe = acp3x_audio_probe, .remove = acp3x_audio_remove, .driver = { .name = "acp3x_rv_i2s", + .pm = &acp3x_pm_ops, }, }; -- 2.7.4

2 1

[alsa-devel] [PATCH 10/11] ASoC: amd: add acp3x system resume pm op
by Vijendar Mukunda 13 Nov '18

13 Nov '18

When system wide suspend happens, ACP will be powered off. When system resumes, all the runtime configuration data for ACP needs to be programmed again. Added 'resume'pm call back to ACP pm ops. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 46 +++++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 710c5c5..2e61cef 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -690,6 +690,51 @@ static int acp3x_audio_remove(struct platform_device *pdev) return 0; } +static int acp3x_resume(struct device *dev) +{ + int status; + u32 val; + struct i2s_dev_data *adata = dev_get_drvdata(dev); + + status = acp3x_init(adata->acp3x_base); + if (status) + return -ENODEV; + + if (adata->play_stream && adata->play_stream->runtime) { + struct i2s_stream_instance *rtd = + adata->play_stream->runtime->private_data; + config_acp3x_dma(rtd, SNDRV_PCM_STREAM_PLAYBACK); + rv_writel((rtd->xfer_resolution << 3), + rtd->acp3x_base + mmACP_BTTDM_ITER); + if (adata->tdm_mode == true) { + rv_writel(adata->tdm_fmt, adata->acp3x_base + + mmACP_BTTDM_TXFRMT); + val = rv_readl(adata->acp3x_base + mmACP_BTTDM_ITER); + rv_writel((val | 0x2), adata->acp3x_base + + mmACP_BTTDM_ITER); + } + } + + if (adata->capture_stream && adata->capture_stream->runtime) { + struct i2s_stream_instance *rtd = + adata->capture_stream->runtime->private_data; + config_acp3x_dma(rtd, SNDRV_PCM_STREAM_CAPTURE); + rv_writel((rtd->xfer_resolution << 3), + rtd->acp3x_base + mmACP_BTTDM_IRER); + if (adata->tdm_mode == true) { + rv_writel(adata->tdm_fmt, adata->acp3x_base + + mmACP_BTTDM_RXFRMT); + val = rv_readl(adata->acp3x_base + mmACP_BTTDM_IRER); + rv_writel((val | 0x2), adata->acp3x_base + + mmACP_BTTDM_IRER); + } + } + + rv_writel(1, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); + return 0; +} + + static int acp3x_pcm_runtime_suspend(struct device *dev) { int status; @@ -721,6 +766,7 @@ static int acp3x_pcm_runtime_resume(struct device *dev) static const struct dev_pm_ops acp3x_pm_ops = { .runtime_suspend = acp3x_pcm_runtime_suspend, .runtime_resume = acp3x_pcm_runtime_resume, + .resume = acp3x_resume, }; static struct platform_driver acp3x_dma_driver = { -- 2.7.4

2 1

[alsa-devel] [PATCH 11/11] ASoC: amd: enable acp3x drivers build
by Vijendar Mukunda 13 Nov '18

13 Nov '18

ACP3x drivers can be built by selecting necessary kernel config option. The patch enables build support of the same. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Tested-by: Ravulapati Vishnu vardhan Rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/Kconfig | 5 +++++ sound/soc/amd/Makefile | 1 + sound/soc/amd/raven/Makefile | 4 ++++ 3 files changed, 10 insertions(+) create mode 100644 sound/soc/amd/raven/Makefile diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 58c1dcb..5f3b560 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -19,3 +19,8 @@ config SND_SOC_AMD_CZ_RT5645_MACH depends on SND_SOC_AMD_ACP && I2C help This option enables machine driver for rt5645. + +config SND_SOC_AMD_ACP3x + tristate "AMD Audio Coprocessor-v3.x support" + help + This option enables ACP v3.x I2S support on AMD platform diff --git a/sound/soc/amd/Makefile b/sound/soc/amd/Makefile index 79b0622..8e1c571 100644 --- a/sound/soc/amd/Makefile +++ b/sound/soc/amd/Makefile @@ -5,3 +5,4 @@ snd-soc-acp-rt5645-mach-objs := acp-rt5645.o obj-$(CONFIG_SND_SOC_AMD_ACP) += acp_audio_dma.o obj-$(CONFIG_SND_SOC_AMD_CZ_DA7219MX98357_MACH) += snd-soc-acp-da7219mx98357-mach.o obj-$(CONFIG_SND_SOC_AMD_CZ_RT5645_MACH) += snd-soc-acp-rt5645-mach.o +obj-$(CONFIG_SND_SOC_AMD_ACP3x) += raven/ diff --git a/sound/soc/amd/raven/Makefile b/sound/soc/amd/raven/Makefile new file mode 100644 index 0000000..8c03e6f --- /dev/null +++ b/sound/soc/amd/raven/Makefile @@ -0,0 +1,4 @@ +snd-pci-acp3x-objs := pci-acp3x.o +snd-acp3x-pcm-dma-objs := acp3x-pcm-dma.o +obj-$(CONFIG_SND_SOC_AMD_ACP3x) += snd-pci-acp3x.o +obj-$(CONFIG_SND_SOC_AMD_ACP3x) += snd-acp3x-pcm-dma.o -- 2.7.4

2 2

[alsa-devel] [PATCH v3 0/8] ASoC: omap-mcbsp: Cleanup and split core/sidetone support
by Peter Ujfalusi 13 Nov '18

13 Nov '18

Hi, Changes since v2: - Split the dma_data addr and maxburst cleanup patches - Added Jarkko's Acked-by tot he series. Changes since v1: - Two patch added to address the code move triggered few warnings from build. One of them is around mcbsp->pdata checks (not valid complaint), the other is that we set -EINVAL to unsigned variables (they were not used in that case) While preparing for merging the davinci and omap directories (I will send the series for that shortly) I have taken some time to do cleanup on the McBSP driver. The mcbsp.c/h files were the result when we moved code from arch/arm/plat-omap to sound/soc/omap/ a long time ago and it contained code for McBSP core and the OMAP3 sidetone functionality. With this series I tried to split the core and sidetone code to separate files. There were two reasons for this: 1. to have the OMAP specific DAI driver files prefixed with omap-* under the new sound/soc/ti/ directory 2. easier to find functions as all sidetone code is in separate file. Regards, Peter --- Peter Ujfalusi (8): ASoC: omap-mcbsp: Clean up dma_data addr initialization code ASoC: omap-mcbsp: Skip dma_data.maxburst initialization ASoC: omap-mcbsp: Clean up the interrupt handlers ASoC: omap-mcbsp: Simplify the mcbsp_start/_stop function parameters ASoC: omap-mcbsp: Move out the FIFO check from set_threshold and get_delay ASoC: omap-mcbsp: Re-arrange files for core McBSP and Sidetone function split ASoC: omap-mcbsp: Remove redundant check for mcbsp->pdata ASoC: omap-mcbsp: No need to initialize max_xx_thres when it is not used sound/soc/omap/Makefile | 2 +- sound/soc/omap/mcbsp.c | 1104 ----------------- sound/soc/omap/{mcbsp.h => omap-mcbsp-priv.h} | 126 +- sound/soc/omap/omap-mcbsp-st.c | 516 ++++++++ sound/soc/omap/omap-mcbsp.c | 852 ++++++++++--- sound/soc/omap/omap-mcbsp.h | 8 +- 6 files changed, 1278 insertions(+), 1330 deletions(-) delete mode 100644 sound/soc/omap/mcbsp.c rename sound/soc/omap/{mcbsp.h => omap-mcbsp-priv.h} (70%) create mode 100644 sound/soc/omap/omap-mcbsp-st.c -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

3 11

[alsa-devel] Applied "ASoC: omap-mcbsp: Skip dma_data.maxburst initialization" to the asoc tree
by Mark Brown 13 Nov '18

13 Nov '18

The patch ASoC: omap-mcbsp: Skip dma_data.maxburst initialization 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 c9ece9c29e263a4bda60b141b7b9dece4256b7dd Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Thu, 8 Nov 2018 09:29:54 +0200 Subject: [PATCH] ASoC: omap-mcbsp: Skip dma_data.maxburst initialization It is configured runtime so no need to initialize it. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Tested-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/omap/mcbsp.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index d6784f2887e4..8c3cedfcb878 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -1035,11 +1035,8 @@ int omap_mcbsp_init(struct platform_device *pdev) mcbsp->dma_data[0].addr = omap_mcbsp_dma_reg_params(mcbsp, SNDRV_PCM_STREAM_PLAYBACK); - mcbsp->dma_data[0].maxburst = 4; - mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, SNDRV_PCM_STREAM_CAPTURE); - mcbsp->dma_data[1].maxburst = 4; mcbsp->fclk = clk_get(&pdev->dev, "fck"); if (IS_ERR(mcbsp->fclk)) { -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: omap-mcbsp: Clean up the interrupt handlers" to the asoc tree
by Mark Brown 13 Nov '18

13 Nov '18

The patch ASoC: omap-mcbsp: Clean up the interrupt handlers 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 d63a7625a6dff8f75604b952de600d24b58b8b54 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Thu, 8 Nov 2018 09:29:55 +0200 Subject: [PATCH] ASoC: omap-mcbsp: Clean up the interrupt handlers The parameter name of dev_id is leftover from the old times when we passed numeric ID as data for the interrupt handlers. The mcbsp_rx and mcbsp_tx is misleading as they are pointers to the mcbsp struct. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Tested-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/omap/mcbsp.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 8c3cedfcb878..fca0b0e9186b 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -110,9 +110,9 @@ static void omap_mcbsp_dump_reg(struct omap_mcbsp *mcbsp) dev_dbg(mcbsp->dev, "***********************\n"); } -static irqreturn_t omap_mcbsp_irq_handler(int irq, void *dev_id) +static irqreturn_t omap_mcbsp_irq_handler(int irq, void *data) { - struct omap_mcbsp *mcbsp = dev_id; + struct omap_mcbsp *mcbsp = data; u16 irqst; irqst = MCBSP_READ(mcbsp, IRQST); @@ -151,37 +151,37 @@ static irqreturn_t omap_mcbsp_irq_handler(int irq, void *dev_id) return IRQ_HANDLED; } -static irqreturn_t omap_mcbsp_tx_irq_handler(int irq, void *dev_id) +static irqreturn_t omap_mcbsp_tx_irq_handler(int irq, void *data) { - struct omap_mcbsp *mcbsp_tx = dev_id; + struct omap_mcbsp *mcbsp = data; u16 irqst_spcr2; - irqst_spcr2 = MCBSP_READ(mcbsp_tx, SPCR2); - dev_dbg(mcbsp_tx->dev, "TX IRQ callback : 0x%x\n", irqst_spcr2); + irqst_spcr2 = MCBSP_READ(mcbsp, SPCR2); + dev_dbg(mcbsp->dev, "TX IRQ callback : 0x%x\n", irqst_spcr2); if (irqst_spcr2 & XSYNC_ERR) { - dev_err(mcbsp_tx->dev, "TX Frame Sync Error! : 0x%x\n", + dev_err(mcbsp->dev, "TX Frame Sync Error! : 0x%x\n", irqst_spcr2); /* Writing zero to XSYNC_ERR clears the IRQ */ - MCBSP_WRITE(mcbsp_tx, SPCR2, MCBSP_READ_CACHE(mcbsp_tx, SPCR2)); + MCBSP_WRITE(mcbsp, SPCR2, MCBSP_READ_CACHE(mcbsp, SPCR2)); } return IRQ_HANDLED; } -static irqreturn_t omap_mcbsp_rx_irq_handler(int irq, void *dev_id) +static irqreturn_t omap_mcbsp_rx_irq_handler(int irq, void *data) { - struct omap_mcbsp *mcbsp_rx = dev_id; + struct omap_mcbsp *mcbsp = data; u16 irqst_spcr1; - irqst_spcr1 = MCBSP_READ(mcbsp_rx, SPCR1); - dev_dbg(mcbsp_rx->dev, "RX IRQ callback : 0x%x\n", irqst_spcr1); + irqst_spcr1 = MCBSP_READ(mcbsp, SPCR1); + dev_dbg(mcbsp->dev, "RX IRQ callback : 0x%x\n", irqst_spcr1); if (irqst_spcr1 & RSYNC_ERR) { - dev_err(mcbsp_rx->dev, "RX Frame Sync Error! : 0x%x\n", + dev_err(mcbsp->dev, "RX Frame Sync Error! : 0x%x\n", irqst_spcr1); /* Writing zero to RSYNC_ERR clears the IRQ */ - MCBSP_WRITE(mcbsp_rx, SPCR1, MCBSP_READ_CACHE(mcbsp_rx, SPCR1)); + MCBSP_WRITE(mcbsp, SPCR1, MCBSP_READ_CACHE(mcbsp, SPCR1)); } return IRQ_HANDLED; -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: omap-mcbsp: Simplify the mcbsp_start/_stop function parameters" to the asoc tree
by Mark Brown 13 Nov '18

13 Nov '18

The patch ASoC: omap-mcbsp: Simplify the mcbsp_start/_stop function parameters 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 59d177f65f50df07f5f0b461c968d2fab7597af1 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Thu, 8 Nov 2018 09:29:56 +0200 Subject: [PATCH] ASoC: omap-mcbsp: Simplify the mcbsp_start/_stop function parameters We either start/stop TX or RX, never both. Move the tx/rx direction selection within the functions. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Tested-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/omap/mcbsp.c | 8 ++++++-- sound/soc/omap/mcbsp.h | 4 ++-- sound/soc/omap/omap-mcbsp.c | 9 ++++----- 3 files changed, 12 insertions(+), 9 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index fca0b0e9186b..7ff22561f00f 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -637,8 +637,10 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) * If no transmitter or receiver is active prior calling, then sample-rate * generator and frame sync are started. */ -void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx) +void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int stream) { + int tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); + int rx = !tx; int enable_srg = 0; u16 w; @@ -694,8 +696,10 @@ void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx) omap_mcbsp_dump_reg(mcbsp); } -void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx) +void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int stream) { + int tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); + int rx = !tx; int idle; u16 w; diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index 46ae1269a698..92472c6ef358 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -339,8 +339,8 @@ u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp); int omap_mcbsp_get_dma_op_mode(struct omap_mcbsp *mcbsp); int omap_mcbsp_request(struct omap_mcbsp *mcbsp); void omap_mcbsp_free(struct omap_mcbsp *mcbsp); -void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx); -void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx); +void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int stream); +void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int stream); /* McBSP functional clock source changing function */ int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index caa24c463b3e..a18b7ecc3a2e 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -201,27 +201,26 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - int err = 0, play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: mcbsp->active++; - omap_mcbsp_start(mcbsp, play, !play); + omap_mcbsp_start(mcbsp, substream->stream); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - omap_mcbsp_stop(mcbsp, play, !play); + omap_mcbsp_stop(mcbsp, substream->stream); mcbsp->active--; break; default: - err = -EINVAL; + return -EINVAL; } - return err; + return 0; } static snd_pcm_sframes_t omap_mcbsp_dai_delay( -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: omap-mcbsp: Move out the FIFO check from set_threshold and get_delay" to the asoc tree
by Mark Brown 13 Nov '18

13 Nov '18

The patch ASoC: omap-mcbsp: Move out the FIFO check from set_threshold and get_delay 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 be51c576e8495fd2cffbababad6de7e0a0a562ba Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Thu, 8 Nov 2018 09:29:57 +0200 Subject: [PATCH] ASoC: omap-mcbsp: Move out the FIFO check from set_threshold and get_delay Check if the McBSP have FIFO in the omap_mcbsp_set_threshold() and omap_mcbsp_dai_delay() delay function to skip calling the lower layer if it is not needed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Tested-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/omap/mcbsp.c | 12 ------------ sound/soc/omap/omap-mcbsp.c | 8 ++++++++ 2 files changed, 8 insertions(+), 12 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 7ff22561f00f..b19168f5c110 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -450,9 +450,6 @@ int omap_st_is_enabled(struct omap_mcbsp *mcbsp) */ void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) { - if (mcbsp->pdata->buffer_size == 0) - return; - if (threshold && threshold <= mcbsp->max_tx_thres) MCBSP_WRITE(mcbsp, THRSH2, threshold - 1); } @@ -464,9 +461,6 @@ void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) */ void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) { - if (mcbsp->pdata->buffer_size == 0) - return; - if (threshold && threshold <= mcbsp->max_rx_thres) MCBSP_WRITE(mcbsp, THRSH1, threshold - 1); } @@ -478,9 +472,6 @@ u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp) { u16 buffstat; - if (mcbsp->pdata->buffer_size == 0) - return 0; - /* Returns the number of free locations in the buffer */ buffstat = MCBSP_READ(mcbsp, XBUFFSTAT); @@ -496,9 +487,6 @@ u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp) { u16 buffstat, threshold; - if (mcbsp->pdata->buffer_size == 0) - return 0; - /* Returns the number of used locations in the buffer */ buffstat = MCBSP_READ(mcbsp, RBUFFSTAT); /* RX threshold */ diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index a18b7ecc3a2e..69a6b8ad6d42 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -71,6 +71,10 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream, struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); int words; + /* No need to proceed further if McBSP does not have FIFO */ + if (mcbsp->pdata->buffer_size == 0) + return; + /* * Configure McBSP threshold based on either: * packet_size, when the sDMA is in packet mode, or based on the @@ -233,6 +237,10 @@ static snd_pcm_sframes_t omap_mcbsp_dai_delay( u16 fifo_use; snd_pcm_sframes_t delay; + /* No need to proceed further if McBSP does not have FIFO */ + if (mcbsp->pdata->buffer_size == 0) + return 0; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_use = omap_mcbsp_get_tx_delay(mcbsp); else -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: omap-mcbsp: Re-arrange files for core McBSP and Sidetone function split" to the asoc tree
by Mark Brown 13 Nov '18

13 Nov '18

The patch ASoC: omap-mcbsp: Re-arrange files for core McBSP and Sidetone function split 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 9c34d023dc35a4b3d005d94db512742dfcdc8f82 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Thu, 8 Nov 2018 09:29:58 +0200 Subject: [PATCH] ASoC: omap-mcbsp: Re-arrange files for core McBSP and Sidetone function split The mcbsp.c was copied a while back from arch/arm/plat-omap/mcbsp.c and it contained a mix of McBSP and McBSP sidetone functions. Create new file structure with the following split: omap-mcbsp.c - McBSP related functions omap-mcbsp-st.c - McBSP sidetone functionality omap-mcbsp-priv.h - Private header for internal use omap-mcbsp.h - Header for user drivers I have tried to do the code move with minimal code change, cleanup patches can be based on the new structure. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Tested-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/omap/Makefile | 2 +- sound/soc/omap/mcbsp.c | 1094 ----------------- sound/soc/omap/{mcbsp.h => omap-mcbsp-priv.h} | 126 +- sound/soc/omap/omap-mcbsp-st.c | 516 ++++++++ sound/soc/omap/omap-mcbsp.c | 838 ++++++++++--- sound/soc/omap/omap-mcbsp.h | 8 +- 6 files changed, 1269 insertions(+), 1315 deletions(-) delete mode 100644 sound/soc/omap/mcbsp.c rename sound/soc/omap/{mcbsp.h => omap-mcbsp-priv.h} (70%) create mode 100644 sound/soc/omap/omap-mcbsp-st.c diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index af50cdd6eea9..d005338dd13c 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -2,7 +2,7 @@ # OMAP Platform Support snd-soc-sdma-objs := sdma-pcm.o snd-soc-omap-dmic-objs := omap-dmic.o -snd-soc-omap-mcbsp-objs := omap-mcbsp.o mcbsp.o +snd-soc-omap-mcbsp-objs := omap-mcbsp.o omap-mcbsp-st.o snd-soc-omap-mcpdm-objs := omap-mcpdm.o snd-soc-omap-hdmi-audio-objs := omap-hdmi-audio.o diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c deleted file mode 100644 index b19168f5c110..000000000000 --- a/sound/soc/omap/mcbsp.c +++ /dev/null @@ -1,1094 +0,0 @@ -/* - * sound/soc/omap/mcbsp.c - * - * Copyright (C) 2004 Nokia Corporation - * Author: Samuel Ortiz <samuel.ortiz(a)nokia.com> - * - * Contact: Jarkko Nikula <jarkko.nikula(a)bitmer.com> - * Peter Ujfalusi <peter.ujfalusi(a)ti.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. - * - * Multichannel mode not supported. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/device.h> -#include <linux/platform_device.h> -#include <linux/interrupt.h> -#include <linux/err.h> -#include <linux/clk.h> -#include <linux/delay.h> -#include <linux/io.h> -#include <linux/slab.h> -#include <linux/pm_runtime.h> - -#include <linux/platform_data/asoc-ti-mcbsp.h> - -#include "mcbsp.h" - -static void omap_mcbsp_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) -{ - void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step; - - if (mcbsp->pdata->reg_size == 2) { - ((u16 *)mcbsp->reg_cache)[reg] = (u16)val; - writew_relaxed((u16)val, addr); - } else { - ((u32 *)mcbsp->reg_cache)[reg] = val; - writel_relaxed(val, addr); - } -} - -static int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, bool from_cache) -{ - void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step; - - if (mcbsp->pdata->reg_size == 2) { - return !from_cache ? readw_relaxed(addr) : - ((u16 *)mcbsp->reg_cache)[reg]; - } else { - return !from_cache ? readl_relaxed(addr) : - ((u32 *)mcbsp->reg_cache)[reg]; - } -} - -static void omap_mcbsp_st_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) -{ - writel_relaxed(val, mcbsp->st_data->io_base_st + reg); -} - -static int omap_mcbsp_st_read(struct omap_mcbsp *mcbsp, u16 reg) -{ - return readl_relaxed(mcbsp->st_data->io_base_st + reg); -} - -#define MCBSP_READ(mcbsp, reg) \ - omap_mcbsp_read(mcbsp, OMAP_MCBSP_REG_##reg, 0) -#define MCBSP_WRITE(mcbsp, reg, val) \ - omap_mcbsp_write(mcbsp, OMAP_MCBSP_REG_##reg, val) -#define MCBSP_READ_CACHE(mcbsp, reg) \ - omap_mcbsp_read(mcbsp, OMAP_MCBSP_REG_##reg, 1) - -#define MCBSP_ST_READ(mcbsp, reg) \ - omap_mcbsp_st_read(mcbsp, OMAP_ST_REG_##reg) -#define MCBSP_ST_WRITE(mcbsp, reg, val) \ - omap_mcbsp_st_write(mcbsp, OMAP_ST_REG_##reg, val) - -static void omap_mcbsp_dump_reg(struct omap_mcbsp *mcbsp) -{ - dev_dbg(mcbsp->dev, "**** McBSP%d regs ****\n", mcbsp->id); - dev_dbg(mcbsp->dev, "DRR2: 0x%04x\n", - MCBSP_READ(mcbsp, DRR2)); - dev_dbg(mcbsp->dev, "DRR1: 0x%04x\n", - MCBSP_READ(mcbsp, DRR1)); - dev_dbg(mcbsp->dev, "DXR2: 0x%04x\n", - MCBSP_READ(mcbsp, DXR2)); - dev_dbg(mcbsp->dev, "DXR1: 0x%04x\n", - MCBSP_READ(mcbsp, DXR1)); - dev_dbg(mcbsp->dev, "SPCR2: 0x%04x\n", - MCBSP_READ(mcbsp, SPCR2)); - dev_dbg(mcbsp->dev, "SPCR1: 0x%04x\n", - MCBSP_READ(mcbsp, SPCR1)); - dev_dbg(mcbsp->dev, "RCR2: 0x%04x\n", - MCBSP_READ(mcbsp, RCR2)); - dev_dbg(mcbsp->dev, "RCR1: 0x%04x\n", - MCBSP_READ(mcbsp, RCR1)); - dev_dbg(mcbsp->dev, "XCR2: 0x%04x\n", - MCBSP_READ(mcbsp, XCR2)); - dev_dbg(mcbsp->dev, "XCR1: 0x%04x\n", - MCBSP_READ(mcbsp, XCR1)); - dev_dbg(mcbsp->dev, "SRGR2: 0x%04x\n", - MCBSP_READ(mcbsp, SRGR2)); - dev_dbg(mcbsp->dev, "SRGR1: 0x%04x\n", - MCBSP_READ(mcbsp, SRGR1)); - dev_dbg(mcbsp->dev, "PCR0: 0x%04x\n", - MCBSP_READ(mcbsp, PCR0)); - dev_dbg(mcbsp->dev, "***********************\n"); -} - -static irqreturn_t omap_mcbsp_irq_handler(int irq, void *data) -{ - struct omap_mcbsp *mcbsp = data; - u16 irqst; - - irqst = MCBSP_READ(mcbsp, IRQST); - dev_dbg(mcbsp->dev, "IRQ callback : 0x%x\n", irqst); - - if (irqst & RSYNCERREN) - dev_err(mcbsp->dev, "RX Frame Sync Error!\n"); - if (irqst & RFSREN) - dev_dbg(mcbsp->dev, "RX Frame Sync\n"); - if (irqst & REOFEN) - dev_dbg(mcbsp->dev, "RX End Of Frame\n"); - if (irqst & RRDYEN) - dev_dbg(mcbsp->dev, "RX Buffer Threshold Reached\n"); - if (irqst & RUNDFLEN) - dev_err(mcbsp->dev, "RX Buffer Underflow!\n"); - if (irqst & ROVFLEN) - dev_err(mcbsp->dev, "RX Buffer Overflow!\n"); - - if (irqst & XSYNCERREN) - dev_err(mcbsp->dev, "TX Frame Sync Error!\n"); - if (irqst & XFSXEN) - dev_dbg(mcbsp->dev, "TX Frame Sync\n"); - if (irqst & XEOFEN) - dev_dbg(mcbsp->dev, "TX End Of Frame\n"); - if (irqst & XRDYEN) - dev_dbg(mcbsp->dev, "TX Buffer threshold Reached\n"); - if (irqst & XUNDFLEN) - dev_err(mcbsp->dev, "TX Buffer Underflow!\n"); - if (irqst & XOVFLEN) - dev_err(mcbsp->dev, "TX Buffer Overflow!\n"); - if (irqst & XEMPTYEOFEN) - dev_dbg(mcbsp->dev, "TX Buffer empty at end of frame\n"); - - MCBSP_WRITE(mcbsp, IRQST, irqst); - - return IRQ_HANDLED; -} - -static irqreturn_t omap_mcbsp_tx_irq_handler(int irq, void *data) -{ - struct omap_mcbsp *mcbsp = data; - u16 irqst_spcr2; - - irqst_spcr2 = MCBSP_READ(mcbsp, SPCR2); - dev_dbg(mcbsp->dev, "TX IRQ callback : 0x%x\n", irqst_spcr2); - - if (irqst_spcr2 & XSYNC_ERR) { - dev_err(mcbsp->dev, "TX Frame Sync Error! : 0x%x\n", - irqst_spcr2); - /* Writing zero to XSYNC_ERR clears the IRQ */ - MCBSP_WRITE(mcbsp, SPCR2, MCBSP_READ_CACHE(mcbsp, SPCR2)); - } - - return IRQ_HANDLED; -} - -static irqreturn_t omap_mcbsp_rx_irq_handler(int irq, void *data) -{ - struct omap_mcbsp *mcbsp = data; - u16 irqst_spcr1; - - irqst_spcr1 = MCBSP_READ(mcbsp, SPCR1); - dev_dbg(mcbsp->dev, "RX IRQ callback : 0x%x\n", irqst_spcr1); - - if (irqst_spcr1 & RSYNC_ERR) { - dev_err(mcbsp->dev, "RX Frame Sync Error! : 0x%x\n", - irqst_spcr1); - /* Writing zero to RSYNC_ERR clears the IRQ */ - MCBSP_WRITE(mcbsp, SPCR1, MCBSP_READ_CACHE(mcbsp, SPCR1)); - } - - return IRQ_HANDLED; -} - -/* - * omap_mcbsp_config simply write a config to the - * appropriate McBSP. - * You either call this function or set the McBSP registers - * by yourself before calling omap_mcbsp_start(). - */ -void omap_mcbsp_config(struct omap_mcbsp *mcbsp, - const struct omap_mcbsp_reg_cfg *config) -{ - dev_dbg(mcbsp->dev, "Configuring McBSP%d phys_base: 0x%08lx\n", - mcbsp->id, mcbsp->phys_base); - - /* We write the given config */ - MCBSP_WRITE(mcbsp, SPCR2, config->spcr2); - MCBSP_WRITE(mcbsp, SPCR1, config->spcr1); - MCBSP_WRITE(mcbsp, RCR2, config->rcr2); - MCBSP_WRITE(mcbsp, RCR1, config->rcr1); - MCBSP_WRITE(mcbsp, XCR2, config->xcr2); - MCBSP_WRITE(mcbsp, XCR1, config->xcr1); - MCBSP_WRITE(mcbsp, SRGR2, config->srgr2); - MCBSP_WRITE(mcbsp, SRGR1, config->srgr1); - MCBSP_WRITE(mcbsp, MCR2, config->mcr2); - MCBSP_WRITE(mcbsp, MCR1, config->mcr1); - MCBSP_WRITE(mcbsp, PCR0, config->pcr0); - if (mcbsp->pdata->has_ccr) { - MCBSP_WRITE(mcbsp, XCCR, config->xccr); - MCBSP_WRITE(mcbsp, RCCR, config->rccr); - } - /* Enable wakeup behavior */ - if (mcbsp->pdata->has_wakeup) - MCBSP_WRITE(mcbsp, WAKEUPEN, XRDYEN | RRDYEN); - - /* Enable TX/RX sync error interrupts by default */ - if (mcbsp->irq) - MCBSP_WRITE(mcbsp, IRQEN, RSYNCERREN | XSYNCERREN | - RUNDFLEN | ROVFLEN | XUNDFLEN | XOVFLEN); -} - -/** - * omap_mcbsp_dma_reg_params - returns the address of mcbsp data register - * @mcbsp: omap_mcbsp struct for the McBSP instance - * @stream: Stream direction (playback/capture) - * - * Returns the address of mcbsp data transmit register or data receive register - * to be used by DMA for transferring/receiving data - */ -static int omap_mcbsp_dma_reg_params(struct omap_mcbsp *mcbsp, - unsigned int stream) -{ - int data_reg; - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (mcbsp->pdata->reg_size == 2) - data_reg = OMAP_MCBSP_REG_DXR1; - else - data_reg = OMAP_MCBSP_REG_DXR; - } else { - if (mcbsp->pdata->reg_size == 2) - data_reg = OMAP_MCBSP_REG_DRR1; - else - data_reg = OMAP_MCBSP_REG_DRR; - } - - return mcbsp->phys_dma_base + data_reg * mcbsp->pdata->reg_step; -} - -static void omap_st_on(struct omap_mcbsp *mcbsp) -{ - unsigned int w; - - if (mcbsp->pdata->force_ick_on) - mcbsp->pdata->force_ick_on(mcbsp->st_data->mcbsp_iclk, true); - - /* Disable Sidetone clock auto-gating for normal operation */ - w = MCBSP_ST_READ(mcbsp, SYSCONFIG); - MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w & ~(ST_AUTOIDLE)); - - /* Enable McBSP Sidetone */ - w = MCBSP_READ(mcbsp, SSELCR); - MCBSP_WRITE(mcbsp, SSELCR, w | SIDETONEEN); - - /* Enable Sidetone from Sidetone Core */ - w = MCBSP_ST_READ(mcbsp, SSELCR); - MCBSP_ST_WRITE(mcbsp, SSELCR, w | ST_SIDETONEEN); -} - -static void omap_st_off(struct omap_mcbsp *mcbsp) -{ - unsigned int w; - - w = MCBSP_ST_READ(mcbsp, SSELCR); - MCBSP_ST_WRITE(mcbsp, SSELCR, w & ~(ST_SIDETONEEN)); - - w = MCBSP_READ(mcbsp, SSELCR); - MCBSP_WRITE(mcbsp, SSELCR, w & ~(SIDETONEEN)); - - /* Enable Sidetone clock auto-gating to reduce power consumption */ - w = MCBSP_ST_READ(mcbsp, SYSCONFIG); - MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w | ST_AUTOIDLE); - - if (mcbsp->pdata->force_ick_on) - mcbsp->pdata->force_ick_on(mcbsp->st_data->mcbsp_iclk, false); -} - -static void omap_st_fir_write(struct omap_mcbsp *mcbsp, s16 *fir) -{ - u16 val, i; - - val = MCBSP_ST_READ(mcbsp, SSELCR); - - if (val & ST_COEFFWREN) - MCBSP_ST_WRITE(mcbsp, SSELCR, val & ~(ST_COEFFWREN)); - - MCBSP_ST_WRITE(mcbsp, SSELCR, val | ST_COEFFWREN); - - for (i = 0; i < 128; i++) - MCBSP_ST_WRITE(mcbsp, SFIRCR, fir[i]); - - i = 0; - - val = MCBSP_ST_READ(mcbsp, SSELCR); - while (!(val & ST_COEFFWRDONE) && (++i < 1000)) - val = MCBSP_ST_READ(mcbsp, SSELCR); - - MCBSP_ST_WRITE(mcbsp, SSELCR, val & ~(ST_COEFFWREN)); - - if (i == 1000) - dev_err(mcbsp->dev, "McBSP FIR load error!\n"); -} - -static void omap_st_chgain(struct omap_mcbsp *mcbsp) -{ - u16 w; - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - - w = MCBSP_ST_READ(mcbsp, SSELCR); - - MCBSP_ST_WRITE(mcbsp, SGAINCR, ST_CH0GAIN(st_data->ch0gain) | \ - ST_CH1GAIN(st_data->ch1gain)); -} - -int omap_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, s16 chgain) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - int ret = 0; - - if (!st_data) - return -ENOENT; - - spin_lock_irq(&mcbsp->lock); - if (channel == 0) - st_data->ch0gain = chgain; - else if (channel == 1) - st_data->ch1gain = chgain; - else - ret = -EINVAL; - - if (st_data->enabled) - omap_st_chgain(mcbsp); - spin_unlock_irq(&mcbsp->lock); - - return ret; -} - -int omap_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, s16 *chgain) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - int ret = 0; - - if (!st_data) - return -ENOENT; - - spin_lock_irq(&mcbsp->lock); - if (channel == 0) - *chgain = st_data->ch0gain; - else if (channel == 1) - *chgain = st_data->ch1gain; - else - ret = -EINVAL; - spin_unlock_irq(&mcbsp->lock); - - return ret; -} - -static int omap_st_start(struct omap_mcbsp *mcbsp) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - - if (st_data->enabled && !st_data->running) { - omap_st_fir_write(mcbsp, st_data->taps); - omap_st_chgain(mcbsp); - - if (!mcbsp->free) { - omap_st_on(mcbsp); - st_data->running = 1; - } - } - - return 0; -} - -int omap_st_enable(struct omap_mcbsp *mcbsp) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - - if (!st_data) - return -ENODEV; - - spin_lock_irq(&mcbsp->lock); - st_data->enabled = 1; - omap_st_start(mcbsp); - spin_unlock_irq(&mcbsp->lock); - - return 0; -} - -static int omap_st_stop(struct omap_mcbsp *mcbsp) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - - if (st_data->running) { - if (!mcbsp->free) { - omap_st_off(mcbsp); - st_data->running = 0; - } - } - - return 0; -} - -int omap_st_disable(struct omap_mcbsp *mcbsp) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - int ret = 0; - - if (!st_data) - return -ENODEV; - - spin_lock_irq(&mcbsp->lock); - omap_st_stop(mcbsp); - st_data->enabled = 0; - spin_unlock_irq(&mcbsp->lock); - - return ret; -} - -int omap_st_is_enabled(struct omap_mcbsp *mcbsp) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - - if (!st_data) - return -ENODEV; - - return st_data->enabled; -} - -/* - * omap_mcbsp_set_rx_threshold configures the transmit threshold in words. - * The threshold parameter is 1 based, and it is converted (threshold - 1) - * for the THRSH2 register. - */ -void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) -{ - if (threshold && threshold <= mcbsp->max_tx_thres) - MCBSP_WRITE(mcbsp, THRSH2, threshold - 1); -} - -/* - * omap_mcbsp_set_rx_threshold configures the receive threshold in words. - * The threshold parameter is 1 based, and it is converted (threshold - 1) - * for the THRSH1 register. - */ -void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) -{ - if (threshold && threshold <= mcbsp->max_rx_thres) - MCBSP_WRITE(mcbsp, THRSH1, threshold - 1); -} - -/* - * omap_mcbsp_get_tx_delay returns the number of used slots in the McBSP FIFO - */ -u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp) -{ - u16 buffstat; - - /* Returns the number of free locations in the buffer */ - buffstat = MCBSP_READ(mcbsp, XBUFFSTAT); - - /* Number of slots are different in McBSP ports */ - return mcbsp->pdata->buffer_size - buffstat; -} - -/* - * omap_mcbsp_get_rx_delay returns the number of free slots in the McBSP FIFO - * to reach the threshold value (when the DMA will be triggered to read it) - */ -u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp) -{ - u16 buffstat, threshold; - - /* Returns the number of used locations in the buffer */ - buffstat = MCBSP_READ(mcbsp, RBUFFSTAT); - /* RX threshold */ - threshold = MCBSP_READ(mcbsp, THRSH1); - - /* Return the number of location till we reach the threshold limit */ - if (threshold <= buffstat) - return 0; - else - return threshold - buffstat; -} - -int omap_mcbsp_request(struct omap_mcbsp *mcbsp) -{ - void *reg_cache; - int err; - - reg_cache = kzalloc(mcbsp->reg_cache_size, GFP_KERNEL); - if (!reg_cache) { - return -ENOMEM; - } - - spin_lock(&mcbsp->lock); - if (!mcbsp->free) { - dev_err(mcbsp->dev, "McBSP%d is currently in use\n", - mcbsp->id); - err = -EBUSY; - goto err_kfree; - } - - mcbsp->free = false; - mcbsp->reg_cache = reg_cache; - spin_unlock(&mcbsp->lock); - - if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->request) - mcbsp->pdata->ops->request(mcbsp->id - 1); - - /* - * Make sure that transmitter, receiver and sample-rate generator are - * not running before activating IRQs. - */ - MCBSP_WRITE(mcbsp, SPCR1, 0); - MCBSP_WRITE(mcbsp, SPCR2, 0); - - if (mcbsp->irq) { - err = request_irq(mcbsp->irq, omap_mcbsp_irq_handler, 0, - "McBSP", (void *)mcbsp); - if (err != 0) { - dev_err(mcbsp->dev, "Unable to request IRQ\n"); - goto err_clk_disable; - } - } else { - err = request_irq(mcbsp->tx_irq, omap_mcbsp_tx_irq_handler, 0, - "McBSP TX", (void *)mcbsp); - if (err != 0) { - dev_err(mcbsp->dev, "Unable to request TX IRQ\n"); - goto err_clk_disable; - } - - err = request_irq(mcbsp->rx_irq, omap_mcbsp_rx_irq_handler, 0, - "McBSP RX", (void *)mcbsp); - if (err != 0) { - dev_err(mcbsp->dev, "Unable to request RX IRQ\n"); - goto err_free_irq; - } - } - - return 0; -err_free_irq: - free_irq(mcbsp->tx_irq, (void *)mcbsp); -err_clk_disable: - if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) - mcbsp->pdata->ops->free(mcbsp->id - 1); - - /* Disable wakeup behavior */ - if (mcbsp->pdata->has_wakeup) - MCBSP_WRITE(mcbsp, WAKEUPEN, 0); - - spin_lock(&mcbsp->lock); - mcbsp->free = true; - mcbsp->reg_cache = NULL; -err_kfree: - spin_unlock(&mcbsp->lock); - kfree(reg_cache); - - return err; -} - -void omap_mcbsp_free(struct omap_mcbsp *mcbsp) -{ - void *reg_cache; - - if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) - mcbsp->pdata->ops->free(mcbsp->id - 1); - - /* Disable wakeup behavior */ - if (mcbsp->pdata->has_wakeup) - MCBSP_WRITE(mcbsp, WAKEUPEN, 0); - - /* Disable interrupt requests */ - if (mcbsp->irq) - MCBSP_WRITE(mcbsp, IRQEN, 0); - - if (mcbsp->irq) { - free_irq(mcbsp->irq, (void *)mcbsp); - } else { - free_irq(mcbsp->rx_irq, (void *)mcbsp); - free_irq(mcbsp->tx_irq, (void *)mcbsp); - } - - reg_cache = mcbsp->reg_cache; - - /* - * Select CLKS source from internal source unconditionally before - * marking the McBSP port as free. - * If the external clock source via MCBSP_CLKS pin has been selected the - * system will refuse to enter idle if the CLKS pin source is not reset - * back to internal source. - */ - if (!mcbsp_omap1()) - omap2_mcbsp_set_clks_src(mcbsp, MCBSP_CLKS_PRCM_SRC); - - spin_lock(&mcbsp->lock); - if (mcbsp->free) - dev_err(mcbsp->dev, "McBSP%d was not reserved\n", mcbsp->id); - else - mcbsp->free = true; - mcbsp->reg_cache = NULL; - spin_unlock(&mcbsp->lock); - - kfree(reg_cache); -} - -/* - * Here we start the McBSP, by enabling transmitter, receiver or both. - * If no transmitter or receiver is active prior calling, then sample-rate - * generator and frame sync are started. - */ -void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int stream) -{ - int tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); - int rx = !tx; - int enable_srg = 0; - u16 w; - - if (mcbsp->st_data) - omap_st_start(mcbsp); - - /* Only enable SRG, if McBSP is master */ - w = MCBSP_READ_CACHE(mcbsp, PCR0); - if (w & (FSXM | FSRM | CLKXM | CLKRM)) - enable_srg = !((MCBSP_READ_CACHE(mcbsp, SPCR2) | - MCBSP_READ_CACHE(mcbsp, SPCR1)) & 1); - - if (enable_srg) { - /* Start the sample generator */ - w = MCBSP_READ_CACHE(mcbsp, SPCR2); - MCBSP_WRITE(mcbsp, SPCR2, w | (1 << 6)); - } - - /* Enable transmitter and receiver */ - tx &= 1; - w = MCBSP_READ_CACHE(mcbsp, SPCR2); - MCBSP_WRITE(mcbsp, SPCR2, w | tx); - - rx &= 1; - w = MCBSP_READ_CACHE(mcbsp, SPCR1); - MCBSP_WRITE(mcbsp, SPCR1, w | rx); - - /* - * Worst case: CLKSRG*2 = 8000khz: (1/8000) * 2 * 2 usec - * REVISIT: 100us may give enough time for two CLKSRG, however - * due to some unknown PM related, clock gating etc. reason it - * is now at 500us. - */ - udelay(500); - - if (enable_srg) { - /* Start frame sync */ - w = MCBSP_READ_CACHE(mcbsp, SPCR2); - MCBSP_WRITE(mcbsp, SPCR2, w | (1 << 7)); - } - - if (mcbsp->pdata->has_ccr) { - /* Release the transmitter and receiver */ - w = MCBSP_READ_CACHE(mcbsp, XCCR); - w &= ~(tx ? XDISABLE : 0); - MCBSP_WRITE(mcbsp, XCCR, w); - w = MCBSP_READ_CACHE(mcbsp, RCCR); - w &= ~(rx ? RDISABLE : 0); - MCBSP_WRITE(mcbsp, RCCR, w); - } - - /* Dump McBSP Regs */ - omap_mcbsp_dump_reg(mcbsp); -} - -void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int stream) -{ - int tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); - int rx = !tx; - int idle; - u16 w; - - /* Reset transmitter */ - tx &= 1; - if (mcbsp->pdata->has_ccr) { - w = MCBSP_READ_CACHE(mcbsp, XCCR); - w |= (tx ? XDISABLE : 0); - MCBSP_WRITE(mcbsp, XCCR, w); - } - w = MCBSP_READ_CACHE(mcbsp, SPCR2); - MCBSP_WRITE(mcbsp, SPCR2, w & ~tx); - - /* Reset receiver */ - rx &= 1; - if (mcbsp->pdata->has_ccr) { - w = MCBSP_READ_CACHE(mcbsp, RCCR); - w |= (rx ? RDISABLE : 0); - MCBSP_WRITE(mcbsp, RCCR, w); - } - w = MCBSP_READ_CACHE(mcbsp, SPCR1); - MCBSP_WRITE(mcbsp, SPCR1, w & ~rx); - - idle = !((MCBSP_READ_CACHE(mcbsp, SPCR2) | - MCBSP_READ_CACHE(mcbsp, SPCR1)) & 1); - - if (idle) { - /* Reset the sample rate generator */ - w = MCBSP_READ_CACHE(mcbsp, SPCR2); - MCBSP_WRITE(mcbsp, SPCR2, w & ~(1 << 6)); - } - - if (mcbsp->st_data) - omap_st_stop(mcbsp); -} - -int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) -{ - struct clk *fck_src; - const char *src; - int r; - - if (fck_src_id == MCBSP_CLKS_PAD_SRC) - src = "pad_fck"; - else if (fck_src_id == MCBSP_CLKS_PRCM_SRC) - src = "prcm_fck"; - else - return -EINVAL; - - fck_src = clk_get(mcbsp->dev, src); - if (IS_ERR(fck_src)) { - dev_err(mcbsp->dev, "CLKS: could not clk_get() %s\n", src); - return -EINVAL; - } - - pm_runtime_put_sync(mcbsp->dev); - - r = clk_set_parent(mcbsp->fclk, fck_src); - if (r) { - dev_err(mcbsp->dev, "CLKS: could not clk_set_parent() to %s\n", - src); - clk_put(fck_src); - return r; - } - - pm_runtime_get_sync(mcbsp->dev); - - clk_put(fck_src); - - return 0; - -} - -#define max_thres(m) (mcbsp->pdata->buffer_size) -#define valid_threshold(m, val) ((val) <= max_thres(m)) -#define THRESHOLD_PROP_BUILDER(prop) \ -static ssize_t prop##_show(struct device *dev, \ - struct device_attribute *attr, char *buf) \ -{ \ - struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \ - \ - return sprintf(buf, "%u\n", mcbsp->prop); \ -} \ - \ -static ssize_t prop##_store(struct device *dev, \ - struct device_attribute *attr, \ - const char *buf, size_t size) \ -{ \ - struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \ - unsigned long val; \ - int status; \ - \ - status = kstrtoul(buf, 0, &val); \ - if (status) \ - return status; \ - \ - if (!valid_threshold(mcbsp, val)) \ - return -EDOM; \ - \ - mcbsp->prop = val; \ - return size; \ -} \ - \ -static DEVICE_ATTR(prop, 0644, prop##_show, prop##_store); - -THRESHOLD_PROP_BUILDER(max_tx_thres); -THRESHOLD_PROP_BUILDER(max_rx_thres); - -static const char *dma_op_modes[] = { - "element", "threshold", -}; - -static ssize_t dma_op_mode_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); - int dma_op_mode, i = 0; - ssize_t len = 0; - const char * const *s; - - dma_op_mode = mcbsp->dma_op_mode; - - for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++) { - if (dma_op_mode == i) - len += sprintf(buf + len, "[%s] ", *s); - else - len += sprintf(buf + len, "%s ", *s); - } - len += sprintf(buf + len, "\n"); - - return len; -} - -static ssize_t dma_op_mode_store(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t size) -{ - struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); - int i; - - i = sysfs_match_string(dma_op_modes, buf); - if (i < 0) - return i; - - spin_lock_irq(&mcbsp->lock); - if (!mcbsp->free) { - size = -EBUSY; - goto unlock; - } - mcbsp->dma_op_mode = i; - -unlock: - spin_unlock_irq(&mcbsp->lock); - - return size; -} - -static DEVICE_ATTR_RW(dma_op_mode); - -static const struct attribute *additional_attrs[] = { - &dev_attr_max_tx_thres.attr, - &dev_attr_max_rx_thres.attr, - &dev_attr_dma_op_mode.attr, - NULL, -}; - -static const struct attribute_group additional_attr_group = { - .attrs = (struct attribute **)additional_attrs, -}; - -static ssize_t st_taps_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - ssize_t status = 0; - int i; - - spin_lock_irq(&mcbsp->lock); - for (i = 0; i < st_data->nr_taps; i++) - status += sprintf(&buf[status], (i ? ", %d" : "%d"), - st_data->taps[i]); - if (i) - status += sprintf(&buf[status], "\n"); - spin_unlock_irq(&mcbsp->lock); - - return status; -} - -static ssize_t st_taps_store(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t size) -{ - struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - int val, tmp, status, i = 0; - - spin_lock_irq(&mcbsp->lock); - memset(st_data->taps, 0, sizeof(st_data->taps)); - st_data->nr_taps = 0; - - do { - status = sscanf(buf, "%d%n", &val, &tmp); - if (status < 0 || status == 0) { - size = -EINVAL; - goto out; - } - if (val < -32768 || val > 32767) { - size = -EINVAL; - goto out; - } - st_data->taps[i++] = val; - buf += tmp; - if (*buf != ',') - break; - buf++; - } while (1); - - st_data->nr_taps = i; - -out: - spin_unlock_irq(&mcbsp->lock); - - return size; -} - -static DEVICE_ATTR_RW(st_taps); - -static const struct attribute *sidetone_attrs[] = { - &dev_attr_st_taps.attr, - NULL, -}; - -static const struct attribute_group sidetone_attr_group = { - .attrs = (struct attribute **)sidetone_attrs, -}; - -static int omap_st_add(struct omap_mcbsp *mcbsp, struct resource *res) -{ - struct omap_mcbsp_st_data *st_data; - int err; - - st_data = devm_kzalloc(mcbsp->dev, sizeof(*mcbsp->st_data), GFP_KERNEL); - if (!st_data) - return -ENOMEM; - - st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick"); - if (IS_ERR(st_data->mcbsp_iclk)) { - dev_warn(mcbsp->dev, - "Failed to get ick, sidetone might be broken\n"); - st_data->mcbsp_iclk = NULL; - } - - st_data->io_base_st = devm_ioremap(mcbsp->dev, res->start, - resource_size(res)); - if (!st_data->io_base_st) - return -ENOMEM; - - err = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); - if (err) - return err; - - mcbsp->st_data = st_data; - return 0; -} - -/* - * McBSP1 and McBSP3 are directly mapped on 1610 and 1510. - * 730 has only 2 McBSP, and both of them are MPU peripherals. - */ -int omap_mcbsp_init(struct platform_device *pdev) -{ - struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - struct resource *res; - int ret = 0; - - spin_lock_init(&mcbsp->lock); - mcbsp->free = true; - - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); - if (!res) - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - - mcbsp->io_base = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(mcbsp->io_base)) - return PTR_ERR(mcbsp->io_base); - - mcbsp->phys_base = res->start; - mcbsp->reg_cache_size = resource_size(res); - - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma"); - if (!res) - mcbsp->phys_dma_base = mcbsp->phys_base; - else - mcbsp->phys_dma_base = res->start; - - /* - * OMAP1, 2 uses two interrupt lines: TX, RX - * OMAP2430, OMAP3 SoC have combined IRQ line as well. - * OMAP4 and newer SoC only have the combined IRQ line. - * Use the combined IRQ if available since it gives better debugging - * possibilities. - */ - mcbsp->irq = platform_get_irq_byname(pdev, "common"); - if (mcbsp->irq == -ENXIO) { - mcbsp->tx_irq = platform_get_irq_byname(pdev, "tx"); - - if (mcbsp->tx_irq == -ENXIO) { - mcbsp->irq = platform_get_irq(pdev, 0); - mcbsp->tx_irq = 0; - } else { - mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx"); - mcbsp->irq = 0; - } - } - - if (!pdev->dev.of_node) { - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx"); - if (!res) { - dev_err(&pdev->dev, "invalid tx DMA channel\n"); - return -ENODEV; - } - mcbsp->dma_req[0] = res->start; - mcbsp->dma_data[0].filter_data = &mcbsp->dma_req[0]; - - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); - if (!res) { - dev_err(&pdev->dev, "invalid rx DMA channel\n"); - return -ENODEV; - } - mcbsp->dma_req[1] = res->start; - mcbsp->dma_data[1].filter_data = &mcbsp->dma_req[1]; - } else { - mcbsp->dma_data[0].filter_data = "tx"; - mcbsp->dma_data[1].filter_data = "rx"; - } - - mcbsp->dma_data[0].addr = omap_mcbsp_dma_reg_params(mcbsp, - SNDRV_PCM_STREAM_PLAYBACK); - mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, - SNDRV_PCM_STREAM_CAPTURE); - - mcbsp->fclk = clk_get(&pdev->dev, "fck"); - if (IS_ERR(mcbsp->fclk)) { - ret = PTR_ERR(mcbsp->fclk); - dev_err(mcbsp->dev, "unable to get fck: %d\n", ret); - return ret; - } - - mcbsp->dma_op_mode = MCBSP_DMA_MODE_ELEMENT; - if (mcbsp->pdata->buffer_size) { - /* - * Initially configure the maximum thresholds to a safe value. - * The McBSP FIFO usage with these values should not go under - * 16 locations. - * If the whole FIFO without safety buffer is used, than there - * is a possibility that the DMA will be not able to push the - * new data on time, causing channel shifts in runtime. - */ - mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10; - mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10; - - ret = sysfs_create_group(&mcbsp->dev->kobj, - &additional_attr_group); - if (ret) { - dev_err(mcbsp->dev, - "Unable to create additional controls\n"); - goto err_thres; - } - } else { - mcbsp->max_tx_thres = -EINVAL; - mcbsp->max_rx_thres = -EINVAL; - } - - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "sidetone"); - if (res) { - ret = omap_st_add(mcbsp, res); - if (ret) { - dev_err(mcbsp->dev, - "Unable to create sidetone controls\n"); - goto err_st; - } - } - - return 0; - -err_st: - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); -err_thres: - clk_put(mcbsp->fclk); - return ret; -} - -void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp) -{ - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - - if (mcbsp->st_data) { - sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); - clk_put(mcbsp->st_data->mcbsp_iclk); - } -} diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/omap-mcbsp-priv.h similarity index 70% rename from sound/soc/omap/mcbsp.h rename to sound/soc/omap/omap-mcbsp-priv.h index 92472c6ef358..7865cda4bf0a 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/omap-mcbsp-priv.h @@ -1,28 +1,15 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* - * sound/soc/omap/mcbsp.h - * * OMAP Multi-Channel Buffered Serial Port * * Contact: Jarkko Nikula <jarkko.nikula(a)bitmer.com> * Peter Ujfalusi <peter.ujfalusi(a)ti.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. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * */ -#ifndef __ASOC_MCBSP_H -#define __ASOC_MCBSP_H + +#ifndef __OMAP_MCBSP_PRIV_H__ +#define __OMAP_MCBSP_PRIV_H__ + +#include <linux/platform_data/asoc-ti-mcbsp.h> #ifdef CONFIG_ARCH_OMAP1 #define mcbsp_omap1() 1 @@ -30,8 +17,6 @@ #define mcbsp_omap1() 0 #endif -#include <sound/dmaengine_pcm.h> - /* McBSP register numbers. Register address offset = num * reg_step */ enum { /* Common registers */ @@ -85,15 +70,6 @@ enum { OMAP_MCBSP_REG_SSELCR, }; -/* OMAP3 sidetone control registers */ -#define OMAP_ST_REG_REV 0x00 -#define OMAP_ST_REG_SYSCONFIG 0x10 -#define OMAP_ST_REG_IRQSTATUS 0x18 -#define OMAP_ST_REG_IRQENABLE 0x1C -#define OMAP_ST_REG_SGAINCR 0x24 -#define OMAP_ST_REG_SFIRCR 0x28 -#define OMAP_ST_REG_SSELCR 0x2C - /************************** McBSP SPCR1 bit definitions ***********************/ #define RRST BIT(0) #define RRDY BIT(1) @@ -202,24 +178,6 @@ enum { #define SIDLEMODE(value) (((value) & 0x3) << 3) #define CLOCKACTIVITY(value) (((value) & 0x3) << 8) -/********************** McBSP SSELCR bit definitions ***********************/ -#define SIDETONEEN BIT(10) - -/********************** McBSP Sidetone SYSCONFIG bit definitions ***********/ -#define ST_AUTOIDLE BIT(0) - -/********************** McBSP Sidetone SGAINCR bit definitions *************/ -#define ST_CH0GAIN(value) ((value) & 0xffff) /* Bits 0:15 */ -#define ST_CH1GAIN(value) (((value) & 0xffff) << 16) /* Bits 16:31 */ - -/********************** McBSP Sidetone SFIRCR bit definitions **************/ -#define ST_FIRCOEFF(value) ((value) & 0xffff) /* Bits 0:15 */ - -/********************** McBSP Sidetone SSELCR bit definitions **************/ -#define ST_SIDETONEEN BIT(0) -#define ST_COEFFWREN BIT(1) -#define ST_COEFFWRDONE BIT(2) - /********************** McBSP DMA operating modes **************************/ #define MCBSP_DMA_MODE_ELEMENT 0 #define MCBSP_DMA_MODE_THRESHOLD 1 @@ -278,16 +236,7 @@ struct omap_mcbsp_reg_cfg { u16 rccr; }; -struct omap_mcbsp_st_data { - void __iomem *io_base_st; - struct clk *mcbsp_iclk; - bool running; - bool enabled; - s16 taps[128]; /* Sidetone filter coefficients */ - int nr_taps; /* Number of filter coefficients in use */ - s16 ch0gain; - s16 ch1gain; -}; +struct omap_mcbsp_st_data; struct omap_mcbsp { struct device *dev; @@ -330,29 +279,46 @@ struct omap_mcbsp { struct pm_qos_request pm_qos_req; }; -void omap_mcbsp_config(struct omap_mcbsp *mcbsp, - const struct omap_mcbsp_reg_cfg *config); -void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold); -void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold); -u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp); -u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp); -int omap_mcbsp_get_dma_op_mode(struct omap_mcbsp *mcbsp); -int omap_mcbsp_request(struct omap_mcbsp *mcbsp); -void omap_mcbsp_free(struct omap_mcbsp *mcbsp); -void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int stream); -void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int stream); - -/* McBSP functional clock source changing function */ -int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id); +static inline void omap_mcbsp_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) +{ + void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step; + + if (mcbsp->pdata->reg_size == 2) { + ((u16 *)mcbsp->reg_cache)[reg] = (u16)val; + writew_relaxed((u16)val, addr); + } else { + ((u32 *)mcbsp->reg_cache)[reg] = val; + writel_relaxed(val, addr); + } +} + +static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, + bool from_cache) +{ + void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step; + + if (mcbsp->pdata->reg_size == 2) { + return !from_cache ? readw_relaxed(addr) : + ((u16 *)mcbsp->reg_cache)[reg]; + } else { + return !from_cache ? readl_relaxed(addr) : + ((u32 *)mcbsp->reg_cache)[reg]; + } +} + +#define MCBSP_READ(mcbsp, reg) \ + omap_mcbsp_read(mcbsp, OMAP_MCBSP_REG_##reg, 0) +#define MCBSP_WRITE(mcbsp, reg, val) \ + omap_mcbsp_write(mcbsp, OMAP_MCBSP_REG_##reg, val) +#define MCBSP_READ_CACHE(mcbsp, reg) \ + omap_mcbsp_read(mcbsp, OMAP_MCBSP_REG_##reg, 1) + /* Sidetone specific API */ -int omap_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, s16 chgain); -int omap_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, s16 *chgain); -int omap_st_enable(struct omap_mcbsp *mcbsp); -int omap_st_disable(struct omap_mcbsp *mcbsp); -int omap_st_is_enabled(struct omap_mcbsp *mcbsp); +int omap_mcbsp_st_init(struct platform_device *pdev); +void omap_mcbsp_st_cleanup(struct platform_device *pdev); -int omap_mcbsp_init(struct platform_device *pdev); -void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp); +int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp); +int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp); -#endif /* __ASOC_MCBSP_H */ +#endif /* __OMAP_MCBSP_PRIV_H__ */ diff --git a/sound/soc/omap/omap-mcbsp-st.c b/sound/soc/omap/omap-mcbsp-st.c new file mode 100644 index 000000000000..1a3fe854e856 --- /dev/null +++ b/sound/soc/omap/omap-mcbsp-st.c @@ -0,0 +1,516 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * McBSP Sidetone support + * + * Copyright (C) 2004 Nokia Corporation + * Author: Samuel Ortiz <samuel.ortiz(a)nokia.com> + * + * Contact: Jarkko Nikula <jarkko.nikula(a)bitmer.com> + * Peter Ujfalusi <peter.ujfalusi(a)ti.com> + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/device.h> +#include <linux/platform_device.h> +#include <linux/interrupt.h> +#include <linux/err.h> +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/slab.h> +#include <linux/pm_runtime.h> + +#include "omap-mcbsp.h" +#include "omap-mcbsp-priv.h" + +/* OMAP3 sidetone control registers */ +#define OMAP_ST_REG_REV 0x00 +#define OMAP_ST_REG_SYSCONFIG 0x10 +#define OMAP_ST_REG_IRQSTATUS 0x18 +#define OMAP_ST_REG_IRQENABLE 0x1C +#define OMAP_ST_REG_SGAINCR 0x24 +#define OMAP_ST_REG_SFIRCR 0x28 +#define OMAP_ST_REG_SSELCR 0x2C + +/********************** McBSP SSELCR bit definitions ***********************/ +#define SIDETONEEN BIT(10) + +/********************** McBSP Sidetone SYSCONFIG bit definitions ***********/ +#define ST_AUTOIDLE BIT(0) + +/********************** McBSP Sidetone SGAINCR bit definitions *************/ +#define ST_CH0GAIN(value) ((value) & 0xffff) /* Bits 0:15 */ +#define ST_CH1GAIN(value) (((value) & 0xffff) << 16) /* Bits 16:31 */ + +/********************** McBSP Sidetone SFIRCR bit definitions **************/ +#define ST_FIRCOEFF(value) ((value) & 0xffff) /* Bits 0:15 */ + +/********************** McBSP Sidetone SSELCR bit definitions **************/ +#define ST_SIDETONEEN BIT(0) +#define ST_COEFFWREN BIT(1) +#define ST_COEFFWRDONE BIT(2) + +struct omap_mcbsp_st_data { + void __iomem *io_base_st; + struct clk *mcbsp_iclk; + bool running; + bool enabled; + s16 taps[128]; /* Sidetone filter coefficients */ + int nr_taps; /* Number of filter coefficients in use */ + s16 ch0gain; + s16 ch1gain; +}; + +static void omap_mcbsp_st_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) +{ + writel_relaxed(val, mcbsp->st_data->io_base_st + reg); +} + +static int omap_mcbsp_st_read(struct omap_mcbsp *mcbsp, u16 reg) +{ + return readl_relaxed(mcbsp->st_data->io_base_st + reg); +} + +#define MCBSP_ST_READ(mcbsp, reg) omap_mcbsp_st_read(mcbsp, OMAP_ST_REG_##reg) +#define MCBSP_ST_WRITE(mcbsp, reg, val) \ + omap_mcbsp_st_write(mcbsp, OMAP_ST_REG_##reg, val) + +static void omap_mcbsp_st_on(struct omap_mcbsp *mcbsp) +{ + unsigned int w; + + if (mcbsp->pdata->force_ick_on) + mcbsp->pdata->force_ick_on(mcbsp->st_data->mcbsp_iclk, true); + + /* Disable Sidetone clock auto-gating for normal operation */ + w = MCBSP_ST_READ(mcbsp, SYSCONFIG); + MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w & ~(ST_AUTOIDLE)); + + /* Enable McBSP Sidetone */ + w = MCBSP_READ(mcbsp, SSELCR); + MCBSP_WRITE(mcbsp, SSELCR, w | SIDETONEEN); + + /* Enable Sidetone from Sidetone Core */ + w = MCBSP_ST_READ(mcbsp, SSELCR); + MCBSP_ST_WRITE(mcbsp, SSELCR, w | ST_SIDETONEEN); +} + +static void omap_mcbsp_st_off(struct omap_mcbsp *mcbsp) +{ + unsigned int w; + + w = MCBSP_ST_READ(mcbsp, SSELCR); + MCBSP_ST_WRITE(mcbsp, SSELCR, w & ~(ST_SIDETONEEN)); + + w = MCBSP_READ(mcbsp, SSELCR); + MCBSP_WRITE(mcbsp, SSELCR, w & ~(SIDETONEEN)); + + /* Enable Sidetone clock auto-gating to reduce power consumption */ + w = MCBSP_ST_READ(mcbsp, SYSCONFIG); + MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w | ST_AUTOIDLE); + + if (mcbsp->pdata->force_ick_on) + mcbsp->pdata->force_ick_on(mcbsp->st_data->mcbsp_iclk, false); +} + +static void omap_mcbsp_st_fir_write(struct omap_mcbsp *mcbsp, s16 *fir) +{ + u16 val, i; + + val = MCBSP_ST_READ(mcbsp, SSELCR); + + if (val & ST_COEFFWREN) + MCBSP_ST_WRITE(mcbsp, SSELCR, val & ~(ST_COEFFWREN)); + + MCBSP_ST_WRITE(mcbsp, SSELCR, val | ST_COEFFWREN); + + for (i = 0; i < 128; i++) + MCBSP_ST_WRITE(mcbsp, SFIRCR, fir[i]); + + i = 0; + + val = MCBSP_ST_READ(mcbsp, SSELCR); + while (!(val & ST_COEFFWRDONE) && (++i < 1000)) + val = MCBSP_ST_READ(mcbsp, SSELCR); + + MCBSP_ST_WRITE(mcbsp, SSELCR, val & ~(ST_COEFFWREN)); + + if (i == 1000) + dev_err(mcbsp->dev, "McBSP FIR load error!\n"); +} + +static void omap_mcbsp_st_chgain(struct omap_mcbsp *mcbsp) +{ + u16 w; + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + w = MCBSP_ST_READ(mcbsp, SSELCR); + + MCBSP_ST_WRITE(mcbsp, SGAINCR, ST_CH0GAIN(st_data->ch0gain) | + ST_CH1GAIN(st_data->ch1gain)); +} + +static int omap_mcbsp_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, + s16 chgain) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + int ret = 0; + + if (!st_data) + return -ENOENT; + + spin_lock_irq(&mcbsp->lock); + if (channel == 0) + st_data->ch0gain = chgain; + else if (channel == 1) + st_data->ch1gain = chgain; + else + ret = -EINVAL; + + if (st_data->enabled) + omap_mcbsp_st_chgain(mcbsp); + spin_unlock_irq(&mcbsp->lock); + + return ret; +} + +static int omap_mcbsp_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, + s16 *chgain) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + int ret = 0; + + if (!st_data) + return -ENOENT; + + spin_lock_irq(&mcbsp->lock); + if (channel == 0) + *chgain = st_data->ch0gain; + else if (channel == 1) + *chgain = st_data->ch1gain; + else + ret = -EINVAL; + spin_unlock_irq(&mcbsp->lock); + + return ret; +} + +static int omap_mcbsp_st_enable(struct omap_mcbsp *mcbsp) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + if (!st_data) + return -ENODEV; + + spin_lock_irq(&mcbsp->lock); + st_data->enabled = 1; + omap_mcbsp_st_start(mcbsp); + spin_unlock_irq(&mcbsp->lock); + + return 0; +} + +static int omap_mcbsp_st_disable(struct omap_mcbsp *mcbsp) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + int ret = 0; + + if (!st_data) + return -ENODEV; + + spin_lock_irq(&mcbsp->lock); + omap_mcbsp_st_stop(mcbsp); + st_data->enabled = 0; + spin_unlock_irq(&mcbsp->lock); + + return ret; +} + +static int omap_mcbsp_st_is_enabled(struct omap_mcbsp *mcbsp) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + if (!st_data) + return -ENODEV; + + return st_data->enabled; +} + +static ssize_t st_taps_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + ssize_t status = 0; + int i; + + spin_lock_irq(&mcbsp->lock); + for (i = 0; i < st_data->nr_taps; i++) + status += sprintf(&buf[status], (i ? ", %d" : "%d"), + st_data->taps[i]); + if (i) + status += sprintf(&buf[status], "\n"); + spin_unlock_irq(&mcbsp->lock); + + return status; +} + +static ssize_t st_taps_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t size) +{ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + int val, tmp, status, i = 0; + + spin_lock_irq(&mcbsp->lock); + memset(st_data->taps, 0, sizeof(st_data->taps)); + st_data->nr_taps = 0; + + do { + status = sscanf(buf, "%d%n", &val, &tmp); + if (status < 0 || status == 0) { + size = -EINVAL; + goto out; + } + if (val < -32768 || val > 32767) { + size = -EINVAL; + goto out; + } + st_data->taps[i++] = val; + buf += tmp; + if (*buf != ',') + break; + buf++; + } while (1); + + st_data->nr_taps = i; + +out: + spin_unlock_irq(&mcbsp->lock); + + return size; +} + +static DEVICE_ATTR_RW(st_taps); + +static const struct attribute *sidetone_attrs[] = { + &dev_attr_st_taps.attr, + NULL, +}; + +static const struct attribute_group sidetone_attr_group = { + .attrs = (struct attribute **)sidetone_attrs, +}; + +int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + if (st_data->enabled && !st_data->running) { + omap_mcbsp_st_fir_write(mcbsp, st_data->taps); + omap_mcbsp_st_chgain(mcbsp); + + if (!mcbsp->free) { + omap_mcbsp_st_on(mcbsp); + st_data->running = 1; + } + } + + return 0; +} + +int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + if (st_data->running) { + if (!mcbsp->free) { + omap_mcbsp_st_off(mcbsp); + st_data->running = 0; + } + } + + return 0; +} + +int omap_mcbsp_st_init(struct platform_device *pdev) +{ + struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); + struct omap_mcbsp_st_data *st_data; + struct resource *res; + int ret; + + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "sidetone"); + if (!res) + return 0; + + st_data = devm_kzalloc(mcbsp->dev, sizeof(*mcbsp->st_data), GFP_KERNEL); + if (!st_data) + return -ENOMEM; + + st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick"); + if (IS_ERR(st_data->mcbsp_iclk)) { + dev_warn(mcbsp->dev, + "Failed to get ick, sidetone might be broken\n"); + st_data->mcbsp_iclk = NULL; + } + + st_data->io_base_st = devm_ioremap(mcbsp->dev, res->start, + resource_size(res)); + if (!st_data->io_base_st) + return -ENOMEM; + + ret = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); + if (ret) + return ret; + + mcbsp->st_data = st_data; + + return 0; +} + +void omap_mcbsp_st_cleanup(struct platform_device *pdev) +{ + struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); + + if (mcbsp->st_data) { + sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); + clk_put(mcbsp->st_data->mcbsp_iclk); + } +} + +static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int max = mc->max; + int min = mc->min; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = min; + uinfo->value.integer.max = max; + return 0; +} + +#define OMAP_MCBSP_ST_CHANNEL_VOLUME(channel) \ +static int \ +omap_mcbsp_set_st_ch##channel##_volume(struct snd_kcontrol *kc, \ + struct snd_ctl_elem_value *uc) \ +{ \ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \ + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); \ + struct soc_mixer_control *mc = \ + (struct soc_mixer_control *)kc->private_value; \ + int max = mc->max; \ + int min = mc->min; \ + int val = uc->value.integer.value[0]; \ + \ + if (val < min || val > max) \ + return -EINVAL; \ + \ + /* OMAP McBSP implementation uses index values 0..4 */ \ + return omap_mcbsp_st_set_chgain(mcbsp, channel, val); \ +} \ + \ +static int \ +omap_mcbsp_get_st_ch##channel##_volume(struct snd_kcontrol *kc, \ + struct snd_ctl_elem_value *uc) \ +{ \ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \ + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); \ + s16 chgain; \ + \ + if (omap_mcbsp_st_get_chgain(mcbsp, channel, &chgain)) \ + return -EAGAIN; \ + \ + uc->value.integer.value[0] = chgain; \ + return 0; \ +} + +OMAP_MCBSP_ST_CHANNEL_VOLUME(0) +OMAP_MCBSP_ST_CHANNEL_VOLUME(1) + +static int omap_mcbsp_st_put_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + u8 value = ucontrol->value.integer.value[0]; + + if (value == omap_mcbsp_st_is_enabled(mcbsp)) + return 0; + + if (value) + omap_mcbsp_st_enable(mcbsp); + else + omap_mcbsp_st_disable(mcbsp); + + return 1; +} + +static int omap_mcbsp_st_get_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + + ucontrol->value.integer.value[0] = omap_mcbsp_st_is_enabled(mcbsp); + return 0; +} + +#define OMAP_MCBSP_SOC_SINGLE_S16_EXT(xname, xmin, xmax, \ + xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = omap_mcbsp_st_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.min = xmin, .max = xmax} } + +#define OMAP_MCBSP_ST_CONTROLS(port) \ +static const struct snd_kcontrol_new omap_mcbsp##port##_st_controls[] = { \ +SOC_SINGLE_EXT("McBSP" #port " Sidetone Switch", 1, 0, 1, 0, \ + omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode), \ +OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP" #port " Sidetone Channel 0 Volume", \ + -32768, 32767, \ + omap_mcbsp_get_st_ch0_volume, \ + omap_mcbsp_set_st_ch0_volume), \ +OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP" #port " Sidetone Channel 1 Volume", \ + -32768, 32767, \ + omap_mcbsp_get_st_ch1_volume, \ + omap_mcbsp_set_st_ch1_volume), \ +} + +OMAP_MCBSP_ST_CONTROLS(2); +OMAP_MCBSP_ST_CONTROLS(3); + +int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id) +{ + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + + if (!mcbsp->st_data) { + dev_warn(mcbsp->dev, "No sidetone data for port\n"); + return 0; + } + + switch (port_id) { + case 2: /* McBSP 2 */ + return snd_soc_add_dai_controls(cpu_dai, + omap_mcbsp2_st_controls, + ARRAY_SIZE(omap_mcbsp2_st_controls)); + case 3: /* McBSP 3 */ + return snd_soc_add_dai_controls(cpu_dai, + omap_mcbsp3_st_controls, + ARRAY_SIZE(omap_mcbsp3_st_controls)); + default: + dev_err(mcbsp->dev, "Port %d not supported\n", port_id); + break; + } + + return -EINVAL; +} +EXPORT_SYMBOL_GPL(omap_mcbsp_st_add_controls); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 69a6b8ad6d42..e48fad11a0cc 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -35,21 +35,12 @@ #include <sound/soc.h> #include <sound/dmaengine_pcm.h> -#include <linux/platform_data/asoc-ti-mcbsp.h> -#include "mcbsp.h" +#include "omap-mcbsp-priv.h" #include "omap-mcbsp.h" #include "sdma-pcm.h" #define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000) -#define OMAP_MCBSP_SOC_SINGLE_S16_EXT(xname, xmin, xmax, \ - xhandler_get, xhandler_put) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = omap_mcbsp_st_info_volsw, \ - .get = xhandler_get, .put = xhandler_put, \ - .private_value = (unsigned long) &(struct soc_mixer_control) \ - {.min = xmin, .max = xmax} } - enum { OMAP_MCBSP_WORD_8 = 0, OMAP_MCBSP_WORD_12, @@ -59,6 +50,702 @@ enum { OMAP_MCBSP_WORD_32, }; +static void omap_mcbsp_dump_reg(struct omap_mcbsp *mcbsp) +{ + dev_dbg(mcbsp->dev, "**** McBSP%d regs ****\n", mcbsp->id); + dev_dbg(mcbsp->dev, "DRR2: 0x%04x\n", MCBSP_READ(mcbsp, DRR2)); + dev_dbg(mcbsp->dev, "DRR1: 0x%04x\n", MCBSP_READ(mcbsp, DRR1)); + dev_dbg(mcbsp->dev, "DXR2: 0x%04x\n", MCBSP_READ(mcbsp, DXR2)); + dev_dbg(mcbsp->dev, "DXR1: 0x%04x\n", MCBSP_READ(mcbsp, DXR1)); + dev_dbg(mcbsp->dev, "SPCR2: 0x%04x\n", MCBSP_READ(mcbsp, SPCR2)); + dev_dbg(mcbsp->dev, "SPCR1: 0x%04x\n", MCBSP_READ(mcbsp, SPCR1)); + dev_dbg(mcbsp->dev, "RCR2: 0x%04x\n", MCBSP_READ(mcbsp, RCR2)); + dev_dbg(mcbsp->dev, "RCR1: 0x%04x\n", MCBSP_READ(mcbsp, RCR1)); + dev_dbg(mcbsp->dev, "XCR2: 0x%04x\n", MCBSP_READ(mcbsp, XCR2)); + dev_dbg(mcbsp->dev, "XCR1: 0x%04x\n", MCBSP_READ(mcbsp, XCR1)); + dev_dbg(mcbsp->dev, "SRGR2: 0x%04x\n", MCBSP_READ(mcbsp, SRGR2)); + dev_dbg(mcbsp->dev, "SRGR1: 0x%04x\n", MCBSP_READ(mcbsp, SRGR1)); + dev_dbg(mcbsp->dev, "PCR0: 0x%04x\n", MCBSP_READ(mcbsp, PCR0)); + dev_dbg(mcbsp->dev, "***********************\n"); +} + +static int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) +{ + struct clk *fck_src; + const char *src; + int r; + + if (fck_src_id == MCBSP_CLKS_PAD_SRC) + src = "pad_fck"; + else if (fck_src_id == MCBSP_CLKS_PRCM_SRC) + src = "prcm_fck"; + else + return -EINVAL; + + fck_src = clk_get(mcbsp->dev, src); + if (IS_ERR(fck_src)) { + dev_err(mcbsp->dev, "CLKS: could not clk_get() %s\n", src); + return -EINVAL; + } + + pm_runtime_put_sync(mcbsp->dev); + + r = clk_set_parent(mcbsp->fclk, fck_src); + if (r) { + dev_err(mcbsp->dev, "CLKS: could not clk_set_parent() to %s\n", + src); + clk_put(fck_src); + return r; + } + + pm_runtime_get_sync(mcbsp->dev); + + clk_put(fck_src); + + return 0; +} + +static irqreturn_t omap_mcbsp_irq_handler(int irq, void *data) +{ + struct omap_mcbsp *mcbsp = data; + u16 irqst; + + irqst = MCBSP_READ(mcbsp, IRQST); + dev_dbg(mcbsp->dev, "IRQ callback : 0x%x\n", irqst); + + if (irqst & RSYNCERREN) + dev_err(mcbsp->dev, "RX Frame Sync Error!\n"); + if (irqst & RFSREN) + dev_dbg(mcbsp->dev, "RX Frame Sync\n"); + if (irqst & REOFEN) + dev_dbg(mcbsp->dev, "RX End Of Frame\n"); + if (irqst & RRDYEN) + dev_dbg(mcbsp->dev, "RX Buffer Threshold Reached\n"); + if (irqst & RUNDFLEN) + dev_err(mcbsp->dev, "RX Buffer Underflow!\n"); + if (irqst & ROVFLEN) + dev_err(mcbsp->dev, "RX Buffer Overflow!\n"); + + if (irqst & XSYNCERREN) + dev_err(mcbsp->dev, "TX Frame Sync Error!\n"); + if (irqst & XFSXEN) + dev_dbg(mcbsp->dev, "TX Frame Sync\n"); + if (irqst & XEOFEN) + dev_dbg(mcbsp->dev, "TX End Of Frame\n"); + if (irqst & XRDYEN) + dev_dbg(mcbsp->dev, "TX Buffer threshold Reached\n"); + if (irqst & XUNDFLEN) + dev_err(mcbsp->dev, "TX Buffer Underflow!\n"); + if (irqst & XOVFLEN) + dev_err(mcbsp->dev, "TX Buffer Overflow!\n"); + if (irqst & XEMPTYEOFEN) + dev_dbg(mcbsp->dev, "TX Buffer empty at end of frame\n"); + + MCBSP_WRITE(mcbsp, IRQST, irqst); + + return IRQ_HANDLED; +} + +static irqreturn_t omap_mcbsp_tx_irq_handler(int irq, void *data) +{ + struct omap_mcbsp *mcbsp = data; + u16 irqst_spcr2; + + irqst_spcr2 = MCBSP_READ(mcbsp, SPCR2); + dev_dbg(mcbsp->dev, "TX IRQ callback : 0x%x\n", irqst_spcr2); + + if (irqst_spcr2 & XSYNC_ERR) { + dev_err(mcbsp->dev, "TX Frame Sync Error! : 0x%x\n", + irqst_spcr2); + /* Writing zero to XSYNC_ERR clears the IRQ */ + MCBSP_WRITE(mcbsp, SPCR2, MCBSP_READ_CACHE(mcbsp, SPCR2)); + } + + return IRQ_HANDLED; +} + +static irqreturn_t omap_mcbsp_rx_irq_handler(int irq, void *data) +{ + struct omap_mcbsp *mcbsp = data; + u16 irqst_spcr1; + + irqst_spcr1 = MCBSP_READ(mcbsp, SPCR1); + dev_dbg(mcbsp->dev, "RX IRQ callback : 0x%x\n", irqst_spcr1); + + if (irqst_spcr1 & RSYNC_ERR) { + dev_err(mcbsp->dev, "RX Frame Sync Error! : 0x%x\n", + irqst_spcr1); + /* Writing zero to RSYNC_ERR clears the IRQ */ + MCBSP_WRITE(mcbsp, SPCR1, MCBSP_READ_CACHE(mcbsp, SPCR1)); + } + + return IRQ_HANDLED; +} + +/* + * omap_mcbsp_config simply write a config to the + * appropriate McBSP. + * You either call this function or set the McBSP registers + * by yourself before calling omap_mcbsp_start(). + */ +static void omap_mcbsp_config(struct omap_mcbsp *mcbsp, + const struct omap_mcbsp_reg_cfg *config) +{ + dev_dbg(mcbsp->dev, "Configuring McBSP%d phys_base: 0x%08lx\n", + mcbsp->id, mcbsp->phys_base); + + /* We write the given config */ + MCBSP_WRITE(mcbsp, SPCR2, config->spcr2); + MCBSP_WRITE(mcbsp, SPCR1, config->spcr1); + MCBSP_WRITE(mcbsp, RCR2, config->rcr2); + MCBSP_WRITE(mcbsp, RCR1, config->rcr1); + MCBSP_WRITE(mcbsp, XCR2, config->xcr2); + MCBSP_WRITE(mcbsp, XCR1, config->xcr1); + MCBSP_WRITE(mcbsp, SRGR2, config->srgr2); + MCBSP_WRITE(mcbsp, SRGR1, config->srgr1); + MCBSP_WRITE(mcbsp, MCR2, config->mcr2); + MCBSP_WRITE(mcbsp, MCR1, config->mcr1); + MCBSP_WRITE(mcbsp, PCR0, config->pcr0); + if (mcbsp->pdata->has_ccr) { + MCBSP_WRITE(mcbsp, XCCR, config->xccr); + MCBSP_WRITE(mcbsp, RCCR, config->rccr); + } + /* Enable wakeup behavior */ + if (mcbsp->pdata->has_wakeup) + MCBSP_WRITE(mcbsp, WAKEUPEN, XRDYEN | RRDYEN); + + /* Enable TX/RX sync error interrupts by default */ + if (mcbsp->irq) + MCBSP_WRITE(mcbsp, IRQEN, RSYNCERREN | XSYNCERREN | + RUNDFLEN | ROVFLEN | XUNDFLEN | XOVFLEN); +} + +/** + * omap_mcbsp_dma_reg_params - returns the address of mcbsp data register + * @mcbsp: omap_mcbsp struct for the McBSP instance + * @stream: Stream direction (playback/capture) + * + * Returns the address of mcbsp data transmit register or data receive register + * to be used by DMA for transferring/receiving data + */ +static int omap_mcbsp_dma_reg_params(struct omap_mcbsp *mcbsp, + unsigned int stream) +{ + int data_reg; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (mcbsp->pdata->reg_size == 2) + data_reg = OMAP_MCBSP_REG_DXR1; + else + data_reg = OMAP_MCBSP_REG_DXR; + } else { + if (mcbsp->pdata->reg_size == 2) + data_reg = OMAP_MCBSP_REG_DRR1; + else + data_reg = OMAP_MCBSP_REG_DRR; + } + + return mcbsp->phys_dma_base + data_reg * mcbsp->pdata->reg_step; +} + +/* + * omap_mcbsp_set_rx_threshold configures the transmit threshold in words. + * The threshold parameter is 1 based, and it is converted (threshold - 1) + * for the THRSH2 register. + */ +static void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) +{ + if (threshold && threshold <= mcbsp->max_tx_thres) + MCBSP_WRITE(mcbsp, THRSH2, threshold - 1); +} + +/* + * omap_mcbsp_set_rx_threshold configures the receive threshold in words. + * The threshold parameter is 1 based, and it is converted (threshold - 1) + * for the THRSH1 register. + */ +static void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) +{ + if (threshold && threshold <= mcbsp->max_rx_thres) + MCBSP_WRITE(mcbsp, THRSH1, threshold - 1); +} + +/* + * omap_mcbsp_get_tx_delay returns the number of used slots in the McBSP FIFO + */ +static u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp) +{ + u16 buffstat; + + /* Returns the number of free locations in the buffer */ + buffstat = MCBSP_READ(mcbsp, XBUFFSTAT); + + /* Number of slots are different in McBSP ports */ + return mcbsp->pdata->buffer_size - buffstat; +} + +/* + * omap_mcbsp_get_rx_delay returns the number of free slots in the McBSP FIFO + * to reach the threshold value (when the DMA will be triggered to read it) + */ +static u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp) +{ + u16 buffstat, threshold; + + /* Returns the number of used locations in the buffer */ + buffstat = MCBSP_READ(mcbsp, RBUFFSTAT); + /* RX threshold */ + threshold = MCBSP_READ(mcbsp, THRSH1); + + /* Return the number of location till we reach the threshold limit */ + if (threshold <= buffstat) + return 0; + else + return threshold - buffstat; +} + +static int omap_mcbsp_request(struct omap_mcbsp *mcbsp) +{ + void *reg_cache; + int err; + + reg_cache = kzalloc(mcbsp->reg_cache_size, GFP_KERNEL); + if (!reg_cache) + return -ENOMEM; + + spin_lock(&mcbsp->lock); + if (!mcbsp->free) { + dev_err(mcbsp->dev, "McBSP%d is currently in use\n", mcbsp->id); + err = -EBUSY; + goto err_kfree; + } + + mcbsp->free = false; + mcbsp->reg_cache = reg_cache; + spin_unlock(&mcbsp->lock); + + if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->request) + mcbsp->pdata->ops->request(mcbsp->id - 1); + + /* + * Make sure that transmitter, receiver and sample-rate generator are + * not running before activating IRQs. + */ + MCBSP_WRITE(mcbsp, SPCR1, 0); + MCBSP_WRITE(mcbsp, SPCR2, 0); + + if (mcbsp->irq) { + err = request_irq(mcbsp->irq, omap_mcbsp_irq_handler, 0, + "McBSP", (void *)mcbsp); + if (err != 0) { + dev_err(mcbsp->dev, "Unable to request IRQ\n"); + goto err_clk_disable; + } + } else { + err = request_irq(mcbsp->tx_irq, omap_mcbsp_tx_irq_handler, 0, + "McBSP TX", (void *)mcbsp); + if (err != 0) { + dev_err(mcbsp->dev, "Unable to request TX IRQ\n"); + goto err_clk_disable; + } + + err = request_irq(mcbsp->rx_irq, omap_mcbsp_rx_irq_handler, 0, + "McBSP RX", (void *)mcbsp); + if (err != 0) { + dev_err(mcbsp->dev, "Unable to request RX IRQ\n"); + goto err_free_irq; + } + } + + return 0; +err_free_irq: + free_irq(mcbsp->tx_irq, (void *)mcbsp); +err_clk_disable: + if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) + mcbsp->pdata->ops->free(mcbsp->id - 1); + + /* Disable wakeup behavior */ + if (mcbsp->pdata->has_wakeup) + MCBSP_WRITE(mcbsp, WAKEUPEN, 0); + + spin_lock(&mcbsp->lock); + mcbsp->free = true; + mcbsp->reg_cache = NULL; +err_kfree: + spin_unlock(&mcbsp->lock); + kfree(reg_cache); + + return err; +} + +static void omap_mcbsp_free(struct omap_mcbsp *mcbsp) +{ + void *reg_cache; + + if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) + mcbsp->pdata->ops->free(mcbsp->id - 1); + + /* Disable wakeup behavior */ + if (mcbsp->pdata->has_wakeup) + MCBSP_WRITE(mcbsp, WAKEUPEN, 0); + + /* Disable interrupt requests */ + if (mcbsp->irq) + MCBSP_WRITE(mcbsp, IRQEN, 0); + + if (mcbsp->irq) { + free_irq(mcbsp->irq, (void *)mcbsp); + } else { + free_irq(mcbsp->rx_irq, (void *)mcbsp); + free_irq(mcbsp->tx_irq, (void *)mcbsp); + } + + reg_cache = mcbsp->reg_cache; + + /* + * Select CLKS source from internal source unconditionally before + * marking the McBSP port as free. + * If the external clock source via MCBSP_CLKS pin has been selected the + * system will refuse to enter idle if the CLKS pin source is not reset + * back to internal source. + */ + if (!mcbsp_omap1()) + omap2_mcbsp_set_clks_src(mcbsp, MCBSP_CLKS_PRCM_SRC); + + spin_lock(&mcbsp->lock); + if (mcbsp->free) + dev_err(mcbsp->dev, "McBSP%d was not reserved\n", mcbsp->id); + else + mcbsp->free = true; + mcbsp->reg_cache = NULL; + spin_unlock(&mcbsp->lock); + + kfree(reg_cache); +} + +/* + * Here we start the McBSP, by enabling transmitter, receiver or both. + * If no transmitter or receiver is active prior calling, then sample-rate + * generator and frame sync are started. + */ +static void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int stream) +{ + int tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); + int rx = !tx; + int enable_srg = 0; + u16 w; + + if (mcbsp->st_data) + omap_mcbsp_st_start(mcbsp); + + /* Only enable SRG, if McBSP is master */ + w = MCBSP_READ_CACHE(mcbsp, PCR0); + if (w & (FSXM | FSRM | CLKXM | CLKRM)) + enable_srg = !((MCBSP_READ_CACHE(mcbsp, SPCR2) | + MCBSP_READ_CACHE(mcbsp, SPCR1)) & 1); + + if (enable_srg) { + /* Start the sample generator */ + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w | (1 << 6)); + } + + /* Enable transmitter and receiver */ + tx &= 1; + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w | tx); + + rx &= 1; + w = MCBSP_READ_CACHE(mcbsp, SPCR1); + MCBSP_WRITE(mcbsp, SPCR1, w | rx); + + /* + * Worst case: CLKSRG*2 = 8000khz: (1/8000) * 2 * 2 usec + * REVISIT: 100us may give enough time for two CLKSRG, however + * due to some unknown PM related, clock gating etc. reason it + * is now at 500us. + */ + udelay(500); + + if (enable_srg) { + /* Start frame sync */ + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w | (1 << 7)); + } + + if (mcbsp->pdata->has_ccr) { + /* Release the transmitter and receiver */ + w = MCBSP_READ_CACHE(mcbsp, XCCR); + w &= ~(tx ? XDISABLE : 0); + MCBSP_WRITE(mcbsp, XCCR, w); + w = MCBSP_READ_CACHE(mcbsp, RCCR); + w &= ~(rx ? RDISABLE : 0); + MCBSP_WRITE(mcbsp, RCCR, w); + } + + /* Dump McBSP Regs */ + omap_mcbsp_dump_reg(mcbsp); +} + +static void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int stream) +{ + int tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); + int rx = !tx; + int idle; + u16 w; + + /* Reset transmitter */ + tx &= 1; + if (mcbsp->pdata->has_ccr) { + w = MCBSP_READ_CACHE(mcbsp, XCCR); + w |= (tx ? XDISABLE : 0); + MCBSP_WRITE(mcbsp, XCCR, w); + } + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w & ~tx); + + /* Reset receiver */ + rx &= 1; + if (mcbsp->pdata->has_ccr) { + w = MCBSP_READ_CACHE(mcbsp, RCCR); + w |= (rx ? RDISABLE : 0); + MCBSP_WRITE(mcbsp, RCCR, w); + } + w = MCBSP_READ_CACHE(mcbsp, SPCR1); + MCBSP_WRITE(mcbsp, SPCR1, w & ~rx); + + idle = !((MCBSP_READ_CACHE(mcbsp, SPCR2) | + MCBSP_READ_CACHE(mcbsp, SPCR1)) & 1); + + if (idle) { + /* Reset the sample rate generator */ + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w & ~(1 << 6)); + } + + if (mcbsp->st_data) + omap_mcbsp_st_stop(mcbsp); +} + +#define max_thres(m) (mcbsp->pdata->buffer_size) +#define valid_threshold(m, val) ((val) <= max_thres(m)) +#define THRESHOLD_PROP_BUILDER(prop) \ +static ssize_t prop##_show(struct device *dev, \ + struct device_attribute *attr, char *buf) \ +{ \ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \ + \ + return sprintf(buf, "%u\n", mcbsp->prop); \ +} \ + \ +static ssize_t prop##_store(struct device *dev, \ + struct device_attribute *attr, \ + const char *buf, size_t size) \ +{ \ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \ + unsigned long val; \ + int status; \ + \ + status = kstrtoul(buf, 0, &val); \ + if (status) \ + return status; \ + \ + if (!valid_threshold(mcbsp, val)) \ + return -EDOM; \ + \ + mcbsp->prop = val; \ + return size; \ +} \ + \ +static DEVICE_ATTR(prop, 0644, prop##_show, prop##_store) + +THRESHOLD_PROP_BUILDER(max_tx_thres); +THRESHOLD_PROP_BUILDER(max_rx_thres); + +static const char * const dma_op_modes[] = { + "element", "threshold", +}; + +static ssize_t dma_op_mode_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); + int dma_op_mode, i = 0; + ssize_t len = 0; + const char * const *s; + + dma_op_mode = mcbsp->dma_op_mode; + + for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++) { + if (dma_op_mode == i) + len += sprintf(buf + len, "[%s] ", *s); + else + len += sprintf(buf + len, "%s ", *s); + } + len += sprintf(buf + len, "\n"); + + return len; +} + +static ssize_t dma_op_mode_store(struct device *dev, + struct device_attribute *attr, const char *buf, + size_t size) +{ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); + int i; + + i = sysfs_match_string(dma_op_modes, buf); + if (i < 0) + return i; + + spin_lock_irq(&mcbsp->lock); + if (!mcbsp->free) { + size = -EBUSY; + goto unlock; + } + mcbsp->dma_op_mode = i; + +unlock: + spin_unlock_irq(&mcbsp->lock); + + return size; +} + +static DEVICE_ATTR_RW(dma_op_mode); + +static const struct attribute *additional_attrs[] = { + &dev_attr_max_tx_thres.attr, + &dev_attr_max_rx_thres.attr, + &dev_attr_dma_op_mode.attr, + NULL, +}; + +static const struct attribute_group additional_attr_group = { + .attrs = (struct attribute **)additional_attrs, +}; + +/* + * McBSP1 and McBSP3 are directly mapped on 1610 and 1510. + * 730 has only 2 McBSP, and both of them are MPU peripherals. + */ +static int omap_mcbsp_init(struct platform_device *pdev) +{ + struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); + struct resource *res; + int ret = 0; + + spin_lock_init(&mcbsp->lock); + mcbsp->free = true; + + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); + if (!res) + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + + mcbsp->io_base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(mcbsp->io_base)) + return PTR_ERR(mcbsp->io_base); + + mcbsp->phys_base = res->start; + mcbsp->reg_cache_size = resource_size(res); + + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma"); + if (!res) + mcbsp->phys_dma_base = mcbsp->phys_base; + else + mcbsp->phys_dma_base = res->start; + + /* + * OMAP1, 2 uses two interrupt lines: TX, RX + * OMAP2430, OMAP3 SoC have combined IRQ line as well. + * OMAP4 and newer SoC only have the combined IRQ line. + * Use the combined IRQ if available since it gives better debugging + * possibilities. + */ + mcbsp->irq = platform_get_irq_byname(pdev, "common"); + if (mcbsp->irq == -ENXIO) { + mcbsp->tx_irq = platform_get_irq_byname(pdev, "tx"); + + if (mcbsp->tx_irq == -ENXIO) { + mcbsp->irq = platform_get_irq(pdev, 0); + mcbsp->tx_irq = 0; + } else { + mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx"); + mcbsp->irq = 0; + } + } + + if (!pdev->dev.of_node) { + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx"); + if (!res) { + dev_err(&pdev->dev, "invalid tx DMA channel\n"); + return -ENODEV; + } + mcbsp->dma_req[0] = res->start; + mcbsp->dma_data[0].filter_data = &mcbsp->dma_req[0]; + + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); + if (!res) { + dev_err(&pdev->dev, "invalid rx DMA channel\n"); + return -ENODEV; + } + mcbsp->dma_req[1] = res->start; + mcbsp->dma_data[1].filter_data = &mcbsp->dma_req[1]; + } else { + mcbsp->dma_data[0].filter_data = "tx"; + mcbsp->dma_data[1].filter_data = "rx"; + } + + mcbsp->dma_data[0].addr = omap_mcbsp_dma_reg_params(mcbsp, + SNDRV_PCM_STREAM_PLAYBACK); + mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, + SNDRV_PCM_STREAM_CAPTURE); + + mcbsp->fclk = clk_get(&pdev->dev, "fck"); + if (IS_ERR(mcbsp->fclk)) { + ret = PTR_ERR(mcbsp->fclk); + dev_err(mcbsp->dev, "unable to get fck: %d\n", ret); + return ret; + } + + mcbsp->dma_op_mode = MCBSP_DMA_MODE_ELEMENT; + if (mcbsp->pdata->buffer_size) { + /* + * Initially configure the maximum thresholds to a safe value. + * The McBSP FIFO usage with these values should not go under + * 16 locations. + * If the whole FIFO without safety buffer is used, than there + * is a possibility that the DMA will be not able to push the + * new data on time, causing channel shifts in runtime. + */ + mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10; + mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10; + + ret = sysfs_create_group(&mcbsp->dev->kobj, + &additional_attr_group); + if (ret) { + dev_err(mcbsp->dev, + "Unable to create additional controls\n"); + goto err_thres; + } + } else { + mcbsp->max_tx_thres = -EINVAL; + mcbsp->max_rx_thres = -EINVAL; + } + + ret = omap_mcbsp_st_init(pdev); + if (ret) + goto err_st; + + return 0; + +err_st: + if (mcbsp->pdata->buffer_size) + sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); +err_thres: + clk_put(mcbsp->fclk); + return ret; +} + /* * Stream DMA parameters. DMA request line and port address are set runtime * since they are different between OMAP1 and later OMAPs @@ -656,132 +1343,6 @@ static const struct snd_soc_component_driver omap_mcbsp_component = { .name = "omap-mcbsp", }; -static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int max = mc->max; - int min = mc->min; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = min; - uinfo->value.integer.max = max; - return 0; -} - -#define OMAP_MCBSP_ST_CHANNEL_VOLUME(channel) \ -static int \ -omap_mcbsp_set_st_ch##channel##_volume(struct snd_kcontrol *kc, \ - struct snd_ctl_elem_value *uc) \ -{ \ - struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \ - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); \ - struct soc_mixer_control *mc = \ - (struct soc_mixer_control *)kc->private_value; \ - int max = mc->max; \ - int min = mc->min; \ - int val = uc->value.integer.value[0]; \ - \ - if (val < min || val > max) \ - return -EINVAL; \ - \ - /* OMAP McBSP implementation uses index values 0..4 */ \ - return omap_st_set_chgain(mcbsp, channel, val); \ -} \ - \ -static int \ -omap_mcbsp_get_st_ch##channel##_volume(struct snd_kcontrol *kc, \ - struct snd_ctl_elem_value *uc) \ -{ \ - struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \ - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); \ - s16 chgain; \ - \ - if (omap_st_get_chgain(mcbsp, channel, &chgain)) \ - return -EAGAIN; \ - \ - uc->value.integer.value[0] = chgain; \ - return 0; \ -} - -OMAP_MCBSP_ST_CHANNEL_VOLUME(0) -OMAP_MCBSP_ST_CHANNEL_VOLUME(1) - -static int omap_mcbsp_st_put_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - u8 value = ucontrol->value.integer.value[0]; - - if (value == omap_st_is_enabled(mcbsp)) - return 0; - - if (value) - omap_st_enable(mcbsp); - else - omap_st_disable(mcbsp); - - return 1; -} - -static int omap_mcbsp_st_get_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - - ucontrol->value.integer.value[0] = omap_st_is_enabled(mcbsp); - return 0; -} - -#define OMAP_MCBSP_ST_CONTROLS(port) \ -static const struct snd_kcontrol_new omap_mcbsp##port##_st_controls[] = { \ -SOC_SINGLE_EXT("McBSP" #port " Sidetone Switch", 1, 0, 1, 0, \ - omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode), \ -OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP" #port " Sidetone Channel 0 Volume", \ - -32768, 32767, \ - omap_mcbsp_get_st_ch0_volume, \ - omap_mcbsp_set_st_ch0_volume), \ -OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP" #port " Sidetone Channel 1 Volume", \ - -32768, 32767, \ - omap_mcbsp_get_st_ch1_volume, \ - omap_mcbsp_set_st_ch1_volume), \ -} - -OMAP_MCBSP_ST_CONTROLS(2); -OMAP_MCBSP_ST_CONTROLS(3); - -int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id) -{ - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - - if (!mcbsp->st_data) { - dev_warn(mcbsp->dev, "No sidetone data for port\n"); - return 0; - } - - switch (port_id) { - case 2: /* McBSP 2 */ - return snd_soc_add_dai_controls(cpu_dai, - omap_mcbsp2_st_controls, - ARRAY_SIZE(omap_mcbsp2_st_controls)); - case 3: /* McBSP 3 */ - return snd_soc_add_dai_controls(cpu_dai, - omap_mcbsp3_st_controls, - ARRAY_SIZE(omap_mcbsp3_st_controls)); - default: - dev_err(mcbsp->dev, "Port %d not supported\n", port_id); - break; - } - - return -EINVAL; -} -EXPORT_SYMBOL_GPL(omap_mcbsp_st_add_controls); - static struct omap_mcbsp_platform_data omap2420_pdata = { .reg_step = 4, .reg_size = 2, @@ -893,7 +1454,10 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) if (pm_qos_request_active(&mcbsp->pm_qos_req)) pm_qos_remove_request(&mcbsp->pm_qos_req); - omap_mcbsp_cleanup(mcbsp); + if (mcbsp->pdata->buffer_size) + sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); + + omap_mcbsp_st_cleanup(pdev); clk_put(mcbsp->fclk); diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index 2e3369c27be3..7911d24898c9 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -22,8 +22,10 @@ * */ -#ifndef __OMAP_I2S_H__ -#define __OMAP_I2S_H__ +#ifndef __OMAP_MCBSP_H__ +#define __OMAP_MCBSP_H__ + +#include <sound/dmaengine_pcm.h> /* Source clocks for McBSP sample rate generator */ enum omap_mcbsp_clksrg_clk { @@ -41,4 +43,4 @@ enum omap_mcbsp_div { int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id); -#endif +#endif /* __OMAP_MCBSP_H__ */ -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: omap-mcbsp: Remove redundant check for mcbsp->pdata" to the asoc tree
by Mark Brown 13 Nov '18

13 Nov '18

The patch ASoC: omap-mcbsp: Remove redundant check for mcbsp->pdata 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 465d85b3976a15c9e0e205ff50be64efa11b7af3 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Thu, 8 Nov 2018 09:29:59 +0200 Subject: [PATCH] ASoC: omap-mcbsp: Remove redundant check for mcbsp->pdata The driver will not probe if the pdata is not provided or created. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Tested-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/omap/omap-mcbsp.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index e48fad11a0cc..8cab1151d3bb 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -324,7 +324,7 @@ static int omap_mcbsp_request(struct omap_mcbsp *mcbsp) mcbsp->reg_cache = reg_cache; spin_unlock(&mcbsp->lock); - if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->request) + if(mcbsp->pdata->ops && mcbsp->pdata->ops->request) mcbsp->pdata->ops->request(mcbsp->id - 1); /* @@ -361,7 +361,7 @@ static int omap_mcbsp_request(struct omap_mcbsp *mcbsp) err_free_irq: free_irq(mcbsp->tx_irq, (void *)mcbsp); err_clk_disable: - if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) + if(mcbsp->pdata->ops && mcbsp->pdata->ops->free) mcbsp->pdata->ops->free(mcbsp->id - 1); /* Disable wakeup behavior */ @@ -382,7 +382,7 @@ static void omap_mcbsp_free(struct omap_mcbsp *mcbsp) { void *reg_cache; - if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) + if(mcbsp->pdata->ops && mcbsp->pdata->ops->free) mcbsp->pdata->ops->free(mcbsp->id - 1); /* Disable wakeup behavior */ -- 2.19.1

1 0

[alsa-devel] Applied "ASoC: omap-mcbsp: No need to initialize max_xx_thres when it is not used" to the asoc tree
by Mark Brown 13 Nov '18

13 Nov '18

The patch ASoC: omap-mcbsp: No need to initialize max_xx_thres when it is not used 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 f15c19af929d380d0acb2a8d8277d688351bd5de Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Thu, 8 Nov 2018 09:30:00 +0200 Subject: [PATCH] ASoC: omap-mcbsp: No need to initialize max_xx_thres when it is not used Initializing to -EINVAL is not correct as the variables are unsigned and if buffer_size is 0 then they are not used anyway. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Tested-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/omap/omap-mcbsp.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 8cab1151d3bb..e0ef4774c710 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -727,9 +727,6 @@ static int omap_mcbsp_init(struct platform_device *pdev) "Unable to create additional controls\n"); goto err_thres; } - } else { - mcbsp->max_tx_thres = -EINVAL; - mcbsp->max_rx_thres = -EINVAL; } ret = omap_mcbsp_st_init(pdev); -- 2.19.1

1 0

[alsa-devel] [PATCH] ASoC: rockchip: add missing slave_config setting for I2S
by Katsuhiro Suzuki 13 Nov '18

13 Nov '18

This patch adds missing prepare_sleve_config that is needed for setup the DMA slave channel for I2S. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/rockchip/rockchip_pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c index 9e7b5fa4cf59..4ac78d7a4b2d 100644 --- a/sound/soc/rockchip/rockchip_pcm.c +++ b/sound/soc/rockchip/rockchip_pcm.c @@ -33,6 +33,7 @@ static const struct snd_pcm_hardware snd_rockchip_hardware = { static const struct snd_dmaengine_pcm_config rk_dmaengine_pcm_config = { .pcm_hardware = &snd_rockchip_hardware, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, .prealloc_buffer_size = 32 * 1024, }; -- 2.19.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Fix dma-unsafe read of scratch registers
by Richard Fitzgerald 13 Nov '18

13 Nov '18

Stack memory isn't DMA-safe so it isn't safe to use either regmap_raw_read or regmap_bulk_read to read into stack memory. The two functions to read the scratch registers were using stack memory and regmap_raw_read. It's not worth allocating memory just for this trivial read, and it isn't time-critical. A simple regmap_read for each register is sufficient. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 37 ++++++++++++++++++++----------------- 1 file changed, 20 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index a53dc174bbf0..66501b8dc46f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -765,38 +765,41 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *mem, static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) { - u16 scratch[4]; + unsigned int scratch[4]; + unsigned int addr = dsp->base + ADSP2_SCRATCH0; + unsigned int i; int ret; - ret = regmap_raw_read(dsp->regmap, dsp->base + ADSP2_SCRATCH0, - scratch, sizeof(scratch)); - if (ret) { - adsp_err(dsp, "Failed to read SCRATCH regs: %d\n", ret); - return; + for (i = 0; i < ARRAY_SIZE(scratch); ++i) { + ret = regmap_read(dsp->regmap, addr + i, &scratch[i]); + if (ret) { + adsp_err(dsp, "Failed to read SCRATCH%u: %d\n", i, ret); + return; + } } adsp_dbg(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n", - be16_to_cpu(scratch[0]), - be16_to_cpu(scratch[1]), - be16_to_cpu(scratch[2]), - be16_to_cpu(scratch[3])); + scratch[0], scratch[1], scratch[2], scratch[3]); } static void wm_adsp2v2_show_fw_status(struct wm_adsp *dsp) { - u32 scratch[2]; + unsigned int scratch[2]; int ret; - ret = regmap_raw_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH0_1, - scratch, sizeof(scratch)); - + ret = regmap_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH0_1, + &scratch[0]); if (ret) { - adsp_err(dsp, "Failed to read SCRATCH regs: %d\n", ret); + adsp_err(dsp, "Failed to read SCRATCH0_1: %d\n", ret); return; } - scratch[0] = be32_to_cpu(scratch[0]); - scratch[1] = be32_to_cpu(scratch[1]); + ret = regmap_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH2_3, + &scratch[1]); + if (ret) { + adsp_err(dsp, "Failed to read SCRATCH2_3: %d\n", ret); + return; + } adsp_dbg(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n", scratch[0] & 0xFFFF, -- 2.11.0

2 3

Re: [alsa-devel] Realtek ALC294 Intel Sunrise Point-LP HD Audio driver - sound does not work
by Takashi Iwai 13 Nov '18

13 Nov '18

On Tue, 13 Nov 2018 19:11:18 +0100, Tomasz PrzybyX2 wrote: > > Hello Takashi, > > > Does the headphone work? > > No. For making the headphone working, you don't need any jack detection, in general. You can disable the auto-mute mode. And, at best, avoid PulseAudio for the basic testing in this level. If both headphone and speaker don't work even with the statically enabled routing, I have no idea what's wrong. You really need to ping ASUS for a better help. It's vendor-specific things, after all. thanks, Takashi > > Internal speakers are muted > Headphones and external speakers connected via JACK output combo are > muted . Even with `options snd_hda_intel index=0 > model=dell-headset-multi` custom flag > > QUOTE from arch wiki > https://wiki.archlinux.org/index.php/Advanced_Linux_Sound_Architecture#Corr… > --- > Correctly detect microphone plugged in a 4-pin 3.5mm (TRRS) jack > On some modern laptops you may have a combined 3.5mm headset jack, > instead of two separated ones, which may not be correctly detected by > default. To make ALSA correctly detect plug-in status on your 3.5mm > jack, you could put the following line into your > /etc/modprobe.d/alsa-base.conf : > > # options snd_hda_intel index=0 model=[YOUR_MODEL_SETTING] > For complete list of options put in [YOUR_MODEL_SETTING], see > HD-Audio-Models.txt. Tested with Gigabyte Aero15 2017 (P65 Model), > with ALC255 and set model to dell-headset-multi. > > ---- > > See main bug report > > https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1784485 > > Others forum issue reports: > > https://forum.manjaro.org/t/alc294-analog-audio-card-detected-but-no-sound-… > > https://forum.ubuntuusers.de/topic/asus-zenbook-lautsprecher-ohne-funktion/ > > > One more report from me: > ``` > > [manjaro@manjaro ~]$ pactl list sinks > Sink #0 > State: SUSPENDED > Name: alsa_output.pci-0000_00_1f.3.analog-stereo > Description: Built-in Audio Analog Stereo > Driver: module-alsa-card.c > Sample Specification: s16le 2ch 44100Hz > Channel Map: front-left,front-right > Owner Module: 6 > Mute: no > Volume: front-left: 30129 / 46% / -20.25 dB, front-right: > 30129 / 46% / -20.25 dB > balance 0.00 > Base Volume: 65536 / 100% / 0.00 dB > Monitor Source: alsa_output.pci-0000_00_1f.3.analog-stereo.monitor > Latency: 0 usec, configured 0 usec > Flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY > Properties: > alsa.resolution_bits = "16" > device.api = "alsa" > device.class = "sound" > alsa.class = "generic" > alsa.subclass = "generic-mix" > alsa.name = "ALC294 Analog" > alsa.id = "ALC294 Analog" > alsa.subdevice = "0" > alsa.subdevice_name = "subdevice #0" > alsa.device = "0" > alsa.card = "0" > alsa.card_name = "HDA Intel PCH" > alsa.long_card_name = "HDA Intel PCH at 0xef528000 irq 130" > alsa.driver_name = "snd_hda_intel" > device.bus_path = "pci-0000:00:1f.3" > sysfs.path = "/devices/pci0000:00/0000:00:1f.3/sound/card0" > device.bus = "pci" > device.vendor.id = "8086" > device.vendor.name = "Intel Corporation" > device.product.id = "9d71" > device.product.name = "Sunrise Point-LP HD Audio" > device.form_factor = "internal" > device.string = "front:0" > device.buffering.buffer_size = "352800" > device.buffering.fragment_size = "176400" > device.access_mode = "mmap+timer" > device.profile.name = "analog-stereo" > device.profile.description = "Analog Stereo" > device.description = "Built-in Audio Analog Stereo" > alsa.mixer_name = "Realtek ALC294" > alsa.components = "HDA:10ec0294,10431a00,00100004 > HDA:8086280b,80860101,00100000" > module-udev-detect.discovered = "1" > device.icon_name = "audio-card-pci" > Ports: > analog-output-speaker: Speakers (priority: 10000, not > available) > analog-output-headphones: Headphones (priority: 9000, > available) > Active Port: analog-output-headphones > Formats: > pcm > [manjaro@manjaro ~]$ > > ``` > > > This is common issue. > > Tomasz > > W dniu 13.11.2018 o 18:37, Takashi Iwai pisze: > > On Sat, 10 Nov 2018 14:42:01 +0100, > > Tomasz PrzybyX2 wrote: > >> > >> Hello Devs > >> > >> Lots of users with Intel Sunrise Point-LP HD Audio driver (ALC294) > >> (snd_hda_intel) have major issue. Not sound at all. Only crackling > >> from external speakers OR silence from internal notebook speakers. > >> Tried fix with extra boot flags does not work at all. > > > > Does the headphone work? If yes, it's possibly either a wrong pin > > config or some extra stuff needed to drive the speaker amp. > > In the latter case, only vendors know the details. > > > > > > Takashi > > > >> > >> Also affected Asus VivoBook R542UF > >> > >> ``` > >> Machine: > >> Type: Laptop System: ASUSTeK product: VivoBook 15_ASUS Laptop X542UF > >> v: 1.0 serial: <filter> > >> Mobo: ASUSTeK model: X542UF v: 1.0 serial: <filter> > >> UEFI: American Megatrends v: X542UF.303 date: 03/15/2018 > >> Audio: > >> Device-1: Intel Sunrise Point-LP HD Audio driver: snd_hda_intel > >> v: kernel bus ID: 00:1f.3 > >> Sound Server: ALSA Kernel: 4.19 > >> ``` > >> ``` > >> aplay -l > >> **** List of PLAYBACK Hardware Devices **** > >> card 0: PCH [HDA Intel PCH], device 0: ALC294 Analog [ALC294 Analog] > >> Subdevices: 1/1 > >> Subdevice #0: subdevice #0 > >> card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] > >> Subdevices: 1/1 > >> Subdevice #0: subdevice #0 > >> card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] > >> Subdevices: 1/1 > >> Subdevice #0: subdevice #0 > >> card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] > >> Subdevices: 1/1 > >> Subdevice #0: subdevice #0 > >> card 0: PCH [HDA Intel PCH], device 9: HDMI 3 [HDMI 3] > >> Subdevices: 1/1 > >> Subdevice #0: subdevice #0 > >> card 0: PCH [HDA Intel PCH], device 10: HDMI 4 [HDMI 4] > >> Subdevices: 1/1 > >> Subdevice #0: subdevice #0 > >> ``` > >> Dmesg parts > >> ``` > >> [ 12.342528] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC294: > >> line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:speaker > >> [ 12.342531] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 > >> (0x0/0x0/0x0/0x0/0x0) > >> [ 12.342533] snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 > >> (0x21/0x0/0x0/0x0/0x0) > >> [ 12.342535] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 > >> [ 12.342536] snd_hda_codec_realtek hdaudioC0D0: inputs: > >> [ 12.342538] snd_hda_codec_realtek hdaudioC0D0: Mic=0x1b > >> ``` > >> ``` > >> [ 12.634171] input: HDA Intel PCH Headphone as > >> /devices/pci0000:00/0000:00:1f.3/sound/card0/input18 > >> [ 12.634389] input: HDA Intel PCH HDMI/DP,pcm=3 as > >> /devices/pci0000:00/0000:00:1f.3/sound/card0/input19 > >> [ 12.634563] input: HDA Intel PCH HDMI/DP,pcm=7 as > >> /devices/pci0000:00/0000:00:1f.3/sound/card0/input20 > >> [ 12.634737] input: HDA Intel PCH HDMI/DP,pcm=8 as > >> /devices/pci0000:00/0000:00:1f.3/sound/card0/input21 > >> [ 12.634905] input: HDA Intel PCH HDMI/DP,pcm=9 as > >> /devices/pci0000:00/0000:00:1f.3/sound/card0/input22 > >> [ 12.635045] input: HDA Intel PCH HDMI/DP,pcm=10 as > >> /devices/pci0000:00/0000:00:1f.3/sound/card0/input23 > >> ``` > >> > >> Regards > >> > >> Tomasz > >> > >> _______________________________________________ > >> Alsa-devel mailing list > >> Alsa-devel(a)alsa-project.org > >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >

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[alsa-devel] [sound:for-linus 5/5] sound/pci//hda/patch_ca0132.c:8416:3: error: implicit declaration of function 'pci_iounmap'; did you mean 'pcim_iounmap'?
by kbuild test robot 13 Nov '18

13 Nov '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: d99501b8575dc1248bacf1b58d2241cb4b265d49 commit: d99501b8575dc1248bacf1b58d2241cb4b265d49 [5/5] ALSA: hda/ca0132 - Call pci_iounmap() instead of iounmap() config: sh-allyesconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout d99501b8575dc1248bacf1b58d2241cb4b265d49 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): sound/pci//hda/patch_ca0132.c: In function 'ca0132_free': >> sound/pci//hda/patch_ca0132.c:8416:3: error: implicit declaration of function 'pci_iounmap'; did you mean 'pcim_iounmap'? [-Werror=implicit-function-declaration] pci_iounmap(codec->bus->pci, spec->mem_base); ^~~~~~~~~~~ pcim_iounmap sound/pci//hda/patch_ca0132.c: In function 'patch_ca0132': sound/pci//hda/patch_ca0132.c:8799:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'? [-Werror=implicit-function-declaration] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^~~~~~~~~ pcim_iomap sound/pci//hda/patch_ca0132.c:8799:18: warning: assignment makes pointer from integer without a cast [-Wint-conversion] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^ cc1: some warnings being treated as errors vim +8416 sound/pci//hda/patch_ca0132.c 8386 8387 static void ca0132_free(struct hda_codec *codec) 8388 { 8389 struct ca0132_spec *spec = codec->spec; 8390 8391 cancel_delayed_work_sync(&spec->unsol_hp_work); 8392 snd_hda_power_up(codec); 8393 switch (spec->quirk) { 8394 case QUIRK_SBZ: 8395 sbz_exit_chip(codec); 8396 break; 8397 case QUIRK_ZXR: 8398 zxr_exit_chip(codec); 8399 break; 8400 case QUIRK_R3D: 8401 r3d_exit_chip(codec); 8402 break; 8403 case QUIRK_AE5: 8404 ae5_exit_chip(codec); 8405 break; 8406 case QUIRK_R3DI: 8407 r3di_gpio_shutdown(codec); 8408 break; 8409 } 8410 8411 snd_hda_sequence_write(codec, spec->base_exit_verbs); 8412 ca0132_exit_chip(codec); 8413 8414 snd_hda_power_down(codec); 8415 if (spec->mem_base) > 8416 pci_iounmap(codec->bus->pci, spec->mem_base); 8417 kfree(spec->spec_init_verbs); 8418 kfree(codec->spec); 8419 } 8420 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 17/17] ASoc: bcm2835-i2s: Switch to SPDX identifier
by Stefan Wahren 13 Nov '18

13 Nov '18

Adopt the SPDX license identifier headers to ease license compliance management. Cc: Florian Kauer <florian.kauer(a)koalo.de> Signed-off-by: Stefan Wahren <stefan.wahren(a)i2se.com> --- sound/soc/bcm/bcm2835-i2s.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index d5f73a8..0e4a0e9 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -1,3 +1,4 @@ +// SPDX-License-Identifier: GPL-2.0 /* * ALSA SoC I2S Audio Layer for Broadcom BCM2835 SoC * @@ -20,15 +21,6 @@ * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver * Author: Timur Tabi <timur(a)freescale.com> * Copyright 2007-2010 Freescale Semiconductor, Inc. - * - * 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. - * - * 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. */ #include <linux/bitops.h> -- 2.7.4

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[alsa-devel] snd_pcm_hw_params resets microphone boost
by sonntex 13 Nov '18

13 Nov '18

I have small program that captures audio from a microphone. As my microphone is very sensitive, I prefer to mute the microphone boost before capturing. I open alsamixer and set volume of the microphone boost to zero, save settings and exit. Then I execute my program. I see that it resets volume of the microphone boost in snd_pcm_hw_params. What should I do resolve this problem? Where can I find default value and change it?

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[alsa-devel] Realtek ALC294 Intel Sunrise Point-LP HD Audio driver - sound does not work
by Tomasz Przybył 13 Nov '18

13 Nov '18

Hello Devs Lots of users with Intel Sunrise Point-LP HD Audio driver (ALC294) (snd_hda_intel) have major issue. Not sound at all. Only crackling from external speakers OR silence from internal notebook speakers. Tried fix with extra boot flags does not work at all. Also affected Asus VivoBook R542UF ``` Machine: Type: Laptop System: ASUSTeK product: VivoBook 15_ASUS Laptop X542UF v: 1.0 serial: <filter> Mobo: ASUSTeK model: X542UF v: 1.0 serial: <filter> UEFI: American Megatrends v: X542UF.303 date: 03/15/2018 Audio: Device-1: Intel Sunrise Point-LP HD Audio driver: snd_hda_intel v: kernel bus ID: 00:1f.3 Sound Server: ALSA Kernel: 4.19 ``` ``` aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC294 Analog [ALC294 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 10: HDMI 4 [HDMI 4] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` Dmesg parts ``` [ 12.342528] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC294: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:speaker [ 12.342531] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 12.342533] snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 12.342535] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 12.342536] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 12.342538] snd_hda_codec_realtek hdaudioC0D0: Mic=0x1b ``` ``` [ 12.634171] input: HDA Intel PCH Headphone as /devices/pci0000:00/0000:00:1f.3/sound/card0/input18 [ 12.634389] input: HDA Intel PCH HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input19 [ 12.634563] input: HDA Intel PCH HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input20 [ 12.634737] input: HDA Intel PCH HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input21 [ 12.634905] input: HDA Intel PCH HDMI/DP,pcm=9 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input22 [ 12.635045] input: HDA Intel PCH HDMI/DP,pcm=10 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input23 ``` Regards Tomasz

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[alsa-devel] [PATCH v5 2/3] ALSA: hda: fix front speakers on Huawei MBXP.
by Ayman Bagabas 13 Nov '18

13 Nov '18

This patch solves bug 200501 'Only 2 of 4 speakers playing sound.' https://bugzilla.kernel.org/show_bug.cgi?id=200501 It enables the front speakers on Huawei Matebook X Pro laptops. These laptops come with Dolby Atmos sound system and these pins configuration enables the front speakers. Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fa61674a5605..135c0826859b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5493,6 +5493,7 @@ enum { ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, + ALC256_FIXUP_HUAWEI_MBXP_PINS, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, }; @@ -5764,6 +5765,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x12, 0x90a60130}, + {0x13, 0x40000000}, + {0x14, 0x90170110}, + {0x18, 0x411111f0}, + {0x19, 0x04a11040}, + {0x1a, 0x411111f0}, + {0x1b, 0x90170112}, + {0x1d, 0x40759a05}, + {0x1e, 0x411111f0}, + {0x21, 0x04211020}, + { } + } + }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_x101_headset_mic, @@ -6594,6 +6611,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ #if 0 -- 2.17.2

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[alsa-devel] using the softvol plugin with min_dB > 0
by Guillermo Rodriguez Garcia 13 Nov '18

13 Nov '18

Hi all, I have a board with a tlv320aic3106 codec and a digital mic. The tlv320aic3106 does not support hardware gain for the digital mic, so I am using the softvol plugin to provide this via software. I would like to setup softvol so that it applies a gain in a range that goes from approx +10dB to approx +40dB. The problem is that softvol only accepts negative values for min_dB, i.e. assumes that you *always* want to do attenuation, and that you might or might not want to have gain (max_dB can be zero or negative). Is there any reason for this, or is it just historical ? (I am aware that the original softvol plugin only supported attenuation, and that support for gain was added later) Is it possible to setup softvol to accept a positive min_dB value ? Best regards, Guillermo Rodriguez Garcia guille.rodriguez(a)gmail.com

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[alsa-devel] Help with extplug transfer callback
by GitHub issues - opened 13 Nov '18

13 Nov '18

alsa-project/alsa-lib issue #2 was opened from dchabot: I've tried posting the following at alsa-users mailing list, but keep getting hit with "awaiting moderator approval" (I am registered to the list). I'm trying to create an extplug and having trouble deciphering how the source and destination buffer reading/writing is supposed to be done in the transfer callback. My goal is to create an extplug that takes a stereo data stream, divides that into 6 or 8 output streams, and applies FIR filters to each output stream. With the code I currently have (https://github.com/dchabot/extplug), trying to play a 1kHz tone results in a garbled, noisy mess. There is a tone in there, but it is a mess. I started from the skeleton code here - http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins.html. Note that the snd_pcm_extplug_create call here should be passing the "slave" variable, not "conf". That tripped me up for the better part of an hour. I based my initial transfer callback (snd_pcm_extplug_callback_t.transfer member) on that found in the speex plugin. Initially I didn't notice that speex operated with single in/out channels, so you can imagine how that went. Looking at another example of an extplug (https://github.com/dpapavas/alsaloudness), I noted that I should be iterating over each channel for src_areas and dst_areas. I just cannot see how this should be done properly. Is documentation for this critical step missing or simply hard to find. Seems kinda important to know how buffer operations should be done... At this point, all I want to do is copy input channels to output channels. I'm grappling with the fact that this is more complicated than I initially thought, or perhaps more difficult than it should be. My transfer callback currently looks like this: ``` /* copied from pcm-plugins/speex/pcm-speex.c */ static inline void* area_addr(const snd_pcm_channel_area_t *area, snd_pcm_uframes_t offset) { unsigned int bitofs = area->first + area->step * offset; return (char *) area->addr + bitofs / 8; } static snd_pcm_sframes_t transfer_callback(snd_pcm_extplug_t *ext, const snd_pcm_channel_area_t *dst_areas, snd_pcm_uframes_t dst_offset, const snd_pcm_channel_area_t *src_areas, snd_pcm_uframes_t src_offset, snd_pcm_uframes_t size) { int C = ext->channels; short *src, *dst; for(int i = 0; i < C; i++) { src = area_addr(&src_areas[i], src_offset); dst = area_addr(&dst_areas[i], dst_offset); memcpy(dst, src, size); } return size; } ``` Need a little help please. Issue URL : https://github.com/alsa-project/alsa-lib/issues/2 Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] [PATCH 1/4] README: update Vinod's email
by Vinod Koul 13 Nov '18

13 Nov '18

Update email address in README from old invalid id Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- README | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/README b/README index 689ca75e07fd..8ea6c26f5381 100644 --- a/README +++ b/README @@ -1,7 +1,7 @@ README for tinycompress ======================= - vinod.koul(a)linux.intel.com - ========================== + vkoul(a)kernel.org + ================ 1. WHAT tinycompress is a userspace library for anyone who wants to use the ALSA -- 2.14.5

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[alsa-devel] [PATCH] tinycompress: Update headers to include AUDIOCODEC_BESPOKE
by Charles Keepax 13 Nov '18

13 Nov '18

commit dbb6b94339e8 ("ALSA: compress: Add SND_AUDIOCODEC_BESPOKE") added an additional CODEC type for bespoke CODECs. Update the local headers to include this. Reported-by: Andrew Ford <andrew.ford(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- include/sound/compress_params.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/include/sound/compress_params.h b/include/sound/compress_params.h index 9802292..7286556 100644 --- a/include/sound/compress_params.h +++ b/include/sound/compress_params.h @@ -38,8 +38,9 @@ #define SND_AUDIOCODEC_IEC61937 ((__u32) 0x0000000B) #define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C) #define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D) +#define SND_AUDIOCODEC_BESPOKE ((__u32) 0x0000000E) -#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_G729 +#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_BESPOKE #define SND_AUDIOPROFILE_PCM ((__u32) 0x00000001) #define SND_AUDIOCHANMODE_MP3_MONO ((__u32) 0x00000001) -- 2.11.0

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[alsa-devel] [PATCH - dmix v5 0/1] pcm: dmix: Add option to allow alignment of slave pointers
by twischer＠de.adit-jv.com 13 Nov '18

13 Nov '18

On 11/12/18 17:20, Takashi Iwai wrote: > Please provide your own sign-off when you submitting a patch, too. > > The subject line should be adjusted better to match with the change; > it's about adding an option to allow alignment of hwptr reporting. > > About the changes: > the term "hw_ptr_filter" might sound confusing. IMO, this is about > the (hw)ptr alignment, not about filtering. > Also, a too long option string can easily lead to a misspell :) > > After all, we do either "no" alignment, "roundup" or "rounddown", or > "auto" choice. > > > thanks, > > Takashi Subject changed. Sign-off added. Option changed to hw_ptr_alignment. Option values changed to no, roundup, rounddown, auto Best regards Timo

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[alsa-devel] [PATCH] ALSA: oxfw: add support for APOGEE duet FireWire
by Takashi Sakamoto 13 Nov '18

13 Nov '18

This commit adds support for APOGEE duet FireWire, launched 2007, already discontinued. This model uses Oxford Semiconductor FW971 as its communication engine. Below is information on Configuration ROM of this unit. The unit supports some AV/C commands defined by Audio subunit specification and vendor dependent commands. $ ./hinawa-config-rom-printer /dev/fw1 { 'bus-info': { 'adj': False, 'bmc': False, 'chip_ID': 42949742248, 'cmc': False, 'cyc_clk_acc': 255, 'generation': 0, 'imc': False, 'isc': True, 'link_spd': 3, 'max_ROM': 0, 'max_rec': 64, 'name': '1394', 'node_vendor_ID': 987, 'pmc': False}, 'root-directory': [ ['VENDOR', 987], ['DESCRIPTOR', 'Apogee Electronics'], ['MODEL', 122333], ['DESCRIPTOR', 'Duet'], [ 'NODE_CAPABILITIES', { 'addressing': {'64': True, 'fix': True, 'prv': False}, 'misc': {'int': False, 'ms': False, 'spt': True}, 'state': { 'atn': False, 'ded': False, 'drq': True, 'elo': False, 'init': False, 'lst': True, 'off': False}, 'testing': {'bas': False, 'ext': False}}], [ 'UNIT', [ ['SPECIFIER_ID', 41005], ['VERSION', 65537], ['MODEL', 122333], ['DESCRIPTOR', 'Duet']]]]} Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 1 + sound/firewire/oxfw/oxfw.c | 8 ++++++++ 2 files changed, 9 insertions(+) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 8a146b039276..44cedb65bb88 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -41,6 +41,7 @@ config SND_OXFW * Mackie(Loud) U.420/U.420d * TASCAM FireOne * Stanton Controllers & Systems 1 Deck/Mixer + * APOGEE duet FireWire To compile this driver as a module, choose M here: the module will be called snd-oxfw. diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index afb78d90384b..3d27f3378d5d 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -20,6 +20,7 @@ #define VENDOR_LACIE 0x00d04b #define VENDOR_TASCAM 0x00022e #define OUI_STANTON 0x001260 +#define OUI_APOGEE 0x0003db #define MODEL_SATELLITE 0x00200f @@ -397,6 +398,13 @@ static const struct ieee1394_device_id oxfw_id_table[] = { .vendor_id = OUI_STANTON, .model_id = 0x002000, }, + // APOGEE, duet FireWire + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_APOGEE, + .model_id = 0x01dddd, + }, { } }; MODULE_DEVICE_TABLE(ieee1394, oxfw_id_table); -- 2.19.1

2 1

[alsa-devel] [PATCH - dmix v4 0/1] pcm: dmix: Align slave_hw_ptr to slave period
by twischer＠de.adit-jv.com 12 Nov '18

12 Nov '18

Hi Takashi, in the following we are providing the patch as requested by you: On 11/6/18 17:32, Takashi Iwai wrote: > I guess we may provide an option for choosing one of the following > modes: > > - current default mode; > no latency, but wakeup may be delayed, and may require 2*periods > > - appl_ptr round up to the next period; > the current workaround for periods <= 2. > This imposes the start latency. > > - round down to the current period start; > No latency, but will drop a few samples at start. > > ... and ideally, provide a mode "auto" for choosing the appropriate > one depending on the situation by a smart guess. Please provide your feedback. Best regards Timo

2 2

[alsa-devel] [PATCH] ALSA: hda/ca0132 - Call pci_iounmap() instead of iounmap()
by Takashi Iwai 12 Nov '18

12 Nov '18

We need to call pci_iounmap() instead of iounmap() for the regions obtained via pci_iomap() call for some archs that need special treatment. Fixes: aa31704fd81c ("ALSA: hda/ca0132: Add PCI region2 iomap for SBZ") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 0a24037184c3..bdbbc51e8d18 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -8413,7 +8413,7 @@ static void ca0132_free(struct hda_codec *codec) snd_hda_power_down(codec); if (spec->mem_base) - iounmap(spec->mem_base); + pci_iounmap(codec->bus->pci, spec->mem_base); kfree(spec->spec_init_verbs); kfree(codec->spec); } -- 2.19.1

1 0

[alsa-devel] [PATCH 0/3] constify snd_soc_dai_ops structures
by Julia Lawall 12 Nov '18

12 Nov '18

The snd_soc_dai_ops structures are only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structures const as well. Done with the help of Coccinelle. --- drivers/soundwire/intel.c | 4 ++-- sound/soc/codecs/ak4458.c | 2 +- sound/soc/codecs/ak5558.c | 2 +- sound/soc/codecs/hdac_hda.c | 2 +- sound/soc/codecs/tas6424.c | 2 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 8 ++++---- 6 files changed, 10 insertions(+), 10 deletions(-)

3 6

[alsa-devel] [PATCH] ALSA: hda/realtek - Add quirk entry for HP Pavilion 15
by Takashi Iwai 12 Nov '18

12 Nov '18

HP Pavilion 15 (103c:820d) with ALC295 codec requires the quirk for the mute LED control over mic3 pin. Added the corresponding quirk entry. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201653 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fa61674a5605..970bc44a378b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6481,6 +6481,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC), + SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), SND_PCI_QUIRK(0x103c, 0x82bf, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), -- 2.19.1

1 0

[alsa-devel] [PATCH v4 3/3] ALSA: hda: add support for Huawei WMI micmute LED
by Ayman Bagabas 11 Nov '18

11 Nov '18

Some of Huawei laptops come with a LED in the micmute key. This patch enables and disable this LED accordingly. This enables micmute LED for: 1. Matebook X (19e5:3200), (19e5:3201) 2. Matebook X Pro (19e5:3204) Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> --- sound/pci/hda/huawei_wmi_helper.c | 47 +++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 14 ++++++++- 2 files changed, 60 insertions(+), 1 deletion(-) create mode 100644 sound/pci/hda/huawei_wmi_helper.c diff --git a/sound/pci/hda/huawei_wmi_helper.c b/sound/pci/hda/huawei_wmi_helper.c new file mode 100644 index 000000000000..7c91f914ffba --- /dev/null +++ b/sound/pci/hda/huawei_wmi_helper.c @@ -0,0 +1,47 @@ +// SPDX-License-Identifier: GPL-2.0 +/* Helper functions for Huawei WMI micmute LED; + * to be included from codec driver + */ + +#if IS_ENABLED(CONFIG_HUAWEI_LAPTOP) +#include <linux/platform_data/x86/huawei_wmi.h> + +static int (*huawei_wmi_micmute_led_set_func)(bool); + +static void update_huawei_wmi_micmute_led(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + huawei_wmi_micmute_led_set_func(spec->micmute_led.led_value); +} + +static void alc_fixup_huawei_wmi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + bool removefunc = false; + + if (action == HDA_FIXUP_ACT_PROBE) { + if (!huawei_wmi_micmute_led_set_func) + huawei_wmi_micmute_led_set_func = symbol_request(huawei_wmi_micmute_led_set); + if (!huawei_wmi_micmute_led_set_func) { + codec_warn(codec, "Failed to find huawei_wmi symbol huawei_wmi_micmute_led_set\n"); + return; + } + + removefunc = (huawei_wmi_micmute_led_set_func(false) < 0) + || (snd_hda_gen_add_micmute_led(codec, update_huawei_wmi_micmute_led) < 0); + } + + if (huawei_wmi_micmute_led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { + symbol_put(huawei_wmi_micmute_led_set); + huawei_wmi_micmute_led_set_func = NULL; + } +} + +#else + +static void alc_fixup_huawei_wmi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ +} + +#endif diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 135c0826859b..5431dee40f4f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5374,6 +5374,9 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" +/* for alc_fixup_huawei_micmute_led */ +#include "huawei_wmi_helper.c" + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -5494,6 +5497,7 @@ enum { ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, ALC256_FIXUP_HUAWEI_MBXP_PINS, + ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, }; @@ -5765,6 +5769,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_huawei_wmi + }, [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5779,7 +5787,9 @@ static const struct hda_fixup alc269_fixups[] = { {0x1e, 0x411111f0}, {0x21, 0x04211020}, { } - } + }, + .chained = true, + .chain_id = ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, @@ -6611,6 +6621,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x19e5, 0x3200, "Huawei MBX", ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED), + SND_PCI_QUIRK(0x19e5, 0x3201, "Huawei MBX", ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- 2.17.2

3 2

[alsa-devel] [PATCH 00/17] bcm2835: Treewide switch to SPDX identifier
by Stefan Wahren 10 Nov '18

10 Nov '18

After fixing the module license of some bcm2835 drivers in "Fix MODULE_LICENSE of bcm2835 drivers", we can finally switch all outstanding bcm2835 / Raspberry Pi drivers to SPDX identifier and drop the unnecessary license text. I omit the staging area, because there are proposed a lot of changes for vc04_services. I hope it is okay to have the binding header integrated in the tree-wise patches instead of a separate one. This should avoid conflicts while merging. Stefan Wahren (17): ARM: dts: bcm2835-rpi-zero: Switch to SPDX identifier ARM: mach-bcm: Switch bcm2835 and platsmp to SPDX identifier hw_random: bcm2835-rng: Switch to SPDX identifier clk: bcm2835: Switch to SPDX identifier clocksource: bcm2835: Switch to SPDX identifier dma: bcm2835: Switch to SPDX identifier firmware: raspberrypi: Switch to SPDX identifier i2c: bcm2835: Switch to SPDX identifier irqchip: bcm283x: Switch to SPDX identifier mailbox: bcm2835: Switch to SPDX identifier mmc: bcm2835: Switch to SPDX identifier pinctrl: bcm2835: Switch to SPDX identifier pwm: bcm2835: Switch to SPDX identifier soc: bcm: Switch raspberrypi-power to SPDX identifier spi: bcm2835: Switch to SPDX identifier thermal: bcm2835: Switch to SPDX identifier ASoc: bcm2835-i2s: Switch to SPDX identifier arch/arm/boot/dts/bcm2835-rpi-zero-w.dts | 8 +------- arch/arm/boot/dts/bcm2835-rpi-zero.dts | 8 +------- arch/arm/mach-bcm/board_bcm2835.c | 11 +---------- arch/arm/mach-bcm/platsmp.c | 10 +--------- arch/arm/mach-bcm/platsmp.h | 6 +----- drivers/char/hw_random/bcm2835-rng.c | 7 ++----- drivers/clk/bcm/clk-bcm2835-aux.c | 11 +---------- drivers/clk/bcm/clk-bcm2835.c | 12 +----------- drivers/clocksource/bcm2835_timer.c | 15 +-------------- drivers/dma/bcm2835-dma.c | 11 +---------- drivers/firmware/raspberrypi.c | 5 +---- drivers/i2c/busses/i2c-bcm2835.c | 10 +--------- drivers/irqchip/irq-bcm2835.c | 11 +---------- drivers/irqchip/irq-bcm2836.c | 11 +---------- drivers/mailbox/bcm2835-mailbox.c | 8 +------- drivers/mmc/host/bcm2835.c | 13 +------------ drivers/pinctrl/bcm/pinctrl-bcm2835.c | 11 +---------- drivers/pwm/pwm-bcm2835.c | 5 +---- drivers/soc/bcm/raspberrypi-power.c | 5 +---- drivers/spi/spi-bcm2835.c | 11 +---------- drivers/spi/spi-bcm2835aux.c | 11 +---------- drivers/thermal/broadcom/bcm2835_thermal.c | 11 +---------- include/dt-bindings/clock/bcm2835-aux.h | 10 +--------- include/dt-bindings/clock/bcm2835.h | 10 +--------- include/dt-bindings/pinctrl/bcm2835.h | 8 +------- include/dt-bindings/power/raspberrypi-power.h | 5 +---- include/soc/bcm2835/raspberrypi-firmware.h | 5 +---- sound/soc/bcm/bcm2835-i2s.c | 10 +--------- 28 files changed, 29 insertions(+), 230 deletions(-) -- 2.7.4

1 0

[alsa-devel] [PATCH v4 2/3] ALSA: hda: fix front speakers on Huawei MBXP.
by Ayman Bagabas 10 Nov '18

10 Nov '18

This patch solves bug 200501 'Only 2 of 4 speakers playing sound.' https://bugzilla.kernel.org/show_bug.cgi?id=200501 It enables the front speakers on Huawei Matebook X Pro laptops. These laptops come with Dolby Atmos sound system and these pins configuration enables the front speakers. Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fa61674a5605..135c0826859b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5493,6 +5493,7 @@ enum { ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, + ALC256_FIXUP_HUAWEI_MBXP_PINS, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, }; @@ -5764,6 +5765,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x12, 0x90a60130}, + {0x13, 0x40000000}, + {0x14, 0x90170110}, + {0x18, 0x411111f0}, + {0x19, 0x04a11040}, + {0x1a, 0x411111f0}, + {0x1b, 0x90170112}, + {0x1d, 0x40759a05}, + {0x1e, 0x411111f0}, + {0x21, 0x04211020}, + { } + } + }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_x101_headset_mic, @@ -6594,6 +6611,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ #if 0 -- 2.17.2

1 0

[alsa-devel] [PATCH] ASoC: Intel: Power down links before turning off display audio power
by Pierre-Louis Bossart 09 Nov '18

09 Nov '18

On certain platforms, Display HDMI HDA codec was not going to sleep state after the use when links are powered down after turning off the display power. As per the HW recommendation, links are powered down before turning off the display power to ensure that the codec goes to sleep state. This patch was updated from an earlier version submitted upstream [1] which conflicted with the changes merged for HDaudio codec support with the Intel DSP. [1] https://patchwork.kernel.org/patch/10540213/ Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/hdac_hdmi.c | 13 +++++++------ sound/soc/intel/skylake/skl.c | 12 ++++++------ 2 files changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 4e9854889a95..e0fe682cea04 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -2187,11 +2187,6 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) */ snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - err = snd_hdac_display_power(bus, false); - if (err < 0) { - dev_err(dev, "Cannot turn on display power on i915\n"); - return err; - } hlink = snd_hdac_ext_bus_get_link(bus, dev_name(dev)); if (!hlink) { @@ -2201,7 +2196,13 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) snd_hdac_ext_bus_link_put(bus, hlink); - return 0; + err = snd_hdac_display_power(bus, false); + if (err < 0) { + dev_err(dev, "Cannot turn off display power on i915\n"); + return err; + } + + return err; } static int hdac_hdmi_runtime_resume(struct device *dev) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 29225623b4b4..1586c97d9450 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -815,6 +815,12 @@ static void skl_probe_work(struct work_struct *work) } } + /* + * we are done probing so decrement link counts + */ + list_for_each_entry(hlink, &bus->hlink_list, list) + snd_hdac_ext_bus_link_put(bus, hlink); + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { err = snd_hdac_display_power(bus, false); if (err < 0) { @@ -824,12 +830,6 @@ static void skl_probe_work(struct work_struct *work) } } - /* - * we are done probing so decrement link counts - */ - list_for_each_entry(hlink, &bus->hlink_list, list) - snd_hdac_ext_bus_link_put(bus, hlink); - /* configure PM */ pm_runtime_put_noidle(bus->dev); pm_runtime_allow(bus->dev); -- 2.17.1

1 1

[alsa-devel] [PATCH] ASoC: sun8i-codec: fix crash on module removal
by Vasily Khoruzhick 09 Nov '18

09 Nov '18

drvdata is actually sun8i_codec, not snd_soc_card, so it crashes when calling snd_soc_card_get_drvdata(). Drop card and scodec vars anyway since we don't need to disable/unprepare clocks - it's already done by calling runtime_suspend() Drop clk_disable_unprepare() calls for the same reason. Fixes: 36c684936fae7 ("ASoC: Add sun8i digital audio codec") Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- sound/soc/sunxi/sun8i-codec.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index 522a72fde78d..c4196d4e5915 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -605,16 +605,10 @@ static int sun8i_codec_probe(struct platform_device *pdev) static int sun8i_codec_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - struct sun8i_codec *scodec = snd_soc_card_get_drvdata(card); - pm_runtime_disable(&pdev->dev); if (!pm_runtime_status_suspended(&pdev->dev)) sun8i_codec_runtime_suspend(&pdev->dev); - clk_disable_unprepare(scodec->clk_module); - clk_disable_unprepare(scodec->clk_bus); - return 0; } -- 2.19.1

3 2

[alsa-devel] [PATCH] ASoC: rsnd: use ring buffer for rsnd_mod_name()
by Kuninori Morimoto 09 Nov '18

09 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit c0ea089dbad4 ("ASoC: rsnd: rsnd_mod_name() handles both name and ID") merged "name" and "ID" on rsnd_mod_name() to handle sub-ID (= for CTU/BUSIF). Then, it decided to share static char to avoid pointless memory. But, it doesn't work correctry in below case, because last called name will be used. dev_xxx(dev, "%s is connected to %s\n", rsnd_mod_name(mod_a), /* ssiu[00] */ rsnd_mod_name(mod_b)); /* ssi[0] */ -> rcar_sound ec500000.sound: ssi[0] is connected to ssi[0] ~~~~~~ ~~~~~~ We still don't want to have pointless memory, so let's use ring buffer. 16byte x 5 is very enough for this purpose. dev_xxx(dev, "%s is connected to %s\n", rsnd_mod_name(mod_a), /* ssiu[00] */ rsnd_mod_name(mod_b)); /* ssi[0] */ -> rcar_sound ec500000.sound: ssiu[00] is connected to ssi[0] ~~~~~~~~ ~~~~~~ Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 12f559e..56469ac 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -137,10 +137,17 @@ struct dma_chan *rsnd_mod_dma_req(struct rsnd_dai_stream *io, return mod->ops->dma_req(io, mod); } +#define MOD_NAME_NUM 5 #define MOD_NAME_SIZE 16 char *rsnd_mod_name(struct rsnd_mod *mod) { - static char name[MOD_NAME_SIZE]; + static char names[MOD_NAME_NUM][MOD_NAME_SIZE]; + static int num; + char *name = names[num]; + + num++; + if (num >= MOD_NAME_NUM) + num = 0; /* * Let's use same char to avoid pointlessness memory -- 2.7.4

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[alsa-devel] [PATCH] ALSA: hda/ca0132 - Optimize for non-PCI configuration
by Takashi Iwai 09 Nov '18

09 Nov '18

All the recent support of Creative boards and onboard audio depend on PCI, but they can't be trimmed easily even if you build without CONFIG_PCI, since the quirk is detected dynamically and the code has many branches with the flag check like spec->quirk type or spec->use_alt_functions. This patch makes these checks static for CONFIG_PCI=n case so that the compiler optimizes out. The access to flags are replaced with macros that are replaced with a static value for CONFIG_PCI=n. The macros look slightly ugly for avoiding compiler warnings wrt unused variables, and some additional default-case handlings for another compiler warnings, but the rest are very straightforward changes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_ca0132.c | 205 +++++++++++++++++++++-------------- 1 file changed, 123 insertions(+), 82 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 0a24037184c3..9c0b94ba786e 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1081,6 +1081,18 @@ enum { QUIRK_AE5, }; +#ifdef CONFIG_PCI +#define ca0132_quirk(spec) ((spec)->quirk) +#define ca0132_use_pci_mmio(spec) ((spec)->use_pci_mmio) +#define ca0132_use_alt_functions(spec) ((spec)->use_alt_functions) +#define ca0132_use_alt_controls(spec) ((spec)->use_alt_controls) +#else +#define ca0132_quirk(spec) ({ (void)(spec); QUIRK_NONE; }) +#define ca0132_use_alt_functions(spec) ({ (void)(spec); false; }) +#define ca0132_use_pci_mmio(spec) ({ (void)(spec); false; }) +#define ca0132_use_alt_controls(spec) ({ (void)(spec); false; }) +#endif + static const struct hda_pintbl alienware_pincfgs[] = { { 0x0b, 0x90170110 }, /* Builtin Speaker */ { 0x0c, 0x411111f0 }, /* N/A */ @@ -3100,7 +3112,7 @@ static void dspload_post_setup(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; codec_dbg(codec, "---- dspload_post_setup ------\n"); - if (!spec->use_alt_functions) { + if (!ca0132_use_alt_functions(spec)) { /*set DSP speaker to 2.0 configuration*/ chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x18), 0x08080080); chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x19), 0x3f800000); @@ -3332,7 +3344,7 @@ static void ca0132_gpio_init(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_AE5: snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00); @@ -3343,6 +3355,8 @@ static void ca0132_gpio_init(struct hda_codec *codec) snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00); snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x5B); break; + default: + break; } } @@ -3352,7 +3366,7 @@ static void ca0132_gpio_setup(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, 0x07); @@ -3371,6 +3385,8 @@ static void ca0132_gpio_setup(struct hda_codec *codec) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0x0C); break; + default: + break; } } @@ -4171,7 +4187,7 @@ static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec) switch (spec->cur_out_type) { case SPEAKER_OUT: - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: ca0113_mmio_gpio_set(codec, 7, false); ca0113_mmio_gpio_set(codec, 4, true); @@ -4202,10 +4218,12 @@ static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec) chipio_set_control_param(codec, 0x0d, 0xa4); chipio_write(codec, 0x18b03c, 0x00000012); break; + default: + break; } break; case HEADPHONE_OUT: - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: ca0113_mmio_gpio_set(codec, 7, true); ca0113_mmio_gpio_set(codec, 4, true); @@ -4237,10 +4255,12 @@ static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec) chipio_set_control_param(codec, 0x0d, 0xa1); chipio_write(codec, 0x18b03c, 0x00000012); break; + default: + break; } break; case SURROUND_OUT: - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: ca0113_mmio_gpio_set(codec, 7, false); ca0113_mmio_gpio_set(codec, 4, true); @@ -4271,6 +4291,8 @@ static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec) chipio_set_control_param(codec, 0x0d, 0xa4); chipio_write(codec, 0x18b03c, 0x00000012); break; + default: + break; } break; } @@ -4445,7 +4467,7 @@ static void ca0132_unsol_hp_delayed(struct work_struct *work) to_delayed_work(work), struct ca0132_spec, unsol_hp_work); struct hda_jack_tbl *jack; - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_select_out(spec->codec); else ca0132_select_out(spec->codec); @@ -4529,14 +4551,14 @@ static int ca0132_alt_set_vipsource(struct hda_codec *codec, int val) chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) chipio_set_conn_rate(codec, 0x0F, SR_96_000); if (spec->in_enum_val == REAR_LINE_IN) tmp = FLOAT_ZERO; else { - if (spec->quirk == QUIRK_SBZ) + if (ca0132_quirk(spec) == QUIRK_SBZ) tmp = FLOAT_THREE; else tmp = FLOAT_ONE; @@ -4548,7 +4570,7 @@ static int ca0132_alt_set_vipsource(struct hda_codec *codec, int val) codec_dbg(codec, "%s: on.", __func__); chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_16_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_16_000); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) chipio_set_conn_rate(codec, 0x0F, SR_16_000); if (spec->effects_switch[VOICE_FOCUS - EFFECT_START_NID]) @@ -4644,7 +4666,7 @@ static int ca0132_alt_select_in(struct hda_codec *codec) switch (spec->cur_mic_type) { case REAR_MIC: - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_R3D: ca0113_mmio_gpio_set(codec, 0, false); @@ -4668,14 +4690,14 @@ static int ca0132_alt_select_in(struct hda_codec *codec) chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) chipio_set_conn_rate(codec, 0x0F, SR_96_000); dspio_set_uint_param(codec, 0x80, 0x00, tmp); chipio_set_stream_control(codec, 0x03, 1); chipio_set_stream_control(codec, 0x04, 1); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: chipio_write(codec, 0x18B098, 0x0000000C); chipio_write(codec, 0x18B09C, 0x0000000C); @@ -4688,12 +4710,14 @@ static int ca0132_alt_select_in(struct hda_codec *codec) chipio_write(codec, 0x18B098, 0x0000000C); chipio_write(codec, 0x18B09C, 0x0000004C); break; + default: + break; } ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val); break; case REAR_LINE_IN: ca0132_mic_boost_set(codec, 0); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_R3D: ca0113_mmio_gpio_set(codec, 0, false); @@ -4704,28 +4728,32 @@ static int ca0132_alt_select_in(struct hda_codec *codec) case QUIRK_AE5: ca0113_mmio_command_set(codec, 0x48, 0x28, 0x00); break; + default: + break; } chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) chipio_set_conn_rate(codec, 0x0F, SR_96_000); tmp = FLOAT_ZERO; dspio_set_uint_param(codec, 0x80, 0x00, tmp); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_AE5: chipio_write(codec, 0x18B098, 0x00000000); chipio_write(codec, 0x18B09C, 0x00000000); break; + default: + break; } chipio_set_stream_control(codec, 0x03, 1); chipio_set_stream_control(codec, 0x04, 1); break; case FRONT_MIC: - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_R3D: ca0113_mmio_gpio_set(codec, 0, true); @@ -4747,7 +4775,7 @@ static int ca0132_alt_select_in(struct hda_codec *codec) chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) chipio_set_conn_rate(codec, 0x0F, SR_96_000); dspio_set_uint_param(codec, 0x80, 0x00, tmp); @@ -4755,7 +4783,7 @@ static int ca0132_alt_select_in(struct hda_codec *codec) chipio_set_stream_control(codec, 0x03, 1); chipio_set_stream_control(codec, 0x04, 1); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: chipio_write(codec, 0x18B098, 0x0000000C); chipio_write(codec, 0x18B09C, 0x000000CC); @@ -4764,6 +4792,8 @@ static int ca0132_alt_select_in(struct hda_codec *codec) chipio_write(codec, 0x18B098, 0x0000000C); chipio_write(codec, 0x18B09C, 0x0000004C); break; + default: + break; } ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val); break; @@ -4858,7 +4888,7 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val) val = 0; /* If Voice Focus on SBZ, set to two channel. */ - if ((nid == VOICE_FOCUS) && (spec->use_pci_mmio) + if ((nid == VOICE_FOCUS) && ca0132_use_pci_mmio(spec) && (spec->cur_mic_type != REAR_LINE_IN)) { if (spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID]) { @@ -4877,7 +4907,7 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val) * For SBZ noise reduction, there's an extra command * to module ID 0x47. No clue why. */ - if ((nid == NOISE_REDUCTION) && (spec->use_pci_mmio) + if ((nid == NOISE_REDUCTION) && ca0132_use_pci_mmio(spec) && (spec->cur_mic_type != REAR_LINE_IN)) { if (spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID]) { @@ -4893,7 +4923,7 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val) } /* If rear line in disable effects. */ - if (spec->use_alt_functions && + if (ca0132_use_alt_functions(spec) && spec->in_enum_val == REAR_LINE_IN) val = 0; } @@ -4923,7 +4953,7 @@ static int ca0132_pe_switch_set(struct hda_codec *codec) codec_dbg(codec, "ca0132_pe_switch_set: val=%ld\n", spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]); - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_select_out(codec); i = OUT_EFFECT_START_NID - EFFECT_START_NID; @@ -4983,7 +5013,7 @@ static int ca0132_cvoice_switch_set(struct hda_codec *codec) /* set correct vipsource */ oldval = stop_mic1(codec); - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ret |= ca0132_alt_set_vipsource(codec, 1); else ret |= ca0132_set_vipsource(codec, 1); @@ -5052,7 +5082,7 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID]; if (!auto_jack) { - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_select_out(codec); else ca0132_select_out(codec); @@ -5069,7 +5099,7 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, } if (nid == VNID_HP_ASEL) { - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_select_out(codec); else ca0132_select_out(codec); @@ -5783,7 +5813,7 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol, /* mic boost */ if (nid == spec->input_pins[0]) { spec->cur_mic_boost = *valp; - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { if (spec->in_enum_val != REAR_LINE_IN) changed = ca0132_mic_boost_set(codec, *valp); } else { @@ -6079,7 +6109,7 @@ static int add_fx_switch(struct hda_codec *codec, hda_nid_t nid, /* If using alt_controls, add FX: prefix. But, don't add FX: * prefix to OutFX or InFX enable controls. */ - if ((spec->use_alt_controls) && (nid <= IN_EFFECT_END_NID)) + if (ca0132_use_alt_controls(spec) && (nid <= IN_EFFECT_END_NID)) sprintf(namestr, "FX: %s %s Switch", pfx, dirstr[dir]); else sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); @@ -6356,7 +6386,7 @@ static int ca0132_build_controls(struct hda_codec *codec) return err; } /* Setup vmaster with surround slaves for desktop ca0132 devices */ - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { snd_hda_set_vmaster_tlv(codec, spec->dacs[0], HDA_OUTPUT, spec->tlv); snd_hda_add_vmaster(codec, "Master Playback Volume", @@ -6376,7 +6406,7 @@ static int ca0132_build_controls(struct hda_codec *codec) num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT; for (i = 0; i < num_fx; i++) { /* Desktop cards break if Echo Cancellation is used. */ - if (spec->use_pci_mmio) { + if (ca0132_use_pci_mmio(spec)) { if (i == (ECHO_CANCELLATION - IN_EFFECT_START_NID + OUT_EFFECTS_COUNT)) continue; @@ -6393,7 +6423,7 @@ static int ca0132_build_controls(struct hda_codec *codec) * EQ presets, and Smart Volume presets. Also, change names to add FX * prefix, and change PlayEnhancement and CrystalVoice to match. */ - if (spec->use_alt_controls) { + if (ca0132_use_alt_controls(spec)) { err = ca0132_alt_add_svm_enum(codec); if (err < 0) return err; @@ -6447,7 +6477,7 @@ static int ca0132_build_controls(struct hda_codec *codec) * to select the new outputs and inputs, plus add the new mic boost * setting control. */ - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { err = ca0132_alt_add_output_enum(codec); if (err < 0) return err; @@ -6458,14 +6488,14 @@ static int ca0132_build_controls(struct hda_codec *codec) * ZxR only has microphone input, there is no front panel * header on the card, and aux-in is handled by the DBPro board. */ - if (spec->quirk != QUIRK_ZXR) { + if (ca0132_quirk(spec) != QUIRK_ZXR) { err = ca0132_alt_add_input_enum(codec); if (err < 0) return err; } } - if (spec->quirk == QUIRK_AE5) { + if (ca0132_quirk(spec) == QUIRK_AE5) { err = ae5_add_headphone_gain_enum(codec); if (err < 0) return err; @@ -6474,7 +6504,7 @@ static int ca0132_build_controls(struct hda_codec *codec) return err; } - if (spec->quirk == QUIRK_ZXR) { + if (ca0132_quirk(spec) == QUIRK_ZXR) { err = zxr_add_headphone_gain_switch(codec); if (err < 0) return err; @@ -6504,7 +6534,7 @@ static int ca0132_build_controls(struct hda_codec *codec) return err; } - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_add_chmap_ctls(codec); return 0; @@ -6582,7 +6612,7 @@ static int ca0132_build_pcms(struct hda_codec *codec) info = snd_hda_codec_pcm_new(codec, "CA0132 Analog"); if (!info) return -ENOMEM; - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { info->own_chmap = true; info->stream[SNDRV_PCM_STREAM_PLAYBACK].chmap = ca0132_alt_chmaps; @@ -6596,7 +6626,7 @@ static int ca0132_build_pcms(struct hda_codec *codec) info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0]; /* With the DSP enabled, desktops don't use this ADC. */ - if (!spec->use_alt_functions) { + if (!ca0132_use_alt_functions(spec)) { info = snd_hda_codec_pcm_new(codec, "CA0132 Analog Mic-In2"); if (!info) return -ENOMEM; @@ -6794,7 +6824,7 @@ static void ca0132_init_dmic(struct hda_codec *codec) * Bit 6: set to select Data2, clear for Data1 * Bit 7: set to enable DMic, clear for AMic */ - if (spec->quirk == QUIRK_ALIENWARE_M17XR4) + if (ca0132_quirk(spec) == QUIRK_ALIENWARE_M17XR4) val = 0x33; else val = 0x23; @@ -6876,7 +6906,7 @@ static void ca0132_alt_init_analog_mics(struct hda_codec *codec) /* Mic 1 Setup */ chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); - if (spec->quirk == QUIRK_R3DI) { + if (ca0132_quirk(spec) == QUIRK_R3DI) { chipio_set_conn_rate(codec, 0x0F, SR_96_000); tmp = FLOAT_ONE; } else @@ -6886,7 +6916,7 @@ static void ca0132_alt_init_analog_mics(struct hda_codec *codec) /* Mic 2 setup (not present on desktop cards) */ chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000); chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) chipio_set_conn_rate(codec, 0x0F, SR_96_000); tmp = FLOAT_ZERO; dspio_set_uint_param(codec, 0x80, 0x01, tmp); @@ -6948,7 +6978,7 @@ static void sbz_chipio_startup_data(struct hda_codec *codec) chipio_set_stream_channels(codec, 0x0C, 6); chipio_set_stream_control(codec, 0x0C, 1); /* No clue what these control */ - if (spec->quirk == QUIRK_SBZ) { + if (ca0132_quirk(spec) == QUIRK_SBZ) { chipio_write_no_mutex(codec, 0x190030, 0x0001e0c0); chipio_write_no_mutex(codec, 0x190034, 0x0001e1c1); chipio_write_no_mutex(codec, 0x190038, 0x0001e4c2); @@ -6961,7 +6991,7 @@ static void sbz_chipio_startup_data(struct hda_codec *codec) chipio_write_no_mutex(codec, 0x190054, 0x0001edc9); chipio_write_no_mutex(codec, 0x190058, 0x0001eaca); chipio_write_no_mutex(codec, 0x19005c, 0x0001ebcb); - } else if (spec->quirk == QUIRK_ZXR) { + } else if (ca0132_quirk(spec) == QUIRK_ZXR) { chipio_write_no_mutex(codec, 0x190038, 0x000140c2); chipio_write_no_mutex(codec, 0x19003c, 0x000141c3); chipio_write_no_mutex(codec, 0x190040, 0x000150c4); @@ -6991,7 +7021,7 @@ static void ca0132_alt_dsp_scp_startup(struct hda_codec *codec) * why this is, but multiple tests have confirmed it. */ for (i = 0; i < 2; i++) { - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_AE5: tmp = 0x00000003; @@ -7020,6 +7050,8 @@ static void ca0132_alt_dsp_scp_startup(struct hda_codec *codec) tmp = 0x00000000; dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); break; + default: + break; } msleep(100); } @@ -7042,7 +7074,7 @@ static void ca0132_alt_dsp_initial_mic_setup(struct hda_codec *codec) chipio_set_stream_control(codec, 0x03, 1); chipio_set_stream_control(codec, 0x04, 1); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: chipio_write(codec, 0x18b098, 0x0000000c); chipio_write(codec, 0x18b09C, 0x0000000c); @@ -7051,6 +7083,8 @@ static void ca0132_alt_dsp_initial_mic_setup(struct hda_codec *codec) chipio_write(codec, 0x18b098, 0x0000000c); chipio_write(codec, 0x18b09c, 0x0000004c); break; + default: + break; } } @@ -7272,7 +7306,7 @@ static void r3d_setup_defaults(struct hda_codec *codec) /* Set speaker source? */ dspio_set_uint_param(codec, 0x32, 0x00, tmp); - if (spec->quirk == QUIRK_R3DI) + if (ca0132_quirk(spec) == QUIRK_R3DI) r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADED); /* Setup effect defaults */ @@ -7419,7 +7453,7 @@ static void ca0132_init_flags(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { chipio_set_control_flag(codec, CONTROL_FLAG_DSP_96KHZ, 1); chipio_set_control_flag(codec, CONTROL_FLAG_DAC_96KHZ, 1); chipio_set_control_flag(codec, CONTROL_FLAG_ADC_B_96KHZ, 1); @@ -7452,7 +7486,7 @@ static void ca0132_init_params(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000); chipio_set_conn_rate(codec, 0x0B, SR_48_000); chipio_set_control_param(codec, CONTROL_PARAM_SPDIF1_SOURCE, 0); @@ -7489,7 +7523,7 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec) * can use the default firmware, but I'll leave the option in case * it needs it again. */ - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_R3D: case QUIRK_AE5: @@ -7563,7 +7597,7 @@ static void ca0132_download_dsp(struct hda_codec *codec) } /* For codecs using alt functions, this is already done earlier */ - if (spec->dsp_state == DSP_DOWNLOADED && (!spec->use_alt_functions)) + if (spec->dsp_state == DSP_DOWNLOADED && !ca0132_use_alt_functions(spec)) ca0132_set_dsp_msr(codec, true); } @@ -7600,7 +7634,7 @@ static void amic_callback(struct hda_codec *codec, struct hda_jack_callback *cb) { struct ca0132_spec *spec = codec->spec; - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_select_in(codec); else ca0132_select_mic(codec); @@ -7615,7 +7649,7 @@ static void ca0132_init_unsol(struct hda_codec *codec) snd_hda_jack_detect_enable_callback(codec, UNSOL_TAG_DSP, ca0132_process_dsp_response); /* Front headphone jack detection */ - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) snd_hda_jack_detect_enable_callback(codec, spec->unsol_tag_front_hp, hp_callback); } @@ -7705,7 +7739,7 @@ static void ca0132_init_chip(struct hda_codec *codec) mutex_init(&spec->chipio_mutex); spec->cur_out_type = SPEAKER_OUT; - if (!spec->use_alt_functions) + if (!ca0132_use_alt_functions(spec)) spec->cur_mic_type = DIGITAL_MIC; else spec->cur_mic_type = REAR_MIC; @@ -7731,7 +7765,7 @@ static void ca0132_init_chip(struct hda_codec *codec) * Sets defaults for the effect slider controls, only for alternative * ca0132 codecs. Also sets x-bass crossover frequency to 80hz. */ - if (spec->use_alt_controls) { + if (ca0132_use_alt_controls(spec)) { spec->xbass_xover_freq = 8; for (i = 0; i < EFFECT_LEVEL_SLIDERS; i++) spec->fx_ctl_val[i] = effect_slider_defaults[i]; @@ -7746,7 +7780,7 @@ static void ca0132_init_chip(struct hda_codec *codec) * the daughter board. So, there is no input enum control, and we need * to make sure that spec->in_enum_val is set properly. */ - if (spec->quirk == QUIRK_ZXR) + if (ca0132_quirk(spec) == QUIRK_ZXR) spec->in_enum_val = REAR_MIC; #ifdef ENABLE_TUNING_CONTROLS @@ -8087,27 +8121,27 @@ static void ca0132_mmio_init(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - if (spec->quirk == QUIRK_AE5) + if (ca0132_quirk(spec) == QUIRK_AE5) writel(0x00000001, spec->mem_base + 0x400); else writel(0x00000000, spec->mem_base + 0x400); - if (spec->quirk == QUIRK_AE5) + if (ca0132_quirk(spec) == QUIRK_AE5) writel(0x00000001, spec->mem_base + 0x408); else writel(0x00000000, spec->mem_base + 0x408); - if (spec->quirk == QUIRK_AE5) + if (ca0132_quirk(spec) == QUIRK_AE5) writel(0x00000001, spec->mem_base + 0x40c); else writel(0x00000000, spec->mem_base + 0x40C); - if (spec->quirk == QUIRK_ZXR) + if (ca0132_quirk(spec) == QUIRK_ZXR) writel(0x00880640, spec->mem_base + 0x01C); else writel(0x00880680, spec->mem_base + 0x01C); - if (spec->quirk == QUIRK_AE5) + if (ca0132_quirk(spec) == QUIRK_AE5) writel(0x00000080, spec->mem_base + 0xC0C); else writel(0x00000083, spec->mem_base + 0xC0C); @@ -8115,7 +8149,7 @@ static void ca0132_mmio_init(struct hda_codec *codec) writel(0x00000030, spec->mem_base + 0xC00); writel(0x00000000, spec->mem_base + 0xC04); - if (spec->quirk == QUIRK_AE5) + if (ca0132_quirk(spec) == QUIRK_AE5) writel(0x00000000, spec->mem_base + 0xC0C); else writel(0x00000003, spec->mem_base + 0xC0C); @@ -8124,7 +8158,7 @@ static void ca0132_mmio_init(struct hda_codec *codec) writel(0x00000003, spec->mem_base + 0xC0C); writel(0x00000003, spec->mem_base + 0xC0C); - if (spec->quirk == QUIRK_AE5) + if (ca0132_quirk(spec) == QUIRK_AE5) writel(0x00000001, spec->mem_base + 0xC08); else writel(0x000000C1, spec->mem_base + 0xC08); @@ -8135,7 +8169,7 @@ static void ca0132_mmio_init(struct hda_codec *codec) writel(0x000000C1, spec->mem_base + 0xC08); writel(0x00000080, spec->mem_base + 0xC04); - if (spec->quirk == QUIRK_AE5) { + if (ca0132_quirk(spec) == QUIRK_AE5) { writel(0x00000000, spec->mem_base + 0x42c); writel(0x00000000, spec->mem_base + 0x46c); writel(0x00000000, spec->mem_base + 0x4ac); @@ -8210,7 +8244,7 @@ static void ca0132_alt_init(struct hda_codec *codec) ca0132_alt_vol_setup(codec); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: codec_dbg(codec, "SBZ alt_init"); ca0132_gpio_init(codec); @@ -8247,6 +8281,8 @@ static void ca0132_alt_init(struct hda_codec *codec) snd_hda_sequence_write(codec, spec->chip_init_verbs); snd_hda_sequence_write(codec, spec->desktop_init_verbs); break; + default: + break; } } @@ -8273,7 +8309,7 @@ static int ca0132_init(struct hda_codec *codec) spec->dsp_reload = true; spec->dsp_state = DSP_DOWNLOAD_INIT; } else { - if (spec->quirk == QUIRK_SBZ) + if (ca0132_quirk(spec) == QUIRK_SBZ) sbz_dsp_startup_check(codec); return 0; } @@ -8283,12 +8319,12 @@ static int ca0132_init(struct hda_codec *codec) spec->dsp_state = DSP_DOWNLOAD_INIT; spec->curr_chip_addx = INVALID_CHIP_ADDRESS; - if (spec->use_pci_mmio) + if (ca0132_use_pci_mmio(spec)) ca0132_mmio_init(codec); snd_hda_power_up_pm(codec); - if (spec->quirk == QUIRK_AE5) + if (ca0132_quirk(spec) == QUIRK_AE5) ae5_register_set(codec); ca0132_init_unsol(codec); @@ -8297,14 +8333,14 @@ static int ca0132_init(struct hda_codec *codec) snd_hda_sequence_write(codec, spec->base_init_verbs); - if (spec->use_alt_functions) + if (ca0132_use_alt_functions(spec)) ca0132_alt_init(codec); ca0132_download_dsp(codec); ca0132_refresh_widget_caps(codec); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_R3DI: case QUIRK_R3D: r3d_setup_defaults(codec); @@ -8333,7 +8369,7 @@ static int ca0132_init(struct hda_codec *codec) init_input(codec, cfg->dig_in_pin, spec->dig_in); - if (!spec->use_alt_functions) { + if (!ca0132_use_alt_functions(spec)) { snd_hda_sequence_write(codec, spec->chip_init_verbs); snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, VENDOR_CHIPIO_PARAM_EX_ID_SET, 0x0D); @@ -8341,11 +8377,11 @@ static int ca0132_init(struct hda_codec *codec) VENDOR_CHIPIO_PARAM_EX_VALUE_SET, 0x20); } - if (spec->quirk == QUIRK_SBZ) + if (ca0132_quirk(spec) == QUIRK_SBZ) ca0132_gpio_setup(codec); snd_hda_sequence_write(codec, spec->spec_init_verbs); - if (spec->use_alt_functions) { + if (ca0132_use_alt_functions(spec)) { ca0132_alt_select_out(codec); ca0132_alt_select_in(codec); } else { @@ -8390,7 +8426,7 @@ static void ca0132_free(struct hda_codec *codec) cancel_delayed_work_sync(&spec->unsol_hp_work); snd_hda_power_up(codec); - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: sbz_exit_chip(codec); break; @@ -8406,6 +8442,8 @@ static void ca0132_free(struct hda_codec *codec) case QUIRK_R3DI: r3di_gpio_shutdown(codec); break; + default: + break; } snd_hda_sequence_write(codec, spec->base_exit_verbs); @@ -8460,12 +8498,12 @@ static void ca0132_config(struct hda_codec *codec) spec->multiout.dac_nids = spec->dacs; spec->multiout.num_dacs = 3; - if (!spec->use_alt_functions) + if (!ca0132_use_alt_functions(spec)) spec->multiout.max_channels = 2; else spec->multiout.max_channels = 6; - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_ALIENWARE: codec_dbg(codec, "%s: QUIRK_ALIENWARE applied.\n", __func__); snd_hda_apply_pincfgs(codec, alienware_pincfgs); @@ -8490,9 +8528,11 @@ static void ca0132_config(struct hda_codec *codec) codec_dbg(codec, "%s: QUIRK_AE5 applied.\n", __func__); snd_hda_apply_pincfgs(codec, r3di_pincfgs); break; + default: + break; } - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_ALIENWARE: spec->num_outputs = 2; spec->out_pins[0] = 0x0b; /* speaker out */ @@ -8653,7 +8693,7 @@ static int ca0132_prepare_verbs(struct hda_codec *codec) * Since desktop cards use pci_mmio, this can be used to determine * whether or not to use these verbs instead of a separate bool. */ - if (spec->use_pci_mmio) + if (ca0132_use_pci_mmio(spec)) spec->desktop_init_verbs = ca0132_init_verbs1; spec->spec_init_verbs = kcalloc(NUM_SPEC_VERBS, sizeof(struct hda_verb), @@ -8728,11 +8768,10 @@ static int patch_ca0132(struct hda_codec *codec) spec->quirk = quirk->value; else spec->quirk = QUIRK_NONE; - - if (spec->quirk == QUIRK_SBZ) + if (ca0132_quirk(spec) == QUIRK_SBZ) sbz_detect_quirk(codec); - if (spec->quirk == QUIRK_ZXR_DBPRO) + if (ca0132_quirk(spec) == QUIRK_ZXR_DBPRO) codec->patch_ops = dbpro_patch_ops; else codec->patch_ops = ca0132_patch_ops; @@ -8745,7 +8784,7 @@ static int patch_ca0132(struct hda_codec *codec) spec->num_mixers = 1; /* Set which mixers each quirk uses. */ - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: spec->mixers[0] = desktop_mixer; snd_hda_codec_set_name(codec, "Sound Blaster Z"); @@ -8774,7 +8813,7 @@ static int patch_ca0132(struct hda_codec *codec) } /* Setup whether or not to use alt functions/controls/pci_mmio */ - switch (spec->quirk) { + switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_R3D: case QUIRK_AE5: @@ -8795,6 +8834,7 @@ static int patch_ca0132(struct hda_codec *codec) break; } +#ifdef CONFIG_PCI if (spec->use_pci_mmio) { spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); if (spec->mem_base == NULL) { @@ -8802,6 +8842,7 @@ static int patch_ca0132(struct hda_codec *codec) spec->quirk = QUIRK_NONE; } } +#endif spec->base_init_verbs = ca0132_base_init_verbs; spec->base_exit_verbs = ca0132_base_exit_verbs; -- 2.19.1

1 0

[alsa-devel] KMSAN: uninit-value in linear_transfer (2)
by syzbot 09 Nov '18

09 Nov '18

Hello, syzbot found the following crash on: HEAD commit: 7438a3b20295 kmsan: print user address when reporting info.. git tree: https://github.com/google/kmsan.git/master console output: https://syzkaller.appspot.com/x/log.txt?x=15b42133400000 kernel config: https://syzkaller.appspot.com/x/.config?x=8df5fc509a1b351b dashboard link: https://syzkaller.appspot.com/bug?extid=1cb36954e127c98dd037 compiler: clang version 8.0.0 (trunk 343298) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=12be9825400000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=13e2c425400000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+1cb36954e127c98dd037(a)syzkaller.appspotmail.com ================================================================== BUG: KMSAN: uninit-value in __arch_swab32 arch/x86/include/uapi/asm/swab.h:10 [inline] BUG: KMSAN: uninit-value in __fswab32 include/uapi/linux/swab.h:59 [inline] BUG: KMSAN: uninit-value in do_convert sound/core/oss/linear.c:50 [inline] BUG: KMSAN: uninit-value in convert sound/core/oss/linear.c:81 [inline] BUG: KMSAN: uninit-value in linear_transfer+0x92d/0xca0 sound/core/oss/linear.c:110 CPU: 1 PID: 6835 Comm: syz-executor866 Not tainted 4.19.0+ #78 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x32d/0x480 lib/dump_stack.c:113 kmsan_report+0x19f/0x300 mm/kmsan/kmsan.c:911 __msan_warning+0x76/0xd0 mm/kmsan/kmsan_instr.c:415 __arch_swab32 arch/x86/include/uapi/asm/swab.h:10 [inline] __fswab32 include/uapi/linux/swab.h:59 [inline] do_convert sound/core/oss/linear.c:50 [inline] convert sound/core/oss/linear.c:81 [inline] linear_transfer+0x92d/0xca0 sound/core/oss/linear.c:110 snd_pcm_plug_read_transfer+0x3bf/0x590 sound/core/oss/pcm_plugin.c:659 snd_pcm_oss_read2 sound/core/oss/pcm_oss.c:1480 [inline] snd_pcm_oss_read1 sound/core/oss/pcm_oss.c:1518 [inline] snd_pcm_oss_read+0xe6d/0x1cb0 sound/core/oss/pcm_oss.c:2758 __vfs_read+0x1e2/0xb10 fs/read_write.c:416 vfs_read+0x380/0x6b0 fs/read_write.c:452 ksys_read fs/read_write.c:578 [inline] __do_sys_read fs/read_write.c:588 [inline] __se_sys_read+0x17a/0x370 fs/read_write.c:586 __x64_sys_read+0x4a/0x70 fs/read_write.c:586 do_syscall_64+0xcf/0x110 arch/x86/entry/common.c:291 entry_SYSCALL_64_after_hwframe+0x63/0xe7 RIP: 0033:0x446379 Code: e8 0c e8 ff ff 48 83 c4 18 c3 0f 1f 80 00 00 00 00 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 0f 83 7b 08 fc ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:00007fcd5b97cda8 EFLAGS: 00000246 ORIG_RAX: 0000000000000000 RAX: ffffffffffffffda RBX: 00000000006dac38 RCX: 0000000000446379 RDX: 0000000000000184 RSI: 0000000020002180 RDI: 0000000000000003 RBP: 00000000006dac30 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00000000006dac3c R13: 7073642f7665642f R14: 00800000c0045002 R15: 0000000000000001 Uninit was stored to memory at: kmsan_save_stack_with_flags mm/kmsan/kmsan.c:252 [inline] kmsan_save_stack mm/kmsan/kmsan.c:267 [inline] kmsan_internal_chain_origin+0x136/0x240 mm/kmsan/kmsan.c:569 kmsan_memcpy_origins+0x13d/0x1b0 mm/kmsan/kmsan.c:393 __msan_memcpy+0x6f/0x80 mm/kmsan/kmsan_instr.c:242 do_convert sound/core/oss/linear.c:48 [inline] convert sound/core/oss/linear.c:81 [inline] linear_transfer+0x74c/0xca0 sound/core/oss/linear.c:110 snd_pcm_plug_read_transfer+0x3bf/0x590 sound/core/oss/pcm_plugin.c:659 snd_pcm_oss_read2 sound/core/oss/pcm_oss.c:1480 [inline] snd_pcm_oss_read1 sound/core/oss/pcm_oss.c:1518 [inline] snd_pcm_oss_read+0xe6d/0x1cb0 sound/core/oss/pcm_oss.c:2758 __vfs_read+0x1e2/0xb10 fs/read_write.c:416 vfs_read+0x380/0x6b0 fs/read_write.c:452 ksys_read fs/read_write.c:578 [inline] __do_sys_read fs/read_write.c:588 [inline] __se_sys_read+0x17a/0x370 fs/read_write.c:586 __x64_sys_read+0x4a/0x70 fs/read_write.c:586 do_syscall_64+0xcf/0x110 arch/x86/entry/common.c:291 entry_SYSCALL_64_after_hwframe+0x63/0xe7 Uninit was created at: kmsan_save_stack_with_flags mm/kmsan/kmsan.c:252 [inline] kmsan_internal_alloc_meta_for_pages+0x155/0x740 mm/kmsan/kmsan.c:689 kmsan_alloc_page+0x77/0xe0 mm/kmsan/kmsan_hooks.c:320 __alloc_pages_nodemask+0x12cc/0x6640 mm/page_alloc.c:4416 alloc_pages_current+0x584/0x7e0 mm/mempolicy.c:2093 alloc_pages include/linux/gfp.h:511 [inline] __vmalloc_area_node mm/vmalloc.c:1689 [inline] __vmalloc_node_range+0x879/0x12a0 mm/vmalloc.c:1752 __vmalloc_node mm/vmalloc.c:1797 [inline] __vmalloc_node_flags mm/vmalloc.c:1811 [inline] vmalloc+0xd8/0xf0 mm/vmalloc.c:1833 snd_pcm_plugin_alloc+0x255/0xc80 sound/core/oss/pcm_plugin.c:71 snd_pcm_plug_alloc+0x281/0x600 sound/core/oss/pcm_plugin.c:137 snd_pcm_oss_change_params_locked+0x5e40/0x6e30 sound/core/oss/pcm_oss.c:1039 snd_pcm_oss_change_params sound/core/oss/pcm_oss.c:1107 [inline] snd_pcm_oss_get_active_substream+0x4f7/0x5a0 sound/core/oss/pcm_oss.c:1124 snd_pcm_oss_get_rate sound/core/oss/pcm_oss.c:1774 [inline] snd_pcm_oss_set_rate sound/core/oss/pcm_oss.c:1766 [inline] snd_pcm_oss_ioctl+0x4adb/0x8860 sound/core/oss/pcm_oss.c:2614 do_vfs_ioctl+0xf77/0x2d30 fs/ioctl.c:46 ksys_ioctl fs/ioctl.c:702 [inline] __do_sys_ioctl fs/ioctl.c:709 [inline] __se_sys_ioctl+0x1da/0x270 fs/ioctl.c:707 __x64_sys_ioctl+0x4a/0x70 fs/ioctl.c:707 do_syscall_64+0xcf/0x110 arch/x86/entry/common.c:291 entry_SYSCALL_64_after_hwframe+0x63/0xe7 ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#bug-status-tracking for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

3 4

[alsa-devel] [PATCH v2 0/7] ASoC: omap-mcbsp: Cleanup and split core/sidetone support
by Peter Ujfalusi 08 Nov '18

08 Nov '18

Hi, Changes since v1: - Two patch added to address the code move triggered few warnings from build. One of them is around mcbsp->pdata checks (not valid complaint), the other is that we set -EINVAL to unsigned variables (they were not used in that case) While preparing for merging the davinci and omap directories (I will send the series for that shortly) I have taken some time to do cleanup on the McBSP driver. The mcbsp.c/h files were the result when we moved code from arch/arm/plat-omap to sound/soc/omap/ a long time ago and it contained code for McBSP core and the OMAP3 sidetone functionality. With this series I tried to split the core and sidetone code to separate files. There were two reasons for this: 1. to have the OMAP specific DAI driver files prefixed with omap-* under the new sound/soc/ti/ directory 2. easier to find functions as all sidetone code is in separate file. Regards, Peter --- Peter Ujfalusi (7): ASoC: omap-mcbsp: Clean up dma_data addr initialization code ASoC: omap-mcbsp: Clean up the interrupt handlers ASoC: omap-mcbsp: Simplify the mcbsp_start/_stop function parameters ASoC: omap-mcbsp: Move out the FIFO check from set_threshold and get_delay ASoC: omap-mcbsp: Re-arrange files for core McBSP and Sidetone function split ASoC: omap-mcbsp: Remove redundant check for mcbsp->pdata ASoC: omap-mcbsp: No need to initialize max_xx_thres when it is not used sound/soc/omap/Makefile | 2 +- sound/soc/omap/mcbsp.c | 1104 ----------------- sound/soc/omap/{mcbsp.h => omap-mcbsp-priv.h} | 126 +- sound/soc/omap/omap-mcbsp-st.c | 516 ++++++++ sound/soc/omap/omap-mcbsp.c | 852 ++++++++++--- sound/soc/omap/omap-mcbsp.h | 8 +- 6 files changed, 1278 insertions(+), 1330 deletions(-) delete mode 100644 sound/soc/omap/mcbsp.c rename sound/soc/omap/{mcbsp.h => omap-mcbsp-priv.h} (70%) create mode 100644 sound/soc/omap/omap-mcbsp-st.c -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

3 10

[alsa-devel] High CPU usage in mpd
by GitHub issues - opened 08 Nov '18

08 Nov '18

alsa-project/alsa-utils issue #2 was opened from buttjake: I've ran into a strange bug in mpd where I get really high cpu usage. All the details are in the following two short threads. https://github.com/MusicPlayerDaemon/MPD/issues/391#issuecomment-435698235 https://bbs.archlinux.org/viewtopic.php?id=241722 ( buttcake is me ) I can also solve the bug by adding ``` audio_output { type "alsa" device "hw:0,0" } ``` to the mpd.conf Issue URL : https://github.com/alsa-project/alsa-utils/issues/2 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [PATCH] ASoC: fsl_ssi: Change to use DEFINE_SHOW_ATTRIBUTE macro
by Yangtao Li 07 Nov '18

07 Nov '18

Use DEFINE_SHOW_ATTRIBUTE macro to simplify the code. Signed-off-by: Yangtao Li <tiny.windzz(a)gmail.com> --- sound/soc/fsl/fsl_ssi_dbg.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 1255dfe19eef..6f6294149476 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -124,17 +124,7 @@ static int fsl_ssi_stats_show(struct seq_file *s, void *unused) return 0; } -static int fsl_ssi_stats_open(struct inode *inode, struct file *file) -{ - return single_open(file, fsl_ssi_stats_show, inode->i_private); -} - -static const struct file_operations fsl_ssi_stats_ops = { - .open = fsl_ssi_stats_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; +DEFINE_SHOW_ATTRIBUTE(fsl_ssi_stats); int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev) { @@ -144,7 +134,7 @@ int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev) ssi_dbg->dbg_stats = debugfs_create_file("stats", 0444, ssi_dbg->dbg_dir, ssi_dbg, - &fsl_ssi_stats_ops); + &fsl_ssi_stats_fops); if (!ssi_dbg->dbg_stats) { debugfs_remove(ssi_dbg->dbg_dir); return -ENOMEM; -- 2.17.0

2 1

[alsa-devel] UCM profile dirs without card config
by Takashi Iwai 07 Nov '18

07 Nov '18

Hi, some of the recently added UCM profiles are the files to be included by others, and they are not card targets. Unfortunately alsaucm doesn't know about it, and it aborts with error, e.g. ==== % alsaucm ALSA lib utils.c:67:(uc_mgr_config_load) could not open configuration file /usr/share/alsa/ucm/bytcr/bytcr.conf ALSA lib parser.c:1427:(load_master_config) error: could not parse configuration for card bytcr alsaucm: unable to obtain card list: No such file or directory ==== The similar error is found in nau8824, rt5640 and rt5651. After putting a placeholder card config file (e.g. bytcr.conf), the error is gone. But certainly we don't want to allow user to choose this as the proper card. So, I guess we need to put some flag to each such directory indicating that it's no card config. For example, putting /usr/share/alsa/ucm/bytcr/.nocard file or such... Not sure whether it's wise to have a file starting with dot, though. Maybe clearer to be "nocard" or any other visible one? Comments / suggestions appreciated. thanks, Takashi

1 0

[alsa-devel] Applied "ASoC: fsl_ssi: Change to use DEFINE_SHOW_ATTRIBUTE macro" to the asoc tree
by Mark Brown 07 Nov '18

07 Nov '18

The patch ASoC: fsl_ssi: Change to use DEFINE_SHOW_ATTRIBUTE macro 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 d3ff06b03c106171cdc7a718c4de724ce6ae346d Mon Sep 17 00:00:00 2001 From: Yangtao Li <tiny.windzz(a)gmail.com> Date: Mon, 5 Nov 2018 09:37:30 -0500 Subject: [PATCH] ASoC: fsl_ssi: Change to use DEFINE_SHOW_ATTRIBUTE macro Use DEFINE_SHOW_ATTRIBUTE macro to simplify the code. Signed-off-by: Yangtao Li <tiny.windzz(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_ssi_dbg.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 1255dfe19eef..6f6294149476 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -124,17 +124,7 @@ static int fsl_ssi_stats_show(struct seq_file *s, void *unused) return 0; } -static int fsl_ssi_stats_open(struct inode *inode, struct file *file) -{ - return single_open(file, fsl_ssi_stats_show, inode->i_private); -} - -static const struct file_operations fsl_ssi_stats_ops = { - .open = fsl_ssi_stats_open, - .read = seq_read, - .llseek = seq_lseek, - .release = single_release, -}; +DEFINE_SHOW_ATTRIBUTE(fsl_ssi_stats); int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev) { @@ -144,7 +134,7 @@ int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev) ssi_dbg->dbg_stats = debugfs_create_file("stats", 0444, ssi_dbg->dbg_dir, ssi_dbg, - &fsl_ssi_stats_ops); + &fsl_ssi_stats_fops); if (!ssi_dbg->dbg_stats) { debugfs_remove(ssi_dbg->dbg_dir); return -ENOMEM; -- 2.19.0.rc2

1 0

Re: [alsa-devel] Build regressions/improvements in v4.20-rc1
by Geert Uytterhoeven 07 Nov '18

07 Nov '18

On Mon, Nov 5, 2018 at 11:07 PM Geert Uytterhoeven <geert(a)linux-m68k.org> wrote: > Below is the list of build error/warning regressions/improvements in > v4.20-rc1[1] compared to v4.19[2]. > > Summarized: > - build errors: +3/-0 > - build warnings: +449/-2712 > > Happy fixing! ;-) > > Thanks to the linux-next team for providing the build service. > > [1] http://kisskb.ellerman.id.au/kisskb/branch/linus/head/651022382c7f8da46cb48… (all 240 configs) > [2] http://kisskb.ellerman.id.au/kisskb/branch/linus/head/84df9525b0c27f3ebc2eb… (all 240 configs) > > > *** ERRORS *** > > 3 error regressions: > + /kisskb/src/arch/s390/kernel/nospec-branch.c: error: macro "memcpy" passed 6 arguments, but takes just 3: => 130:61 s390-all{mod,yes}config > + /kisskb/src/sound/pci/hda/patch_ca0132.c: error: implicit declaration of function 'pci_iomap' [-Werror=implicit-function-declaration]: => 8799:3 sh4-all{mod,yes}config Looks like d9b84a15892c0233 ("ALSA: hda: Fix implicit definition of pci_iomap() on SH") is not sufficient? > + {standard input}: Error: Unrecognized opcode: `slbfee.': => 89, 37 powerpc-all{mod,yes}config Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

5 8

[alsa-devel] [PATCH 0/1] slimbus: simplify getting .driver_data
by Wolfram Sang 07 Nov '18

07 Nov '18

I got tired of fixing this in Renesas drivers manually, so I took the big hammer. Remove this cumbersome code pattern which got copy-pasted too much already: - struct platform_device *pdev = to_platform_device(dev); - struct ep93xx_keypad *keypad = platform_get_drvdata(pdev); + struct ep93xx_keypad *keypad = dev_get_drvdata(dev); A branch, tested by buildbot, can be found here: git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git coccinelle/get_drvdata I have been asked if it couldn't be done for dev_set_drvdata as well. I checked it and did not find one occasion where it could be simplified like this. Not much of a surprise because driver_data is usually set in probe() functions which access struct platform_device in many other ways. I am open for other comments, suggestions, too, of course. Here is the cocci-script I created: @@ struct device* d; identifier pdev; expression *ptr; @@ ( - struct platform_device *pdev = to_platform_device(d); | - struct platform_device *pdev; ... - pdev = to_platform_device(d); ) <... when != pdev - &pdev->dev + d ...> ptr = - platform_get_drvdata(pdev) + dev_get_drvdata(d) <... when != pdev - &pdev->dev + d ...> Kind regards, Wolfram Wolfram Sang (1): slimbus: qcom-ctrl: simplify getting .driver_data drivers/slimbus/qcom-ctrl.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) -- 2.19.0

2 2

[alsa-devel] [PATCH 0/2] ASoC: qdsp6: few routing fixes
by Srinivas Kandagatla 07 Nov '18

07 Nov '18

Jimmy and Rohit found a issue w.r.t backend dais being powered on all the time on there device. Looks like there was a bug in the driver since start. This patchset fixes it and also fixes another bug where asm routes are added each time at dai probe. Thanks, srini Srinivas Kandagatla (2): ASoC: qdsp6: q6asm-dai: Only add routing once. ASoC: qdsp6: q6afe-dai: Fix the dai widgets sound/soc/qcom/qdsp6/q6afe-dai.c | 208 +++++++++++++++---------------- sound/soc/qcom/qdsp6/q6asm-dai.c | 33 ----- sound/soc/qcom/qdsp6/q6routing.c | 19 +++ 3 files changed, 123 insertions(+), 137 deletions(-) -- 2.19.1

3 5

[alsa-devel] [PATCH] ASoC: ak5558: Remove redundant snd_soc_component_read32 calls
by Axel Lin 06 Nov '18

06 Nov '18

snd_soc_component_update_bits() will only update the mask bits, so remove the redundant snd_soc_component_read32(). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/ak5558.c | 17 +++++------------ 1 file changed, 5 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index 60f1f12c81ea..8179512129d3 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -130,16 +130,12 @@ static int ak5558_hw_params(struct snd_pcm_substream *substream, u8 bits; int pcm_width = max(params_physical_width(params), ak5558->slot_width); - /* set master/slave audio interface */ - bits = snd_soc_component_read32(component, AK5558_02_CONTROL1); - bits &= ~AK5558_BITS; - switch (pcm_width) { case 16: - bits |= AK5558_DIF_24BIT_MODE; + bits = AK5558_DIF_24BIT_MODE; break; case 32: - bits |= AK5558_DIF_32BIT_MODE; + bits = AK5558_DIF_32BIT_MODE; break; default: return -EINVAL; @@ -168,18 +164,15 @@ static int ak5558_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) } /* set master/slave audio interface */ - format = snd_soc_component_read32(component, AK5558_02_CONTROL1); - format &= ~AK5558_DIF; - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - format |= AK5558_DIF_I2S_MODE; + format = AK5558_DIF_I2S_MODE; break; case SND_SOC_DAIFMT_LEFT_J: - format |= AK5558_DIF_MSB_MODE; + format = AK5558_DIF_MSB_MODE; break; case SND_SOC_DAIFMT_DSP_B: - format |= AK5558_DIF_MSB_MODE; + format = AK5558_DIF_MSB_MODE; break; default: return -EINVAL; -- 2.17.1

3 2

[alsa-devel] Applied "ASoC: pcm3168a: remove read-only status register from snd_kcontrol_new" to the asoc tree
by Mark Brown 06 Nov '18

06 Nov '18

The patch ASoC: pcm3168a: remove read-only status register from snd_kcontrol_new 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 223bc10b84970fd772c105b550beeef3ac3502be Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 6 Nov 2018 04:33:02 +0000 Subject: [PATCH] ASoC: pcm3168a: remove read-only status register from snd_kcontrol_new "DACn Zero Flag" and "ADCn Overflow Flag" are read-only status register. These information are needed for driver not for user. Let's remove these from snd_kcontrol_new. There is no detail explanation for these register on datasheet, but, judging from its behavior, it seems these register needs clock to read. Otherwise, I2C returns error. Therefor, amixer command will be failed under non working timing. Without this patch, user often fail amixer command. This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 TDM sound. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/pcm3168a.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 5f4327f5b725..115f21658222 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -133,10 +133,6 @@ static const struct snd_kcontrol_new pcm3168a_snd_controls[] = { SOC_DOUBLE("DAC2 Invert Switch", PCM3168A_DAC_INV, 2, 3, 1, 0), SOC_DOUBLE("DAC3 Invert Switch", PCM3168A_DAC_INV, 4, 5, 1, 0), SOC_DOUBLE("DAC4 Invert Switch", PCM3168A_DAC_INV, 6, 7, 1, 0), - SOC_DOUBLE_STS("DAC1 Zero Flag", PCM3168A_DAC_ZERO, 0, 1, 1, 0), - SOC_DOUBLE_STS("DAC2 Zero Flag", PCM3168A_DAC_ZERO, 2, 3, 1, 0), - SOC_DOUBLE_STS("DAC3 Zero Flag", PCM3168A_DAC_ZERO, 4, 5, 1, 0), - SOC_DOUBLE_STS("DAC4 Zero Flag", PCM3168A_DAC_ZERO, 6, 7, 1, 0), SOC_ENUM("DAC Volume Control Type", pcm3168a_dac_volume_type), SOC_ENUM("DAC Volume Rate Multiplier", pcm3168a_dac_att_mult), SOC_ENUM("DAC De-Emphasis", pcm3168a_dac_demp), @@ -176,9 +172,6 @@ static const struct snd_kcontrol_new pcm3168a_snd_controls[] = { SOC_DOUBLE("ADC1 Mute Switch", PCM3168A_ADC_MUTE, 0, 1, 1, 0), SOC_DOUBLE("ADC2 Mute Switch", PCM3168A_ADC_MUTE, 2, 3, 1, 0), SOC_DOUBLE("ADC3 Mute Switch", PCM3168A_ADC_MUTE, 4, 5, 1, 0), - SOC_DOUBLE_STS("ADC1 Overflow Flag", PCM3168A_ADC_OV, 0, 1, 1, 0), - SOC_DOUBLE_STS("ADC2 Overflow Flag", PCM3168A_ADC_OV, 2, 3, 1, 0), - SOC_DOUBLE_STS("ADC3 Overflow Flag", PCM3168A_ADC_OV, 4, 5, 1, 0), SOC_ENUM("ADC Volume Control Type", pcm3168a_adc_volume_type), SOC_ENUM("ADC Volume Rate Multiplier", pcm3168a_adc_att_mult), SOC_ENUM("ADC Overflow Flag Polarity", pcm3168a_adc_ov_pol), -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH] ASoC: omap: Remove unused machine driver for AM3517-evm
by Peter Ujfalusi 06 Nov '18

06 Nov '18

The audio support is done via simple-audio-card via DT, the custom driver is no longer needed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/Kconfig | 9 --- sound/soc/omap/Makefile | 2 - sound/soc/omap/am3517evm.c | 141 ------------------------------------- 3 files changed, 152 deletions(-) delete mode 100644 sound/soc/omap/am3517evm.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 6dccea6fdaeb..3d418482d242 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -78,15 +78,6 @@ config SND_OMAP_SOC_OSK5912 help Say Y if you want to add support for SoC audio on osk5912. -config SND_OMAP_SOC_AM3517EVM - tristate "SoC Audio support for OMAP3517 / AM3517 EVM" - depends on SND_SDMA_SOC && MACH_OMAP3517EVM && I2C - select SND_OMAP_SOC_MCBSP - select SND_SOC_TLV320AIC23_I2C - help - Say Y if you want to add support for SoC audio on the OMAP3517 / AM3517 - EVM. - config SND_OMAP_SOC_OMAP_TWL4030 tristate "SoC Audio support for TI SoC based boards with twl4030 codec" depends on TWL4030_CORE && SND_SDMA_SOC diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 8922aa5a840c..d005338dd13c 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -17,7 +17,6 @@ snd-soc-n810-objs := n810.o snd-soc-rx51-objs := rx51.o snd-soc-ams-delta-objs := ams-delta.o snd-soc-osk5912-objs := osk5912.o -snd-soc-am3517evm-objs := am3517evm.o snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o snd-soc-omap-twl4030-objs := omap-twl4030.o snd-soc-omap3pandora-objs := omap3pandora.o @@ -26,7 +25,6 @@ obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o -obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_TWL4030) += snd-soc-omap-twl4030.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c deleted file mode 100644 index d5651026ec10..000000000000 --- a/sound/soc/omap/am3517evm.c +++ /dev/null @@ -1,141 +0,0 @@ -/* - * am3517evm.c -- ALSA SoC support for OMAP3517 / AM3517 EVM - * - * Author: Anuj Aggarwal <anuj.aggarwal(a)ti.com> - * - * Based on sound/soc/omap/beagle.c by Steve Sakoman - * - * Copyright (C) 2009 Texas Instruments Incorporated - * - * 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 version 2. - * - * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind, - * whether express or implied; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - */ - -#include <linux/clk.h> -#include <linux/platform_device.h> -#include <linux/module.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> -#include <linux/platform_data/asoc-ti-mcbsp.h> - -#include "omap-mcbsp.h" - -#include "../codecs/tlv320aic23.h" - -#define CODEC_CLOCK 12000000 - -static int am3517evm_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; - - /* Set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, - CODEC_CLOCK, SND_SOC_CLOCK_IN); - if (ret < 0) - printk(KERN_ERR "can't set codec system clock\n"); - - return ret; -} - -static const struct snd_soc_ops am3517evm_ops = { - .hw_params = am3517evm_hw_params, -}; - -/* am3517evm machine dapm widgets */ -static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { - SND_SOC_DAPM_HP("Line Out", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), - SND_SOC_DAPM_MIC("Mic In", NULL), -}; - -static const struct snd_soc_dapm_route audio_map[] = { - /* Line Out connected to LLOUT, RLOUT */ - {"Line Out", NULL, "LOUT"}, - {"Line Out", NULL, "ROUT"}, - - {"LLINEIN", NULL, "Line In"}, - {"RLINEIN", NULL, "Line In"}, - - {"MICIN", NULL, "Mic In"}, -}; - -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link am3517evm_dai = { - .name = "TLV320AIC23", - .stream_name = "AIC23", - .cpu_dai_name = "omap-mcbsp.1", - .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "omap-mcbsp.1", - .codec_name = "tlv320aic23-codec.2-001a", - .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &am3517evm_ops, -}; - -/* Audio machine driver */ -static struct snd_soc_card snd_soc_am3517evm = { - .name = "am3517evm", - .owner = THIS_MODULE, - .dai_link = &am3517evm_dai, - .num_links = 1, - - .dapm_widgets = tlv320aic23_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), -}; - -static struct platform_device *am3517evm_snd_device; - -static int __init am3517evm_soc_init(void) -{ - int ret; - - if (!machine_is_omap3517evm()) - return -ENODEV; - pr_info("OMAP3517 / AM3517 EVM SoC init\n"); - - am3517evm_snd_device = platform_device_alloc("soc-audio", -1); - if (!am3517evm_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(am3517evm_snd_device, &snd_soc_am3517evm); - - ret = platform_device_add(am3517evm_snd_device); - if (ret) - goto err1; - - return 0; - -err1: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(am3517evm_snd_device); - - return ret; -} - -static void __exit am3517evm_soc_exit(void) -{ - platform_device_unregister(am3517evm_snd_device); -} - -module_init(am3517evm_soc_init); -module_exit(am3517evm_soc_exit); - -MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal(a)ti.com>"); -MODULE_DESCRIPTION("ALSA SoC OMAP3517 / AM3517 EVM"); -MODULE_LICENSE("GPL v2"); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH] ASoC: pxa: change ac97 dependencies
by Arnd Bergmann 06 Nov '18

06 Nov '18

Enabling both the old AC97_BUS code and the new AC97_BUS_COMPAT causes problems because both modules provide an exported snd_ac97_reset() function. I had tried to fix the problem of having both coexist earlier, but my patch only prevented them from being built-in. This is because of a special Kconfig feature that lets a symbol have a dependency on another one being disabled, but still allow both to be loadable modules. Changing the dependency to =n avoids that problem, now we can only build the new driver if the old one is completely disabled. If we could figure out a way to let rename one of the reset functions and have each driver link to exactly the old or the compat code, that would also work, but I could not find if that's possible. Fixes: bec5ecdf41d4 ("ASoC: pxa: avoid AC97_BUS build warning") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/pxa/Kconfig | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 943b44de1464..6075fc32e6b9 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -79,7 +79,7 @@ config SND_PXA2XX_SOC_TOSA tristate "SoC AC97 Audio support for Tosa" depends on SND_PXA2XX_SOC && MACH_TOSA depends on MFD_TC6393XB - depends on !AC97_BUS + depends on AC97_BUS=n select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -89,7 +89,7 @@ config SND_PXA2XX_SOC_TOSA config SND_PXA2XX_SOC_E740 tristate "SoC AC97 Audio support for e740" depends on SND_PXA2XX_SOC && MACH_E740 - depends on !AC97_BUS + depends on AC97_BUS=n select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -99,7 +99,7 @@ config SND_PXA2XX_SOC_E740 config SND_PXA2XX_SOC_E750 tristate "SoC AC97 Audio support for e750" depends on SND_PXA2XX_SOC && MACH_E750 - depends on !AC97_BUS + depends on AC97_BUS=n select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -109,7 +109,7 @@ config SND_PXA2XX_SOC_E750 config SND_PXA2XX_SOC_E800 tristate "SoC AC97 Audio support for e800" depends on SND_PXA2XX_SOC && MACH_E800 - depends on !AC97_BUS + depends on AC97_BUS=n select SND_SOC_WM9712 select SND_PXA2XX_SOC_AC97 help @@ -120,7 +120,7 @@ config SND_PXA2XX_SOC_EM_X270 tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300" depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \ MACH_CM_X300) - depends on !AC97_BUS + depends on AC97_BUS=n select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -131,7 +131,7 @@ config SND_PXA2XX_SOC_PALM27X bool "SoC Audio support for Palm T|X, T5, E2 and LifeDrive" depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \ MACH_PALMT5 || MACH_PALMTE2) - depends on !AC97_BUS + depends on AC97_BUS=n select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -161,7 +161,7 @@ config SND_SOC_TTC_DKB config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE - depends on !AC97_BUS + depends on AC97_BUS=n select SND_PXA2XX_SOC_AC97 select SND_PXA_SOC_SSP select SND_SOC_WM9713 @@ -201,7 +201,7 @@ config SND_PXA2XX_SOC_MAGICIAN config SND_PXA2XX_SOC_MIOA701 tristate "SoC Audio support for MIO A701" depends on SND_PXA2XX_SOC && MACH_MIOA701 - depends on !AC97_BUS + depends on AC97_BUS=n select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9713 help -- 2.18.0

2 1

[alsa-devel] Applied "ASoC: acpi: define common interface for machine driver configuration" to the asoc tree
by Mark Brown 06 Nov '18

06 Nov '18

The patch ASoC: acpi: define common interface for machine driver configuration 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 8679284b37e9e2d529d60d1acc4133af4aa5dd34 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Thu, 1 Nov 2018 16:34:48 -0500 Subject: [PATCH] ASoC: acpi: define common interface for machine driver configuration The machine drivers may need information provided by the platform driver. Currently the information is passed using pdata specific to each plaform driver. This prevents other drivers, such as SOF, from reusing machine drivers directly. Add a new structure which contains the required fields. This proposal requires a bit more work on the platform side but this generic interface helps reuse code directly. Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-acpi.h | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index e45b2330d16a..5154c6359609 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -37,6 +37,19 @@ snd_soc_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], struct snd_soc_acpi_mach * snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines); +/** + * snd_soc_acpi_mach_params: interface for machine driver configuration + * + * @acpi_ipc_irq_index: used for BYT-CR detection + * @platform: string used for HDaudio codec support + * @codec_mask: used for HDAudio support + */ +struct snd_soc_acpi_mach_params { + u32 acpi_ipc_irq_index; + const char *platform; + u32 codec_mask; +}; + /** * snd_soc_acpi_mach: ACPI-based machine descriptor. Most of the fields are * related to the hardware, except for the firmware and topology file names. @@ -68,6 +81,7 @@ struct snd_soc_acpi_mach { struct snd_soc_acpi_mach * (*machine_quirk)(void *arg); const void *quirk_data; void *pdata; + struct snd_soc_acpi_mach_params mach_params; const char *sof_fw_filename; const char *sof_tplg_filename; const char *asoc_plat_name; -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: Intel: use standard interface for Hdaudio machine driver" to the asoc tree
by Mark Brown 06 Nov '18

06 Nov '18

The patch ASoC: Intel: use standard interface for Hdaudio machine driver 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 842bb5135f1016151235413726e4956210a79664 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Thu, 1 Nov 2018 16:34:49 -0500 Subject: [PATCH] ASoC: Intel: use standard interface for Hdaudio machine driver Don't rely on internal Skylake-specific data structures, use generic interface to let other drivers use the same machine driver as is, e.g. SOF to support HDaudio codecs and HDMI outputs. Tested on LeafHill CRB board, no regression seen with this change. Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 20 ++++++++++---------- sound/soc/intel/skylake/skl.c | 10 +++------- 2 files changed, 13 insertions(+), 17 deletions(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index b415dd4c85f5..b6f287fa9505 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -12,8 +12,8 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include "../../codecs/hdac_hdmi.h" -#include "../skylake/skl.h" #include "skl_hda_dsp_common.h" static const struct snd_soc_dapm_widget skl_hda_widgets[] = { @@ -101,17 +101,17 @@ static struct snd_soc_card hda_soc_card = { #define IDISP_ROUTE_COUNT (IDISP_DAI_COUNT * 2) #define IDISP_CODEC_MASK 0x4 -static int skl_hda_fill_card_info(struct skl_machine_pdata *pdata) +static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) { struct snd_soc_card *card = &hda_soc_card; struct snd_soc_dai_link *dai_link; u32 codec_count, codec_mask; int i, num_links, num_route; - codec_mask = pdata->codec_mask; + codec_mask = mach_params->codec_mask; codec_count = hweight_long(codec_mask); - if (codec_count == 1 && pdata->codec_mask & IDISP_CODEC_MASK) { + if (codec_count == 1 && codec_mask & IDISP_CODEC_MASK) { num_links = IDISP_DAI_COUNT; num_route = IDISP_ROUTE_COUNT; } else if (codec_count == 2 && codec_mask & IDISP_CODEC_MASK) { @@ -127,14 +127,14 @@ static int skl_hda_fill_card_info(struct skl_machine_pdata *pdata) card->num_dapm_routes = num_route; for_each_card_prelinks(card, i, dai_link) - dai_link->platform_name = pdata->platform; + dai_link->platform_name = mach_params->platform; return 0; } static int skl_hda_audio_probe(struct platform_device *pdev) { - struct skl_machine_pdata *pdata; + struct snd_soc_acpi_mach *mach; struct skl_hda_private *ctx; int ret; @@ -146,11 +146,11 @@ static int skl_hda_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - pdata = dev_get_drvdata(&pdev->dev); - if (!pdata) + mach = dev_get_drvdata(&pdev->dev); + if (!mach) return -EINVAL; - ret = skl_hda_fill_card_info(pdata); + ret = skl_hda_fill_card_info(&mach->mach_params); if (ret < 0) { dev_err(&pdev->dev, "Unsupported HDAudio/iDisp configuration found\n"); return ret; @@ -158,7 +158,7 @@ static int skl_hda_audio_probe(struct platform_device *pdev) ctx->pcm_count = hda_soc_card.num_links; ctx->dai_index = 1; /* hdmi codec dai name starts from index 1 */ - ctx->platform_name = pdata->platform; + ctx->platform_name = mach->mach_params.platform; hda_soc_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&hda_soc_card, ctx); diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 29225623b4b4..ebab04b51bc1 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -525,7 +525,6 @@ static int skl_machine_device_register(struct skl *skl) { struct snd_soc_acpi_mach *mach = skl->mach; struct hdac_bus *bus = skl_to_bus(skl); - struct skl_machine_pdata *pdata; struct platform_device *pdev; int ret; @@ -542,12 +541,9 @@ static int skl_machine_device_register(struct skl *skl) return -EIO; } - if (mach->pdata) { - pdata = (struct skl_machine_pdata *)mach->pdata; - pdata->platform = dev_name(bus->dev); - pdata->codec_mask = bus->codec_mask; - dev_set_drvdata(&pdev->dev, mach->pdata); - } + mach->mach_params.platform = dev_name(bus->dev); + mach->mach_params.codec_mask = bus->codec_mask; + dev_set_drvdata(&pdev->dev, mach); skl->i2s_dev = pdev; -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: Intel: use standard interface for Atom machine drivers" to the asoc tree
by Mark Brown 06 Nov '18

06 Nov '18

The patch ASoC: Intel: use standard interface for Atom machine drivers 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 3ee1cd4f81e15f51638db80fb9f1371b3bdf05ba Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Thu, 1 Nov 2018 16:34:50 -0500 Subject: [PATCH] ASoC: Intel: use standard interface for Atom machine drivers Don't rely on internal Atom/SST-specific data structures, use generic interface to let other drivers use the same machine drivers as is, e.g. SOF to support BYT-CR devices Tested-by: Hans de Goede <hdegoede(a)redhat.com> Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst_acpi.c | 4 ++++ sound/soc/intel/boards/bytcr_rt5640.c | 6 +----- sound/soc/intel/boards/bytcr_rt5651.c | 6 +----- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +----- 4 files changed, 7 insertions(+), 15 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index c90b04cc071d..ac542535b9d5 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -341,6 +341,10 @@ static int sst_acpi_probe(struct platform_device *pdev) byt_rvp_platform_data.res_info = &bytcr_res_info; } + /* update machine parameters */ + mach->mach_params.acpi_ipc_irq_index = + pdata->res_info->acpi_ipc_irq_index; + plat_dev = platform_device_register_data(dev, pdata->platform, -1, NULL, 0); if (IS_ERR(plat_dev)) { diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 8587bd3d1cc1..09591144ea7d 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -29,7 +29,6 @@ #include <linux/input.h> #include <linux/slab.h> #include <asm/cpu_device_id.h> -#include <asm/platform_sst_audio.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -1152,10 +1151,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) * (will be overridden if DMI quirk is detected) */ if (is_valleyview()) { - struct sst_platform_info *p_info = mach->pdata; - const struct sst_res_info *res_info = p_info->res_info; - - if (res_info->acpi_ipc_irq_index == 0) + if (mach->mach_params.acpi_ipc_irq_index == 0) is_bytcr = true; } diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index c44298130720..e528995668b7 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -32,7 +32,6 @@ #include <linux/slab.h> #include <asm/cpu_device_id.h> #include <asm/intel-family.h> -#include <asm/platform_sst_audio.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -920,10 +919,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) * (will be overridden if DMI quirk is detected) */ if (x86_match_cpu(baytrail_cpu_ids)) { - struct sst_platform_info *p_info = mach->pdata; - const struct sst_res_info *res_info = p_info->res_info; - - if (res_info->acpi_ipc_irq_index == 0) + if (mach->mach_params.acpi_ipc_irq_index == 0) is_bytcr = true; } diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index f5a5ea6a093c..250a356a0cbf 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -27,7 +27,6 @@ #include <linux/dmi.h> #include <linux/slab.h> #include <asm/cpu_device_id.h> -#include <asm/platform_sst_audio.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -585,10 +584,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) * (will be overridden if DMI quirk is detected) */ if (is_valleyview()) { - struct sst_platform_info *p_info = mach->pdata; - const struct sst_res_info *res_info = p_info->res_info; - - if (res_info->acpi_ipc_irq_index == 0) + if (mach->mach_params.acpi_ipc_irq_index == 0) is_bytcr = true; } -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: Intel: boards: fix Skylake typo" to the asoc tree
by Mark Brown 06 Nov '18

06 Nov '18

The patch ASoC: Intel: boards: fix Skylake typo 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 48bf41a2be51ef3f67b60f85bf75cecfb57884ba Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Thu, 1 Nov 2018 16:34:51 -0500 Subject: [PATCH] ASoC: Intel: boards: fix Skylake typo s/skylaye/skylake Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +++--- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 ++-- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 4 ++-- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 4 ++-- 4 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 3fa1c3ca6d37..3bc78e3a91ee 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -441,7 +441,7 @@ static int kabylake_refcap_startup(struct snd_pcm_substream *substream) } -static struct snd_soc_ops skylaye_refcap_ops = { +static struct snd_soc_ops skylake_refcap_ops = { .startup = kabylake_refcap_startup, }; @@ -525,7 +525,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .dpcm_capture = 1, .nonatomic = 1, .dynamic = 1, - .ops = &skylaye_refcap_ops, + .ops = &skylake_refcap_ops, }, [KBL_DPCM_AUDIO_DMIC_CP] = { .name = "Kbl Audio DMIC cap", @@ -736,7 +736,7 @@ static struct snd_soc_dai_link kabylake_max98927_dais[] = { .dpcm_capture = 1, .nonatomic = 1, .dynamic = 1, - .ops = &skylaye_refcap_ops, + .ops = &skylake_refcap_ops, }, [KBL_DPCM_AUDIO_DMIC_CP] = { .name = "Kbl Audio DMIC cap", diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 99e1320c485f..6ea969c0a5fb 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -586,7 +586,7 @@ static int kabylake_refcap_startup(struct snd_pcm_substream *substream) &constraints_16000); } -static struct snd_soc_ops skylaye_refcap_ops = { +static struct snd_soc_ops skylake_refcap_ops = { .startup = kabylake_refcap_startup, }; @@ -655,7 +655,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .dpcm_capture = 1, .nonatomic = 1, .dynamic = 1, - .ops = &skylaye_refcap_ops, + .ops = &skylake_refcap_ops, }, [KBL_DPCM_AUDIO_DMIC_CP] = { .name = "Kbl Audio DMIC cap", diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index d31482b8c9bb..552958ce736d 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -400,7 +400,7 @@ static int skylake_refcap_startup(struct snd_pcm_substream *substream) &constraints_16000); } -static const struct snd_soc_ops skylaye_refcap_ops = { +static const struct snd_soc_ops skylake_refcap_ops = { .startup = skylake_refcap_startup, }; @@ -447,7 +447,7 @@ static struct snd_soc_dai_link skylake_dais[] = { .dpcm_capture = 1, .nonatomic = 1, .dynamic = 1, - .ops = &skylaye_refcap_ops, + .ops = &skylake_refcap_ops, }, [SKL_DPCM_AUDIO_DMIC_CP] = { .name = "Skl Audio DMIC cap", diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index e877bb60beb1..f985b30a1d0e 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -449,7 +449,7 @@ static int skylake_refcap_startup(struct snd_pcm_substream *substream) &constraints_16000); } -static const struct snd_soc_ops skylaye_refcap_ops = { +static const struct snd_soc_ops skylake_refcap_ops = { .startup = skylake_refcap_startup, }; @@ -496,7 +496,7 @@ static struct snd_soc_dai_link skylake_dais[] = { .dpcm_capture = 1, .nonatomic = 1, .dynamic = 1, - .ops = &skylaye_refcap_ops, + .ops = &skylake_refcap_ops, }, [SKL_DPCM_AUDIO_DMIC_CP] = { .name = "Skl Audio DMIC cap", -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: Intel: remove GFP_ATOMIC, use GFP_KERNEL" to the asoc tree
by Mark Brown 06 Nov '18

06 Nov '18

The patch ASoC: Intel: remove GFP_ATOMIC, use GFP_KERNEL 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 972b0d456e645ea8fd3fdc70f95f0e41c27c0870 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Thu, 1 Nov 2018 16:34:52 -0500 Subject: [PATCH] ASoC: Intel: remove GFP_ATOMIC, use GFP_KERNEL GFP_ATOMIC is not required on any Intel drivers, use GFP_KERNEL instead. A first cleanup was merged in April but missed a number occurrences and new ones were added by copy/paste inertia. While we are at it, make checkpatch happy with a sizeof(*msg) Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst_pvt.c | 4 ++-- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- 5 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index af93244b4868..00a37a09dc9b 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -166,11 +166,11 @@ int sst_create_ipc_msg(struct ipc_post **arg, bool large) { struct ipc_post *msg; - msg = kzalloc(sizeof(struct ipc_post), GFP_ATOMIC); + msg = kzalloc(sizeof(*msg), GFP_KERNEL); if (!msg) return -ENOMEM; if (large) { - msg->mailbox_data = kzalloc(SST_MAILBOX_SIZE, GFP_ATOMIC); + msg->mailbox_data = kzalloc(SST_MAILBOX_SIZE, GFP_KERNEL); if (!msg->mailbox_data) { kfree(msg); return -ENOMEM; diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 51f0d45d6f8f..9de64f447e7b 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -403,7 +403,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) const char *i2c_name; int i; - drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) return -ENOMEM; diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index c4b94e2617c5..c74c4f17316f 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -603,7 +603,7 @@ static int geminilake_audio_probe(struct platform_device *pdev) { struct glk_card_private *ctx; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 3bc78e3a91ee..58eb0fe69978 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -935,7 +935,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index b6f287fa9505..15c502d6774d 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -140,7 +140,7 @@ static int skl_hda_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "%s: entry\n", __func__); - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: Intel: common: add SOF information for APL RVP" to the asoc tree
by Mark Brown 06 Nov '18

06 Nov '18

The patch ASoC: Intel: common: add SOF information for APL RVP 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 97bb91ae2f3325fd06ea2c28fb8b5b4e023b4b5d Mon Sep 17 00:00:00 2001 From: Bard liao <bard.liao(a)intel.com> Date: Thu, 1 Nov 2018 16:34:53 -0500 Subject: [PATCH] ASoC: Intel: common: add SOF information for APL RVP Add firmware/topology information for APL RVP Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Bard liao <bard.liao(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c index f39386e540d3..2756fa4ab552 100644 --- a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c @@ -19,6 +19,9 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[] = { .id = "INT343A", .drv_name = "bxt_alc298s_i2s", .fw_filename = "intel/dsp_fw_bxtn.bin", + .sof_fw_filename = "intel/sof-apl.ri", + .sof_tplg_filename = "intel/sof-apl-rt298.tplg", + .asoc_plat_name = "0000:00:0e.0", }, { .id = "DLGS7219", -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH v2 0/7] ASoC: Intel: cleanups
by Pierre-Louis Bossart 06 Nov '18

06 Nov '18

This patchset contain 4 different cleanups: a) change of pdata definition to avoid using SST-specific structures and allow for machine driver reuse w/ SOF. No new features, just different data structures. b) removal of remaining GFP_ATOMIC usages c) Cleanup of machine driver tables for APL (missing SOF information and quirk for APL RVP/LeafHill CRB boards). These boards are used outside of Intel so it's worth sharing the fixes for the general public. d) typo v2: Added Andy Shevchenko's Reviewed-by tag removed global variable w/ .driver_data in DMI quirk Bard liao (1): ASoC: Intel: common: add SOF information for APL RVP Pierre-Louis Bossart (6): ASoC: acpi: define common interface for machine driver configuration ASoC: Intel: use standard interface for Hdaudio machine driver ASoC: Intel: use standard interface for Atom machine drivers ASoC: Intel: boards: fix Skylake typo ASoC: Intel: remove GFP_ATOMIC, use GFP_KERNEL ASoC: Intel: common: add quirk for APL RVP boards include/sound/soc-acpi.h | 14 ++++++++ sound/soc/intel/atom/sst/sst_acpi.c | 4 +++ sound/soc/intel/atom/sst/sst_pvt.c | 4 +-- sound/soc/intel/boards/bytcr_rt5640.c | 6 +--- sound/soc/intel/boards/bytcr_rt5651.c | 6 +--- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +--- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 ++--- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 +-- sound/soc/intel/boards/skl_hda_dsp_generic.c | 22 ++++++------ .../soc/intel/boards/skl_nau88l25_max98357a.c | 4 +-- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 4 +-- .../intel/common/soc-acpi-intel-bxt-match.c | 36 +++++++++++++++++++ sound/soc/intel/skylake/skl.c | 10 ++---- 15 files changed, 85 insertions(+), 47 deletions(-) -- 2.17.1

2 8

[alsa-devel] [PATCH 0/3] audio-graph-card: remove endpoint bidirectional check
by Kuninori Morimoto 06 Nov '18

06 Nov '18

Hi Mark These are cleanup for audio-graph-card. It removes unneeded error check code Kuninori Morimoto (3): ASoC: audio-graph-card: remove endpoint bidirectional check ASoC: audio-graph-scu-card: remove endpoint bidirectional check ASoC: audio-graph-scu-card: remove error check which never happen sound/soc/generic/audio-graph-card.c | 9 --------- sound/soc/generic/audio-graph-scu-card.c | 19 +------------------ 2 files changed, 1 insertion(+), 27 deletions(-) -- 2.7.4

2 6

[alsa-devel] [PATCH 00/10] ASoC: rsnd: add TDM Split mode
by Kuninori Morimoto 06 Nov '18

06 Nov '18

Hi Mark Cc: Jiada These patches are for TDM Split mode support for Renesas rsnd driver. 1) - 6) : tidyup / prepare for TDM Split mode suport 7), 9) : DT binding patch for TDM Split mode 8), 10) : TDM Split mode support Kuninori Morimoto (10): 1) ASoC: rsnd: remove endpoint bidirectional check 2) ASoC: rsnd: rename rsnd_runtime_is_ssi_xxx() 3) ASoC: rsnd: use defined io_playback/io_capture 4) ASoC: rsnd: move HDMI information from ssi.c to core.c 5) ASoC: rsnd: handle DPCM converted rate/chan under core 6) ASoC: rsnd: tidyup Document for DT binding 7) ASoC: rsnd: add SSIU BUSIF support for Document 8) ASoC: rsnd: add SSIU BUSIF support 9) ASoC: rsnd: add TDM Split mode support for Document 10) ASoC: rsnd: add TDM Split mode support .../devicetree/bindings/sound/renesas,rsnd.txt | 388 ++++++++++++--------- sound/soc/sh/rcar/core.c | 112 +++++- sound/soc/sh/rcar/ctu.c | 45 --- sound/soc/sh/rcar/dma.c | 37 +- sound/soc/sh/rcar/rsnd.h | 30 +- sound/soc/sh/rcar/src.c | 40 +-- sound/soc/sh/rcar/ssi.c | 115 ++---- sound/soc/sh/rcar/ssiu.c | 212 +++++++++-- 8 files changed, 604 insertions(+), 375 deletions(-) -- 2.7.4 Best regards --- Kuninori Morimoto

2 19

[alsa-devel] Applied "ASoC: rsnd: add TDM Split mode support for Document" to the asoc tree
by Mark Brown 06 Nov '18

06 Nov '18

The patch ASoC: rsnd: add TDM Split mode support for Document 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 cf704dc83fd6600758e98ac71769eaeb00dfc770 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 6 Nov 2018 05:21:22 +0000 Subject: [PATCH] ASoC: rsnd: add TDM Split mode support for Document This patch adds TDM Split mode support. rsnd driver is assuming audio-graph-scu-card is used for Sound Card. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/renesas,rsnd.txt | 91 +++++++++++++++++++ 1 file changed, 91 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 6ed1a3658ffd..453e8ec2ac78 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -188,6 +188,25 @@ In DMA mode, you can select not to use SSIU by using "no-busif" via SSI. SSIU handles BUSIF which will be used for TDM Split mode. This driver is assuming that audio-graph card will be used. +TDM Split mode merges 4 sounds. You can see 4 sound interface on system, +and these sounds will be merged SSIU/SSI. + + aplay -D plughw:0,0 xxxx.wav & + aplay -D plughw:0,1 xxxx.wav & + aplay -D plughw:0,2 xxxx.wav & + aplay -D plughw:0,3 xxxx.wav + + 2ch 8ch + [MEM] -> [SSIU 30] -+-> [SSIU 3] --> [Codec] + 2ch | + [MEM] -> [SSIU 31] -+ + 2ch | + [MEM] -> [SSIU 32] -+ + 2ch | + [MEM] -> [SSIU 33] -+ + +see "Example: simple sound card for TDM Split" + ============================================= * SSI (Serial Sound Interface) ============================================= @@ -648,6 +667,78 @@ rsnd_tdm: sound { }; }; +============================================= +Example: simple sound card for TDM Split +============================================= + +sound_card: sound { + compatible = "audio-graph-scu-card"; + prefix = "xxxx"; + routing = "xxxx Playback", "DAI0 Playback", + "xxxx Playback", "DAI1 Playback", + "xxxx Playback", "DAI2 Playback", + "xxxx Playback", "DAI3 Playback"; + convert-channels = <8>; /* TDM Split */ + + dais = <&rsnd_port0 /* playback ch1/ch2 */ + &rsnd_port1 /* playback ch3/ch4 */ + &rsnd_port2 /* playback ch5/ch6 */ + &rsnd_port3 /* playback ch7/ch8 */ + >; +}; + +audio-codec { + ... + port { + codec_0: endpoint@1 { + remote-endpoint = <&rsnd_ep0>; + }; + codec_1: endpoint@2 { + remote-endpoint = <&rsnd_ep1>; + }; + codec_2: endpoint@3 { + remote-endpoint = <&rsnd_ep2>; + }; + codec_3: endpoint@4 { + remote-endpoint = <&rsnd_ep3>; + }; + }; +}; + +&rcar_sound { + ... + ports { + rsnd_port0: port@0 { + rsnd_ep0: endpoint { + remote-endpoint = <&codec_0>; + ... + playback = <&ssiu30 &ssi3>; + }; + }; + rsnd_port1: port@1 { + rsnd_ep1: endpoint { + remote-endpoint = <&codec_1>; + ... + playback = <&ssiu31 &ssi3>; + }; + }; + rsnd_port2: port@2 { + rsnd_ep2: endpoint { + remote-endpoint = <&codec_2>; + ... + playback = <&ssiu32 &ssi3>; + }; + }; + rsnd_port3: port@3 { + rsnd_ep3: endpoint { + remote-endpoint = <&codec_3>; + ... + playback = <&ssiu33 &ssi3>; + }; + }; + }; +}; + ============================================= Example: simple sound card for Multi channel ============================================= -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH - dmix v3 0/1] pcm: dmix: Align slave_hw_ptr to slave period boundary
by Timo Wischer 06 Nov '18

06 Nov '18

On 10/30/18 14:21, Takashi Iwai wrote: > Yes, and is this correct? Suppose you start a stream at the position > one sample before the next period (psize * N + (psize-1)). Then > slave_hw_ptr is psize * N. At the next moment, the dpcm->hw.ptr > reaches to psize * (N+1), and snd_pcm_dmix_sync_ptr() gets called. > Then slave_hw_ptr will be updated to psize * (N+1) although only one > sample has been processed? I have created a spreadsheet to simulate the behavior of dmix (see attachment). I used this sheet to test different corner cases with different implementations. Column B-D describes 3 corner cases with the original implementation. Column E-H describes the corner cases with patch v2 applied and column I-L describes the corner cases with patch v3. With patch v3 I was not able to find a corner case which would fail. All our internal audio test are also passing fine. The corner cases are described in the form N*period+period-1 (N+1)period+1 (N+2)period+1 The first line describes the value of dmix->spcm->hw.ptr when snd_pcm_dmix_start() will be called. The second line describes the hw_ptr when poll() returns for the first time and the third line describes the hw_ptr when poll() returns for the second time. In column B is the issue case described when the patch is not applied. snd_pcm_avail() returns 2 frames only (B27) but the buffer only contains 3 frames (B18). Therefore up to 5 frames are available (a buffer size of 8 frames is choosen). The issue case of patch v2 is shown in column H. snd_pcm_avail() returns 7 frames (H46) but the buffer contains still 3 frames (H37). Therefore there are only 5 frames free to overwrite. Do you see any corner cases which I missed or any other drawbacks? Best regards and thanks for your time Timo

3 5

[alsa-devel] [PATCH] ASoC: pxa-ssp: make suspend call resilient against multiple invocations
by Daniel Mack 06 Nov '18

06 Nov '18

The suspend callback is called twice when the machine is suspended, once through the ASoC layer and once through the platform device layer. Make sure not to confuse the clock layer with unbalanced calls, and bail out early if we are already suspended. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/pxa/pxa-ssp.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index adcf8ba9d287..fd93706702a6 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -51,6 +51,7 @@ struct ssp_priv { uint32_t cr1; uint32_t to; uint32_t psp; + bool suspended; #endif }; @@ -142,6 +143,9 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); struct ssp_device *ssp = priv->ssp; + if (priv->suspended) + return 0; + if (!cpu_dai->active) clk_prepare_enable(ssp->clk); @@ -152,6 +156,8 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) pxa_ssp_disable(ssp); clk_disable_unprepare(ssp->clk); + priv->suspended = true; + return 0; } @@ -161,6 +167,9 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) struct ssp_device *ssp = priv->ssp; uint32_t sssr = SSSR_ROR | SSSR_TUR | SSSR_BCE; + if (!priv->suspended) + return 0; + clk_prepare_enable(ssp->clk); __raw_writel(sssr, ssp->mmio_base + SSSR); @@ -174,6 +183,8 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) else clk_disable_unprepare(ssp->clk); + priv->suspended = false; + return 0; } -- 2.17.2

2 1

[alsa-devel] [PATCH] fix the kernel panic due to wrong use the dev memory API
by He, Bo 06 Nov '18

06 Nov '18

skl->dais is allocated with devm_kcalloc, can't free with the krealloc. Memory allocated with devm API is automatically freed on driver detach, Like all other devres resources. Refer to drivers/base/devres.c devm_kmalloc for more details. Signed-off-by: he, bo <bo.he(a)intel.com> --- sound/soc/intel/skylake/skl-pcm.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 823e391..928d314 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1438,7 +1438,8 @@ int skl_platform_register(struct device *dev) } if (!skl->use_tplg_pcm) { - dais = krealloc(skl->dais, sizeof(skl_fe_dai) + + devm_kfree(dev, skl->dais); + dais = devm_kcalloc(dev, skl->dais, sizeof(skl_fe_dai) + sizeof(skl_platform_dai), GFP_KERNEL); if (!dais) { ret = -ENOMEM; @@ -1472,7 +1473,5 @@ int skl_platform_unregister(struct device *dev) } } - kfree(skl->dais); - return 0; } -- 2.7.4

3 4

[alsa-devel] [PATCH] ALSA: hda - Fix incorrect clearance of thinkpad_acpi hooks
by Takashi Iwai 06 Nov '18

06 Nov '18

Since the commit c647f806b8c2 ("ALSA: hda - Allow multiple ADCs for mic mute LED controls") we allow enabling the mic mute LED with multiple ADCs. The commit changed the function return value to be zero or a negative error, while this change was overlooked in the thinkpad_acpi helper code where it still expects a positive return value for success. This eventually leads to a NULL dereference on a system that has only a mic mute LED. This patch corrects the return value check in the corresponding code as well. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201621 Fixes: c647f806b8c2 ("ALSA: hda - Allow multiple ADCs for mic mute LED controls") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/thinkpad_helper.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 97f49b751e6e..568575b72f2f 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -58,8 +58,8 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, removefunc = false; } if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0 && - snd_hda_gen_add_micmute_led(codec, - update_tpacpi_micmute) > 0) + !snd_hda_gen_add_micmute_led(codec, + update_tpacpi_micmute)) removefunc = false; } -- 2.19.1

1 0

[alsa-devel] [PATCH] ASoC: skl: always select SND_SOC_HDAC_HDA
by Arnd Bergmann 06 Nov '18

06 Nov '18

The skylake sound support is written to work both with or without CONFIG_SND_SOC_HDAC_HDA, and uses an #ifdef to decide whether it should link against that. However, this fails with SND_SOC_ALL_CODECS=m or SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH=m when the Skylake support itself is built-in, with this link error: sound/soc/intel/skylake/skl.o: In function `skl_probe': skl.c:(.text+0x56c): undefined reference to `snd_soc_hdac_hda_get_ops' Using an explicit 'select' here simplifies the logic and prevents it from happening, at the cost of always including the compile time dependency. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/intel/Kconfig | 1 + sound/soc/intel/skylake/skl.c | 2 -- 2 files changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 0caa1f4eb94d..c21ce7624af1 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -109,6 +109,7 @@ config SND_SOC_INTEL_SKYLAKE depends on PCI && ACPI select SND_HDA_EXT_CORE select SND_HDA_DSP_LOADER + select SND_SOC_HDAC_HDA select SND_SOC_TOPOLOGY select SND_SOC_INTEL_SST select SND_SOC_ACPI_INTEL_MATCH diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 29225623b4b4..1069ee265ce5 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -870,9 +870,7 @@ static int skl_create(struct pci_dev *pci, hbus = skl_to_hbus(skl); bus = skl_to_bus(skl); -#if IS_ENABLED(CONFIG_SND_SOC_HDAC_HDA) ext_ops = snd_soc_hdac_hda_get_ops(); -#endif snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, ext_ops); bus->use_posbuf = 1; skl->pci = pci; -- 2.18.0

5 10

[alsa-devel] plughw behaviour not consistent with hw
by Stefano Antonelli 06 Nov '18

06 Nov '18

Hello list, I'm working on an application that requires a period time of 10ms. The hardware at the other end of the bus is a ulaw codec running at 8kHz. The application runs at 16kHz S16_LE and I was hoping to use the plughw device to do sample rate conversion and ulaw encode/decode. This way the application doesn't need to change to accommodate this ulaw codec. However, it's not working and I don't know how to go about debugging. I'm hoping I can get some pointers here. The problem is best illustrated by this output, first using plughw: alsa_read: 10 alsa_read: 20 alsa_read: 0 alsa_read: 20 alsa_read: 0 alsa_read: 20 alsa_read: 0 alsa_read: 20 alsa_read: 0 alsa_read: 20 It would appear that plughw gives the correct period time on the first call, but then subsequently buffers 2 periods before returning. Using hw with matching sample rate, sample format, and number of samples: alsa_read: 10 alsa_read: 10 alsa_read: 10 alsa_read: 10 alsa_read: 10 alsa_r ead: 10 alsa_read: 10 alsa_read: 10 alsa_read: 10 alsa_read: 10 This code snippet generated the above output: for (i = 0; i < 10; ++i) { gettimeofday(&time1, NULL); err = snd_pcm_readi (capture_handle, buf, NSMP); if (err != NSMP) { printf ("read from audio interface failed " " (%s)\n", snd_strerror (err)); exit (1); } gettimeofday(&time2, NULL); printf("alsa_read: %d\n", getTimeDiff(time1, time2)); } I have tried using the patch from here (without the environment variable switch): http://mailman.alsa-project.org/pipermail/alsa-devel/2008-October/01170 3.html to print out the configuration of plughw's plugin chain, but it generates no output. I've verified the md5sum of the library after compilation and it matches the one I compiled. And I ran ldconfig after replacing libasound.so. So I don't know what's going on there. I definitely call snd_pcm_hw_params() when setting up the capture device. This is using alsa-lib-1.0.25. Does anyone know why plughw might be acting the way it is and how I can get it to deliver 10ms periods? Equally appreciated would be any tips on how to debug this problem. Thanks, Stef

2 1

[alsa-devel] [PATCH][RESEND] ASoC: pcm3168a: remove read-only status register from snd_kcontrol_new
by Kuninori Morimoto 06 Nov '18

06 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> "DACn Zero Flag" and "ADCn Overflow Flag" are read-only status register. These information are needed for driver not for user. Let's remove these from snd_kcontrol_new. There is no detail explanation for these register on datasheet, but, judging from its behavior, it seems these register needs clock to read. Otherwise, I2C returns error. Therefor, amixer command will be failed under non working timing. Without this patch, user often fail amixer command. This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 TDM sound. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- > Mark 2weeks passed. repost this patch sound/soc/codecs/pcm3168a.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 5f4327f..115f2165 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -133,10 +133,6 @@ static const struct snd_kcontrol_new pcm3168a_snd_controls[] = { SOC_DOUBLE("DAC2 Invert Switch", PCM3168A_DAC_INV, 2, 3, 1, 0), SOC_DOUBLE("DAC3 Invert Switch", PCM3168A_DAC_INV, 4, 5, 1, 0), SOC_DOUBLE("DAC4 Invert Switch", PCM3168A_DAC_INV, 6, 7, 1, 0), - SOC_DOUBLE_STS("DAC1 Zero Flag", PCM3168A_DAC_ZERO, 0, 1, 1, 0), - SOC_DOUBLE_STS("DAC2 Zero Flag", PCM3168A_DAC_ZERO, 2, 3, 1, 0), - SOC_DOUBLE_STS("DAC3 Zero Flag", PCM3168A_DAC_ZERO, 4, 5, 1, 0), - SOC_DOUBLE_STS("DAC4 Zero Flag", PCM3168A_DAC_ZERO, 6, 7, 1, 0), SOC_ENUM("DAC Volume Control Type", pcm3168a_dac_volume_type), SOC_ENUM("DAC Volume Rate Multiplier", pcm3168a_dac_att_mult), SOC_ENUM("DAC De-Emphasis", pcm3168a_dac_demp), @@ -176,9 +172,6 @@ static const struct snd_kcontrol_new pcm3168a_snd_controls[] = { SOC_DOUBLE("ADC1 Mute Switch", PCM3168A_ADC_MUTE, 0, 1, 1, 0), SOC_DOUBLE("ADC2 Mute Switch", PCM3168A_ADC_MUTE, 2, 3, 1, 0), SOC_DOUBLE("ADC3 Mute Switch", PCM3168A_ADC_MUTE, 4, 5, 1, 0), - SOC_DOUBLE_STS("ADC1 Overflow Flag", PCM3168A_ADC_OV, 0, 1, 1, 0), - SOC_DOUBLE_STS("ADC2 Overflow Flag", PCM3168A_ADC_OV, 2, 3, 1, 0), - SOC_DOUBLE_STS("ADC3 Overflow Flag", PCM3168A_ADC_OV, 4, 5, 1, 0), SOC_ENUM("ADC Volume Control Type", pcm3168a_adc_volume_type), SOC_ENUM("ADC Volume Rate Multiplier", pcm3168a_adc_att_mult), SOC_ENUM("ADC Overflow Flag Polarity", pcm3168a_adc_ov_pol), -- 2.7.4

1 0

[alsa-devel] aplay: fix description about options for process duration
by GitHub pull_request - opened 06 Nov '18

06 Nov '18

alsa-project/alsa-utils pull request #1 was opened from takaswie: Hi, At present, aplay has two options for duration to process PCM frames. However, the recent one ('--samples') option is not described yet in its manual. Furthermore, '-s' option was used for '--sleep-min' option once and description for this option was left as is. This can puzzle uses. This patchset improves description about options for process duration. Request URL : https://github.com/alsa-project/alsa-utils/pull/1 Patch URL : https://github.com/alsa-project/alsa-utils/pull/1.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [PATCH v2 0/3] Huawei laptops WMI & sound fixes
by Ayman Bagabas 05 Nov '18

05 Nov '18

This patch set fixes some of the issues with Huawei laptops. [PATCH v2 1/3] The first patch adds support for missing hotkeys on some models. Some hotkeys, like brightness keys, work out of the box on these models. [PATCH v2 2/3] This one enables the front speakers on the Huawei Matebook X Pro (MBXP). This solves bug 200501 https://bugzilla.kernel.org/show_bug.cgi?id=200501 It simply uses the pins configurations generated by hdajackretast using the settings posted on the bug page https://imgur.com/a/N1xsCVZ [PATCH v2 3/3] This enables the micmute LED on Huawei laptops. It calls an WMI method, using PATCH #1, to turn the micmute LED on/off. Ayman Bagabas (3): x86: add support for Huawei WMI hotkeys. ALSA: hda: fix front speakers on Huawei MBXP. ALSA: hda: add support for Huawei WMI MicMute LED drivers/platform/x86/Kconfig | 13 ++ drivers/platform/x86/Makefile | 1 + drivers/platform/x86/huawei_wmi.c | 224 ++++++++++++++++++++++++++++++ include/linux/huawei_wmi.h | 7 + sound/pci/hda/huawei_wmi_helper.c | 48 +++++++ sound/pci/hda/patch_realtek.c | 28 ++++ 6 files changed, 321 insertions(+) create mode 100644 drivers/platform/x86/huawei_wmi.c create mode 100644 include/linux/huawei_wmi.h create mode 100644 sound/pci/hda/huawei_wmi_helper.c -- 2.17.2

4 13

[alsa-devel] linux-4.20-rc1/sound/soc/stm/stm32_sai_sub.c:343: Faulty error check
by David Binderman 05 Nov '18

05 Nov '18

Hello there, linux-4.20-rc1/sound/soc/stm/stm32_sai_sub.c:343]: (style) Checking if unsigned variable 'div' is less than zero. Source code is div = stm32_sai_get_clk_div(sai, parent_rate, rate); if (div < 0) return div; Suggest store return value from function in int local variable, test it and only then assign it into an unsigned variable. Regards David Binderman

2 1

[alsa-devel] [PATCH] vga_switcheroo: Fix missing gpu_bound call at audio client registration
by Takashi Iwai 05 Nov '18

05 Nov '18

The commit 37a3a98ef601 ("ALSA: hda - Enable runtime PM only for discrete GPU") added a new ops gpu_bound to be called when GPU gets bound. The patch overlooked, however, that vga_switcheroo_enable() is called only once at GPU is bound. When an audio client is registered after that point, it would miss the gpu_bound call. This leads to the unexpected lack of runtime PM in HD-audio side. For addressing that regression, just call gpu_bound callback manually at vga_switcheroo_register_audio_client() when the GPU was already bound. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201615 Fixes: 37a3a98ef601 ("ALSA: hda - Enable runtime PM only for discrete GPU") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- drivers/gpu/vga/vga_switcheroo.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/drivers/gpu/vga/vga_switcheroo.c b/drivers/gpu/vga/vga_switcheroo.c index cf2a18571d48..a132c37d7334 100644 --- a/drivers/gpu/vga/vga_switcheroo.c +++ b/drivers/gpu/vga/vga_switcheroo.c @@ -380,6 +380,9 @@ int vga_switcheroo_register_audio_client(struct pci_dev *pdev, mutex_unlock(&vgasr_mutex); return -EINVAL; } + /* notify if GPU has been already bound */ + if (ops->gpu_bound) + ops->gpu_bound(pdev, id); } mutex_unlock(&vgasr_mutex); -- 2.19.1

2 6

[alsa-devel] [PATCH 0/5] ASoC: omap-mcbsp: Cleanup and split core/sidetone support
by Peter Ujfalusi 05 Nov '18

05 Nov '18

Hi, While preparing for merging the davinci and omap directories (I will send the series for that shortly) I have taken some time to do cleanup on the McBSP driver. The mcbsp.c/h files were the result when we moved code from arch/arm/plat-omap to sound/soc/omap/ a long time ago and it contained code for McBSP core and the OMAP3 sidetone functionality. With this series I tried to split the core and sidetone code to separate files. There were two reasons for this: 1. to have the OMAP specific DAI driver files prefixed with omap-* under the new sound/soc/ti/ directory 2. easier to find functions as all sidetone code is in separate file. Regards, Peter --- Peter Ujfalusi (5): ASoC: omap-mcbsp: Clean up dma_data addr initialization code ASoC: omap-mcbsp: Clean up the interrupt handlers ASoC: omap-mcbsp: Simplify the mcbsp_start/_stop function parameters ASoC: omap-mcbsp: Move out the FIFO check from set_threshold and get_delay ASoC: omap-mcbsp: Re-arrange files for core McBSP and Sidetone function split sound/soc/omap/Makefile | 2 +- sound/soc/omap/mcbsp.c | 1104 ----------------- sound/soc/omap/{mcbsp.h => omap-mcbsp-priv.h} | 126 +- sound/soc/omap/omap-mcbsp-st.c | 516 ++++++++ sound/soc/omap/omap-mcbsp.c | 855 ++++++++++--- sound/soc/omap/omap-mcbsp.h | 8 +- 6 files changed, 1281 insertions(+), 1330 deletions(-) delete mode 100644 sound/soc/omap/mcbsp.c rename sound/soc/omap/{mcbsp.h => omap-mcbsp-priv.h} (70%) create mode 100644 sound/soc/omap/omap-mcbsp-st.c -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

1 5

[alsa-devel] [PATCH] ASoC: sunxi: rename SND_SUNXI_ADDA_PR_REGMAP to SND_SUN8I_ADDA_PR_REGMAP
by Vasily Khoruzhick 05 Nov '18

05 Nov '18

SND_SUN50I_CODEC_ANALOG selects SND_SUNXI_ADDA_PR_REGMAP which is leftover of renaming SND_SUNXI_ADDA_PR_REGMAP to SND_SUN8I_ADDA_PR_REGMAP. Replace it with SND_SUN8I_ADDA_PR_REGMAP to fix possible link errors for some configurations: sound/soc/sunxi/sun50i-codec-analog.o: In function `sun50i_codec_analog_probe': sun50i-codec-analog.c:(.text+0x62): undefined reference to `sun8i_adda_pr_regmap_init' Fixes: 42371f327df0 ("ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls") Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- sound/soc/sunxi/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 66aad0d3f9c7..8134c3c94229 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -31,7 +31,7 @@ config SND_SUN8I_CODEC_ANALOG config SND_SUN50I_CODEC_ANALOG tristate "Allwinner sun50i Codec Analog Controls Support" depends on (ARM64 && ARCH_SUNXI) || COMPILE_TEST - select SND_SUNXI_ADDA_PR_REGMAP + select SND_SUN8I_ADDA_PR_REGMAP help Say Y or M if you want to add support for the analog controls for the codec embedded in Allwinner A64 SoC. -- 2.19.1

3 2

[alsa-devel] [PATCH] ASoC: stm32: sai: fix less than zero comparison on unsigned int
by Colin King 05 Nov '18

05 Nov '18

From: Colin Ian King <colin.king(a)canonical.com> Currently, the comparison of div < 0 is always false because div is an unsigned int. Fix this by making div an int. Detected by CoverityScan, CID#1475309 ("Unsigned compared against 0") Fixes: 8307b2afd386 "(ASoC: stm32: sai: set sai as mclk clock provider") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/stm/stm32_sai_sub.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 211589b0b2ef..d4825700b63f 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -336,8 +336,7 @@ static int stm32_sai_mclk_set_rate(struct clk_hw *hw, unsigned long rate, { struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); struct stm32_sai_sub_data *sai = mclk->sai_data; - unsigned int div; - int ret; + int div, ret; div = stm32_sai_get_clk_div(sai, parent_rate, rate); if (div < 0) -- 2.19.1

2 1

[alsa-devel] [PATCH] ASoC: smd845: constify snd_soc_ops structure
by Julia Lawall 05 Nov '18

05 Nov '18

The snd_soc_ops structure can be const as it is only stored in the ops field of a snd_soc_dai_link structure and this field is const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/soc/qcom/sdm845.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 9effbecc571f..8d0cdff64377 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -171,7 +171,7 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream) } } -static struct snd_soc_ops sdm845_be_ops = { +static const struct snd_soc_ops sdm845_be_ops = { .hw_params = sdm845_snd_hw_params, .startup = sdm845_snd_startup, .shutdown = sdm845_snd_shutdown,

2 1

[alsa-devel] [PATCH] [v2] ASoC: wm97xx: fix uninitialized regmap pointer problem
by Arnd Bergmann 05 Nov '18

05 Nov '18

gcc notices that without either the ac97 bus or the pdata, we never initialize the regmap pointer, which leads to an uninitialized variable access: sound/soc/codecs/wm9712.c: In function 'wm9712_soc_probe': sound/soc/codecs/wm9712.c:666:2: error: 'regmap' may be used uninitialized in this function [-Werror=maybe-uninitialized] Since that configuration is invalid, it's better to return an error here. I tried to avoid adding complexity to the conditions, and turned the #ifdef into a regular if(IS_ENABLED()) check for readability. This in turn requires moving some header file declarations out of an #ifdef. The same code is used in three drivers, all of which I'm changing the same way. Fixes: 2ed1a8e0ce8d ("ASoC: wm9712: add ac97 new bus support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- v2: fix a missing check that Charles pointed out --- include/sound/soc.h | 2 +- sound/soc/codecs/wm9705.c | 10 ++++------ sound/soc/codecs/wm9712.c | 10 ++++------ sound/soc/codecs/wm9713.c | 10 ++++------ 4 files changed, 13 insertions(+), 19 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 70c10a8f3e90..3e0ac310a3df 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -553,12 +553,12 @@ static inline void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, } #endif -#ifdef CONFIG_SND_SOC_AC97_BUS struct snd_ac97 *snd_soc_alloc_ac97_component(struct snd_soc_component *component); struct snd_ac97 *snd_soc_new_ac97_component(struct snd_soc_component *component, unsigned int id, unsigned int id_mask); void snd_soc_free_ac97_component(struct snd_ac97 *ac97); +#ifdef CONFIG_SND_SOC_AC97_BUS int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, struct platform_device *pdev); diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index ccdf088461b7..54c306707c02 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -325,8 +325,7 @@ static int wm9705_soc_probe(struct snd_soc_component *component) if (wm9705->mfd_pdata) { wm9705->ac97 = wm9705->mfd_pdata->ac97; regmap = wm9705->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { wm9705->ac97 = snd_soc_new_ac97_component(component, WM9705_VENDOR_ID, WM9705_VENDOR_ID_MASK); if (IS_ERR(wm9705->ac97)) { @@ -339,7 +338,8 @@ static int wm9705_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9705->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_set_drvdata(component, wm9705->ac97); @@ -350,14 +350,12 @@ static int wm9705_soc_probe(struct snd_soc_component *component) static void wm9705_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9705_priv *wm9705 = snd_soc_component_get_drvdata(component); - if (!wm9705->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9705->mfd_pdata) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9705->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9705 = { diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index e873baa9e778..01949eaba4fd 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -642,8 +642,7 @@ static int wm9712_soc_probe(struct snd_soc_component *component) if (wm9712->mfd_pdata) { wm9712->ac97 = wm9712->mfd_pdata->ac97; regmap = wm9712->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { int ret; wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID, @@ -660,7 +659,8 @@ static int wm9712_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9712->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_init_regmap(component, regmap); @@ -673,14 +673,12 @@ static int wm9712_soc_probe(struct snd_soc_component *component) static void wm9712_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component); - if (!wm9712->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9712->mfd_pdata) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9712->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9712 = { diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 643863bb32e0..5a2fdf4f69bf 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1214,8 +1214,7 @@ static int wm9713_soc_probe(struct snd_soc_component *component) if (wm9713->mfd_pdata) { wm9713->ac97 = wm9713->mfd_pdata->ac97; regmap = wm9713->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { wm9713->ac97 = snd_soc_new_ac97_component(component, WM9713_VENDOR_ID, WM9713_VENDOR_ID_MASK); if (IS_ERR(wm9713->ac97)) @@ -1225,7 +1224,8 @@ static int wm9713_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9713->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_init_regmap(component, regmap); @@ -1238,14 +1238,12 @@ static int wm9713_soc_probe(struct snd_soc_component *component) static void wm9713_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9713_priv *wm9713 = snd_soc_component_get_drvdata(component); - if (!wm9713->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9713->mfd_pdata) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9713->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9713 = { -- 2.18.0

3 2

[alsa-devel] [PATCH] ASoC: Intel: mrfld: fix uninitialized variable access
by Arnd Bergmann 05 Nov '18

05 Nov '18

Randconfig testing revealed a very old bug, with gcc-8: sound/soc/intel/atom/sst/sst_loader.c: In function 'sst_load_fw': sound/soc/intel/atom/sst/sst_loader.c:357:5: error: 'fw' may be used uninitialized in this function [-Werror=maybe-uninitialized] if (fw == NULL) { ^ sound/soc/intel/atom/sst/sst_loader.c:354:25: note: 'fw' was declared here const struct firmware *fw; We must check the return code of request_firmware() before we look at the pointer result that may be uninitialized when the function fails. Fixes: 9012c9544eea ("ASoC: Intel: mrfld - Add DSP load and management") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/intel/atom/sst/sst_loader.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index 27413ebae956..b8c456753f01 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -354,14 +354,14 @@ static int sst_request_fw(struct intel_sst_drv *sst) const struct firmware *fw; retval = request_firmware(&fw, sst->firmware_name, sst->dev); - if (fw == NULL) { - dev_err(sst->dev, "fw is returning as null\n"); - return -EINVAL; - } if (retval) { dev_err(sst->dev, "request fw failed %d\n", retval); return retval; } + if (fw == NULL) { + dev_err(sst->dev, "fw is returning as null\n"); + return -EINVAL; + } mutex_lock(&sst->sst_lock); retval = sst_cache_and_parse_fw(sst, fw); mutex_unlock(&sst->sst_lock); -- 2.18.0

3 2

[alsa-devel] Applied "ASoC: nau8825: remove unnecessary unlikely()" to the asoc tree
by Mark Brown 05 Nov '18

05 Nov '18

The patch ASoC: nau8825: remove unnecessary unlikely() 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 0b6277e6343e192aaa7d452ab933281eb0d420dc Mon Sep 17 00:00:00 2001 From: Yangtao Li <tiny.windzz(a)gmail.com> Date: Sun, 4 Nov 2018 07:55:29 -0500 Subject: [PATCH] ASoC: nau8825: remove unnecessary unlikely() WARN_ON() already contains an unlikely(), so it's not necessary to use unlikely. Signed-off-by: Yangtao Li <tiny.windzz(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/nau8825.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index b9fed99d8b5e..7bbcbf5f05c8 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -424,10 +424,8 @@ static u32 nau8825_xtalk_sidetone(u32 sig_org, u32 sig_cros) { u32 gain, sidetone; - if (unlikely(sig_org == 0) || unlikely(sig_cros == 0)) { - WARN_ON(1); + if (WARN_ON(sig_org == 0 || sig_cros == 0)) return 0; - } sig_org = nau8825_intlog10_dec3(sig_org); sig_cros = nau8825_intlog10_dec3(sig_cros); -- 2.19.0.rc2

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[alsa-devel] [PATCH] ASoC: AMD: constify regulator_desc structure
by Julia Lawall 05 Nov '18

05 Nov '18

The regulator_desc structure can be const as it is only passed as the second argument of devm_regulator_register and the corresponding parameter is declared as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 3f813ea5210a..a5daad973ce5 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -403,7 +403,7 @@ static struct regulator_config acp_da7219_cfg = { static struct regulator_ops acp_da7219_ops = { }; -static struct regulator_desc acp_da7219_desc = { +static const struct regulator_desc acp_da7219_desc = { .name = "reg-fixed-1.8V", .type = REGULATOR_VOLTAGE, .owner = THIS_MODULE,

2 1

[alsa-devel] [PATCH] ASoC: omap-mcbsp: On OMAP1 and OMAP2420 only 16bit format is supported
by Peter Ujfalusi 05 Nov '18

05 Nov '18

For samples sizes of 20/24/32 bit the DRR2/DXR2 registers must be used to receive or transmit the most significant part of the 32bit sample. We can not trick the system by using the DRR2/DXR2 register offset and use 32bit element size since these (and other McBSP) registers are 16bit ones. We would need support for port_window in cyclic mode from the sDMA driver, but it is not straight forward as we would need to have the sDMA frame to cover the 32bit (DRR2+DRR1/DXR2+DXR1), but the frames must cover the ALSA period to be able to receive periodic interrupts. Since the 32bit samples are not working (DMA timeout), just remove it in case we have McBSP with reg_size of 2. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, I have noticed that on Nokia n810 32bit samples just don't work at all for audio Regards, Peter sound/soc/omap/omap-mcbsp.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d0ebb6b9bfac..caa24c463b3e 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -862,6 +862,11 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) if (ret) return ret; + if (mcbsp->pdata->reg_size == 2) { + omap_mcbsp_dai.playback.formats = SNDRV_PCM_FMTBIT_S16_LE; + omap_mcbsp_dai.capture.formats = SNDRV_PCM_FMTBIT_S16_LE; + } + ret = devm_snd_soc_register_component(&pdev->dev, &omap_mcbsp_component, &omap_mcbsp_dai, 1); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

3 4

[alsa-devel] [PATCH] ASoC: pcm3168a: cleanup hw_constraint for LEFT_J/I2S
by Kuninori Morimoto 05 Nov '18

05 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> hw constraint for LEFT_J/I2S are same, but had duplicated code. This code cleanup these. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/pcm3168a.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index e1c3a18..115f2165 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -497,6 +497,10 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, unsigned int fmt; unsigned int sample_min; unsigned int channel_max; + unsigned int channel_maxs[] = { + 6, /* rx */ + 8 /* tx */ + }; if (tx) fmt = pcm3168a->dac_fmt; @@ -521,18 +525,9 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, channel_max = 2; break; case PCM3168A_FMT_LEFT_J: - sample_min = 24; - if (tx) - channel_max = 8; - else - channel_max = 6; - break; case PCM3168A_FMT_I2S: sample_min = 24; - if (tx) - channel_max = 8; - else - channel_max = 6; + channel_max = channel_maxs[tx]; break; default: sample_min = 24; -- 2.7.4

2 1

[alsa-devel] [PATCH 0/3] ASoC: rsnd: prepare patches for BUSIF support
by Kuninori Morimoto 05 Nov '18

05 Nov '18

Hi Mark These are prepare/cleanup for for Renesas rsnd driver BUSIF support. I will post BUSIF support patch soon, but it needs Jiada Wang <jiada_wang(a)mentor.com> 's check. If it was OK, I will focus to TDM split mode. Kuninori Morimoto (3): ASoC: rsnd: move .get_status under rsnd_mod_ops ASoC: rsnd: add .get_id/.get_id_sub ASoC: rsnd: rsnd_mod_name() handles both name and ID sound/soc/sh/rcar/adg.c | 2 +- sound/soc/sh/rcar/cmd.c | 11 ++-- sound/soc/sh/rcar/core.c | 72 +++++++++++++++++------ sound/soc/sh/rcar/ctu.c | 23 +++++++- sound/soc/sh/rcar/dma.c | 47 +++++++-------- sound/soc/sh/rcar/dvc.c | 3 +- sound/soc/sh/rcar/gen.c | 12 ++-- sound/soc/sh/rcar/mix.c | 3 +- sound/soc/sh/rcar/rsnd.h | 21 +++---- sound/soc/sh/rcar/src.c | 29 +++++----- sound/soc/sh/rcar/ssi.c | 147 +++++++++++++++++++++++------------------------ sound/soc/sh/rcar/ssiu.c | 37 ++++++------ 12 files changed, 232 insertions(+), 175 deletions(-) -- 2.7.4 Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: intel: cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0
by Hans de Goede 05 Nov '18

05 Nov '18

Some boards such as the Swanky model Chromebooks use pmc_plt_clk_0 for the mclk instead of pmc_plt_clk_3. This commit adds a DMI based quirk for this. This fixing audio no longer working on these devices after commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") that commit fixes us unnecessary keeping unused clocks on, but in case of the Swanky that was breaking audio support since we were not using the right clock in the cht_bsw_max98090_ti machine driver. Cc: stable(a)vger.kernel.org Fixes: 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") Reported-and-tested-by: Dean Wallace <duffydack73(a)gmail.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 32 ++++++++++++++++++-- 1 file changed, 29 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index db6976f4ddaa..9d9f6e41d81c 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -19,6 +19,7 @@ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ +#include <linux/dmi.h> #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> @@ -35,6 +36,8 @@ #define CHT_PLAT_CLK_3_HZ 19200000 #define CHT_CODEC_DAI "HiFi" +#define QUIRK_PMC_PLT_CLK_0 0x01 + struct cht_mc_private { struct clk *mclk; struct snd_soc_jack jack; @@ -385,11 +388,29 @@ static struct snd_soc_card snd_soc_card_cht = { .num_controls = ARRAY_SIZE(cht_mc_controls), }; +static const struct dmi_system_id cht_max98090_quirk_table[] = { + { + /* Swanky model Chromebook (Toshiba Chromebook 2) */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Swanky"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + {} +}; + static int snd_cht_mc_probe(struct platform_device *pdev) { + const struct dmi_system_id *dmi_id; struct device *dev = &pdev->dev; int ret_val = 0; struct cht_mc_private *drv; + const char *mclk_name; + int quirks = 0; + + dmi_id = dmi_first_match(cht_max98090_quirk_table); + if (dmi_id) + quirks = (unsigned long)dmi_id->driver_data; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) @@ -411,11 +432,16 @@ static int snd_cht_mc_probe(struct platform_device *pdev) snd_soc_card_cht.dev = &pdev->dev; snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); - drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (quirks & QUIRK_PMC_PLT_CLK_0) + mclk_name = "pmc_plt_clk_0"; + else + mclk_name = "pmc_plt_clk_3"; + + drv->mclk = devm_clk_get(&pdev->dev, mclk_name); if (IS_ERR(drv->mclk)) { dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(drv->mclk)); + "Failed to get MCLK from %s: %ld\n", + mclk_name, PTR_ERR(drv->mclk)); return PTR_ERR(drv->mclk); } -- 2.19.0

2 1

[alsa-devel] [PATCH] ASoC: qdsp6: q6afe: Fix wrong MI2S SD line mask
by Rohit kumar 05 Nov '18

05 Nov '18

SD line mask for MI2S starts from BIT 0 instead of BIT 1. Fix all bit mask for MI2S SD lines. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- sound/soc/qcom/qdsp6/q6afe.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 000775b..829b5e987 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -49,14 +49,14 @@ #define AFE_PORT_I2S_SD1 0x2 #define AFE_PORT_I2S_SD2 0x3 #define AFE_PORT_I2S_SD3 0x4 -#define AFE_PORT_I2S_SD0_MASK BIT(0x1) -#define AFE_PORT_I2S_SD1_MASK BIT(0x2) -#define AFE_PORT_I2S_SD2_MASK BIT(0x3) -#define AFE_PORT_I2S_SD3_MASK BIT(0x4) -#define AFE_PORT_I2S_SD0_1_MASK GENMASK(2, 1) -#define AFE_PORT_I2S_SD2_3_MASK GENMASK(4, 3) -#define AFE_PORT_I2S_SD0_1_2_MASK GENMASK(3, 1) -#define AFE_PORT_I2S_SD0_1_2_3_MASK GENMASK(4, 1) +#define AFE_PORT_I2S_SD0_MASK BIT(0x0) +#define AFE_PORT_I2S_SD1_MASK BIT(0x1) +#define AFE_PORT_I2S_SD2_MASK BIT(0x2) +#define AFE_PORT_I2S_SD3_MASK BIT(0x3) +#define AFE_PORT_I2S_SD0_1_MASK GENMASK(1, 0) +#define AFE_PORT_I2S_SD2_3_MASK GENMASK(3, 2) +#define AFE_PORT_I2S_SD0_1_2_MASK GENMASK(2, 0) +#define AFE_PORT_I2S_SD0_1_2_3_MASK GENMASK(3, 0) #define AFE_PORT_I2S_QUAD01 0x5 #define AFE_PORT_I2S_QUAD23 0x6 #define AFE_PORT_I2S_6CHS 0x7 -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 2

[alsa-devel] [PATCH] ALSA: usb-audio: Define registers for CM6206
by Linus Walleij 05 Nov '18

05 Nov '18

The register map for CM6206 is known and we can define what the values written to the different registers actually mean. I tested to print the hex values before/after this change, there is no functional change. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- sound/usb/quirks.c | 129 +++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 124 insertions(+), 5 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 8a945ece9869..3d0f09108c98 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -19,6 +19,7 @@ #include <linux/usb.h> #include <linux/usb/audio.h> #include <linux/usb/midi.h> +#include <linux/bits.h> #include <sound/control.h> #include <sound/core.h> @@ -668,15 +669,133 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev) } /* - * C-Media CM6206 is based on CM106 with two additional - * registers that are not documented in the data sheet. - * Values here are chosen based on sniffing USB traffic - * under Windows. + * CM6206 registers from the CM6206 datasheet rev 2.1 */ +#define CM6206_REG0_DMA_MASTER BIT(15) +#define CM6206_REG0_SPDIFO_RATE_48K (2 << 12) +#define CM6206_REG0_SPDIFO_RATE_96K (7 << 12) +/* Bit 4 thru 11 is the S/PDIF category code */ +#define CM6206_REG0_SPDIFO_CAT_CODE_GENERAL (0 << 4) +#define CM6206_REG0_SPDIFO_EMPHASIS_CD BIT(3) +#define CM6206_REG0_SPDIFO_COPYRIGHT_NA BIT(2) +#define CM6206_REG0_SPDIFO_NON_AUDIO BIT(1) +#define CM6206_REG0_SPDIFO_PRO_FORMAT BIT(0) + +#define CM6206_REG1_TEST_SEL_CLK BIT(14) +#define CM6206_REG1_PLLBIN_EN BIT(13) +#define CM6206_REG1_SOFT_MUTE_EN BIT(12) +#define CM6206_REG1_GPIO4_OUT BIT(11) +#define CM6206_REG1_GPIO4_OE BIT(10) +#define CM6206_REG1_GPIO3_OUT BIT(9) +#define CM6206_REG1_GPIO3_OE BIT(8) +#define CM6206_REG1_GPIO2_OUT BIT(7) +#define CM6206_REG1_GPIO2_OE BIT(6) +#define CM6206_REG1_GPIO1_OUT BIT(5) +#define CM6206_REG1_GPIO1_OE BIT(4) +#define CM6206_REG1_SPDIFO_INVALID BIT(3) +#define CM6206_REG1_SPDIF_LOOP_EN BIT(2) +#define CM6206_REG1_SPDIFO_DIS BIT(1) +#define CM6206_REG1_SPDIFI_MIX BIT(0) + +#define CM6206_REG2_DRIVER_ON BIT(15) +#define CM6206_REG2_HEADP_SEL_SIDE_CHANNELS (0 << 13) +#define CM6206_REG2_HEADP_SEL_SURROUND_CHANNELS (1 << 13) +#define CM6206_REG2_HEADP_SEL_CENTER_SUBW (2 << 13) +#define CM6206_REG2_HEADP_SEL_FRONT_CHANNELS (3 << 13) +#define CM6206_REG2_MUTE_HEADPHONE_RIGHT BIT(12) +#define CM6206_REG2_MUTE_HEADPHONE_LEFT BIT(11) +#define CM6206_REG2_MUTE_REAR_SURROUND_RIGHT BIT(10) +#define CM6206_REG2_MUTE_REAR_SURROUND_LEFT BIT(9) +#define CM6206_REG2_MUTE_SIDE_SURROUND_RIGHT BIT(8) +#define CM6206_REG2_MUTE_SIDE_SURROUND_LEFT BIT(7) +#define CM6206_REG2_MUTE_SUBWOOFER BIT(6) +#define CM6206_REG2_MUTE_CENTER BIT(5) +#define CM6206_REG2_MUTE_RIGHT_FRONT BIT(3) +#define CM6206_REG2_MUTE_LEFT_FRONT BIT(3) +#define CM6206_REG2_EN_BTL BIT(2) +#define CM6206_REG2_MCUCLKSEL_1_5_MHZ (0) +#define CM6206_REG2_MCUCLKSEL_3_MHZ (1) +#define CM6206_REG2_MCUCLKSEL_6_MHZ (2) +#define CM6206_REG2_MCUCLKSEL_12_MHZ (3) + +/* Bit 11..13 sets the sensitivity to FLY tuner volume control VP/VD signal */ +#define CM6206_REG3_FLYSPEED_DEFAULT (2 << 11) +#define CM6206_REG3_VRAP25EN BIT(10) +#define CM6206_REG3_MSEL1 BIT(9) +#define CM6206_REG3_SPDIFI_RATE_44_1K BIT(0 << 7) +#define CM6206_REG3_SPDIFI_RATE_48K BIT(2 << 7) +#define CM6206_REG3_SPDIFI_RATE_32K BIT(3 << 7) +#define CM6206_REG3_PINSEL BIT(6) +#define CM6206_REG3_FOE BIT(5) +#define CM6206_REG3_ROE BIT(4) +#define CM6206_REG3_CBOE BIT(3) +#define CM6206_REG3_LOSE BIT(2) +#define CM6206_REG3_HPOE BIT(1) +#define CM6206_REG3_SPDIFI_CANREC BIT(0) + +#define CM6206_REG5_DA_RSTN BIT(13) +#define CM6206_REG5_AD_RSTN BIT(12) +#define CM6206_REG5_SPDIFO_AD2SPDO BIT(12) +#define CM6206_REG5_SPDIFO_SEL_FRONT (0 << 9) +#define CM6206_REG5_SPDIFO_SEL_SIDE_SUR (1 << 9) +#define CM6206_REG5_SPDIFO_SEL_CEN_LFE (2 << 9) +#define CM6206_REG5_SPDIFO_SEL_REAR_SUR (3 << 9) +#define CM6206_REG5_CODECM BIT(8) +#define CM6206_REG5_EN_HPF BIT(7) +#define CM6206_REG5_T_SEL_DSDA4 BIT(6) +#define CM6206_REG5_T_SEL_DSDA3 BIT(5) +#define CM6206_REG5_T_SEL_DSDA2 BIT(4) +#define CM6206_REG5_T_SEL_DSDA1 BIT(3) +#define CM6206_REG5_T_SEL_DSDAD_NORMAL 0 +#define CM6206_REG5_T_SEL_DSDAD_FRONT 4 +#define CM6206_REG5_T_SEL_DSDAD_S_SURROUND 5 +#define CM6206_REG5_T_SEL_DSDAD_CEN_LFE 6 +#define CM6206_REG5_T_SEL_DSDAD_R_SURROUND 7 + static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) { int err = 0, reg; - int val[] = {0x2004, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000}; + int val[] = { + /* + * Values here are chosen based on sniffing USB traffic + * under Windows. + * + * REG0: DAC is master, sample rate 48kHz, no copyright + */ + CM6206_REG0_SPDIFO_RATE_48K | + CM6206_REG0_SPDIFO_COPYRIGHT_NA, + /* + * REG1: PLL binary search enable, soft mute enable. + */ + CM6206_REG1_PLLBIN_EN | + CM6206_REG1_SOFT_MUTE_EN | + /* + * REG2: enable output drivers, + * select front channels to the headphone output, + * then mute the headphone channels, run the MCU + * at 1.5 MHz. + */ + CM6206_REG2_DRIVER_ON | + CM6206_REG2_HEADP_SEL_FRONT_CHANNELS | + CM6206_REG2_MUTE_HEADPHONE_RIGHT | + CM6206_REG2_MUTE_HEADPHONE_LEFT, + /* + * REG3: default flyspeed, set 2.5V mic bias + * enable all line out ports and enable SPDIF + */ + CM6206_REG3_FLYSPEED_DEFAULT | + CM6206_REG3_VRAP25EN | + CM6206_REG3_FOE | + CM6206_REG3_ROE | + CM6206_REG3_CBOE | + CM6206_REG3_LOSE | + CM6206_REG3_HPOE | + CM6206_REG3_SPDIFI_CANREC, + /* REG4 is just a bunch of GPIO lines */ + 0x0000, + /* REG5: de-assert AD/DA reset signals */ + CM6206_REG5_DA_RSTN | + CM6206_REG5_AD_RSTN }; for (reg = 0; reg < ARRAY_SIZE(val); reg++) { err = snd_usb_cm106_write_int_reg(dev, reg, val[reg]); -- 2.19.1

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Re: [alsa-devel] [PATCH] ASoC: wm8996: fix small typo
by Mark Brown 05 Nov '18

05 Nov '18

On Fri, Nov 02, 2018 at 09:30:06AM +0800, sh liu wrote: > From e9b923690675ca8fa883fd25dcead5b457856735 Mon Sep 17 00:00:00 2001 > From: liush <liush(a)allwinnertech.com> > Date: Fri, 2 Nov 2018 08:57:00 +0800 > Subject: [PATCH] ASoC: wm8996: fix small typo > > atleast -> at least > > Change-Id: Icc970b438166daef13518b7d1a62b13eb8752f5f > Signed-off-by: liush <liush(a)allwinnertech.com> This patch is corrupted and can't be applied: > *codec, int fll_id, int source, > if (i2c->irq) > timeout *= 10; > else > - /* ensure timeout of atleast 1 jiffies */ > + /* ensure timeout of at least 1 jiffies */ > timeout = timeout/2 ? : 1; > It looks like tabs have been replaced by spaces in whatever you're using to send it.

1 0

[alsa-devel] [PATCH] SoC: nau8825: remove unnecessary unlikely()
by Yangtao Li 04 Nov '18

04 Nov '18

WARN_ON() already contains an unlikely(), so it's not necessary to use unlikely. Signed-off-by: Yangtao Li <tiny.windzz(a)gmail.com> --- sound/soc/codecs/nau8825.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index b9fed99d8b5e..7bbcbf5f05c8 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -424,10 +424,8 @@ static u32 nau8825_xtalk_sidetone(u32 sig_org, u32 sig_cros) { u32 gain, sidetone; - if (unlikely(sig_org == 0) || unlikely(sig_cros == 0)) { - WARN_ON(1); + if (WARN_ON(sig_org == 0 || sig_cros == 0)) return 0; - } sig_org = nau8825_intlog10_dec3(sig_org); sig_cros = nau8825_intlog10_dec3(sig_cros); -- 2.17.0

1 0

[alsa-devel] Trying to get Roland TD-30 soundcard to work on Linux
by Cupertino Miranda 02 Nov '18

02 Nov '18

Hi, I have the Rolands TD-30 edrum module that includes a USB sound card in it. The card lists but no capture or recording devices work, although they are listed. I am new to audio device drivers and for that reason need some guidance to contribute to such driver. A full log can be found at: www.alsa-project.org/db/?f=e1a28cbd7e9f1201d3b0cbe44d8ec1908272a1d4 Looking forward to ear from you. Best regards, Cupertino Miranda

1 0

[alsa-devel] [PATCH 1/2] ALSA: x86: Fix runtime PM for hdmi-lpe-audio
by Ville Syrjala 02 Nov '18

02 Nov '18

From: Ville Syrjälä <ville.syrjala(a)linux.intel.com> Commit 46e831abe864 ("drm/i915/lpe: Mark LPE audio runtime pm as "no callbacks"") broke runtime PM with lpe audio. We can no longer runtime suspend the GPU since the sysfs power/control for the lpe-audio device no longer exists and the device is considered always active. We can fix this by not marking the device as active. Cc: Chris Wilson <chris(a)chris-wilson.co.uk> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Fixes: 46e831abe864 ("drm/i915/lpe: Mark LPE audio runtime pm as "no callbacks"") Signed-off-by: Ville Syrjälä <ville.syrjala(a)linux.intel.com> --- sound/x86/intel_hdmi_audio.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 83d76c345940..bbed4acaafd1 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1877,7 +1877,6 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) pm_runtime_use_autosuspend(&pdev->dev); pm_runtime_mark_last_busy(&pdev->dev); - pm_runtime_set_active(&pdev->dev); dev_dbg(&pdev->dev, "%s: handle pending notification\n", __func__); for_each_port(card_ctx, port) { -- 2.18.1

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[alsa-devel] Pull request testing (please ignore)
by GitHub pull_request-opened 02 Nov '18

02 Nov '18

alsa-project/alsa-lib pull request #1 was opened from perexg: Comment body Issue URL : https://github.com/alsa-project/alsa-lib/pull/1 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] [PATCH] ASoC: wm97xx: fix uninitialized regmap pointer problem
by Arnd Bergmann 02 Nov '18

02 Nov '18

gcc notices that without either the ac97 bus or the pdata, we never initialize the regmap pointer, which leads to an uninitialized variable access: sound/soc/codecs/wm9712.c: In function 'wm9712_soc_probe': sound/soc/codecs/wm9712.c:666:2: error: 'regmap' may be used uninitialized in this function [-Werror=maybe-uninitialized] Since that configuration is invalid, it's better to return an error here. I tried to avoid adding complexity to the conditions, and turned the #ifdef into a regular if(IS_ENABLED()) check for readability. This in turn requires moving some header file declarations out of an #ifdef. The same code is used in three drivers, all of which I'm changing the same way. Fixes: 2ed1a8e0ce8d ("ASoC: wm9712: add ac97 new bus support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- include/sound/soc.h | 2 +- sound/soc/codecs/wm9705.c | 10 ++++------ sound/soc/codecs/wm9712.c | 10 ++++------ sound/soc/codecs/wm9713.c | 10 ++++------ 4 files changed, 13 insertions(+), 19 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 70c10a8f3e90..3e0ac310a3df 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -553,12 +553,12 @@ static inline void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, } #endif -#ifdef CONFIG_SND_SOC_AC97_BUS struct snd_ac97 *snd_soc_alloc_ac97_component(struct snd_soc_component *component); struct snd_ac97 *snd_soc_new_ac97_component(struct snd_soc_component *component, unsigned int id, unsigned int id_mask); void snd_soc_free_ac97_component(struct snd_ac97 *ac97); +#ifdef CONFIG_SND_SOC_AC97_BUS int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, struct platform_device *pdev); diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index ccdf088461b7..54c306707c02 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -325,8 +325,7 @@ static int wm9705_soc_probe(struct snd_soc_component *component) if (wm9705->mfd_pdata) { wm9705->ac97 = wm9705->mfd_pdata->ac97; regmap = wm9705->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { wm9705->ac97 = snd_soc_new_ac97_component(component, WM9705_VENDOR_ID, WM9705_VENDOR_ID_MASK); if (IS_ERR(wm9705->ac97)) { @@ -339,7 +338,8 @@ static int wm9705_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9705->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_set_drvdata(component, wm9705->ac97); @@ -350,14 +350,12 @@ static int wm9705_soc_probe(struct snd_soc_component *component) static void wm9705_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9705_priv *wm9705 = snd_soc_component_get_drvdata(component); - if (!wm9705->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9705->mfd_pdata) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9705->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9705 = { diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index e873baa9e778..01949eaba4fd 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -642,8 +642,7 @@ static int wm9712_soc_probe(struct snd_soc_component *component) if (wm9712->mfd_pdata) { wm9712->ac97 = wm9712->mfd_pdata->ac97; regmap = wm9712->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { int ret; wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID, @@ -660,7 +659,8 @@ static int wm9712_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9712->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_init_regmap(component, regmap); @@ -673,14 +673,12 @@ static int wm9712_soc_probe(struct snd_soc_component *component) static void wm9712_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component); - if (!wm9712->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9712->mfd_pdata) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9712->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9712 = { diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 643863bb32e0..b2bb465c8e5a 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1214,8 +1214,7 @@ static int wm9713_soc_probe(struct snd_soc_component *component) if (wm9713->mfd_pdata) { wm9713->ac97 = wm9713->mfd_pdata->ac97; regmap = wm9713->mfd_pdata->regmap; - } else { -#ifdef CONFIG_SND_SOC_AC97_BUS + } else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { wm9713->ac97 = snd_soc_new_ac97_component(component, WM9713_VENDOR_ID, WM9713_VENDOR_ID_MASK); if (IS_ERR(wm9713->ac97)) @@ -1225,7 +1224,8 @@ static int wm9713_soc_probe(struct snd_soc_component *component) snd_soc_free_ac97_component(wm9713->ac97); return PTR_ERR(regmap); } -#endif + } else { + return -ENXIO; } snd_soc_component_init_regmap(component, regmap); @@ -1238,14 +1238,12 @@ static int wm9713_soc_probe(struct snd_soc_component *component) static void wm9713_soc_remove(struct snd_soc_component *component) { -#ifdef CONFIG_SND_SOC_AC97_BUS struct wm9713_priv *wm9713 = snd_soc_component_get_drvdata(component); - if (!wm9713->mfd_pdata) { + if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) { snd_soc_component_exit_regmap(component); snd_soc_free_ac97_component(wm9713->ac97); } -#endif } static const struct snd_soc_component_driver soc_component_dev_wm9713 = { -- 2.18.0

2 2

[alsa-devel] [PATCH v2 0/1] Introduce dmic mode switch delay parameter
by Jenny TC 02 Nov '18

02 Nov '18

Some DMICs need clock to be running for a specified duration as per the DMIC spec to complete the mode transition like sleep to normal, off to normal etc. v1: First patch set v2: 1). Removed ACPI device entry since a) The DMIC vendor doesn't have an ACPI vendor ID b) The ACPI enumeration creates one more device and conflicts with the existing skl.c dmic platform device registration. 2). Updated commit message Jenny TC (1): ASoC: dmic: Introduce mode switch delay sound/soc/codecs/dmic.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) -- 1.9.1

3 3

[alsa-devel] [PATCH alsa-utils] topology/topology.c: drop unneeded <dlfcn.h> include
by Thomas Petazzoni 02 Nov '18

02 Nov '18

This include is not used/needed and prevents building on systems that don't provide <dlfcn.h>. Signed-off-by: Thomas Petazzoni <thomas.petazzoni(a)bootlin.com> --- topology/topology.c | 1 - 1 file changed, 1 deletion(-) diff --git a/topology/topology.c b/topology/topology.c index 097c255..b72e341 100644 --- a/topology/topology.c +++ b/topology/topology.c @@ -27,7 +27,6 @@ #include <errno.h> #include <string.h> #include <sys/stat.h> -#include <dlfcn.h> #include <getopt.h> #include <assert.h> -- 2.14.4

2 1

[alsa-devel] [PATCH alsa-lib] utils/alsa.m4: conditionally enable libdl in AM_PATH_ALSA m4 macro
by Thomas Petazzoni 02 Nov '18

02 Nov '18

From: Thomas De Schampheleire <thomas.de.schampheleire(a)gmail.com> The AM_PATH_ALSA macro in utils/alsa.m4 unconditionally uses -ldl. This breaks compilation of alsa-utils (and probably other packages using this macro) for targets that do not support dynamic loading. This patch updates the macro to check if dlopen is available, and use that result to conditionally add -ldl to the list of libraries. Signed-off-by: Thomas De Schampheleire <thomas.de.schampheleire(a)gmail.com> Signed-off-by: Jörg Krause <joerg.krause(a)embedded.rocks> Signed-off-by: Thomas Petazzoni <thomas.petazzoni(a)bootlin.com> --- utils/alsa.m4 | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/utils/alsa.m4 b/utils/alsa.m4 index e12310df..a5c5a292 100644 --- a/utils/alsa.m4 +++ b/utils/alsa.m4 @@ -44,6 +44,8 @@ if test "$alsa_inc_prefix" != "" ; then fi AC_MSG_RESULT($ALSA_CFLAGS) +AC_CHECK_LIB(c, dlopen, LIBDL="", [AC_CHECK_LIB(dl, dlopen, LIBDL="-ldl")]) + dnl add any special lib dirs AC_MSG_CHECKING(for ALSA LDFLAGS) if test "$alsa_prefix" != "" ; then @@ -52,7 +54,7 @@ if test "$alsa_prefix" != "" ; then fi dnl add the alsa library -ALSA_LIBS="$ALSA_LIBS -lasound -lm -ldl -lpthread" +ALSA_LIBS="$ALSA_LIBS -lasound -lm $LIBDL -lpthread" LIBS="$ALSA_LIBS $LIBS" AC_MSG_RESULT($ALSA_LIBS) -- 2.14.4

2 1

Re: [alsa-devel] [PATCH] ASoC: wm8996: fix small typo
by Charles Keepax 02 Nov '18

02 Nov '18

On Fri, Nov 02, 2018 at 09:30:06AM +0800, sh liu wrote: > >From e9b923690675ca8fa883fd25dcead5b457856735 Mon Sep 17 00:00:00 2001 > From: liush <liush(a)allwinnertech.com> > Date: Fri, 2 Nov 2018 08:57:00 +0800 > Subject: [PATCH] ASoC: wm8996: fix small typo > > atleast -> at least > > Change-Id: Icc970b438166daef13518b7d1a62b13eb8752f5f > Signed-off-by: liush <liush(a)allwinnertech.com> > --- Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Thanks, Charles

1 0

[alsa-devel] Bug report - Dell-WD15-Dock not working caused by ucm profile in alsa-lib 1.1.7
by René Santing 02 Nov '18

02 Nov '18

Hello Alsa developers, I would like to file a bug report about the Dell-WD15-Dock. This dock stopped working for me on archlinux/kde. The device was not picked up at all by pulse audio after some updates I installed last week. By trial and error I found out it was caused by the ucm profile that was added in alsa-lib 1.1.7. Running the following restored functionality. sudo mv /usr/share/alsa/ucm/Dell-WD15-Dock/Dell-WD15-Dock.conf /usr/share/alsa/ucm/Dell-WD15-Dock/Dell-WD15-Dock.bak pulseaudio -k I uploaded 3 alsa-info.sh profiles. 1.1.6 http://www.alsa-project.org/db/?f=a9a4c53d1e417c3d2ef304fe0778581c35b7a95e (working) 1.1.7 http://www.alsa-project.org/db/?f=bb474a687692e25f5e54c475f16f7ce4ef210f7a (not working) 1.1.7 http://www.alsa-project.org/db/?f=586dd059aaa08e839eb9df5f7b49990a5f4ad1a5 (working but moved the wd15 ucm profile) https://alsa-project.org/main/index.php/Bug_Tracking pointed me to this mailing list to submit this report. If this is the wrong place or there is more information need could you let me know so I can send it to the proper channels? Best regards, René Santing

2 9

[alsa-devel] [PATCH alsa-lib] Don't use fork() on noMMU platforms
by Thomas Petazzoni 02 Nov '18

02 Nov '18

fork() is not available on noMMU platforms, but in the specific case of pcm_direct.c, vfork() can be used instead. This commit adds an autoconf check for fork(), and falls back to vfork() is fork() is not available. Signed-off-by: Thomas Petazzoni <thomas.petazzoni(a)bootlin.com> --- configure.ac | 2 ++ src/pcm/pcm_direct.c | 10 +++++++++- 2 files changed, 11 insertions(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 4c9d860f..ab659490 100644 --- a/configure.ac +++ b/configure.ac @@ -51,6 +51,8 @@ dnl Checks for library functions. AC_PROG_GCC_TRADITIONAL AC_CHECK_FUNCS([uselocale]) +AC_CHECK_FUNC([fork]) + SAVE_LIBRARY_VERSION AC_SUBST(LIBTOOL_VERSION_INFO) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 2b07eff9..4dc3ea26 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -431,13 +431,21 @@ int snd_pcm_direct_server_create(snd_pcm_direct_t *dmix) close(dmix->server_fd); return ret; } - + +#ifdef HAVE_FORK ret = fork(); +#else + ret = vfork(); +#endif if (ret < 0) { close(dmix->server_fd); return ret; } else if (ret == 0) { +#ifdef HAVE_FORK ret = fork(); +#else + ret = vfork(); +#endif if (ret == 0) server_job(dmix); _exit(EXIT_SUCCESS); -- 2.14.4

2 1

[alsa-devel] [PATCH 1/2] CHROMIUM: ASoC: rt5663: Add documentation for power supply support
by Cheng-Yi Chiang 02 Nov '18

02 Nov '18

rt5663 codec driver will support setting CPVDD and AVDD power supply from device tree. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Documentation/devicetree/bindings/sound/rt5663.txt | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5663.txt b/Documentation/devicetree/bindings/sound/rt5663.txt index 23386446c63d6..d4058dfde0392 100644 --- a/Documentation/devicetree/bindings/sound/rt5663.txt +++ b/Documentation/devicetree/bindings/sound/rt5663.txt @@ -36,6 +36,9 @@ Optional properties: "realtek,impedance_sensing_num" is 2. It means that there are 2 ranges of impedance in the impedance sensing function. +- avdd-supply: Power supply for AVDD, providing 1.8V. +- cpvdd-supply: Power supply for CPVDD, providing 3.5V. + Pins on the device (for linking into audio routes) for RT5663: * IN1P @@ -51,4 +54,6 @@ rt5663: codec@12 { compatible = "realtek,rt5663"; reg = <0x12>; interrupts = <7 IRQ_TYPE_EDGE_FALLING>; + avdd-supply = <&pp1800_a_alc5662>; + cpvdd-supply = <&pp3500_a_alc5662>; }; -- 2.19.1.568.g152ad8e336-goog

1 2

[alsa-devel] [PATCH] ASoC: wm8996: fix small typo
by lshua312 02 Nov '18

02 Nov '18

From e9b923690675ca8fa883fd25dcead5b457856735 Mon Sep 17 00:00:00 2001 From: liush <liush(a)allwinnertech.com> Date: Fri, 2 Nov 2018 08:57:00 +0800 Subject: [PATCH] ASoC: wm8996: fix small typo atleast -> at least Change-Id: Icc970b438166daef13518b7d1a62b13eb8752f5f Signed-off-by: liush <liush(a)allwinnertech.com> --- sound/soc/codecs/wm8996.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 8affa49..d039835 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2109,7 +2109,7 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, if (i2c->irq) timeout *= 10; else - /* ensure timeout of atleast 1 jiffies */ + /* ensure timeout of at least 1 jiffies */ timeout = timeout/2 ? : 1; for (retry = 0; retry < 10; retry++) { -- 1.9.1

1 0

[alsa-devel] [PATCH 0/7] ASoC: Intel: cleanups
by Pierre-Louis Bossart 01 Nov '18

01 Nov '18

This patchset contain 4 different cleanups: a) change of pdata definition to avoid using SST-specific structures and allow for machine driver reuse w/ SOF. No new features, just different data structures. b) removal of remaining GFP_ATOMIC usages c) Cleanup of machine driver tables for APL (missing SOF information and quirk for APL RVP/LeafHill CRB boards). These boards are used outside of Intel so it's worth sharing the fixes for the general public. d) typo Bard liao (1): ASoC: Intel: common: add SOF information for APL RVP Pierre-Louis Bossart (6): ASoC: acpi: define common interface for machine driver configuration ASoC: Intel: use standard interface for Hdaudio machine driver ASoC: Intel: use standard interface for Atom machine drivers ASoC: Intel: boards: fix Skylake typo ASoC: Intel: remove GFP_ATOMIC, use GFP_KERNEL ASoC: Intel: common: add quirk for APL RVP boards include/sound/soc-acpi.h | 14 +++++++ sound/soc/intel/atom/sst/sst_acpi.c | 4 ++ sound/soc/intel/atom/sst/sst_pvt.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +-- sound/soc/intel/boards/bytcr_rt5651.c | 6 +-- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +-- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 ++-- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 22 +++++------ .../soc/intel/boards/skl_nau88l25_max98357a.c | 4 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 4 +- .../intel/common/soc-acpi-intel-bxt-match.c | 38 +++++++++++++++++++ sound/soc/intel/skylake/skl.c | 10 ++--- 15 files changed, 87 insertions(+), 47 deletions(-) -- 2.17.1

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[alsa-devel] [USB-Audio - PX USB, playback] snd-usb-audio: probe of 2-3:1.0 failed with error -22 after kernel upgrade
by Mutz Heuer 01 Nov '18

01 Nov '18

USB-Audio Connection on Bowers & Wilkens PX Bluetooth Headphones started working when upgrading kernel from 4.13 to 4.15 series. Then it suddenly stopped working again on newer kernels. The bug report has been generated on the latest working mainline kernel 4.15.7. All newer kernels have some usb quirks in place, which should fix the playback sample rates to 48 kHz. The quirk seems not to be working correctly - it seems to hinder the audio system to recognize the USB-device as a sound device. The Headphones work on kernels before 4,15.7 when manually configuring pulseaudio to a default samplerate of 48 kHz. PX Headphones are on the newest firmware level. See https://patchwork.kernel.org/patch/10265345/. Dropping that patch and recompiling the kernel leads to a situation where alsa and pulseaudio seem to to be working again. Adding default-sample-rate = 48000 to /etc/pulse/daemon.conf leads to working headphones. Maybe the patch has to be adjusted to newer firmware? A quirk seems to be really necessary to avoid the specific patch to daemon.conf. More details can be found in https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1773697 What would be the best way to proceed?

2 4

[alsa-devel] [PATCH 3/3] ALSA:hda: add support for Huawei WMI MicMute LED
by Ayman Bagabas 01 Nov '18

01 Nov '18

Some of Huawei laptops come with a LED in the mic mute key. This patch enables and disable this LED when the internal microphone status is changed. Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> --- include/linux/huawei_wmi.h | 7 ++++ sound/pci/hda/huawei_wmi_helper.c | 66 +++++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 12 ++++++ 3 files changed, 85 insertions(+) create mode 100644 include/linux/huawei_wmi.h create mode 100644 sound/pci/hda/huawei_wmi_helper.c diff --git a/include/linux/huawei_wmi.h b/include/linux/huawei_wmi.h new file mode 100644 index 000000000000..69b656c5029b --- /dev/null +++ b/include/linux/huawei_wmi.h @@ -0,0 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __HUAWEI_WMI_H__ +#define __HUAWEI_WMI_H__ + +int huawei_wmi_micmute_led_set(bool on); + +#endif diff --git a/sound/pci/hda/huawei_wmi_helper.c b/sound/pci/hda/huawei_wmi_helper.c new file mode 100644 index 000000000000..57256f51fd88 --- /dev/null +++ b/sound/pci/hda/huawei_wmi_helper.c @@ -0,0 +1,66 @@ +// SPDX-License-Identifier: GPL-2.0 +/* Helper functions for Huawei WMI Mic Mute LED; + * to be included from codec driver + */ + +#if IS_ENABLED(CONFIG_HUAWEI_LAPTOP) +#include <linux/huawei_wmi.h> + +static int (*huawei_wmi_micmute_led_set_func)(bool); + +static void update_huawei_wmi_micmute_led(struct hda_codec *codec, + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (!ucontrol || !huawei_wmi_micmute_led_set_func) + return; + if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) { + /* TODO: How do I verify if it's a mono or stereo here? */ + bool val = ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]; + huawei_wmi_micmute_led_set_func(!val); + } +} + +static void alc_fixup_huawei_wmi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct hda_gen_spec *spec = codec->spec; + bool removefunc = false; + + codec_info(codec, "In alc_fixup_huawei_wmi\n"); + + if (action == HDA_FIXUP_ACT_PROBE) { + if (!huawei_wmi_micmute_led_set_func) + huawei_wmi_micmute_led_set_func = symbol_request(huawei_wmi_micmute_led_set); + if (!huawei_wmi_micmute_led_set_func) { + codec_warn(codec, "Failed to find huawei_wmi symbol huawei_wmi_micmute_led_set\n"); + return; + } + + removefunc = true; + if (huawei_wmi_micmute_led_set_func(false) >= 0) { + if (spec->num_adc_nids > 1 && !spec->dyn_adc_switch) + codec_dbg(codec, "Skipping micmute LED control due to several ADCs"); + else { + spec->cap_sync_hook = update_huawei_wmi_micmute_led; + removefunc = false; + } + } + codec_info(codec, "In alc_fixup_huawei_wmi IF\n"); + + } + + if (huawei_wmi_micmute_led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { + symbol_put(huawei_wmi_micmute_led_set); + huawei_wmi_micmute_led_set_func = NULL; + } +} + +#else + +static void hda_fixup_huawei_wmi(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ +} + +#endif diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4f7a39c7883c..0dab00c0df50 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5371,6 +5371,9 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, /* for dell wmi mic mute led */ #include "dell_wmi_helper.c" +/* for huawei wmi mic mute led */ +#include "huawei_wmi_helper.c" + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -5494,6 +5497,7 @@ enum { ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, ALC256_FIXUP_HUAWEI_MBXP_PINS, + ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, }; @@ -6348,6 +6352,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_huawei_wmi, + }, [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -6363,6 +6371,8 @@ static const struct hda_fixup alc269_fixups[] = { {0x21, 0x04211020}, { }, }, + .chained = true, + .chain_id = ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED }, [ALC295_FIXUP_HP_X360] = { .type = HDA_FIXUP_FUNC, @@ -6610,6 +6620,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511e, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), + SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ @@ -6775,6 +6786,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, .name = "alc274-dell-aio"}, {.id = ALC255_FIXUP_DUMMY_LINEOUT_VERB, .name = "alc255-dummy-lineout"}, {.id = ALC255_FIXUP_DELL_HEADSET_MIC, .name = "alc255-dell-headset"}, + {.id = ALC256_FIXUP_HUAWEI_WMI_MICMUTE_LED, .name = "alc256-huawei-micmute"}, {.id = ALC295_FIXUP_HP_X360, .name = "alc295-hp-x360"}, {} }; -- 2.17.2

2 2

[alsa-devel] [PATCH 2/3] ALSA:hda: fix front speakers on Huawei MBXP.
by Ayman Bagabas 01 Nov '18

01 Nov '18

This patch enables the front speakers on Huawei Matebook X Pro laptops. These laptops come with Dolby Atmos sound system and these pins configuration enables the front speakers. Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3ac7ba9b342d..4f7a39c7883c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5493,6 +5493,7 @@ enum { ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, + ALC256_FIXUP_HUAWEI_MBXP_PINS, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, }; @@ -6347,6 +6348,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x12, 0x90a60130}, + {0x13, 0x40000000}, + {0x14, 0x90170110}, + {0x18, 0x411111f0}, + {0x19, 0x04a11040}, + {0x1a, 0x411111f0}, + {0x1b, 0x90170112}, + {0x1d, 0x40759a05}, + {0x1e, 0x411111f0}, + {0x21, 0x04211020}, + { }, + }, + }, [ALC295_FIXUP_HP_X360] = { .type = HDA_FIXUP_FUNC, .v.func = alc295_fixup_hp_top_speakers, @@ -6592,6 +6609,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x511e, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), + SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- 2.17.2

3 2

[alsa-devel] Bug report - snd_hda_intel - snd_pcm_delay() returned a value that is exceptionally large -
by Sergey Popol 31 Oct '18

31 Oct '18

Hello, After a few hours my ubuntu workstation becomes unresponsive (requiring a Sysreq reisub to reboot): ================================================================= $ cat /var/log/syslog | grep 20:25 Oct 31 20:25:01 soulwax CRON[1587]: (root) CMD (command -v debian-sa1 > /dev/null && debian-sa1 1 1) Oct 31 20:25:28 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: snd_pcm_delay() returned a value that is exceptionally large: -417964 bytes (-2369 ms). Oct 31 20:25:28 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Oct 31 20:25:29 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event3 - Logitech USB Optical Mouse: SYN_DROPPED event - some input events have been lost. Oct 31 20:25:31 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: snd_pcm_dump(): Oct 31 20:25:32 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: Soft volume PCM Oct 31 20:25:34 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: Control: PCM Playback Volume Oct 31 20:25:34 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: min_dB: -51 Oct 31 20:25:35 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: max_dB: 0 Oct 31 20:25:36 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event4 - Logitech USB Keyboard: SYN_DROPPED event - some input events have been lost. Oct 31 20:25:36 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: resolution: 256 Oct 31 20:25:37 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: Its setup is: Oct 31 20:25:37 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: stream : PLAYBACK Oct 31 20:25:37 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (**) Option "fd" "21" Oct 31 20:25:37 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event2 - Power Button: device removed Oct 31 20:25:38 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: access : MMAP_INTERLEAVED Oct 31 20:25:38 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (**) Option "fd" "24" Oct 31 20:25:39 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event5 - Video Bus: device removed Oct 31 20:25:39 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (**) Option "fd" "25" Oct 31 20:25:39 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event1 - Power Button: device removed Oct 31 20:25:40 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (**) Option "fd" "26" Oct 31 20:25:40 soulwax kernel: [35165.869886] rfkill: input handler enabled Oct 31 20:25:40 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event0 - Sleep Button: device removed Oct 31 20:25:40 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (**) Option "fd" "27" Oct 31 20:25:40 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event3 - Logitech USB Optical Mouse: device removed Oct 31 20:25:41 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (**) Option "fd" "28" Oct 31 20:25:41 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event4 - Logitech USB Keyboard: device removed Oct 31 20:25:41 soulwax pulseaudio[1824]: [alsa-sink-ALC283 Analog] alsa-util.c: format : S16_LE Oct 31 20:25:42 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (**) Option "fd" "29" Oct 31 20:25:42 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event6 - Logitech USB Keyboard: device removed Oct 31 20:25:43 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (**) Option "fd" "30" Oct 31 20:25:54 soulwax /usr/lib/gdm3/gdm-x-session[1673]: (II) event7 - ITE8713 CIR transceiver: device removed ================================================================= http://www.alsa-project.org/db/?f=d5e5889f64b41a208f4dad2c1ac4b49a9b9ec558 Hope that can help. Sergei

1 0

[alsa-devel] [PATCH 3/3] ALSA:hda: add support for Huawei WMI MicMute LED
by ayman.bagabas＠gmail.com 31 Oct '18

31 Oct '18

1 0

[alsa-devel] [PATCH] ALSA:hda: fix front speakers on Huawei MBXP.
by ayman.bagabas＠gmail.com 31 Oct '18

31 Oct '18

1 0

[alsa-devel] [PATCH] ASoC: rsnd: fixup clock start checker
by Kuninori Morimoto 31 Oct '18

31 Oct '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 4d230d12710646 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") fixuped clock start timing. But it exchanged clock start checker from ssi->usrcnt to ssi->rate. Current rsnd_ssi_master_clk_start() is called from .prepare, but some player (for example GStreamer) might calls it many times. In such case, the checker might returns error even though it was not error. It should check ssi->usrcnt instead of ssi->rate. This patch fixup it. Without this patch, GStreamer can't switch 48kHz / 44.1kHz. Reported-by: Yusuke Goda <yusuke.goda.sx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Yusuke Goda <yusuke.goda.sx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 95acc8b..dc85fff 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -300,7 +300,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_ssi_is_multi_slave(mod, io)) return 0; - if (ssi->rate) { + if (ssi->usrcnt > 1) { if (ssi->rate != rate) { dev_err(dev, "SSI parent/child should use same rate\n"); return -EINVAL; -- 2.7.4

2 1

[alsa-devel] Bug in hdmi codec driver in DPCM
by Rohit Kumar 31 Oct '18

31 Oct '18

Hi, We are facing crash in hdmi codec driver when we use DPCM based dai-links. Crash is happening in hdmi_codec_eld_chmap() while accessing chmap_info struct during hdmi_codec_startup(). However, chmap_info struct is allocated in hdmi_codec_pcm_new() which is only called in non-DPCM based dai-links. Below is the crash stack: [61.635493] Unable to handle kernel NULL pointer dereference at virtual address 00000018 ..... [61.666696]CM = 0, WnR = 1 [61.669778] user pgtable: 4k pages, 39-bit VAs, pgd = ffffffc0d6633000 [61.676526] [0000000000000018] *pgd=0000000153fc8003, *pud=0000000153fc8003, *pmd=0000000000000000 [61.685793] Internal error: Oops: 96000046 [#1] PREEMPT SMP [61.722955] CPU: 7 PID: 2238 Comm: aplay Not tainted 4.14.72 #21 .. [61.740269] PC is at hdmi_codec_startup+0x124/0x164 [61.745308] LR is at hdmi_codec_startup+0xe4/0x164 .. We would probably need to move the implementation on pcm_new() to some other function to make it compatible with DPCM. Can you please provide suggestions in solving the issue. Thanks, Rohit

2 2

[alsa-devel] [PATCH] ASoC: stm32: sai: fix invalid use of sizeof in stm32_sai_add_mclk_provider()
by Wei Yongjun 30 Oct '18

30 Oct '18

sizeof() when applied to a pointer typed expression gives the size of the pointer, not that of the pointed data. Fixes: 8307b2afd386 ("ASoC: stm32: sai: set sai as mclk clock provider") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/stm/stm32_sai_sub.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index ea05cc9..211589b 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -390,7 +390,7 @@ static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai) char *mclk_name, *p, *s = (char *)pname; int ret, i = 0; - mclk = devm_kzalloc(dev, sizeof(mclk), GFP_KERNEL); + mclk = devm_kzalloc(dev, sizeof(*mclk), GFP_KERNEL); if (!mclk) return -ENOMEM;

2 1

[alsa-devel] [PATCH v2] ASoC: AMD: Fix race condition between register access
by Agrawal, Akshu 30 Oct '18

30 Oct '18

During simultaneous running of playback and capture, we got hit by incorrect value write on common register. This was due to race condition between 2 streams. Fixing this by locking the common register access. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: Added 2 helper functions, removed locking in ch enable/disable as they are separate registers. sound/soc/amd/acp-pcm-dma.c | 63 ++++++++++++++++++++++++++++++++++----------- 1 file changed, 48 insertions(+), 15 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 0ac4b5b..5be9c2d 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -121,6 +121,9 @@ .periods_max = CAPTURE_MAX_NUM_PERIODS, }; +/* Lock to protect access to registers */ +static DEFINE_SPINLOCK(lock); + static u32 acp_reg_read(void __iomem *acp_mmio, u32 reg) { return readl(acp_mmio + (reg * 4)); @@ -131,6 +134,33 @@ static void acp_reg_write(u32 val, void __iomem *acp_mmio, u32 reg) writel(val, acp_mmio + (reg * 4)); } +static void acp_reg_write_srbm_targ(void __iomem *acp_mmio, + u32 addr, u32 data) +{ + unsigned long flags; + + spin_lock_irqsave(&lock, flags); + acp_reg_write(addr, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); + acp_reg_write(data, acp_mmio, mmACP_SRBM_Targ_Idx_Data); + spin_unlock_irqrestore(&lock, flags); +} + +static void acp_reg_read_mod_write(void __iomem *acp_mmio, u32 addr, + u32 mask, bool set) +{ + u32 val; + unsigned long flags; + + spin_lock_irqsave(&lock, flags); + val = acp_reg_read(acp_mmio, addr); + if (set) + val |= mask; + else + val &= ~mask; + acp_reg_write(val, acp_mmio, addr); + spin_unlock_irqrestore(&lock, flags); +} + /* * Configure a given dma channel parameters - enable/disable, * number of descriptors, priority @@ -172,15 +202,12 @@ static void config_dma_descriptor_in_sram(void __iomem *acp_mmio, sram_offset = (descr_idx * sizeof(acp_dma_dscr_transfer_t)); /* program the source base address. */ - acp_reg_write(sram_offset, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); - acp_reg_write(descr_info->src, acp_mmio, mmACP_SRBM_Targ_Idx_Data); + acp_reg_write_srbm_targ(acp_mmio, sram_offset, descr_info->src); /* program the destination base address. */ - acp_reg_write(sram_offset + 4, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); - acp_reg_write(descr_info->dest, acp_mmio, mmACP_SRBM_Targ_Idx_Data); - + acp_reg_write_srbm_targ(acp_mmio, sram_offset + 4, descr_info->dest); /* program the number of bytes to be transferred for this descriptor. */ - acp_reg_write(sram_offset + 8, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); - acp_reg_write(descr_info->xfer_val, acp_mmio, mmACP_SRBM_Targ_Idx_Data); + acp_reg_write_srbm_targ(acp_mmio, sram_offset + 8, + descr_info->xfer_val); } static void pre_config_reset(void __iomem *acp_mmio, u16 ch_num) @@ -309,8 +336,12 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, u32 low; u32 high; u32 offset; + unsigned long flags; offset = ACP_DAGB_GRP_SRBM_SRAM_BASE_OFFSET + (pte_offset * 8); + + spin_lock_irqsave(&lock, flags); + for (page_idx = 0; page_idx < (num_of_pages); page_idx++) { /* Load the low address of page int ACP SRAM through SRBM */ acp_reg_write((offset + (page_idx * 8)), @@ -333,6 +364,8 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, /* Move to next physically contiguos page */ pg++; } + + spin_unlock_irqrestore(&lock, flags); } static void config_acp_dma(void __iomem *acp_mmio, @@ -870,29 +903,29 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, } } if (adata->asic_type == CHIP_STONEY) { - val = acp_reg_read(adata->acp_mmio, - mmACP_I2S_16BIT_RESOLUTION_EN); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: - val |= ACP_I2S_BT_16BIT_RESOLUTION_EN; + val = ACP_I2S_BT_16BIT_RESOLUTION_EN; break; case I2S_SP_INSTANCE: default: - val |= ACP_I2S_SP_16BIT_RESOLUTION_EN; + val = ACP_I2S_SP_16BIT_RESOLUTION_EN; } } else { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: - val |= ACP_I2S_BT_16BIT_RESOLUTION_EN; + val = ACP_I2S_BT_16BIT_RESOLUTION_EN; break; case I2S_SP_INSTANCE: default: - val |= ACP_I2S_MIC_16BIT_RESOLUTION_EN; + val = ACP_I2S_MIC_16BIT_RESOLUTION_EN; } } - acp_reg_write(val, adata->acp_mmio, - mmACP_I2S_16BIT_RESOLUTION_EN); + acp_reg_read_mod_write(adata->acp_mmio, + mmACP_I2S_16BIT_RESOLUTION_EN, + val, + true); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: AMD: Fix race condition between register access
by Agrawal, Akshu 30 Oct '18

30 Oct '18

During simultaneous running of playback and capture, we got hit by incorrect value write on common register. This was due to race condition between 2 streams. Fixing this by locking the common register access. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 0ac4b5b..993a7db 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -121,6 +121,9 @@ .periods_max = CAPTURE_MAX_NUM_PERIODS, }; +/* Lock to protect access to registers */ +static DEFINE_SPINLOCK(lock); + static u32 acp_reg_read(void __iomem *acp_mmio, u32 reg) { return readl(acp_mmio + (reg * 4)); @@ -168,9 +171,12 @@ static void config_dma_descriptor_in_sram(void __iomem *acp_mmio, acp_dma_dscr_transfer_t *descr_info) { u32 sram_offset; + unsigned long flags; sram_offset = (descr_idx * sizeof(acp_dma_dscr_transfer_t)); + spin_lock_irqsave(&lock, flags); + /* program the source base address. */ acp_reg_write(sram_offset, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); acp_reg_write(descr_info->src, acp_mmio, mmACP_SRBM_Targ_Idx_Data); @@ -181,6 +187,8 @@ static void config_dma_descriptor_in_sram(void __iomem *acp_mmio, /* program the number of bytes to be transferred for this descriptor. */ acp_reg_write(sram_offset + 8, acp_mmio, mmACP_SRBM_Targ_Idx_Addr); acp_reg_write(descr_info->xfer_val, acp_mmio, mmACP_SRBM_Targ_Idx_Data); + + spin_unlock_irqrestore(&lock, flags); } static void pre_config_reset(void __iomem *acp_mmio, u16 ch_num) @@ -309,8 +317,12 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, u32 low; u32 high; u32 offset; + unsigned long flags; offset = ACP_DAGB_GRP_SRBM_SRAM_BASE_OFFSET + (pte_offset * 8); + + spin_lock_irqsave(&lock, flags); + for (page_idx = 0; page_idx < (num_of_pages); page_idx++) { /* Load the low address of page int ACP SRAM through SRBM */ acp_reg_write((offset + (page_idx * 8)), @@ -333,6 +345,8 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, /* Move to next physically contiguos page */ pg++; } + + spin_unlock_irqrestore(&lock, flags); } static void config_acp_dma(void __iomem *acp_mmio, @@ -367,6 +381,7 @@ static void acp_dma_cap_channel_enable(void __iomem *acp_mmio, u16 cap_channel) { u32 val, ch_reg, imr_reg, res_reg; + unsigned long flags; switch (cap_channel) { case CAP_CHANNEL1: @@ -381,6 +396,8 @@ static void acp_dma_cap_channel_enable(void __iomem *acp_mmio, imr_reg = mmACP_I2SMICSP_IMR0; break; } + spin_lock_irqsave(&lock, flags); + val = acp_reg_read(acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); if (val & ACP_I2S_MIC_16BIT_RESOLUTION_EN) { @@ -393,12 +410,15 @@ static void acp_dma_cap_channel_enable(void __iomem *acp_mmio, val &= ~ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK; acp_reg_write(val, acp_mmio, imr_reg); acp_reg_write(0x1, acp_mmio, ch_reg); + + spin_unlock_irqrestore(&lock, flags); } static void acp_dma_cap_channel_disable(void __iomem *acp_mmio, u16 cap_channel) { u32 val, ch_reg, imr_reg; + unsigned long flags; switch (cap_channel) { case CAP_CHANNEL1: @@ -411,11 +431,15 @@ static void acp_dma_cap_channel_disable(void __iomem *acp_mmio, ch_reg = mmACP_I2SMICSP_RER0; break; } + spin_lock_irqsave(&lock, flags); + val = acp_reg_read(acp_mmio, imr_reg); val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXDAM_MASK; val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK; acp_reg_write(val, acp_mmio, imr_reg); acp_reg_write(0x0, acp_mmio, ch_reg); + + spin_unlock_irqrestore(&lock, flags); } /* Start a given DMA channel transfer */ @@ -847,6 +871,7 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, int status; uint64_t size; u32 val = 0; + unsigned long flags; struct page *pg; struct snd_pcm_runtime *runtime; struct audio_substream_data *rtd; @@ -870,6 +895,8 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, } } if (adata->asic_type == CHIP_STONEY) { + spin_lock_irqsave(&lock, flags); + val = acp_reg_read(adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -893,6 +920,8 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, } acp_reg_write(val, adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); + + spin_unlock_irqrestore(&lock, flags); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: fix oops w/ for_each_rtd_codec_dai_rollback() macro
by Pierre-Louis Bossart 30 Oct '18

30 Oct '18

A kernel oops happens on an error case (usual missing BE mixer configuration required by Intel SST driver). Git bisect points to this macro and an operator precedence issue. for (; ((i--) >= 0) && ((dai) = rtd->codec_dais[i]);) The initial code replaced by this macro was while (--i >= 0) { codec_dai = rtd->codec_dais[i]; Fix the C operator precedence difference by reverting to pre-decrement Fixes: 0b7990e3897 ('ASoC: add for_each_rtd_codec_dai() macro') Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/soc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f1dab1f4b194..70c10a8f3e90 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1192,7 +1192,7 @@ struct snd_soc_pcm_runtime { ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \ (i)++) #define for_each_rtd_codec_dai_rollback(rtd, i, dai) \ - for (; ((i--) >= 0) && ((dai) = rtd->codec_dais[i]);) + for (; ((--i) >= 0) && ((dai) = rtd->codec_dais[i]);) /* mixer control */ -- 2.17.1

2 1

[alsa-devel] [PATCH - dmix v2 1/1] pcm: dmix: Align slave_hw_ptr to slave period boundary
by twischer＠de.adit-jv.com 30 Oct '18

30 Oct '18

From: Laxmi Devi <Laxmi.Devi(a)in.bosch.com> These changes are required due to the kernel commit 07b7acb51d283d8469696c906b91f1882696a4d4 ("ASoC: rsnd: update pointer more accurate") Issue is that snd_pcm_wait() goes back to waiting because the hw_ptr is not period aligned. Therefore snd_pcm_wait() will block for a longer time as required. With these rcar driver changes the exact position of the dma is returned. During snd_pcm_start they read hw_ptr as reference, and this hw_ptr is now not period aligned, and is a little ahead over the period while it is read. Therefore when the avail is calculated during snd_pcm_wait(), it is missing the avail_min by a few frames. Consider the below example: Considering the application is trying to write 0x120 frames and the period_size = 0x60, avail_min = 0x120 and buffersize = 0x360 : rsnd_pointer=0x12c -> dma pointer at the end of second write during snd_pcm_dmix_start(). Since another 0x120 buffer is available, application writes 0x120 and goes to snd_pcm_wait(). It is woken up after 3 snd_pcm_period_elapsed() to see rsnd_pointer=0x248. So hw_ptr = new_slave_hw_ptr - reference_slave_hw_ptr = 0x248 - 0x12c = 0x11c. It needs 4 more frames to be able to write. And so it goes back to waiting. But since 3 snd_pcm_period_elapsed(), 3 periods should be available and it should have been able to write. If rsnd_pointer during the start was 0x120 which is 3 periods then 0x248 - 0x120 = 128 it could go on with write. Signed-off-by: Laxmi Devi <Laxmi.Devi(a)in.bosch.com> --- > On 10/29/18 16:54, Takashi Iwai wrote: > The problem is that aligning the start essentially imposes an > artificial long latency, and changes the behavior out of sudden. Now, we are only align the salve_hw_ptr which also solves our delay issue. But this change should not increase the latency because the write position is given by slave_appl_ptr. Do you see any other drawbacks? Best regards Timo diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index a6a8f3a..eaa0b0f 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -560,6 +560,8 @@ static int snd_pcm_dmix_hwsync(snd_pcm_t *pcm) static void reset_slave_ptr(snd_pcm_t *pcm, snd_pcm_direct_t *dmix) { dmix->slave_appl_ptr = dmix->slave_hw_ptr = *dmix->spcm->hw.ptr; + dmix->slave_hw_ptr = ((dmix->slave_hw_ptr / dmix->slave_period_size) + * dmix->slave_period_size); if (pcm->buffer_size > pcm->period_size * 2) return; /* If we have too litte periods, better to align the start position -- 2.7.4

3 3

[alsa-devel] [PATCH] ALSA: firewire-lib: fix insufficient PCM rule for period/buffer size
by Takashi Sakamoto 30 Oct '18

30 Oct '18

In a former commit, PCM constraint based on LCM of SYT_INTERVAL was obsoleted with PCM rule. However, the new PCM rule brings -EINVAL in some cases that max/min values of size of buffer/period is not multiples of one of values of SYT_INTERVAL. For example, pulseaudio always fail to configure PCM substream. This commit changes strategy for the PCM rule. Although the buggy rules had a single dependency (rate from period, period from rate, rate from buffer, buffer from rate), a revised rule has double dependencies (period from period/rate, buffer from buffer/rate). A step of value is calculated with table of SYT_INTERVAL and list of available rates. This prevents interval template which brings -EINVAL to a call of snd_interval_refine(). Fixes: 5950229582bc('ALSA: firewire-lib: add PCM rules to obsolete PCM constraints based on LCM of SYT_INTERVAL') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 57 ++++++----------------------------- 1 file changed, 9 insertions(+), 48 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index fcd965f1d69e..9be76c808fcc 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -146,53 +146,22 @@ static int apply_constraint_to_size(struct snd_pcm_hw_params *params, struct snd_interval *s = hw_param_interval(params, rule->var); const struct snd_interval *r = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval t = { - .min = s->min, .max = s->max, .integer = 1, - }; + struct snd_interval t = {0}; + unsigned int step = 0; int i; for (i = 0; i < CIP_SFC_COUNT; ++i) { - unsigned int rate = amdtp_rate_table[i]; - unsigned int step = amdtp_syt_intervals[i]; - - if (!snd_interval_test(r, rate)) - continue; - - t.min = roundup(t.min, step); - t.max = rounddown(t.max, step); + if (snd_interval_test(r, amdtp_rate_table[i])) + step = max(step, amdtp_syt_intervals[i]); } - if (snd_interval_checkempty(&t)) - return -EINVAL; + t.min = roundup(s->min, step); + t.max = rounddown(s->max, step); + t.integer = 1; return snd_interval_refine(s, &t); } -static int apply_constraint_to_rate(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_interval *r = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - const struct snd_interval *s = hw_param_interval_c(params, rule->deps[0]); - struct snd_interval t = { - .min = UINT_MAX, .max = 0, .integer = 1, - }; - int i; - - for (i = 0; i < CIP_SFC_COUNT; ++i) { - unsigned int step = amdtp_syt_intervals[i]; - unsigned int rate = amdtp_rate_table[i]; - - if (s->min % step || s->max % step) - continue; - - t.min = min(t.min, rate); - t.max = max(t.max, rate); - } - - return snd_interval_refine(r, &t); -} - /** * amdtp_stream_add_pcm_hw_constraints - add hw constraints for PCM substream * @s: the AMDTP stream, which must be initialized. @@ -250,24 +219,16 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, */ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, apply_constraint_to_size, NULL, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) goto end; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - apply_constraint_to_rate, NULL, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); - if (err < 0) - goto end; err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, apply_constraint_to_size, NULL, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) goto end; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - apply_constraint_to_rate, NULL, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1); - if (err < 0) - goto end; end: return err; } -- 2.19.1

2 1

[alsa-devel] [PATCH] ALSA: ca0106: Disable IZD on SB0570 DAC to fix audio pops
by Alex Stanoev 29 Oct '18

29 Oct '18

The Creative Audigy SE (SB0570) card currently exhibits an audible pop whenever playback is stopped or resumed, or during silent periods of an audio stream. Initialise the IZD bit to the 0 to eliminate these pops. The Infinite Zero Detection (IZD) feature on the DAC causes the output to be shunted to Vcap after 2048 samples of silence. This discharges the AC coupling capacitor through the output and causes the aforementioned pop/click noise. The behaviour of the IZD bit is described on page 15 of the WM8768GEDS datasheet: "With IZD=1, applying MUTE for 1024 consecutive input samples will cause all outputs to be connected directly to VCAP. This also happens if 2048 consecutive zero input samples are applied to all 6 channels, and IZD=0. It will be removed as soon as any channel receives a non-zero input". I believe the second sentence might be referring to IZD=1 instead of IZD=0 given the observed behaviour of the card. This change should make the DAC initialisation consistent with Creative's Windows driver, as this popping persists when initialising the card in Linux and soft rebooting into Windows, but is not present on a cold boot to Windows. Signed-off-by: Alex Stanoev <alex(a)astanoev.com> --- sound/pci/ca0106/ca0106.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h index 04402c14cb23..9847b669cf3c 100644 --- a/sound/pci/ca0106/ca0106.h +++ b/sound/pci/ca0106/ca0106.h @@ -582,7 +582,7 @@ #define SPI_PL_BIT_R_R (2<<7) /* right channel = right */ #define SPI_PL_BIT_R_C (3<<7) /* right channel = (L+R)/2 */ #define SPI_IZD_REG 2 -#define SPI_IZD_BIT (1<<4) /* infinite zero detect */ +#define SPI_IZD_BIT (0<<4) /* infinite zero detect */ #define SPI_FMT_REG 3 #define SPI_FMT_BIT_RJ (0<<0) /* right justified mode */ -- 2.19.1

2 1

[alsa-devel] [PATCH] pcm: dmix: Align slave_hw_ptr to slave period boundary
by twischer＠de.adit-jv.com 29 Oct '18

29 Oct '18

From: Laxmi Devi <Laxmi.Devi(a)in.bosch.com> These changes are required due to the kernel commit 07b7acb51d283d8469696c906b91f1882696a4d4 ("ASoC: rsnd: update pointer more accurate") Issue is that snd_pcm_wait() goes back to waiting because the hw_ptr is not period aligned. Therefore snd_pcm_wait() will block for a longer time as required. With these rcar driver changes the exact position of the dma is returned. During snd_pcm_start they read hw_ptr as reference, and this hw_ptr is now not period aligned, and is a little ahead over the period while it is read. Therefore when the avail is calculated during snd_pcm_wait(), it is missing the avail_min by a few frames. Consider the below example: Considering the application is trying to write 0x120 frames and the period_size = 0x60, avail_min = 0x120 and buffersize = 0x360 : rsnd_pointer=0x12c -> dma pointer at the end of second write during snd_pcm_dmix_start(). Since another 0x120 buffer is available, application writes 0x120 and goes to snd_pcm_wait(). It is woken up after 3 snd_pcm_period_elapsed() to see rsnd_pointer=0x248. So hw_ptr = new_slave_hw_ptr - reference_slave_hw_ptr = 0x248 - 0x12c = 0x11c. It needs 4 more frames to be able to write. And so it goes back to waiting. But since 3 snd_pcm_period_elapsed(), 3 periods should be available and it should have been able to write. If rsnd_pointer during the start was 0x120 which is 3 periods then 0x248 - 0x120 = 128 it could go on with write. Signed-off-by: Laxmi Devi <Laxmi.Devi(a)in.bosch.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- We are not completely sure if this is the best approach but we had also no better idea to solve this problem (with similar CPU usage). Therefore if anyone has a better solution please feel free to describe this one. src/pcm/pcm_dmix.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index a6a8f3a..d3e9319 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -560,11 +560,10 @@ static int snd_pcm_dmix_hwsync(snd_pcm_t *pcm) static void reset_slave_ptr(snd_pcm_t *pcm, snd_pcm_direct_t *dmix) { dmix->slave_appl_ptr = dmix->slave_hw_ptr = *dmix->spcm->hw.ptr; - if (pcm->buffer_size > pcm->period_size * 2) - return; - /* If we have too litte periods, better to align the start position - * to the period boundary so that the interrupt can be handled properly - * at the right time. + dmix->slave_hw_ptr = ((dmix->slave_hw_ptr / dmix->slave_period_size) + * dmix->slave_period_size); + /* Better to align the start position to the period boundary so that + * the interrupt can be handled properly at the right time. */ dmix->slave_appl_ptr = ((dmix->slave_appl_ptr + dmix->slave_period_size - 1) / dmix->slave_period_size) * dmix->slave_period_size; -- 2.7.4

2 1

[alsa-devel] alsaloop problems; ALSA streaming tutorial (resend)
by frederik＠ofb.net 29 Oct '18

29 Oct '18

Dear Mark (cc'ing ALSA-user, ALSA-devel) Thank you for your tutorial http://www.pogo.org.uk/~mark/trx/streaming-desktop-audio.html I found it helpful and clearly-written (although missing a "}" brace and I think it should be "-P" instead of "-p"...). I got as far as trying the 'alsaloop' example. It seems very finicky, sometimes I am able to play a test.wav cleanly through the loopback device, and sometimes there are a lot of crackles and I get errors like playback hw:0: change avail_min=2968 playback hw:0: change avail_min=2972 playback hw:0: change avail_min=2976 underrun for playback hw:0 last write before 9.8370ms, queued 10.0000ms/0.0000ms -> missing -0.1630ms expected 10.0000ms, processing 0.1910ms, max missing 0.0443ms last wake 0.0350ms, last check 0.0350ms, avail_min 32.0000ms max buf 40.0000ms, pfilled 0.0000ms, cfilled 10.0000ms job started before 0.0110ms For 'aplay' invocations where there are errors, it generally never recovers; the whole (10 second) wav file is played with crackles and errors being printed. I tried to figure out if this is due to some different rate settings being chosen at random with each 'aplay' invocation, but all the alsaloop debugging output looks the same between working and non-working runs of aplay (except for the above underrun errors). I also tried various settings for -A and -S in 'alsaloop', but nothing would alleviate this problem. The only thing that fixes it is to change the latency '-t' setting to a higher value than the default 10ms. However this seems like a bad solution, given that it is able (50% of the time) to play a whole sound file at the lower latency with no errors; I would rather have it recover from the underruns when they happen, than increase the latency for every invocation. Enabling realtime via the Arch Wiki info (a bit tricky as I had to edit some of the files by hand) didn't change this behavior. I also encountered a problem when trying to make alsaloop output to a dmix device. I wanted to have it output to dmix so that I could use my audio normally while experimenting with alsaloop. I got a "Poll FD initialization failed." error: playback plug:hw0mix/capture hubcap sync type: CAPTRATESHIFT New pitch for playback plug:hw0mix/capture hubcap: 1.00000000 (min/max samples = 0/0) underrun for playback plug:hw0mix last write before 19.7440ms, queued -1.0000ms/-1.0000ms -> missing -1.0000ms expected 10.0000ms, processing 0.0290ms, max missing 0.0000ms last wake 1539883911239.1819ms, last check 19.6460ms, avail_min 2.5000ms max buf 250.0000ms, pfilled 0.0000ms, cfilled 0.0000ms job started before 0.0310ms Poll FD initialization failed. Is this error understood? I would have thought that enough others would also have been wanting to use alsaloop in such a simple configuration, to warrant a better error message. I'm somewhat confused by the alsaloop manual page, my version says it is from "5 Aug 2010" but it has glaring errors, it says "-B" is a synonym for "--buffer" but alsaloop tells me "alsaloop: invalid option -- 'B'". Also it says that I can use "-c" to set both channels and rate. Are these typos really unfixed for eight years? Here are the commands I used: $ alsaloop -C hubcap -P hw:0 -v -v -U |& tee alsaloloop.out ... $ aplay -v test.wav |& tee aplay.out ... $ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version k4.14.71-1-lts. The earlier version of this email had output files attached, but they caused the message to be held for "moderator approval". Here are some HTTP versions of the same files: alsaloop.out: http://ix.io/1pUj aplay.out: http://ix.io/1pUk asound-remote (the config): http://ix.io/1pUl I think I'll investigate network audio with JACK next but I wanted to share my experiences in case there is something I'm doing wrong here, or in case there are bugs that can be fixed in ALSA. Thank you, Frederick

2 2

Re: [alsa-devel] [linux-sunxi] [BUG] On Allwinner A20 Board Cubietruck distorted SPDIF opical output
by Markus Mitsch 27 Oct '18

27 Oct '18

hi. i posted my answer wrong so here is my finding: i made the output work by disabling the sigma delta modulation. after compiling the kernel i can output through spdif with any rate. greetings markus 2018-05-26 18:38 GMT+02:00 Markus Mitsch <markusmitsch86(a)googlemail.com>: > 2018-05-26 16:57 GMT+02:00 Markus Mitsch <markusmitsch86(a)googlemail.com>: >> 2018-05-23 17:12 GMT+02:00 Code Kipper <codekipper(a)gmail.com>: >>> Hi All, >>> I can see that the problem was introduced somewhere here >>> >>> DOESN'T WORK >>> ffc3eb6f3a83 Merge tag 'sunxi-clk-for-4.15' of >>> https://git.kernel.org/pub/scm/linux/kernel/git/sunxi/linux into >>> clk-next >>> ae74ac082886 Merge tag 'clk-v4.15-exynos-pm' of >>> git://git.kernel.org/pub/scm/linux/kernel/git/snawrocki/clk into >>> clk-next >>> faa865f18cb7 Merge tag 'clk-v4.15-samsung' of >>> git://git.kernel.org/pub/scm/linux/kernel/git/snawrocki/clk into >>> clk-next >>> f09a6b86fdae Merge tag 'clk-renesas-for-v4.15-tag1' of >>> git://git.kernel.org/pub/scm/linux/kernel/git/geert/renesas-drivers >>> into clk-next >>> 4328a2186e51 clk: sunxi-ng: sun4i: Export video PLLs >>> 553c7d5ba2fe clk: sunxi-ng: Add A83T display clocks >>> 7679eb20353d clk: samsung: Add a separate driver for Exynos4412 ISP clocks >>> 8ca8ac102484 clk: samsung: Add dt bindings for Exynos4412 ISP clock controller >>> 75920aac275a clk: samsung: Instantiate Exynos4412 ISP clocks only when available >>> 24ea78a09f89 clk: sunxi-ng: sun8i: a23: Use sigma-delta modulation for audio PLL >>> ee6501d69217 clk: sunxi-ng: sun6i: Use sigma-delta modulation for audio PLL >>> 042f7f8f9715 clk: sunxi-ng: sun5i: Use sigma-delta modulation for audio PLL >>> de3448519194 clk: sunxi-ng: sun4i: Use sigma-delta modulation for audio PLL >>> a5e3e2b2ef85 clk: sunxi-ng: sun8i: h3: Use sigma-delta modulation for audio PLL >>> 392ba5fafcdf clk: sunxi-ng: nm: Add support for sigma-delta modulation >>> 05d2eaac96d4 clk: sunxi-ng: Add sigma-delta modulation support >>> 4cdbc40d64d4 clk: sunxi-ng: nm: Check if requested rate is supported >>> by fractional clock >>> d51fe3ba9773 clk: sunxi-ng: sun5i: Fix bit offset of audio PLL post-divider >>> 4e8975cbb516 clk: samsung: exynos5433: mark PM functions as __maybe_unused >>> a4f21e9ceb5c clk: samsung: Remove obsolete clkdev alias support >>> 45d882daf88a clk: samsung: Add explicit MPLL, EPLL clkdev aliases in >>> S3C2443 driver >>> cd05417f728b clk: samsung: Rework clkdev alias handling in S3C2443 driver >>> efea8d377184 clk: samsung: Rework clkdev alias handling in Exynos5440 driver >>> 36ba48240b19 clk: samsung: Drop useless alias in Exynos5420 clk driver >>> 29964890f31c clk: samsung: Remove clkdev alias support in Exynos5250 clk driver >>> 6de08891c896 clk: samsung: Remove double assignment of CLK_ARM_CLK in >>> Exynos4 driver >>> 58f4a5ff3a62 clk: samsung: Remove clkdev alias support in Exynos4 clk driver >>> c9194fb623b0 clk: samsung: Remove support for obsolete Exynos4212 CPU clock >>> d5cd103b06f9 clk: samsung: Remove support for Exynos4212 SoCs in >>> Exynos CLKOUT driver >>> 7c4f63ec94a1 clk: sunxi-ng: a83t: Fix invalid csi-mclk mux offset >>> WORKS >>> 0a4e632b6f9d Merge branch 'clk-fixes' into clk-next >>> >>> BR, >>> CK >>> >>> >>> On 22 May 2018 at 21:27, Code Kipper <codekipper(a)gmail.com> wrote: >>>> Hi Markus, >>>> I've been able to reproduce this on my A31 board with linux-next(plays >>>> at 44100 but not at 96K)so it is an issue. I also tested the same code >>>> on my A64 and it worked fine, The blocks are very similar so it maybe >>>> a clocking issue, I'll look into this. >>>> BR, >>>> CK >>>> >>>> On 21 May 2018 at 20:17, Markus Mitsch <markusmitsch86(a)googlemail.com> wrote: >>>>> 2018-05-21 15:17 GMT+02:00 Maxime Ripard <maxime.ripard(a)bootlin.com>: >>>>>> On Mon, May 21, 2018 at 02:43:55PM +0200, Markus Mitsch wrote: >>>>>>> 2018-05-21 11:43 GMT+02:00 Markus Mitsch <markusmitsch86(a)googlemail.com>: >>>>>>> >>> Hello, >>>>>>> >>> >>>>>>> >>> i have a new problem. if i want to output sound from optical i get >>>>>>> >>> distorted sound. At 44.1 khz its like an old vinly, but at 48 khz and >>>>>>> >>> higher it is very bad. thanks in advance for your time. >>>>>>> > >>>>>>> >> Hi Markus, >>>>>>> >> I haven't tested it for a while but when I did it worked as high as >>>>>>> >> 96K with no issues. Does it work ok with the original cubietruck >>>>>>> >> software?, >>>>>>> >> CK >>>>>>> > >>>>>>> > Hello, >>>>>>> > i am using arch linux with mainline kernel 4.16.8. parhaps i can try >>>>>>> > today with official image. >>>>>>> > mfg >>>>>>> >>>>>>> Hi. >>>>>>> So, with debian image from 2014 with 3.4.79 kernel the sound is clear >>>>>>> up to 96 khz. >>>>>> >>>>>> You said that it is a new problem, when was it introduced? Could you >>>>>> bisect the issue? >>>>>> >>>>>> Maxime >>>>> >>>>> Hello. >>>>> I dont know since when this problem appears. When saying "new" i >>>>> refered to my older post. I tried to look into the dts file and the >>>>> code >>>>> but i am afraid this is a bit too high for me. I can say that the >>>>> analog output works, even at high sampling rates. >>>>> And I have read something about a clock source could be the problem. >>>>> I'm sorry I can't tell you any more. >>>>> >>>>> mfg >>>>> Markus >> >> Hello, >> i tried undoing part of the commit >> de3448519194 clk: sunxi-ng: sun4i: Use sigma-delta modulation for audio PLL >> by setting the value for the divider from 1 to 4 again on both places. >> with no success. >> >> markus > > i now had a little success. the output works!! > i disabled the sigma-delta support by making > ccu_sdm_helper_enable in drivers/clk/sunxi-ng/ccu_sdm.c return imediately. > > markus

2 2

[alsa-devel] [PATCH] ALSA: dice: fix to wait for releases of all ALSA character devices
by Takashi Sakamoto 27 Oct '18

27 Oct '18

In a development period for Linux kernel v4.20, drivers in ALSA firewire stack were changed to wait for releases of all ALSA character devices at .remove callback of bus driver. However, ALSA dice driver is partly out of this change. This bug can bring fault to user process which holds the last of character device in unplugging. This commit fixes the driver to wait in the callback. Fixes: 61ccc6f6b27c ('ALSA: firewire: block .remove callback of bus driver till all of ALSA character devices are released') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 0f6dbcffe711..ed50b222d36e 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -240,8 +240,8 @@ static void dice_remove(struct fw_unit *unit) cancel_delayed_work_sync(&dice->dwork); if (dice->registered) { - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(dice->card); + // Block till all of ALSA character devices are released. + snd_card_free(dice->card); } mutex_destroy(&dice->mutex); -- 2.19.1

2 1

[alsa-devel] QA issue with media-sound/alsa-utils-1.1.7 Gentoo package
by Francesco Turco 27 Oct '18

27 Oct '18

The Portage package manager returns the following QA warning when installing the media-sound/alsa-utils-1.1.7 Gentoo package: * QA Notice: Package triggers severe warnings which indicate that it * may exhibit random runtime failures. * ../alsamixer/volume_mapping.c:116:15: warning: incompatible implicit declaration of built-in function ‘exp10’ * ../alsamixer/volume_mapping.c:151:14: warning: incompatible implicit declaration of built-in function ‘exp10’ * Please do not file a Gentoo bug and instead report the above QA * issues directly to the upstream developers of this software. * Homepage: https://alsa-project.org/ -- https://fturco.gitlab.io/

1 0

[alsa-devel] [PATCH 0/2] Introduce dmic mode switch delay parameter
by Jenny TC 25 Oct '18

25 Oct '18

Some DMICs need clock to be running for a specified duration as per the DMIC spec to complete the mode transition like sleep to mormal, off to normal etc. Jenny TC (2): ASoC: dmic: Enable ACPI device entry ASoC: dmic: Introduce mode switch delay sound/soc/codecs/dmic.c | 38 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 38 insertions(+) -- 1.9.1

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[alsa-devel] [PATCH] alsa-utils: fix identation indentation on speaker-test.c
by Eduardo Abinader 25 Oct '18

25 Oct '18

Signed-off-by: Eduardo Abinader <eduardoabinader(a)gmail.com> --- speaker-test/speaker-test.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/speaker-test/speaker-test.c b/speaker-test/speaker-test.c index 773af0a..4aac30c 100644 --- a/speaker-test/speaker-test.c +++ b/speaker-test/speaker-test.c @@ -1207,7 +1207,7 @@ int main(int argc, char *argv[]) { gettimeofday(&tv1, NULL); for(chn = 0; chn < channels; chn++) { - int channel = get_speaker_channel(chn); + int channel = get_speaker_channel(chn); printf(" %d - %s\n", channel, get_channel_name(channel)); err = write_loop(handle, channel, ((rate*3)/period_size), frames); -- 2.17.1

2 1

[alsa-devel] [PATCH v2 - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin
by Andre Guedes 25 Oct '18

25 Oct '18

Hi all, This new version addresses the feedback from v1 and fixes a bug on capture mode. Code-wise, the changes are very trivial, no major diffs. Below follows the highlights: * Standard headers like linux/if_ether.h are now checked by configure script. * The 'static' modifier was added to array variables in aaf_hw_constraint(). * aaf_close() callback was simplified. * aaf_rx_pdu() was modified so it doesn't return an error in case the number of bytes read by recv() is different from aaf->pdu_size. This is an expected situation and it means the received AVTPDU doesn't belong to the AAF stream the plugin is interested in so it should simply drop that AVTPDU. This fixes a bug when running multiple streams (with different AVPTU sizes) concurrently. Additionally, this new version improves the instructions in doc/aaf.txt in order to make it easier to test the plugin. Anyone should be able to test it just by copying and pasting the instructions. To recap what has been discussed about the plugin so far, check the comments in RFC and PATCH series archive [1-3]. All versions of this series can be also found in my alsa-plugins tree in github [4]. Regards, Andre [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-August/139494.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-September/140290.… [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-October/140667.ht… [4] https://github.com/aguedes/alsa-plugins Andre Guedes (7): aaf: Introduce plugin skeleton aaf: Load configuration parameters aaf: Implement Playback mode support aaf: Prepare for Capture mode support aaf: Implement Capture mode support aaf: Implement dump() ioplug callback aaf: Add support for direct read/write transfers Makefile.am | 3 + aaf/Makefile.am | 9 + aaf/pcm_aaf.c | 1200 +++++++++++++++++++++++++++++++++++++++++++++++++++++++ configure.ac | 11 + doc/aaf.txt | 157 ++++++++ 5 files changed, 1380 insertions(+) create mode 100644 aaf/Makefile.am create mode 100644 aaf/pcm_aaf.c create mode 100644 doc/aaf.txt -- 2.14.4

3 11

[alsa-devel] KASAN: use-after-free Read in snd_seq_timer_interrupt
by syzbot 24 Oct '18

24 Oct '18

Hello, syzbot found the following crash on: HEAD commit: 44786880df19 Merge branch 'parisc-4.20-1' of git://git.ker.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=178c8bad400000 kernel config: https://syzkaller.appspot.com/x/.config?x=5071a2e3cb82c943 dashboard link: https://syzkaller.appspot.com/bug?extid=d5136d4d3240cbe45a2a compiler: gcc (GCC) 8.0.1 20180413 (experimental) Unfortunately, I don't have any reproducer for this crash yet. IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+d5136d4d3240cbe45a2a(a)syzkaller.appspotmail.com ================================================================== BUG: KASAN: use-after-free in snd_seq_timer_interrupt+0x364/0x3e0 sound/core/seq/seq_timer.c:145 Read of size 8 at addr ffff8801cf71c118 by task syz-executor0/24085 CPU: 1 PID: 24085 Comm: syz-executor0 Not tainted 4.19.0+ #300 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: <IRQ> __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x244/0x39d lib/dump_stack.c:113 print_address_description.cold.7+0x9/0x1ff mm/kasan/report.c:256 kasan_report_error mm/kasan/report.c:354 [inline] kasan_report.cold.8+0x242/0x309 mm/kasan/report.c:412 __asan_report_load8_noabort+0x14/0x20 mm/kasan/report.c:433 snd_seq_timer_interrupt+0x364/0x3e0 sound/core/seq/seq_timer.c:145 snd_timer_interrupt+0x9ff/0x1660 sound/core/timer.c:855 snd_timer_s_function+0xcd/0x130 sound/core/timer.c:1094 call_timer_fn+0x272/0x920 kernel/time/timer.c:1326 expire_timers kernel/time/timer.c:1363 [inline] __run_timers+0x7e5/0xc70 kernel/time/timer.c:1682 run_timer_softirq+0x52/0xb0 kernel/time/timer.c:1695 __do_softirq+0x308/0xb7e kernel/softirq.c:292 invoke_softirq kernel/softirq.c:373 [inline] irq_exit+0x17f/0x1c0 kernel/softirq.c:413 exiting_irq arch/x86/include/asm/apic.h:536 [inline] smp_apic_timer_interrupt+0x1cb/0x760 arch/x86/kernel/apic/apic.c:1061 apic_timer_interrupt+0xf/0x20 arch/x86/entry/entry_64.S:801 </IRQ> RIP: 0010:arch_local_irq_restore arch/x86/include/asm/paravirt.h:761 [inline] RIP: 0010:__raw_spin_unlock_irqrestore include/linux/spinlock_api_smp.h:160 [inline] RIP: 0010:_raw_spin_unlock_irqrestore+0xaf/0xd0 kernel/locking/spinlock.c:184 Code: 30 f6 31 89 48 b8 00 00 00 00 00 fc ff df 48 89 fa 48 c1 ea 03 80 3c 02 00 75 28 48 83 3d 28 cd 7c 01 00 74 15 48 89 df 57 9d <0f> 1f 44 00 00 eb ad e8 7c 52 4b f9 eb bb 0f 0b 0f 0b e8 8a 54 0e RSP: 0018:ffff880181eaf508 EFLAGS: 00000282 ORIG_RAX: ffffffffffffff13 RAX: dffffc0000000000 RBX: 0000000000000282 RCX: 0000000000000000 RDX: 1ffffffff1263ec6 RSI: 0000000000000001 RDI: 0000000000000282 RBP: ffff880181eaf518 R08: ffff88018378a140 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: ffff8801c0b94ac0 R13: ffff8801c0b94300 R14: ffff8801daf2cc40 R15: 0000000000000001 try_to_wake_up+0x10a/0x12e0 kernel/sched/core.c:2056 wake_up_process kernel/sched/core.c:2124 [inline] wake_up_q+0xa4/0x100 kernel/sched/core.c:440 futex_wake+0x61f/0x760 kernel/futex.c:1556 do_futex+0x2e4/0x26d0 kernel/futex.c:3533 __do_sys_futex kernel/futex.c:3589 [inline] __se_sys_futex kernel/futex.c:3557 [inline] __x64_sys_futex+0x472/0x6a0 kernel/futex.c:3557 do_syscall_64+0x1b9/0x820 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x457569 Code: fd b3 fb ff c3 66 2e 0f 1f 84 00 00 00 00 00 66 90 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 0f 83 cb b3 fb ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:0000000000a3fbf8 EFLAGS: 00000246 ORIG_RAX: 00000000000000ca RAX: ffffffffffffffda RBX: 000000000072bfa0 RCX: 0000000000457569 RDX: 0000000000000000 RSI: 0000000000000081 RDI: 000000000072bfa8 RBP: 00000000000003b7 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000a3fcb0 R11: 0000000000000246 R12: 000000000072bfac R13: 000000000072bfac R14: 0000000000000003 R15: 0000000000000001 Allocated by task 24073: save_stack+0x43/0xd0 mm/kasan/kasan.c:448 set_track mm/kasan/kasan.c:460 [inline] kasan_kmalloc+0xc7/0xe0 mm/kasan/kasan.c:553 kmem_cache_alloc_trace+0x152/0x750 mm/slab.c:3620 kmalloc include/linux/slab.h:513 [inline] kzalloc include/linux/slab.h:707 [inline] queue_new sound/core/seq/seq_queue.c:113 [inline] snd_seq_queue_alloc+0xab/0x5b0 sound/core/seq/seq_queue.c:185 snd_seq_ioctl_create_queue+0xb0/0x320 sound/core/seq/seq_clientmgr.c:1522 snd_seq_ioctl+0x253/0x440 sound/core/seq/seq_clientmgr.c:2138 vfs_ioctl fs/ioctl.c:46 [inline] file_ioctl fs/ioctl.c:501 [inline] do_vfs_ioctl+0x1de/0x1720 fs/ioctl.c:685 ksys_ioctl+0xa9/0xd0 fs/ioctl.c:702 __do_sys_ioctl fs/ioctl.c:709 [inline] __se_sys_ioctl fs/ioctl.c:707 [inline] __x64_sys_ioctl+0x73/0xb0 fs/ioctl.c:707 do_syscall_64+0x1b9/0x820 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe Freed by task 24072: save_stack+0x43/0xd0 mm/kasan/kasan.c:448 set_track mm/kasan/kasan.c:460 [inline] __kasan_slab_free+0x102/0x150 mm/kasan/kasan.c:521 kasan_slab_free+0xe/0x10 mm/kasan/kasan.c:528 __cache_free mm/slab.c:3498 [inline] kfree+0xcf/0x230 mm/slab.c:3813 queue_delete+0x90/0xb0 sound/core/seq/seq_queue.c:156 snd_seq_queue_client_leave+0x36/0x170 sound/core/seq/seq_queue.c:584 seq_free_client1.part.9+0xf4/0x270 sound/core/seq/seq_clientmgr.c:263 seq_free_client1 sound/core/seq/seq_clientmgr.c:256 [inline] seq_free_client+0x82/0x100 sound/core/seq/seq_clientmgr.c:285 snd_seq_release+0x51/0xe0 sound/core/seq/seq_clientmgr.c:366 __fput+0x385/0xa30 fs/file_table.c:278 ____fput+0x15/0x20 fs/file_table.c:309 task_work_run+0x1e8/0x2a0 kernel/task_work.c:113 tracehook_notify_resume include/linux/tracehook.h:193 [inline] exit_to_usermode_loop+0x318/0x380 arch/x86/entry/common.c:166 prepare_exit_to_usermode arch/x86/entry/common.c:197 [inline] syscall_return_slowpath arch/x86/entry/common.c:268 [inline] do_syscall_64+0x6be/0x820 arch/x86/entry/common.c:293 entry_SYSCALL_64_after_hwframe+0x49/0xbe The buggy address belongs to the object at ffff8801cf71c0c0 which belongs to the cache kmalloc-512 of size 512 The buggy address is located 88 bytes inside of 512-byte region [ffff8801cf71c0c0, ffff8801cf71c2c0) The buggy address belongs to the page: page:ffffea00073dc700 count:1 mapcount:0 mapping:ffff8801da800940 index:0x0 flags: 0x2fffc0000000100(slab) raw: 02fffc0000000100 ffffea00073a4fc8 ffffea000761f4c8 ffff8801da800940 raw: 0000000000000000 ffff8801cf71c0c0 0000000100000006 0000000000000000 kobject: 'loop5' (00000000172d3417): kobject_uevent_env page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8801cf71c000: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc ffff8801cf71c080: fc fc fc fc fc fc fc fc fb fb fb fb fb fb fb fb > ffff8801cf71c100: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff8801cf71c180: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff8801cf71c200: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#bug-status-tracking for how to communicate with syzbot.

1 0

[alsa-devel] [PATCH v4 00/14] ASoC: Add support to WCD9335 Audio Codec
by srinivas.kandagatla＠linaro.org 24 Oct '18

24 Oct '18

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Thankyou for reviewing v3 patchset, here is v4 addressing comments from v3. Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC. It is integrated in multiple Qualcomm SoCs like: MSM8996, MSM8976, and MSM8956 chipsets. WCD9335 had multiple functional blocks, like: Soundwire controller, interrupt mux, pin controller, Audio codec, MBHC, MAD(Mic activity Detection), Ultrasonic proximity and pen detection, Battery-voltage monitoring and Codec processing engine. Currently this patchset has been only tested with SLIMbus interface due to hardware avaiablity, but it can be easily made to work with both SLIMbus and I2C/I2S. This patchset adds very basic playback and capture support witout much fancy features. New features will be added once the basic support is in. This patchset is tested on top of linux-next on DB820c for both playback , capture paths and MBHC. Some parts of the code has been inherited from Qualcomm andriod kernel, so credits to various authors. WCD9335 can be interfaced via I2S/I2C or SLIMbus. Here is my test branch incase someone want to try this out: https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=wcd9335 Thanks, Srini Changes since v3 (https://lkml.org/lkml/2018/9/4/318/) - Updated mbhc dt bindings according to Robs Comments. - Added a cleanup patch to common up fixups. - Addressed various comments on mfd driver by Lee Jones. - Added acks on bindings and irq support. - fixed module loading Srinivas Kandagatla (14): ASoC: dt-bindings: update wcd9335 bindings. mfd: wcd9335: add support to wcd9335 core mfd: wcd9335: add wcd irq support ASoC: wcd9335: add support to wcd9335 codec ASoC: wcd9335: add CLASS-H Controller support ASoC: wcd9335: add basic controls ASoC: wcd9335: add playback dapm widgets ASoC: wcd9335: add capture dapm widgets ASoC: wcd9335: add audio routings ASoC: dt-bindings: Add WCD9335 MBHC specific properties ASoC: wcd9335: add mbhc support ASoC: apq8096: add slim support ASoC: apq8096: add headset JACK support ASoC: qcom: common: move be_hw_fixup to common .../devicetree/bindings/sound/qcom,wcd9335.txt | 30 +- drivers/mfd/Kconfig | 13 + drivers/mfd/Makefile | 4 + drivers/mfd/wcd9335-core.c | 394 ++ include/linux/mfd/wcd9335/registers.h | 640 +++ include/linux/mfd/wcd9335/wcd9335.h | 81 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-clsh-v2.c | 577 +++ sound/soc/codecs/wcd-clsh-v2.h | 49 + sound/soc/codecs/wcd9335.c | 5241 ++++++++++++++++++++ sound/soc/qcom/apq8096.c | 128 +- sound/soc/qcom/common.c | 17 + sound/soc/qcom/sdm845.c | 22 +- 14 files changed, 7168 insertions(+), 35 deletions(-) create mode 100644 drivers/mfd/wcd9335-core.c create mode 100644 include/linux/mfd/wcd9335/registers.h create mode 100644 include/linux/mfd/wcd9335/wcd9335.h create mode 100644 sound/soc/codecs/wcd-clsh-v2.c create mode 100644 sound/soc/codecs/wcd-clsh-v2.h create mode 100644 sound/soc/codecs/wcd9335.c -- 2.9.3

5 21

[alsa-devel] [asoc:for-4.20 519/528] sound/soc/stm/stm32_sai_sub.c:393:26-32: ERROR: application of sizeof to pointer
by kbuild test robot 24 Oct '18

24 Oct '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 6be0f96d799f487f05eb412d362d5a1747d665c2 commit: 8307b2afd386ccce369821daa2196068c47fe8cd [519/528] ASoC: stm32: sai: set sai as mclk clock provider coccinelle warnings: (new ones prefixed by >>) >> sound/soc/stm/stm32_sai_sub.c:393:26-32: ERROR: application of sizeof to pointer -- >> sound/soc/stm/stm32_sai_sub.c:343:5-8: WARNING: Unsigned expression compared with zero: div < 0 Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

3 2

[alsa-devel] Audio miniconference schedule
by Mark Brown 24 Oct '18

24 Oct '18

As previously announced there's an audio miniconference happening on the 21st in Edinburgh: https://blog.sirena.org.uk/2018/09/19/2018-linux-audio-miniconfernce (yes there is a typo in the URL). A number of people have proposed topics so I just went through them and pulled them into a proposed agenda: - Virtualization (Liam) - SoundWire (Liam) - Byte controls (Vinod) - GObject introspection for language bindings (Sakamoto-san) - Integrating the ALSA control core (Sakamoto-san) - AGL (Stephane/Ronan) - Y2038 (broonie) - AVB (Liam) - DSPs (Liam) - Sound Open Firmware What do people think? Any other topics? There's still time to sign up if you're interested in attending, you can sign up right up to the day of the event. It also seems good to mention that there's another audio event going on around ELC: https://blogs.gnome.org/uraeus/2018/09/24/getting-the-team-together-to-revo… which is a hackathon for a new userspace stack called PipeWire - if we have someone attending who knows about that I think it'd be good to talk about it too.

8 24

[alsa-devel] [PATCH - Intervals v2 1/1] interval: Interpret (x x+1] correctly and return x+1
by twischer＠de.adit-jv.com 24 Oct '18

24 Oct '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without this change an interval of (x x+1] will be interpreted as an empty interval but the right value would be x+1. This leads to a failing snd_pcm_hw_params() call which returns -EINVAL. An example issue log is given in the following: snd_pcm_hw_params failed with err -22 (Invalid argument) ACCESS: MMAP_NONINTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 16 CHANNELS: 1 RATE: 16000 PERIOD_TIME: (15999 16000] PERIOD_SIZE: (255 256] PERIOD_BYTES: (510 512] PERIODS: [2 3) BUFFER_TIME: 32000 BUFFER_SIZE: 512 BUFFER_BYTES: 1024 In case of (x x+1) we have to interpret it anyway as a single value of x to compensate rounding issues. For example the period size will result in an interval of (352 353) when the period time is 16ms and the sample rate 22050 Hz (16ms * 22,05 kHz = 352,8 frames). But 352 has to be chosen to allow a buffer size of 705 (32ms * 22,05 kHz = 705,6 frames) which has to be >= 2x period size to avoid Xruns. The buffer size will not end up with an interval of (705 706) simular to the period size because snd_pcm_rate_hw_refine_cchange() calls snd_interval_floor() for the buffer size. Therefore this value will be interpreted as an integer interval instead of a real interval further on. This issue seems to exist since the change of 9bb985c38 ("pcm: snd_interval_refine_first/last: exclude value only if also excluded before") Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- > On 10/18/18 12:57, Takashi Iwai wrote: > This change looks risky. The snd_interval_value() might be called > even if the interval isn't reduced to a single value. Rather check > openmin instead. If I would do "if (i->openmin)" on an interval of (x x+1) x+1 would be returned. But this would result in the second issue which I have tried to describe in the commit message: x has to be returned otherwise it is not guaranteed that buffer_size >= 2x period_size. And this would result in continuously Xruns. Therefore I would like to prefer "if (i->openmin && !i->openmax)" diff --git a/src/pcm/interval_inline.h b/src/pcm/interval_inline.h index a68e292..d9a30b2 100644 --- a/src/pcm/interval_inline.h +++ b/src/pcm/interval_inline.h @@ -51,12 +51,14 @@ INTERVAL_INLINE int snd_interval_single(const snd_interval_t *i) { assert(!snd_interval_empty(i)); return (i->min == i->max || - (i->min + 1 == i->max && i->openmax)); + (i->min + 1 == i->max && (i->openmin || i->openmax))); } INTERVAL_INLINE int snd_interval_value(const snd_interval_t *i) { assert(snd_interval_single(i)); + if (i->openmin && !i->openmax) + return i->max; return i->min; } -- 2.7.4

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Re: [alsa-devel] GFP_ATOMIC in Intel code
by Takashi Iwai 24 Oct '18

24 Oct '18

On Tue, 23 Oct 2018 10:49:27 +0200, Vinod wrote: > > On 22-10-18, 12:22, Pierre-Louis Bossart wrote: > > Looking at the latest for-next branch, by accident I saw a number of > > GFP_ATOMIC references. > > > > With my extended break I don't remember much but didn't we agree that none > > of those were needed? If yes I'll send a patch to fix all this. > > > I thought that someone had fixed up all these instances, Looking back at the git history, "someone" was mostly you :) A few of them below seem to be new codes that copied the bad past examples, some are overlooked one, and some are mailbox codes that aren't 100% sure to safe to replace (although I guess it is). > looks like few > missed, so yes please do fix them up and Welcome back :-) I hope you enjoyed the time-off, Pierre. thanks, Takashi > > > Thanks > > > > -Pierre > > > > > > /data/pbossart/ktest/broonie-next/sound/soc/intel$ git grep ATOMIC > > atom/sst/sst_pvt.c: msg = kzalloc(sizeof(struct ipc_post), GFP_ATOMIC); > > atom/sst/sst_pvt.c: msg->mailbox_data = > > kzalloc(SST_MAILBOX_SIZE, GFP_ATOMIC); > > boards/cht_bsw_rt5672.c: drv = devm_kzalloc(&pdev->dev, sizeof(*drv), > > GFP_ATOMIC); > > boards/glk_rt5682_max98357a.c: ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), > > GFP_ATOMIC); > > boards/kbl_da7219_max98927.c: ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), > > GFP_ATOMIC); > > boards/skl_hda_dsp_generic.c: ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), > > GFP_ATOMIC); > > -- > ~Vinod >

2 1

[alsa-devel] [PATCH - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin
by Andre Guedes 23 Oct '18

23 Oct '18

Hi all, The AAF plugin RFC series v2 has been around on alsa-devel ML for more than two weeks so I'm moving forward and sending it as PATCH series now. The first 5 patches from this series are pretty much the same from the previous version. The news are the top 2 patches which add more functionality to the plugin. Patch 6/7 implements the dump() callback which has been useful for debugging purposes and Patch 7/7 adds support for direct read/write transfers. For general context information about the AAF plugin, I'm quoting below the cover letter from the RFC series v1: "This patchset introduces the AAF plugin to the alsa-plugins project which enables TSN audio applications on Linux. For those not familiarized with TSN, it stands for Time-Sensitive Networking, and it's a set of IEEE technologies (enhancements to IEEE 802.1Q and Ethernet protocols) that provide precise time synchronization, bounded latency and application interoperability to bridged networks. Those technologies enable time-sensitive applications such as audio, video and control loops to run on top of bridged networks, co-existing with regular applications. TSN technologies are a super set of Audio Video Bridging (AVB) technologies also developed by IEEE. AVTP is the protocol defined to transport data in a TSN system, and AAF is one of the formats defined by AVTP to encapsulate audio data. AAF is specified in Chapter 7 from the AVTP spec [1]. This work is part of the effort to enable TSN technologies on upstream Linux ecosystem. All building-blocks required to enable TSN audio applications have been already developed and pushed to upstream projects. Time synchronization features are provided by linuxptp project [2], bounded latency features are provided by Linux Traffic Control subsystem since kernel version 4.15 [3], and AVTP packetization is provided by libavtp [4]. What is currently missing in the ecosystem to enable TSN audio applications is a piece of software that plumbs it all together and interfaces with Linux Audio system properly. That's the point of the AAF plugin introduced here. The AAF plugin is a PCM plugin that uses AVTP protocol to transmit / receive audio data through a TSN capable network. When operating in playback mode, the plugin reads PCM samples from the audio buffer, encapsulates into AVTP packets and transmits to the network, mimicking a typical AVTP Talker. When operating in capture mode, the plugin receives AVTP packets from the network, retrieves the PCM samples, and present them to alsa-lib layer at the presentation time, mimicking a typical AVTP Listener." For further information about what has been previously discussed, please refer to RFC series v1 and v2 archives in [5] and [6]. Finally, all versions of this series can be also found in my alsa-plugins tree in github [7]. Best regards, Andre [1] 1722-2016 - IEEE Standard for a Transport Protocol for Time-Sensitive Applications in Bridged Local Area Networks [2] http://linuxptp.sourceforge.net [3] https://patchwork.ozlabs.org/cover/826678/ [4] https://github.com/AVnu/libavtp [5] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-August/139494.html [6] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-September/140290.… [7] https://github.com/aguedes/alsa-plugins Andre Guedes (7): aaf: Introduce plugin skeleton aaf: Load configuration parameters aaf: Implement Playback mode support aaf: Prepare for Capture mode support aaf: Implement Capture mode support aaf: Implement dump() ioplug callback aaf: Add support for direct read/write transfers Makefile.am | 3 + aaf/Makefile.am | 9 + aaf/pcm_aaf.c | 1197 +++++++++++++++++++++++++++++++++++++++++++++++++++++++ configure.ac | 9 + doc/aaf.txt | 142 +++++++ 5 files changed, 1360 insertions(+) create mode 100644 aaf/Makefile.am create mode 100644 aaf/pcm_aaf.c create mode 100644 doc/aaf.txt -- 2.14.4

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[alsa-devel] [GIT PULL] ASoC fixes for the next release
by Mark Brown 23 Oct '18

23 Oct '18

The following changes since commit 7f91e2af1a4a2c34fc2e8fb046c722e1a9c85399: ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks (2018-10-21 16:46:32 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.0-2 for you to fetch changes up to 6be0f96d799f487f05eb412d362d5a1747d665c2: ASoC: stm32: sai: fix master clock naming (2018-10-23 10:51:45 +0100) ---------------------------------------------------------------- ASoC: Additional fixes for the next release A couple of fixes for build bot issues in the STM32 SAI driver. ---------------------------------------------------------------- Olivier Moysan (2): ASoC: stm32: add clock dependency for sai ASoC: stm32: sai: fix master clock naming sound/soc/stm/Kconfig | 1 + sound/soc/stm/stm32_sai_sub.c | 5 ++--- 2 files changed, 3 insertions(+), 3 deletions(-)

2 1

[alsa-devel] Segfault in pulseaudio since Alsa 1.1.7
by Hyperion 23 Oct '18

23 Oct '18

Hi, Â Since the upgrade to Alsa 1.1.7, I get a segfault at pulseaudio startup. This have been discussed and documented here : [1]https://www.linuxquestions.org/questions/slackware-14/current-fri-oc t-19-puseaudio-segfauts-4175640773/ Â It seems that I found the origin of the bug : reverting these 2 commits fixes the problem : [2]http://git.alsa-project.org/?p=alsa-...792dc809c908cc [3]http://git.alsa-project.org/?p=alsa-...5dc6e1594291e7 Then no more segfault in pulseaudio when accessing the USB soundcard. Â Please could you fix it ? Â Regards jp References Visible links 1. https://www.linuxquestions.org/questions/slackware-14/current-fri-oct-19-pu… 2. http://git.alsa-project.org/?p=alsa-plugins.git;a=commitdiff;h=fc3f5d51062f… 3. http://git.alsa-project.org/?p=alsa-plugins.git;a=commitdiff;h=6b996865126d… Hidden links: 4. https://www.linuxquestions.org/questions/slackware-14/current-fri-oct-19-pu…

2 6

[alsa-devel] [alsa-plugins-1.17] Regression in JACK plugin
by tet 23 Oct '18

23 Oct '18

Hi, after updating to alsa-plugins-1.17 ALSA throws: "ALSA lib pcm_plug.c:1273:(_snd_pcm_plug_open) Unknown field playback_ports My ~/.asourdrc is working in 1.16 and it does not look like the syntax changed for the plugin when looking at the readme. pcm.jack { type plug slave.pcm { type jack capture_ports { 0 system:capture_1 1 system:capture_1 } playback_ports { 0 system:playback_1 1 system:playback_2 } } hint.description "JACK Audio Kit" } pcm.!default { @func getenv vars [ PCM ] default jack } Any ideas?

2 6

[alsa-devel] [RFC PATCH 0/4] Initial prototype of DAPM sample rate domains
by Charles Keepax 23 Oct '18

23 Oct '18

Here is a first pass at adding dapm sample rate domain support. Things are still pretty rough in quite a few places and I expect some major refactoring before they are ready to merge, but things are far enough along for people to look at the approach I am taking to the problem. And should provide enough for discussion at the mini-conference, although hopefully I will get some more done next week as well. The basic high level summary is two new concepts are added to ASoC, one being a sample rate domain and the other being a domain group. The domain groups represent widgets that are limited by hardware restrictions to always run at the same sample rate as each other. Sample rate domains represent a slot on a device for a sample rate. Both of these are currently limited to be within the scope of a single component. The core will track which groups are connected together, then as a group is powered up it will be allocated a domain based on the other groups it is connected to. Domain groups are implemented as a new type of DAPM widget and the domains themselves are a new thing which devices add through their component drivers. As we have previously discussed my intention would be that we also add bridges between domains although I haven't started work on that yet. The current code is limited to within a single component and I think that the bridges will allow better support for propagating rates between components as well as obviously for SRCs. Currently the SRCs on wm5110 are just being handled as points where the domain groups don't connect to another peer. This allows them to function but doesn't accurately convey things like the limitations which input rates support which output rates on the ISRC. Thanks, Charles Charles Keepax (4): ASoC: dapm: Add support for hw_free on CODEC to CODEC links ASoC: dapm: Add support for a rate domain widget ASoC: domain: Add sample rate domain support ASoC: arizona: Add rate domain support include/sound/soc-dapm.h | 14 ++ include/sound/soc-domain.h | 98 +++++++++++ include/sound/soc.h | 8 + sound/soc/Makefile | 2 +- sound/soc/codecs/arizona.c | 255 ++++++++++++++++++++++------ sound/soc/codecs/arizona.h | 76 ++++++++- sound/soc/codecs/wm5110.c | 414 ++++++++++++++++++++++++++++++++++----------- sound/soc/soc-core.c | 8 + sound/soc/soc-dapm.c | 57 ++++++- sound/soc/soc-domain.c | 412 ++++++++++++++++++++++++++++++++++++++++++++ 10 files changed, 1183 insertions(+), 161 deletions(-) create mode 100644 include/sound/soc-domain.h create mode 100644 sound/soc/soc-domain.c -- 2.11.0

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[alsa-devel] [asoc:for-4.20 333/424] sound/pci/hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'?
by kbuild test robot 23 Oct '18

23 Oct '18

Hi Rakesh, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 1c8bc7b3de5e76cb89aacdc7be1475a028af505f commit: 6bae5ea9498926440ffc883f3dbceb0adc65e492 [333/424] ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers config: sh-allmodconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 6bae5ea9498926440ffc883f3dbceb0adc65e492 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): sound/pci/hda/patch_ca0132.c: In function 'patch_ca0132': >> sound/pci/hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'? [-Werror=implicit-function-declaration] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^~~~~~~~~ pcim_iomap sound/pci/hda/patch_ca0132.c:7650:18: warning: assignment makes pointer from integer without a cast [-Wint-conversion] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^ cc1: some warnings being treated as errors vim +7650 sound/pci/hda/patch_ca0132.c d5c016b56 Gabriele Martino 2015-05-18 7581 95c6e9cb7 Ian Minett 2011-06-15 7582 static int patch_ca0132(struct hda_codec *codec) 95c6e9cb7 Ian Minett 2011-06-15 7583 { 95c6e9cb7 Ian Minett 2011-06-15 7584 struct ca0132_spec *spec; a73d511c4 Ian Minett 2012-12-20 7585 int err; d5c016b56 Gabriele Martino 2015-05-18 7586 const struct snd_pci_quirk *quirk; 95c6e9cb7 Ian Minett 2011-06-15 7587 4e76a8833 Takashi Iwai 2014-02-25 7588 codec_dbg(codec, "patch_ca0132\n"); 95c6e9cb7 Ian Minett 2011-06-15 7589 95c6e9cb7 Ian Minett 2011-06-15 7590 spec = kzalloc(sizeof(*spec), GFP_KERNEL); 95c6e9cb7 Ian Minett 2011-06-15 7591 if (!spec) 95c6e9cb7 Ian Minett 2011-06-15 7592 return -ENOMEM; 95c6e9cb7 Ian Minett 2011-06-15 7593 codec->spec = spec; 993884f6a Chih-Chung Chang 2013-03-25 7594 spec->codec = codec; 95c6e9cb7 Ian Minett 2011-06-15 7595 225068ab2 Takashi Iwai 2015-05-29 7596 codec->patch_ops = ca0132_patch_ops; 225068ab2 Takashi Iwai 2015-05-29 7597 codec->pcm_format_first = 1; 225068ab2 Takashi Iwai 2015-05-29 7598 codec->no_sticky_stream = 1; 225068ab2 Takashi Iwai 2015-05-29 7599 d5c016b56 Gabriele Martino 2015-05-18 7600 /* Detect codec quirk */ d5c016b56 Gabriele Martino 2015-05-18 7601 quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks); d5c016b56 Gabriele Martino 2015-05-18 7602 if (quirk) d5c016b56 Gabriele Martino 2015-05-18 7603 spec->quirk = quirk->value; d5c016b56 Gabriele Martino 2015-05-18 7604 else d5c016b56 Gabriele Martino 2015-05-18 7605 spec->quirk = QUIRK_NONE; d5c016b56 Gabriele Martino 2015-05-18 7606 e24aa0a4c Takashi Iwai 2014-08-10 7607 spec->dsp_state = DSP_DOWNLOAD_INIT; a7e76271b Ian Minett 2012-12-20 7608 spec->num_mixers = 1; 017310fbe Connor McAdams 2018-05-08 7609 017310fbe Connor McAdams 2018-05-08 7610 /* Set which mixers each quirk uses. */ 017310fbe Connor McAdams 2018-05-08 7611 switch (spec->quirk) { 017310fbe Connor McAdams 2018-05-08 7612 case QUIRK_SBZ: e25e34450 Connor McAdams 2018-08-08 7613 spec->mixers[0] = desktop_mixer; 017310fbe Connor McAdams 2018-05-08 7614 snd_hda_codec_set_name(codec, "Sound Blaster Z"); 017310fbe Connor McAdams 2018-05-08 7615 break; e25e34450 Connor McAdams 2018-08-08 7616 case QUIRK_R3D: e25e34450 Connor McAdams 2018-08-08 7617 spec->mixers[0] = desktop_mixer; e25e34450 Connor McAdams 2018-08-08 7618 snd_hda_codec_set_name(codec, "Recon3D"); e25e34450 Connor McAdams 2018-08-08 7619 break; 017310fbe Connor McAdams 2018-05-08 7620 case QUIRK_R3DI: 017310fbe Connor McAdams 2018-05-08 7621 spec->mixers[0] = r3di_mixer; 017310fbe Connor McAdams 2018-05-08 7622 snd_hda_codec_set_name(codec, "Recon3Di"); 017310fbe Connor McAdams 2018-05-08 7623 break; 017310fbe Connor McAdams 2018-05-08 7624 default: a7e76271b Ian Minett 2012-12-20 7625 spec->mixers[0] = ca0132_mixer; 017310fbe Connor McAdams 2018-05-08 7626 break; 017310fbe Connor McAdams 2018-05-08 7627 } a7e76271b Ian Minett 2012-12-20 7628 08eca6b1f Connor McAdams 2018-08-08 7629 /* Setup whether or not to use alt functions/controls/pci_mmio */ 009b8f979 Connor McAdams 2018-05-08 7630 switch (spec->quirk) { 009b8f979 Connor McAdams 2018-05-08 7631 case QUIRK_SBZ: e42c7c731 Connor McAdams 2018-08-08 7632 case QUIRK_R3D: 08eca6b1f Connor McAdams 2018-08-08 7633 spec->use_alt_controls = true; 08eca6b1f Connor McAdams 2018-08-08 7634 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7635 spec->use_pci_mmio = true; 08eca6b1f Connor McAdams 2018-08-08 7636 break; 009b8f979 Connor McAdams 2018-05-08 7637 case QUIRK_R3DI: 47cdf76e4 Connor McAdams 2018-05-08 7638 spec->use_alt_controls = true; 009b8f979 Connor McAdams 2018-05-08 7639 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7640 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7641 break; 009b8f979 Connor McAdams 2018-05-08 7642 default: 47cdf76e4 Connor McAdams 2018-05-08 7643 spec->use_alt_controls = false; 009b8f979 Connor McAdams 2018-05-08 7644 spec->use_alt_functions = false; 08eca6b1f Connor McAdams 2018-08-08 7645 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7646 break; 009b8f979 Connor McAdams 2018-05-08 7647 } 009b8f979 Connor McAdams 2018-05-08 7648 08eca6b1f Connor McAdams 2018-08-08 7649 if (spec->use_pci_mmio) { 08eca6b1f Connor McAdams 2018-08-08 @7650 spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); 08eca6b1f Connor McAdams 2018-08-08 7651 if (spec->mem_base == NULL) { 08eca6b1f Connor McAdams 2018-08-08 7652 codec_warn(codec, "pci_iomap failed! Setting quirk to QUIRK_NONE."); 08eca6b1f Connor McAdams 2018-08-08 7653 spec->quirk = QUIRK_NONE; 08eca6b1f Connor McAdams 2018-08-08 7654 } 08eca6b1f Connor McAdams 2018-08-08 7655 } 08eca6b1f Connor McAdams 2018-08-08 7656 5aaca44d8 Ian Minett 2012-12-20 7657 spec->base_init_verbs = ca0132_base_init_verbs; 5aaca44d8 Ian Minett 2012-12-20 7658 spec->base_exit_verbs = ca0132_base_exit_verbs; 5aaca44d8 Ian Minett 2012-12-20 7659 993884f6a Chih-Chung Chang 2013-03-25 7660 INIT_DELAYED_WORK(&spec->unsol_hp_work, ca0132_unsol_hp_delayed); 993884f6a Chih-Chung Chang 2013-03-25 7661 95c6e9cb7 Ian Minett 2011-06-15 7662 ca0132_init_chip(codec); 95c6e9cb7 Ian Minett 2011-06-15 7663 95c6e9cb7 Ian Minett 2011-06-15 7664 ca0132_config(codec); 95c6e9cb7 Ian Minett 2011-06-15 7665 d5c016b56 Gabriele Martino 2015-05-18 7666 err = ca0132_prepare_verbs(codec); d5c016b56 Gabriele Martino 2015-05-18 7667 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7668 goto error; d5c016b56 Gabriele Martino 2015-05-18 7669 a73d511c4 Ian Minett 2012-12-20 7670 err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); a73d511c4 Ian Minett 2012-12-20 7671 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7672 goto error; a73d511c4 Ian Minett 2012-12-20 7673 95c6e9cb7 Ian Minett 2011-06-15 7674 return 0; cc91ceaf3 Takashi Iwai 2017-09-04 7675 cc91ceaf3 Takashi Iwai 2017-09-04 7676 error: cc91ceaf3 Takashi Iwai 2017-09-04 7677 ca0132_free(codec); cc91ceaf3 Takashi Iwai 2017-09-04 7678 return err; 95c6e9cb7 Ian Minett 2011-06-15 7679 } 95c6e9cb7 Ian Minett 2011-06-15 7680 :::::: The code at line 7650 was first introduced by commit :::::: 08eca6b1f1468a4021bac7b3929fd3eb491e2629 ALSA: hda/ca0132 - Add bool variable to enable/disable pci region2 mmio :::::: TO: Connor McAdams <conmanx360(a)gmail.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

4 7

[alsa-devel] [PATCH 0/2] ASoC: stm32: sai: fix kbuild errors
by Olivier Moysan 23 Oct '18

23 Oct '18

These patches fix following compilation errors: - >> sound/soc/stm/stm32_sai_sub.c:260:16: error: field 'hw' has incomplete type - >> sound/soc/stm/stm32_sai_sub.c:412:3: warning: 'strncat' specified bound 6 equals source length [-Wstringop-overflow=] Olivier Moysan (2): ASoC: stm32: add clock dependency for sai ASoC: stm32: sai: fix master clock naming sound/soc/stm/Kconfig | 1 + sound/soc/stm/stm32_sai_sub.c | 5 ++--- 2 files changed, 3 insertions(+), 3 deletions(-) -- 2.7.4

2 4

[alsa-devel] [asoc:for-4.20 519/526] sound/soc/stm/stm32_sai_sub.c:343 stm32_sai_mclk_set_rate() warn: unsigned 'div' is never less than zero.
by kbuild test robot 23 Oct '18

23 Oct '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 7f91e2af1a4a2c34fc2e8fb046c722e1a9c85399 commit: 8307b2afd386ccce369821daa2196068c47fe8cd [519/526] ASoC: stm32: sai: set sai as mclk clock provider smatch warnings: sound/soc/stm/stm32_sai_sub.c:343 stm32_sai_mclk_set_rate() warn: unsigned 'div' is never less than zero. vim +/div +343 sound/soc/stm/stm32_sai_sub.c 333 334 static int stm32_sai_mclk_set_rate(struct clk_hw *hw, unsigned long rate, 335 unsigned long parent_rate) 336 { 337 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 338 struct stm32_sai_sub_data *sai = mclk->sai_data; 339 unsigned int div; 340 int ret; 341 342 div = stm32_sai_get_clk_div(sai, parent_rate, rate); > 343 if (div < 0) 344 return div; 345 346 ret = stm32_sai_set_clk_div(sai, div); 347 if (ret) 348 return ret; 349 350 mclk->freq = rate; 351 352 return 0; 353 } 354 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [GIT PULL] ASoC updates for v5.0
by Mark Brown 23 Oct '18

23 Oct '18

The following changes since commit 35a7f35ad1b150ddf59a41dcac7b2fa32982be0e: Linux 4.19-rc8 (2018-10-15 07:20:24 +0200) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.0 for you to fetch changes up to 65dfb6d6dd2850f3f99f08536d2b14190350c854: Merge branch 'asoc-4.20' into asoc-next (2018-10-21 16:59:23 +0100) ---------------------------------------------------------------- ASoC: Updates for v5.0/v4.20 As ever there's a lot of small and driver specific changes going on here, but we do also have some relatively large changes in the core thanks to the hard work of Charles and Morimoto-san: - More component transitions from Morimoto-san, I think we're about finished with this. Thanks for all the hard work! - Morimoto-san also added a bunch of for_each_foo macros - A bunch of cleanups and fixes for DAPM from Charles. - MCLK support for several different devices, including CS42L51, STM32 SAI, and MAX98373. - Support for Allwinner A64 CODEC analog, Intel boards with DA7219 and MAX98927, Meson AXG PDM inputs, Nuvoton NAU8822, Renesas R8A7744 and TI PCM3060. ---------------------------------------------------------------- Akshu Agrawal (5): ASoC: AMD: Set constraints for DMIC and MAX98357a codec ASoC: AMD: Change MCLK to 48Mhz ASoC: AMD: Fix simultaneous playback and capture on different channel ASoC: AMD: Fix capture unstable in beginning for some runs ASoC: AMD: Add SND_JACK_LINEOUT jack type Alexandru Ardelean (1): ASoC: dt-bindings: add dt bindings for adau1977 audio codec Andreas Dannenberg (2): ASoC: codecs: tas5720: add TAS5722 specific volume control ASoC: codecs: tas5720: add TAS5722 TDM slot width setting support Andreas Färber (3): ASoC: dt-bindings: add max98088 audio codec ASoC: max98088: add OF support ASoC: max98088: Add master clock handling Andrew F. Davis (4): ASoC: tas6424: Print full register name in error message ASoC: tas6424: Add channel fault reporting ASoC: tlv320aic31xx: Add short circuit detection support ASoC: tlv320aic31xx: Add overflow detection support Arnd Bergmann (4): ASoC: pxa: avoid AC97_BUS build warning ASoC: atmel: add SND_SOC_I2C_AND_SPI dependency ASoC: wm9712: fix unused variable warning ASoC: max98988: add I2C dependency Biju Das (1): ASoC: rsnd: Add r8a7744 support Charles Keepax (11): ASoC: dapm: Remove clock framework ifdefs ASoC: dapm: Don't fail creating new DAPM control on NULL pinctrl ASoC: dpcm: Properly initialise hw->rate_max ASoC: dapm: Move error handling to snd_soc_dapm_new_control_unlocked ASoC: dapm: Cosmetic tidy up of snd_soc_dapm_new_control ASoC: dapm: Move connection of CODEC to CODEC DAIs ASoC: dapm: Add support for multi-CODEC CODEC to CODEC links ASoC: dapm: Move CODEC to CODEC params from the widget to the runtime ASoC: dapm: Avoid uninitialised variable warning ASoC: dapm: Add missing return value check for snd_soc_dapm_new_dai ASoC: dapm: Add support for hw_free on CODEC to CODEC links Christoph Hellwig (1): ASoC: intel: don't pass GFP_DMA32 to dma_alloc_coherent Codrin Ciubotariu (2): ASoC: Add driver for PROTO Audio CODEC (with a WM8731) ASoC: mikroe-proto: dt-bindings: add DT bindings for PROTO board Colin Ian King (2): ASoC: hdac_hdmi: remove redundant check for !port condition ASoC: sgtl5000: avoid division by zero if lo_vag is zero Dan Carpenter (4): ASoC: dapm: Fix a couple uninitialized ret variables ASoC: qdsp6: q6asm-dai: clean up a return ASoC: qdsp6: q6asm-dai: checking NULL vs IS_ERR() ASoC: Intel: kbl_da7219_max98927: minor white space clean up Daniel Mack (6): ASoC: pxa-ssp: enable and disable extclk if given ASoC: add fault detect recovery property to DT bindings ASoC: Add device tree documentation file for wm8782 stereo DAC ASoC: wm8782: add support for regulators ASoC: sta32x: set ->component pointer in private struct ASoC: sta32x: Add support for XTI clock Danny Smith (1): ASoC: adau17x1: Implemented safeload support David Lin (1): ASoC: nau8822: new codec driver Dmytro Prokopchuk (1): ASoC: rsnd: fixup SSI clock during suspend/resume modes Dragos Tarcatu (3): ASoC: rsnd: add support for 16/24 bit slot widths ASoC: rsnd: add support for the DSP_A/DSP_B formats ASoC: rsnd: add support for 8 bit S8 format Fabrizio Castro (1): ASoC: rsnd: Add r8a774a1 support Grant Grundler (1): ASoC: max98373: usleep_range() needs include/delay.h Gustavo A. R. Silva (2): ASoC: es8328: Fix fall-through annotations ASoC: hisilicon: fix fall-through annotations Hans de Goede (3): ASoC: rt5670: Add quirk for Thinkpad 8 tablet ASoC: Intel: common: Add quirk for Thinkpad 8 tablet ASoC: Intel: cht-bsw-rt5672: Add key-mappings for the headset buttons Hiroyuki Yokoyama (1): ASoC: rsnd: Add device tree binding for r8a77990 Jerome Brunet (7): ASoC: meson: imply clock and reset controllers ASoC: meson: axg-fifo: report interrupt request failure ASoC: meson: axg-tdm: restrict formats depending on slot width ASoC: dmic: add Kconfig prompt for the generic dmic codec. ASoC: dmic: add DT module alias ASoC: meson: add axg pdm input DT binding documentation ASoC: meson: add axg pdm input Jiada Wang (8): ASoC: rsnd: add warning message to rsnd_kctrl_accept_runtime() ASoC: rsnd: remove is_play parameter from hw_rule function ASoC: rsnd: ssi: Fix issue in dma data address assignment ASoC: rsnd: ssi: Check runtime channel number rather than hw_params ASoc: rsnd: dma: Calculate dma address with consider of BUSIF ASoc: rsnd: dma: Calculate PDMACHCRE with consider of BUSIF ASoC: rsnd: ssiu: Support BUSIF other than BUSIF0 ASoC: rsnd: ssiu: Support to init different BUSIF instance Jon Hunter (1): ASoC: core: Don't schedule DAPM work if already in target state Katsuhiro Suzuki (1): ASoC: rockchip: add missing INTERLEAVED PCM attribute Kirill Marinushkin (3): ASoC: pcm3060: Add codec driver ASoC: pcm3060: Improve stylistics of file comments ASoC: pcm3060: Improve legibility of if-statements Kuninori Morimoto (55): ASoC: simple_card_utils: support snd_soc_dai_link_component style for codec ASoC: simple-card: support snd_soc_dai_link_component style for codec ASoC: simple-scu-card: use simple_dai_props ASoC: simple-scu-card: support snd_soc_dai_link_component style for codec ASoC: audio-graph-card: support snd_soc_dai_link_component style for codec ASoC: audio-graph-scu-card: use simple_dai_props ASoC: audio-graph-scu-card: support snd_soc_dai_link_component style for codec ASoC: simple-card-util: remove dai_link compatible code for codec ASoC: soc-core: use snd_soc_dai_link_component for platform ASoC: simple-card-util: support snd_soc_dai_link_component style for platform ASoC: simple-card: support snd_soc_dai_link_component style for platform ASoC: simple-scu-card: support snd_soc_dai_link_component style for platform ASoC: audio-graph-card: support snd_soc_dai_link_component style for platform ASoC: audio-graph-scu-card: support snd_soc_dai_link_component style for platform ASoC: simple-card-util: remove dai_link compatible code for platform ASoC: rsnd: merge .nolock_start and .prepare ASoC: add for_each_link_codecs() macro ASoC: add for_each_rtd_codec_dai() macro ASoC: rsnd: gen: use tab instead of white-space ASoC: rsnd: don't use %p for dev_dbg() ASoC: atmel_ssc_dai: use devm_snd_soc_register_component() ASoC: bcm: use devm_snd_soc_register_component() ASoC: hdac_hda: use devm_snd_soc_register_component() ASoC: rt5668: use devm_snd_soc_register_component() ASoC: tscs454: use devm_snd_soc_register_component() ASoC: nuc900: use devm_snd_soc_register_component() ASoC: omap: use devm_snd_soc_register_component() ASoC: sh: use devm_snd_soc_register_component() ASoC: txx9: use devm_snd_soc_register_component() ASoC: rt5668: remove empty rt5668_i2c_remove() ASoC: soc-core: avoid nested code on soc_remove_dai() ASoC: soc-core: remove unused num_dai_links ASoC: soc-core: remove dai->driver NULL check ASoC: soc-core: manage platform name under snd_soc_init_platform() ASoC: soc-core: add snd_soc_is_matching_component() ASoC: convert for_each_rtd_codec_dai() for missing part ASoC: rename for_each_rtd_codec_dai_reverse to rollback ASoC: add for_each_card_prelinks() macro ASoC: add for_each_card_links() macro ASoC: add for_each_card_rtds() macro ASoC: add for_each_card_components() macro ASoC: add for_each_comp_order() macro ASoC: add for_each_dpcm_fe() macro ASoC: add for_each_dpcm_be() macro ASoC: tidyup for_each_card_prelinks() dai_link ASoC: add for_each_component() macro ASoC: add for_each_component_dais() macro ASoC: rsnd: use 32bit TDM width as default ASoC: audio-graph-card: enable mclk-fs on codec node ASoC: pcm3168a: add HW constraint for non RIGHT_J ASoC: pcm3168a: add hw constraint for channel ASoC: pcm3168a: add I2S/Left_J TDM support ASoC: pcm3168a: add hw constraint for capture channel ASoC: rsnd: enable TDM settings for SSI parent ASoC: rsnd: tidyup SSICR::SWSP for TDM Linus Walleij (1): ASoC: rt5677-spi: Drop unused GPIO include Lucas Tanure (1): ASoC: wm2000: Remove wm2000_read helper function Mac Chiang (2): ASoC: Intel: Boards: Add KBL Dialog Maxim I2S machine driver ASoC: Intel: common: Add Kabylake Dialog+Maxim machine driver entry Marcel Ziswiler (2): ASoC: tegra_sgtl5000: fix device_node refcounting ASoC: soc-core: fix trivial checkpatch issues Marco Felsch (2): ASoC: dt-bindings: max98088: add external clock binding ASoC: max98988: make it selectable Marcus Cooper (1): ASoC: sun4i-i2s: Add compatibility with A64 codec I2S Mark Brown (10): Merge tag 'v4.19-rc1' into asoc-4.19 Merge tag 'hda-codec-h-move' into asoc-4.20 Merge branch 'asoc-4.19' into asoc-4.20 tas dependency Merge branch 'asoc-4.19' into asoc-4.20 for rcar dep Merge branch 'asoc-4.19' into asoc-4.20 Cirrus conflict ALSA: hda: Fix implicit definition of pci_iomap() on SH Merge branch 'asoc-4.19' into asoc-4.20 for rt5682 deps. Merge remote-tracking branch 'asoc/for-4.19' into asoc-4.20 Merge branch 'asoc-4.19' into asoc-linus Merge branch 'asoc-4.20' into asoc-next Matt Flax (4): ASoC: cs4265: SOC_SINGLE register value error fix ASoC: cs4265: Add native 32bit I2S transport ASoC: cs4265: Add a S/PDIF enable switch ASoC: cs4265: Add a MIC pre. route Matthias Kaehlcke (1): ASoC: soc-utils: Rename dummy_dma_ops to snd_dummy_dma_ops Oder Chiou (2): ASoC: rt5514-spi: Get the period_bytes in the copy work to make sure the value correctly ASoC: rt5663: Remove the boost volume in the beginning of playback Olivier Moysan (5): ASoC: dt-bindings: add mclk provider support to stm32 sai ASoC: dt-bindings: add mclk support to cs42l51 ASoC: stm32: sai: set sai as mclk clock provider ASoC: cs42l51: add mclk support ASoC: cs42l51: fix mclk support Peter Rosin (1): ASoC: atmel: tse850: switch to SPDX license identifier Peter Ujfalusi (1): ASoC: davinci-mcasp: Add support for FIFO usage caused delay reporting Pierre-Louis Bossart (3): ALSA: hda: move hda_codec.h to include/sound ASoC: Intel: common: add table for HDA-based platforms ALSA: hda: move hda_codec.h to include/sound Rakesh Ughreja (5): ASoC: Intel: Boards: Machine driver for SKL+ w/ HDAudio codecs ASoC: Intel: Skylake: use HDAudio if ACPI enumeration fails ASoC: Intel: Skylake: add HDA BE DAIs ASoC: Intel: Skylake: use hda_bus instead of hdac_bus ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers Randy Dunlap (1): ASoC: fix soc-core.c kernel-doc warning Richard Fitzgerald (2): ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate Rob Herring (1): ASoC: Convert to using %pOFn instead of device_node.name Robert Jarzmik (1): ASoC: pxa: switch to new ac97 bus support Robert Rosengren (1): ASoC: adau17x1: Unused exported functions changed to internal Rohit kumar (2): ASoC: Fix UBSAN warning at snd_soc_get/put_volsw_sx() ASoC: q6afe: dt-bindings: Update input range for qcom,sd-lines Ryan Lee (4): ASoC: max98373: Sort Digital Volume in reverse order ASoC: max98373: Sort BDE Limiter Thresh Volume in reverse order ASoC: max98373: Sort max98373_bde_gain_tlv in reverse order ASoC: max98373: Sort DHT Rot Pnt Volume in reverse order Shuming Fan (6): ASoC: rt5682: Update calibration function ASoC: rt5682: Improve HP performance ASoC: rt5682: Remove HP volume control ASoC: rt5682: Update calibration function ASoC: rt5682: Fix the boost volume at the begining of playback ASoC: rt5682: Minor code modification Srinivas Kandagatla (1): ASoC: core: add support to card rebind Takashi Iwai (6): ASoC: intel: skylake: Add missing break in skl_tplg_get_token() ASoC: adau1761: Use the standard fall-through annotation ASoC: pcm186x: Use the standard fall-through annotation ASoC: rt274: Add fall-through annotations ASoC: intel: skylake: Add fall-through annotation ASoC: topology: Use the standard fall-through annotations Tzung-Bi Shih (1): ASoC: ts3a227e: fix a typo in documentation Vasily Khoruzhick (6): ASoC: sun8i-codec: Don't hardcode BCLK / LRCK ratio ASoC: sun8i-codec-analog: split regmap code into separate driver ASoC: dt-binding: Add bindings for Allwinner A64 codec's analog path controls ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64 ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks Yong Zhi (1): ASoC: Intel: hdac_hdmi: Limit sampling rates at dai creation YueHaibing (2): ASoC: qdsp6: q6asm-dai: remove duplicated include from q6asm-dai.c ASoC: qcom: qdsp6: remove duplicated include zhong jiang (5): ASoC: q6core: Use kmemdup to replace kzalloc + memcpy ASoC: skl-topology: Use kmemdup to replace kzalloc + memcpy ASoC: remove unneeded static set .owner field in platform_driver ASoC: remove redundant include ASoC: qcom: qdsp6: remove duplicated include from q6adm.c .../devicetree/bindings/sound/adi,adau1977.txt | 54 + .../devicetree/bindings/sound/amlogic,axg-pdm.txt | 24 + .../devicetree/bindings/sound/cs42l51.txt | 17 + .../devicetree/bindings/sound/maxim,max98088.txt | 23 + .../bindings/sound/mikroe,mikroe-proto.txt | 23 + .../devicetree/bindings/sound/nau8822.txt | 16 + .../devicetree/bindings/sound/pcm3060.txt | 17 + .../devicetree/bindings/sound/qcom,q6afe.txt | 18 +- .../devicetree/bindings/sound/renesas,rsnd.txt | 5 +- .../devicetree/bindings/sound/st,sta32x.txt | 9 + .../devicetree/bindings/sound/st,stm32-sai.txt | 7 + .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + .../bindings/sound/sun50i-codec-analog.txt | 12 + .../devicetree/bindings/sound/ts3a227e.txt | 2 +- Documentation/devicetree/bindings/sound/wm8782.txt | 17 + .../devicetree/bindings/trivial-devices.txt | 1 - .../devicetree/bindings/vendor-prefixes.txt | 1 + MAINTAINERS | 7 + {sound/pci/hda => include/sound}/hda_codec.h | 0 include/sound/simple_card_utils.h | 27 +- include/sound/soc-acpi-intel-match.h | 6 + include/sound/soc-dapm.h | 9 - include/sound/soc-dpcm.h | 10 + include/sound/soc.h | 45 +- sound/arm/Kconfig | 1 - sound/pci/hda/hda_auto_parser.c | 2 +- sound/pci/hda/hda_beep.h | 2 +- sound/pci/hda/hda_bind.c | 14 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_eld.c | 2 +- sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/hda_hwdep.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- sound/pci/hda/hda_jack.c | 2 +- sound/pci/hda/hda_proc.c | 2 +- sound/pci/hda/hda_sysfs.c | 2 +- sound/pci/hda/hda_tegra.c | 2 +- sound/pci/hda/patch_analog.c | 2 +- sound/pci/hda/patch_ca0110.c | 2 +- sound/pci/hda/patch_ca0132.c | 3 +- sound/pci/hda/patch_cirrus.c | 2 +- sound/pci/hda/patch_cmedia.c | 2 +- sound/pci/hda/patch_conexant.c | 2 +- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_si3054.c | 2 +- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/hda/patch_via.c | 2 +- sound/soc/amd/acp-da7219-max98357a.c | 77 +- sound/soc/amd/acp-pcm-dma.c | 30 +- sound/soc/amd/acp.h | 3 +- sound/soc/atmel/Kconfig | 12 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel_ssc_dai.c | 13 +- sound/soc/atmel/mikroe-proto.c | 165 +++ sound/soc/atmel/tse850-pcm5142.c | 78 +- sound/soc/bcm/cygnus-ssp.c | 13 +- sound/soc/codecs/Kconfig | 36 +- sound/soc/codecs/Makefile | 10 + sound/soc/codecs/adau1761.c | 3 +- sound/soc/codecs/adau17x1.c | 86 +- sound/soc/codecs/adau17x1.h | 4 - sound/soc/codecs/cs4265.c | 12 +- sound/soc/codecs/cs42l51.c | 21 + sound/soc/codecs/dmic.c | 1 + sound/soc/codecs/es8328.c | 4 +- sound/soc/codecs/hdac_hda.c | 483 +++++++++ sound/soc/codecs/hdac_hda.h | 24 + sound/soc/codecs/hdac_hdmi.c | 11 +- sound/soc/codecs/max98088.c | 36 + sound/soc/codecs/max98373.c | 47 +- sound/soc/codecs/nau8822.c | 1136 ++++++++++++++++++++ sound/soc/codecs/nau8822.h | 204 ++++ sound/soc/codecs/pcm186x.c | 3 +- sound/soc/codecs/pcm3060-i2c.c | 60 ++ sound/soc/codecs/pcm3060-spi.c | 59 + sound/soc/codecs/pcm3060.c | 295 +++++ sound/soc/codecs/pcm3060.h | 88 ++ sound/soc/codecs/pcm3168a.c | 82 ++ sound/soc/codecs/rt274.c | 2 + sound/soc/codecs/rt5514-spi.c | 14 +- sound/soc/codecs/rt5651.c | 1 - sound/soc/codecs/rt5663.c | 7 +- sound/soc/codecs/rt5668.c | 10 +- sound/soc/codecs/rt5670.c | 12 + sound/soc/codecs/rt5677-spi.c | 1 - sound/soc/codecs/rt5682.c | 86 +- sound/soc/codecs/rt5682.h | 14 + sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/sta32x.c | 30 + sound/soc/codecs/tas5720.c | 103 +- sound/soc/codecs/tas6424.c | 58 +- sound/soc/codecs/tas6424.h | 10 + sound/soc/codecs/tlv320aic31xx.c | 85 ++ sound/soc/codecs/tlv320aic31xx.h | 23 + sound/soc/codecs/tscs454.c | 2 +- sound/soc/codecs/wm2000.c | 54 +- sound/soc/codecs/wm8782.c | 63 ++ sound/soc/codecs/wm8904.c | 1 - sound/soc/codecs/wm8974.c | 1 - sound/soc/codecs/wm9712.c | 3 +- sound/soc/codecs/wm_adsp.c | 26 +- sound/soc/davinci/davinci-mcasp.c | 37 + sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_utils.c | 4 +- sound/soc/fsl/pcm030-audio-fabric.c | 5 +- sound/soc/generic/audio-graph-card.c | 21 +- sound/soc/generic/audio-graph-scu-card.c | 55 +- sound/soc/generic/simple-card-utils.c | 53 +- sound/soc/generic/simple-card.c | 30 +- sound/soc/generic/simple-scu-card.c | 54 +- sound/soc/hisilicon/hi6210-i2s.c | 4 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 4 +- sound/soc/intel/boards/Kconfig | 22 + sound/soc/intel/boards/Makefile | 4 + sound/soc/intel/boards/broadwell.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 4 +- sound/soc/intel/boards/bytcr_rt5651.c | 4 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 9 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 983 +++++++++++++++++ sound/soc/intel/boards/kbl_rt5663_max98927.c | 5 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 5 +- sound/soc/intel/boards/skl_hda_dsp_common.c | 127 +++ sound/soc/intel/boards/skl_hda_dsp_common.h | 38 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 183 ++++ sound/soc/intel/common/Makefile | 3 +- sound/soc/intel/common/soc-acpi-intel-byt-match.c | 7 + sound/soc/intel/common/soc-acpi-intel-hda-match.c | 40 + sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 13 + sound/soc/intel/common/sst-firmware.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 71 +- sound/soc/intel/skylake/skl-topology.c | 4 +- sound/soc/intel/skylake/skl.c | 96 +- sound/soc/intel/skylake/skl.h | 12 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 13 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 14 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 14 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 13 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 12 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 12 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 12 +- sound/soc/meson/Kconfig | 13 + sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-card.c | 16 +- sound/soc/meson/axg-fifo.c | 2 + sound/soc/meson/axg-pdm.c | 654 +++++++++++ sound/soc/meson/axg-tdm-interface.c | 50 +- sound/soc/nuc900/nuc900-ac97.c | 4 +- sound/soc/omap/omap-hdmi-audio.c | 4 +- sound/soc/pxa/Kconfig | 13 +- sound/soc/pxa/pxa-ssp.c | 6 + sound/soc/pxa/pxa2xx-ac97.c | 48 +- sound/soc/qcom/apq8096.c | 7 +- sound/soc/qcom/qdsp6/q6adm.c | 17 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 8 +- sound/soc/qcom/qdsp6/q6asm.c | 1 - sound/soc/qcom/qdsp6/q6core.c | 9 +- sound/soc/qcom/sdm845.c | 7 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 1 - sound/soc/rockchip/rockchip_pcm.c | 3 +- sound/soc/samsung/tm2_wm5110.c | 13 +- sound/soc/sh/hac.c | 3 +- sound/soc/sh/rcar/adg.c | 4 +- sound/soc/sh/rcar/core.c | 124 ++- sound/soc/sh/rcar/ctu.c | 2 +- sound/soc/sh/rcar/dma.c | 109 +- sound/soc/sh/rcar/gen.c | 33 +- sound/soc/sh/rcar/rsnd.h | 63 +- sound/soc/sh/rcar/src.c | 2 +- sound/soc/sh/rcar/ssi.c | 112 +- sound/soc/sh/rcar/ssiu.c | 92 +- sound/soc/soc-compress.c | 4 +- sound/soc/soc-core.c | 582 +++++----- sound/soc/soc-dapm.c | 437 ++++---- sound/soc/soc-ops.c | 4 +- sound/soc/soc-pcm.c | 253 +++-- sound/soc/soc-topology.c | 15 +- sound/soc/soc-utils.c | 4 +- sound/soc/stm/stm32_sai.c | 2 +- sound/soc/stm/stm32_sai.h | 3 + sound/soc/stm/stm32_sai_sub.c | 282 ++++- sound/soc/sunxi/Kconfig | 17 +- sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-i2s.c | 82 +- sound/soc/sunxi/sun50i-codec-analog.c | 444 ++++++++ sound/soc/sunxi/sun8i-adda-pr-regmap.c | 102 ++ sound/soc/sunxi/sun8i-adda-pr-regmap.h | 7 + sound/soc/sunxi/sun8i-codec-analog.c | 79 +- sound/soc/sunxi/sun8i-codec.c | 22 +- sound/soc/tegra/tegra_sgtl5000.c | 17 +- sound/soc/txx9/txx9aclc-ac97.c | 3 +- 193 files changed, 8358 insertions(+), 1542 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau1977.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt create mode 100644 Documentation/devicetree/bindings/sound/cs42l51.txt create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98088.txt create mode 100644 Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt create mode 100644 Documentation/devicetree/bindings/sound/nau8822.txt create mode 100644 Documentation/devicetree/bindings/sound/pcm3060.txt create mode 100644 Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8782.txt rename {sound/pci/hda => include/sound}/hda_codec.h (100%) create mode 100644 sound/soc/atmel/mikroe-proto.c create mode 100644 sound/soc/codecs/hdac_hda.c create mode 100644 sound/soc/codecs/hdac_hda.h create mode 100644 sound/soc/codecs/nau8822.c create mode 100644 sound/soc/codecs/nau8822.h create mode 100644 sound/soc/codecs/pcm3060-i2c.c create mode 100644 sound/soc/codecs/pcm3060-spi.c create mode 100644 sound/soc/codecs/pcm3060.c create mode 100644 sound/soc/codecs/pcm3060.h create mode 100644 sound/soc/intel/boards/kbl_da7219_max98927.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_generic.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-hda-match.c create mode 100644 sound/soc/meson/axg-pdm.c create mode 100644 sound/soc/sunxi/sun50i-codec-analog.c create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.c create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.h

2 1

[alsa-devel] [PATCH] ALSA: hda/ca0132 - Actually fix microphone issue
by Connor McAdams 22 Oct '18

22 Oct '18

This patch fixes the microphone issue for all cards. The previous fix worked on the ZxR, but not on the AE-5 or Z. This patch has been tested to work for all cards. Signed-off-by: Connor McAdams <conmanx360(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 73 ++++++++++++++++++++++---------------------- 1 file changed, 37 insertions(+), 36 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 1a13cea..f0ef52e 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -6983,37 +6983,44 @@ static void sbz_chipio_startup_data(struct hda_codec *codec) static void ca0132_alt_dsp_scp_startup(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - unsigned int tmp; + unsigned int tmp, i; - switch (spec->quirk) { - case QUIRK_SBZ: - case QUIRK_AE5: - tmp = 0x00000003; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - tmp = 0x00000000; - dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp); - tmp = 0x00000001; - dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp); - tmp = 0x00000004; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - tmp = 0x00000005; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - tmp = 0x00000000; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - break; - case QUIRK_R3D: - case QUIRK_R3DI: - tmp = 0x00000000; - dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp); - tmp = 0x00000001; - dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp); - tmp = 0x00000004; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - tmp = 0x00000005; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - tmp = 0x00000000; - dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); - break; + /* + * Gotta run these twice, or else mic works inconsistently. Not clear + * why this is, but multiple tests have confirmed it. + */ + for (i = 0; i < 2; i++) { + switch (spec->quirk) { + case QUIRK_SBZ: + case QUIRK_AE5: + tmp = 0x00000003; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp); + tmp = 0x00000001; + dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp); + tmp = 0x00000004; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + tmp = 0x00000005; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + break; + case QUIRK_R3D: + case QUIRK_R3DI: + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp); + tmp = 0x00000001; + dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp); + tmp = 0x00000004; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + tmp = 0x00000005; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + break; + } + msleep(100); } } @@ -7246,8 +7253,6 @@ static void r3d_setup_defaults(struct hda_codec *codec) int num_fx; int idx, i; - msleep(100); - if (spec->dsp_state != DSP_DOWNLOADED) return; @@ -7292,8 +7297,6 @@ static void sbz_setup_defaults(struct hda_codec *codec) int num_fx; int idx, i; - msleep(100); - if (spec->dsp_state != DSP_DOWNLOADED) return; @@ -7351,8 +7354,6 @@ static void ae5_setup_defaults(struct hda_codec *codec) int num_fx; int idx, i; - msleep(100); - if (spec->dsp_state != DSP_DOWNLOADED) return; -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks" to the asoc tree
by Mark Brown 21 Oct '18

21 Oct '18

The patch ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks 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 7f91e2af1a4a2c34fc2e8fb046c722e1a9c85399 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick <anarsoul(a)gmail.com> Date: Sun, 21 Oct 2018 08:39:11 -0700 Subject: [PATCH] ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks startup() and shutdown() hooks are called for both substreams, so stopping either substream when another is running breaks the latter. E.g. playback breaks if capture is stopped when playback is running. Move code from startup() and shutdown() to resume() and suspend() hooks respectively to fix this issue Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Acked-by: Maxime Ripard <maxime.ripard(a)bootlin.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sunxi/sun4i-i2s.c | 61 +++++++++++++++---------------------- 1 file changed, 25 insertions(+), 36 deletions(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index c63d226e2436..d5ec1a20499d 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -644,40 +644,6 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int sun4i_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - /* Enable the whole hardware block */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN); - - /* Enable the first output line */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_SDO_EN_MASK, - SUN4I_I2S_CTRL_SDO_EN(0)); - - - return clk_prepare_enable(i2s->mod_clk); -} - -static void sun4i_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - clk_disable_unprepare(i2s->mod_clk); - - /* Disable our output lines */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_SDO_EN_MASK, 0); - - /* Disable the whole hardware block */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_GL_EN, 0); -} - static int sun4i_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { @@ -695,8 +661,6 @@ static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = { .hw_params = sun4i_i2s_hw_params, .set_fmt = sun4i_i2s_set_fmt, .set_sysclk = sun4i_i2s_set_sysclk, - .shutdown = sun4i_i2s_shutdown, - .startup = sun4i_i2s_startup, .trigger = sun4i_i2s_trigger, }; @@ -869,6 +833,21 @@ static int sun4i_i2s_runtime_resume(struct device *dev) goto err_disable_clk; } + /* Enable the whole hardware block */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN); + + /* Enable the first output line */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_SDO_EN_MASK, + SUN4I_I2S_CTRL_SDO_EN(0)); + + ret = clk_prepare_enable(i2s->mod_clk); + if (ret) { + dev_err(dev, "Failed to enable module clock\n"); + goto err_disable_clk; + } + return 0; err_disable_clk: @@ -880,6 +859,16 @@ static int sun4i_i2s_runtime_suspend(struct device *dev) { struct sun4i_i2s *i2s = dev_get_drvdata(dev); + clk_disable_unprepare(i2s->mod_clk); + + /* Disable our output lines */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_SDO_EN_MASK, 0); + + /* Disable the whole hardware block */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_GL_EN, 0); + regcache_cache_only(i2s->regmap, true); clk_disable_unprepare(i2s->bus_clk); -- 2.19.0

1 0

[alsa-devel] [PATCH RESEND v2] ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks
by Vasily Khoruzhick 21 Oct '18

21 Oct '18

startup() and shutdown() hooks are called for both substreams, so stopping either substream when another is running breaks the latter. E.g. playback breaks if capture is stopped when playback is running. Move code from startup() and shutdown() to resume() and suspend() hooks respectively to fix this issue Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Acked-by: Maxime Ripard <maxime.ripard(a)bootlin.com> --- v2: - rename from "ASoC: sun4i-i2s: don't try to start up or shut down DAI if it's active" - move code from startup/shutdown hooks into pm_runtime sound/soc/sunxi/sun4i-i2s.c | 61 +++++++++++++++---------------------- 1 file changed, 25 insertions(+), 36 deletions(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index a4aa931ebfae..daa6c08cffbc 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -644,40 +644,6 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int sun4i_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - /* Enable the whole hardware block */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN); - - /* Enable the first output line */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_SDO_EN_MASK, - SUN4I_I2S_CTRL_SDO_EN(0)); - - - return clk_prepare_enable(i2s->mod_clk); -} - -static void sun4i_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - clk_disable_unprepare(i2s->mod_clk); - - /* Disable our output lines */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_SDO_EN_MASK, 0); - - /* Disable the whole hardware block */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_GL_EN, 0); -} - static int sun4i_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { @@ -695,8 +661,6 @@ static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = { .hw_params = sun4i_i2s_hw_params, .set_fmt = sun4i_i2s_set_fmt, .set_sysclk = sun4i_i2s_set_sysclk, - .shutdown = sun4i_i2s_shutdown, - .startup = sun4i_i2s_startup, .trigger = sun4i_i2s_trigger, }; @@ -869,6 +833,21 @@ static int sun4i_i2s_runtime_resume(struct device *dev) goto err_disable_clk; } + /* Enable the whole hardware block */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN); + + /* Enable the first output line */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_SDO_EN_MASK, + SUN4I_I2S_CTRL_SDO_EN(0)); + + ret = clk_prepare_enable(i2s->mod_clk); + if (ret) { + dev_err(dev, "Failed to enable module clock\n"); + goto err_disable_clk; + } + return 0; err_disable_clk: @@ -880,6 +859,16 @@ static int sun4i_i2s_runtime_suspend(struct device *dev) { struct sun4i_i2s *i2s = dev_get_drvdata(dev); + clk_disable_unprepare(i2s->mod_clk); + + /* Disable our output lines */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_SDO_EN_MASK, 0); + + /* Disable the whole hardware block */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_GL_EN, 0); + regcache_cache_only(i2s->regmap, true); clk_disable_unprepare(i2s->bus_clk); -- 2.19.0

1 0

[alsa-devel] [PATCH v2] ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks
by Vasily Khoruzhick 21 Oct '18

21 Oct '18

startup() and shutdown() hooks are called for both substreams, so stopping either substream when another is running breaks the latter. E.g. playback breaks if capture is stopped when playback is running. Move code from startup() and shutdown() to resume() and suspend() hooks respectively to fix this issue Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- v2: - rename from "ASoC: sun4i-i2s: don't try to start up or shut down DAI if it's active" - move code from startup/shutdown hooks into pm_runtime sound/soc/sunxi/sun4i-i2s.c | 61 +++++++++++++++---------------------- 1 file changed, 25 insertions(+), 36 deletions(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index a4aa931ebfae..daa6c08cffbc 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -644,40 +644,6 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int sun4i_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - /* Enable the whole hardware block */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN); - - /* Enable the first output line */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_SDO_EN_MASK, - SUN4I_I2S_CTRL_SDO_EN(0)); - - - return clk_prepare_enable(i2s->mod_clk); -} - -static void sun4i_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); - - clk_disable_unprepare(i2s->mod_clk); - - /* Disable our output lines */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_SDO_EN_MASK, 0); - - /* Disable the whole hardware block */ - regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, - SUN4I_I2S_CTRL_GL_EN, 0); -} - static int sun4i_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { @@ -695,8 +661,6 @@ static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = { .hw_params = sun4i_i2s_hw_params, .set_fmt = sun4i_i2s_set_fmt, .set_sysclk = sun4i_i2s_set_sysclk, - .shutdown = sun4i_i2s_shutdown, - .startup = sun4i_i2s_startup, .trigger = sun4i_i2s_trigger, }; @@ -869,6 +833,21 @@ static int sun4i_i2s_runtime_resume(struct device *dev) goto err_disable_clk; } + /* Enable the whole hardware block */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN); + + /* Enable the first output line */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_SDO_EN_MASK, + SUN4I_I2S_CTRL_SDO_EN(0)); + + ret = clk_prepare_enable(i2s->mod_clk); + if (ret) { + dev_err(dev, "Failed to enable module clock\n"); + goto err_disable_clk; + } + return 0; err_disable_clk: @@ -880,6 +859,16 @@ static int sun4i_i2s_runtime_suspend(struct device *dev) { struct sun4i_i2s *i2s = dev_get_drvdata(dev); + clk_disable_unprepare(i2s->mod_clk); + + /* Disable our output lines */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_SDO_EN_MASK, 0); + + /* Disable the whole hardware block */ + regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, + SUN4I_I2S_CTRL_GL_EN, 0); + regcache_cache_only(i2s->regmap, true); clk_disable_unprepare(i2s->bus_clk); -- 2.19.0

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[alsa-devel] Applied "ASoC: wm2000: Remove wm2000_read helper function" to the asoc tree
by Mark Brown 21 Oct '18

21 Oct '18

The patch ASoC: wm2000: Remove wm2000_read helper function 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 c5d09485def41cab9e75ba23abbf87080183183c Mon Sep 17 00:00:00 2001 From: Lucas Tanure <tanureal(a)opensource.cirrus.com> Date: Fri, 19 Oct 2018 17:44:22 +0100 Subject: [PATCH] ASoC: wm2000: Remove wm2000_read helper function The return type "unsigned int" was used by the wm2000_read() function despite of the aspect that it will eventually return a negative error code. The resulting function doesn't add much to the code, so replace wm2000_read with regmap_read. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm2000.c | 54 ++++++++++++++++++++++----------------- 1 file changed, 30 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index c5ae07234a00..bba330e30162 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -88,19 +88,6 @@ static int wm2000_write(struct i2c_client *i2c, unsigned int reg, return regmap_write(wm2000->regmap, reg, value); } -static unsigned int wm2000_read(struct i2c_client *i2c, unsigned int r) -{ - struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c); - unsigned int val; - int ret; - - ret = regmap_read(wm2000->regmap, r, &val); - if (ret < 0) - return -1; - - return val; -} - static void wm2000_reset(struct wm2000_priv *wm2000) { struct i2c_client *i2c = wm2000->i2c; @@ -115,14 +102,15 @@ static void wm2000_reset(struct wm2000_priv *wm2000) static int wm2000_poll_bit(struct i2c_client *i2c, unsigned int reg, u8 mask) { + struct wm2000_priv *wm2000 = i2c_get_clientdata(i2c); int timeout = 4000; - int val; + unsigned int val; - val = wm2000_read(i2c, reg); + regmap_read(wm2000->regmap, reg, &val); while (!(val & mask) && --timeout) { msleep(1); - val = wm2000_read(i2c, reg); + regmap_read(wm2000->regmap, reg, &val); } if (timeout == 0) @@ -135,6 +123,7 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); unsigned long rate; + unsigned int val; int ret; if (WARN_ON(wm2000->anc_mode != ANC_OFF)) @@ -213,12 +202,17 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) WM2000_MODE_THERMAL_ENABLE); } - ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY); + ret = regmap_read(wm2000->regmap, WM2000_REG_SPEECH_CLARITY, &val); + if (ret != 0) { + dev_err(&i2c->dev, "Unable to read Speech Clarity: %d\n", ret); + regulator_bulk_disable(WM2000_NUM_SUPPLIES, wm2000->supplies); + return ret; + } if (wm2000->speech_clarity) - ret |= WM2000_SPEECH_CLARITY; + val |= WM2000_SPEECH_CLARITY; else - ret &= ~WM2000_SPEECH_CLARITY; - wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret); + val &= ~WM2000_SPEECH_CLARITY; + wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, val); wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33); wm2000_write(i2c, WM2000_REG_SYS_START1, 0x02); @@ -824,7 +818,7 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, const char *filename; const struct firmware *fw = NULL; int ret, i; - int reg; + unsigned int reg; u16 id; wm2000 = devm_kzalloc(&i2c->dev, sizeof(*wm2000), GFP_KERNEL); @@ -860,9 +854,17 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, } /* Verify that this is a WM2000 */ - reg = wm2000_read(i2c, WM2000_REG_ID1); + ret = regmap_read(wm2000->regmap, WM2000_REG_ID1, &reg); + if (ret != 0) { + dev_err(&i2c->dev, "Unable to read ID1: %d\n", ret); + return ret; + } id = reg << 8; - reg = wm2000_read(i2c, WM2000_REG_ID2); + ret = regmap_read(wm2000->regmap, WM2000_REG_ID2, &reg); + if (ret != 0) { + dev_err(&i2c->dev, "Unable to read ID2: %d\n", ret); + return ret; + } id |= reg & 0xff; if (id != 0x2000) { @@ -871,7 +873,11 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, goto err_supplies; } - reg = wm2000_read(i2c, WM2000_REG_REVISON); + ret = regmap_read(wm2000->regmap, WM2000_REG_REVISON, &reg); + if (ret != 0) { + dev_err(&i2c->dev, "Unable to read Revision: %d\n", ret); + return ret; + } dev_info(&i2c->dev, "revision %c\n", reg + 'A'); wm2000->mclk = devm_clk_get(&i2c->dev, "MCLK"); -- 2.19.0

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Re: [alsa-devel] [PATCH] ASoC: wm2000: Remove wm2000_read helper function
by Charles Keepax 21 Oct '18

21 Oct '18

On Fri, Oct 19, 2018 at 05:44:22PM +0100, Lucas Tanure wrote: > The return type "unsigned int" was used by the wm2000_read() > function despite of the aspect that it will eventually return > a negative error code. > The resulting function doesn't add much to the code, so replace > wm2000_read with regmap_read. > > Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> > --- Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Thanks, Charles

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Re: [alsa-devel] Tascam FW-1884 driver
by Takashi Sakamoto 20 Oct '18

20 Oct '18

Hi, (C.C.ed to alsa-devel, ffado-devel and Scott) On Sat, 20 Oct 2018, Cédric Mialaret wrote: > I stumbled up the source code of a linux driver for the Tascam FW-1884 audio interface: > > https://elixir.bootlin.com/linux/v4.19-rc8/source/sound/firewire/tascam/tas… > > a. Does it work ? It, ALSA firewire-tascam driver, works for systems in which Linux kernel is used. It's already included in Linux kernel source since kernel v4.4. > b. Does it implement all functionalities of the FW-1884 ? As of kernel v4.19, it supports below functionalities: - data transmission for PCM frames via ALSA PCM interface - data transmission for MIDI messages via ALSA rawmidi interface Recently I got Scott's help and work to support handling control messages from faders/knobs/buttons on control surface of FW-1884/FW-1082. You can see recent patches in my remote repository: - https://github.com/takaswie/snd-firewire-improve/tree/topic/tascam-userspac… - https://github.com/takaswie/libhinawa/tree/topic/tascam-userspace-take3 You can see our discussion in list: - http://mailman.alsa-project.org/pipermail/alsa-devel/2018-October/140593.ht… > c. Is this the original code from Tascam/Frontier Design? Apparently, no. It comes from my work for reverse engineering, by sniffing packet communications over IEEE 1394 bus. > d. Do you by any chance have the source code of the Windows driver ? I have no such chance, as well as the others. Regards Takashi Sakamoto

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[alsa-devel] [asoc:for-4.20 333/524] sound/pci/hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'?
by kbuild test robot 20 Oct '18

20 Oct '18

Hi Rakesh, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 318e741ee13b5a72f3051d9bb6852b1f4d02d0bb commit: 6bae5ea9498926440ffc883f3dbceb0adc65e492 [333/524] ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers config: sh-allyesconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 6bae5ea9498926440ffc883f3dbceb0adc65e492 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): sound/pci/hda/patch_ca0132.c: In function 'patch_ca0132': >> sound/pci/hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'? [-Werror=implicit-function-declaration] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^~~~~~~~~ pcim_iomap sound/pci/hda/patch_ca0132.c:7650:18: warning: assignment makes pointer from integer without a cast [-Wint-conversion] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^ cc1: some warnings being treated as errors vim +7650 sound/pci/hda/patch_ca0132.c d5c016b56 Gabriele Martino 2015-05-18 7581 95c6e9cb7 Ian Minett 2011-06-15 7582 static int patch_ca0132(struct hda_codec *codec) 95c6e9cb7 Ian Minett 2011-06-15 7583 { 95c6e9cb7 Ian Minett 2011-06-15 7584 struct ca0132_spec *spec; a73d511c4 Ian Minett 2012-12-20 7585 int err; d5c016b56 Gabriele Martino 2015-05-18 7586 const struct snd_pci_quirk *quirk; 95c6e9cb7 Ian Minett 2011-06-15 7587 4e76a8833 Takashi Iwai 2014-02-25 7588 codec_dbg(codec, "patch_ca0132\n"); 95c6e9cb7 Ian Minett 2011-06-15 7589 95c6e9cb7 Ian Minett 2011-06-15 7590 spec = kzalloc(sizeof(*spec), GFP_KERNEL); 95c6e9cb7 Ian Minett 2011-06-15 7591 if (!spec) 95c6e9cb7 Ian Minett 2011-06-15 7592 return -ENOMEM; 95c6e9cb7 Ian Minett 2011-06-15 7593 codec->spec = spec; 993884f6a Chih-Chung Chang 2013-03-25 7594 spec->codec = codec; 95c6e9cb7 Ian Minett 2011-06-15 7595 225068ab2 Takashi Iwai 2015-05-29 7596 codec->patch_ops = ca0132_patch_ops; 225068ab2 Takashi Iwai 2015-05-29 7597 codec->pcm_format_first = 1; 225068ab2 Takashi Iwai 2015-05-29 7598 codec->no_sticky_stream = 1; 225068ab2 Takashi Iwai 2015-05-29 7599 d5c016b56 Gabriele Martino 2015-05-18 7600 /* Detect codec quirk */ d5c016b56 Gabriele Martino 2015-05-18 7601 quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks); d5c016b56 Gabriele Martino 2015-05-18 7602 if (quirk) d5c016b56 Gabriele Martino 2015-05-18 7603 spec->quirk = quirk->value; d5c016b56 Gabriele Martino 2015-05-18 7604 else d5c016b56 Gabriele Martino 2015-05-18 7605 spec->quirk = QUIRK_NONE; d5c016b56 Gabriele Martino 2015-05-18 7606 e24aa0a4c Takashi Iwai 2014-08-10 7607 spec->dsp_state = DSP_DOWNLOAD_INIT; a7e76271b Ian Minett 2012-12-20 7608 spec->num_mixers = 1; 017310fbe Connor McAdams 2018-05-08 7609 017310fbe Connor McAdams 2018-05-08 7610 /* Set which mixers each quirk uses. */ 017310fbe Connor McAdams 2018-05-08 7611 switch (spec->quirk) { 017310fbe Connor McAdams 2018-05-08 7612 case QUIRK_SBZ: e25e34450 Connor McAdams 2018-08-08 7613 spec->mixers[0] = desktop_mixer; 017310fbe Connor McAdams 2018-05-08 7614 snd_hda_codec_set_name(codec, "Sound Blaster Z"); 017310fbe Connor McAdams 2018-05-08 7615 break; e25e34450 Connor McAdams 2018-08-08 7616 case QUIRK_R3D: e25e34450 Connor McAdams 2018-08-08 7617 spec->mixers[0] = desktop_mixer; e25e34450 Connor McAdams 2018-08-08 7618 snd_hda_codec_set_name(codec, "Recon3D"); e25e34450 Connor McAdams 2018-08-08 7619 break; 017310fbe Connor McAdams 2018-05-08 7620 case QUIRK_R3DI: 017310fbe Connor McAdams 2018-05-08 7621 spec->mixers[0] = r3di_mixer; 017310fbe Connor McAdams 2018-05-08 7622 snd_hda_codec_set_name(codec, "Recon3Di"); 017310fbe Connor McAdams 2018-05-08 7623 break; 017310fbe Connor McAdams 2018-05-08 7624 default: a7e76271b Ian Minett 2012-12-20 7625 spec->mixers[0] = ca0132_mixer; 017310fbe Connor McAdams 2018-05-08 7626 break; 017310fbe Connor McAdams 2018-05-08 7627 } a7e76271b Ian Minett 2012-12-20 7628 08eca6b1f Connor McAdams 2018-08-08 7629 /* Setup whether or not to use alt functions/controls/pci_mmio */ 009b8f979 Connor McAdams 2018-05-08 7630 switch (spec->quirk) { 009b8f979 Connor McAdams 2018-05-08 7631 case QUIRK_SBZ: e42c7c731 Connor McAdams 2018-08-08 7632 case QUIRK_R3D: 08eca6b1f Connor McAdams 2018-08-08 7633 spec->use_alt_controls = true; 08eca6b1f Connor McAdams 2018-08-08 7634 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7635 spec->use_pci_mmio = true; 08eca6b1f Connor McAdams 2018-08-08 7636 break; 009b8f979 Connor McAdams 2018-05-08 7637 case QUIRK_R3DI: 47cdf76e4 Connor McAdams 2018-05-08 7638 spec->use_alt_controls = true; 009b8f979 Connor McAdams 2018-05-08 7639 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7640 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7641 break; 009b8f979 Connor McAdams 2018-05-08 7642 default: 47cdf76e4 Connor McAdams 2018-05-08 7643 spec->use_alt_controls = false; 009b8f979 Connor McAdams 2018-05-08 7644 spec->use_alt_functions = false; 08eca6b1f Connor McAdams 2018-08-08 7645 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7646 break; 009b8f979 Connor McAdams 2018-05-08 7647 } 009b8f979 Connor McAdams 2018-05-08 7648 08eca6b1f Connor McAdams 2018-08-08 7649 if (spec->use_pci_mmio) { 08eca6b1f Connor McAdams 2018-08-08 @7650 spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); 08eca6b1f Connor McAdams 2018-08-08 7651 if (spec->mem_base == NULL) { 08eca6b1f Connor McAdams 2018-08-08 7652 codec_warn(codec, "pci_iomap failed! Setting quirk to QUIRK_NONE."); 08eca6b1f Connor McAdams 2018-08-08 7653 spec->quirk = QUIRK_NONE; 08eca6b1f Connor McAdams 2018-08-08 7654 } 08eca6b1f Connor McAdams 2018-08-08 7655 } 08eca6b1f Connor McAdams 2018-08-08 7656 5aaca44d8 Ian Minett 2012-12-20 7657 spec->base_init_verbs = ca0132_base_init_verbs; 5aaca44d8 Ian Minett 2012-12-20 7658 spec->base_exit_verbs = ca0132_base_exit_verbs; 5aaca44d8 Ian Minett 2012-12-20 7659 993884f6a Chih-Chung Chang 2013-03-25 7660 INIT_DELAYED_WORK(&spec->unsol_hp_work, ca0132_unsol_hp_delayed); 993884f6a Chih-Chung Chang 2013-03-25 7661 95c6e9cb7 Ian Minett 2011-06-15 7662 ca0132_init_chip(codec); 95c6e9cb7 Ian Minett 2011-06-15 7663 95c6e9cb7 Ian Minett 2011-06-15 7664 ca0132_config(codec); 95c6e9cb7 Ian Minett 2011-06-15 7665 d5c016b56 Gabriele Martino 2015-05-18 7666 err = ca0132_prepare_verbs(codec); d5c016b56 Gabriele Martino 2015-05-18 7667 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7668 goto error; d5c016b56 Gabriele Martino 2015-05-18 7669 a73d511c4 Ian Minett 2012-12-20 7670 err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); a73d511c4 Ian Minett 2012-12-20 7671 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7672 goto error; a73d511c4 Ian Minett 2012-12-20 7673 95c6e9cb7 Ian Minett 2011-06-15 7674 return 0; cc91ceaf3 Takashi Iwai 2017-09-04 7675 cc91ceaf3 Takashi Iwai 2017-09-04 7676 error: cc91ceaf3 Takashi Iwai 2017-09-04 7677 ca0132_free(codec); cc91ceaf3 Takashi Iwai 2017-09-04 7678 return err; 95c6e9cb7 Ian Minett 2011-06-15 7679 } 95c6e9cb7 Ian Minett 2011-06-15 7680 :::::: The code at line 7650 was first introduced by commit :::::: 08eca6b1f1468a4021bac7b3929fd3eb491e2629 ALSA: hda/ca0132 - Add bool variable to enable/disable pci region2 mmio :::::: TO: Connor McAdams <conmanx360(a)gmail.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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