November 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [question] Sample rate IC followed by DAC
by Matt Flax 06 Nov '16

06 Nov '16

Hi there, I have a sample rate converter IC, which is followed by a DAC. This kind of setup : SoC chip -> Sample rate conv. chip -> DAC chip I also have the reverse for the ADC... but this question is about the DAC path. How do you construct a machine driver for this setup ? Only the Sample rate converter ICs use the SoC I2S bus, the DAC and ADC, sit outside the I2S scope of the SoC... however all chips require I2C control. Unfortunately the silicon only supports an older 3.14 kernel. Can it be something like this : static struct snd_soc_dai_link sr_dai[] = { { .name = "Sample-rate-for-DAC", .stream_name = "playback 0", .cpu_dai_name = "nxp-i2s.0", .platform_name = "nxp-pcm", .codec_dai_name = "sample-rate-in", .codec_name = "sample-rate.4-0012", }, { .name = "DAC", .stream_name = "playback 0", .cpu_dai_name = "?????", .platform_name = "????", .codec_dai_name = "codec-in", .codec_name = "DAC.4-0010", }, }; thanks Matt

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Re: [alsa-devel] Question about the buffer length in the transfer callback of the PCM External Plugin
by Clemens Ladisch 06 Nov '16

06 Nov '16

Bach, Andrea wrote: > What I tried is to copy the buffer I get in chunks of a 64 bytes into > an intermediate buffer for processing and whenever I processed > 64 bytes I copied the result to the output buffer .... but what to do > with the remaining 40 bytes... Leave them in the intermediate buffer, and handle them later when you've got 24 more bytes. > and what to return in the output buffer when the remaining bytes could > not be processed?? Delay the entire chain by 64 bytes, i.e., insert 64 byte at the start of the stream. Then you always have too many bytes instead of too few bytes, and those can be handled with the intermediate buffer. Regards, Clemens

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[alsa-devel] [PATCH] ASoC: tas642x: Add initial support for TAS642x amplifiers.
by Aurelien Chanot 06 Nov '16

06 Nov '16

This patch adds support for Texas Intrument Amplifier TAS642X. Signed-off-by: Aurelien Chanot <chanot.a(a)gmail.com> --- .../devicetree/bindings/sound/tas642x.txt | 27 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas642x.c | 374 +++++++++++++++++++++ sound/soc/codecs/tas642x.h | 32 ++ 5 files changed, 440 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas642x.txt create mode 100644 sound/soc/codecs/tas642x.c create mode 100644 sound/soc/codecs/tas642x.h diff --git a/Documentation/devicetree/bindings/sound/tas642x.txt b/Documentation/devicetree/bindings/sound/tas642x.txt new file mode 100644 index 0000000..57abe6e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas642x.txt @@ -0,0 +1,27 @@ +Texas Instruments TAS642X Audio amplifier + +The TAS642X serial control bus communicates through the I2C protocol only. The +serial bus is also used for periodic codec fault checking/reporting during +audio playback. For more product information please see the links below: + +http://www.ti.com/product/TAS6424-Q1 + + +Required properties: + +- compatible: "ti,tas6424" +- reg: I2C slave address + +Optional properties: + +- chan-swap : swap channels 1-2 with 3-4 +- chan-sel: uses channels 1 to 4 (0) or 5 to 8(1) from the TDM input. + +Example: + +tas6424: tas6424@64 { + status = "okay"; + compatible = "ti,tas6424"; + reg = <0x6c>; + chan-sel = <1>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c67667b..dce371e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -135,6 +135,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TAS5086 if I2C select SND_SOC_TAS571X if I2C select SND_SOC_TAS5720 if I2C + select SND_SOC_TAS642X if I2C select SND_SOC_TFA9879 if I2C select SND_SOC_TLV320AIC23_I2C if I2C select SND_SOC_TLV320AIC23_SPI if SPI_MASTER @@ -812,6 +813,10 @@ config SND_SOC_TAS5720 Enable support for Texas Instruments TAS5720L/M high-efficiency mono Class-D audio power amplifiers. +config SND_SOC_TAS642X + tristate "Texas Instruments TAS6424 speaker amplifier" + depends on I2C + config SND_SOC_TFA9879 tristate "NXP Semiconductors TFA9879 amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 958cd49..3f49516 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -142,6 +142,7 @@ snd-soc-sti-sas-objs := sti-sas.o snd-soc-tas5086-objs := tas5086.o snd-soc-tas571x-objs := tas571x.o snd-soc-tas5720-objs := tas5720.o +snd-soc-tas642x-objs := tas642x.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o @@ -363,6 +364,7 @@ obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o +obj-$(CONFIG_SND_SOC_TAS642X) += snd-soc-tas642x.o obj-$(CONFIG_SND_SOC_TFA9879) += snd-soc-tfa9879.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o diff --git a/sound/soc/codecs/tas642x.c b/sound/soc/codecs/tas642x.c new file mode 100644 index 0000000..16f2c94 --- /dev/null +++ b/sound/soc/codecs/tas642x.c @@ -0,0 +1,374 @@ +/* + * TAS642x amplifier audio driver + * + * Copyright (C) 2016 Witekio + * + * 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. + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/gpio/consumer.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <linux/stddef.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "tas642x.h" + +struct tas642x_chip { + struct snd_soc_codec_driver codec_driver; + const struct regmap_config *regmap_config; + int channel_number; +}; + +struct tas642x_private { + struct device *dev; + const struct tas642x_chip *chip; + struct regmap *regmap; + struct clk *mclk; + unsigned int format; + unsigned int chan_sel; /* 0=1-4; 1=5-8 */ + unsigned int chan_swap; /* 0=No; 1=Yes */ +}; + +static int tas642x_reg_write(void *context, unsigned int reg, + unsigned int value) +{ + struct i2c_client *client = context; + unsigned int size; + uint8_t buf[3]; + int ret; + + size = 2; + buf[0] = reg; + buf[1] = value; + + ret = i2c_master_send(client, buf, size); + if (ret == size) + return 0; + else if (ret < 0) + return ret; + else + return -EIO; +} + +static int tas642x_reg_read(void *context, unsigned int reg, + unsigned int *value) +{ + struct i2c_client *client = context; + uint8_t send_buf, recv_buf[4]; + struct i2c_msg msgs[2]; + unsigned int size; + unsigned int i; + int ret; + + size = 1; + send_buf = reg; + + msgs[0].addr = client->addr; + msgs[0].len = sizeof(send_buf); + msgs[0].buf = &send_buf; + msgs[0].flags = 0; + + msgs[1].addr = client->addr; + msgs[1].len = size; + msgs[1].buf = recv_buf; + msgs[1].flags = I2C_M_RD; + + ret = i2c_transfer(client->adapter, msgs, ARRAY_SIZE(msgs)); + if (ret < 0) + return ret; + else if (ret != ARRAY_SIZE(msgs)) + return -EIO; + + *value = 0; + + for (i = 0; i < size; i++) { + *value <<= 8; + *value |= recv_buf[i]; + } + + return 0; +} + +static int tas6424_set_dai_fmt(struct snd_soc_dai *dai, unsigned int format) +{ + struct tas642x_private *priv = snd_soc_codec_get_drvdata(dai->codec); + + /* set master/slave audio interface. Only support slave */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + pr_err("Only support Slave mode!\n"); + return -EINVAL; + } + + priv->format = format; + + return 0; +} + +static int tas6424_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct tas642x_private *priv = snd_soc_codec_get_drvdata(dai->codec); + u32 val; + + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + val = 0x00; + break; + case SND_SOC_DAIFMT_I2S: + val = 0x04; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = 0x05; + break; + case SND_SOC_DAIFMT_DSP_B: + val = 0x06; + break; + default: + pr_info("Invalid format (0x%x)!!\n", + priv->format & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + if (params_width(params) < 24) + val += 0x10; + + if (priv->chan_sel) + val += 0x20; + + if (priv->chan_swap > 0) + val += 0x08; + + val = regmap_update_bits(priv->regmap, TAS642X_SAP_CTRL, + TAS642X_SAP_FMT_MSK | + TAS642X_SAP_CH_SWP_MSK | + TAS642X_SAP_TDM_SIZ_MSK | + TAS642X_SAP_TDM_SEL_MSK, + val); + pr_info("%s: regmap return 0x%x\n", __func__, val); + return val; +} + +static const struct snd_soc_dai_ops tas6424_dai_ops = { + .set_fmt = tas6424_set_dai_fmt, + .hw_params = tas6424_hw_params, +}; + +static const struct reg_default tas642x_reg_defaults[] = { + { TAS642X_MODE_CTRL, 0x00 }, + { TAS642X_MISC_CTRL1, 0x32 }, + { TAS642X_MISC_CTRL2, 0x62 }, + { TAS642X_SAP_CTRL, 0x04 }, + { TAS642X_CH_STATE, 0xAA }, + { TAS642X_CH1_VOL, 0xCF }, + { TAS642X_CH2_VOL, 0xCF }, + { TAS642X_CH3_VOL, 0xCF }, + { TAS642X_CH4_VOL, 0xCF }, +}; + + +static const struct regmap_config tas642x_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0xff, + .reg_read = tas642x_reg_read, + .reg_write = tas642x_reg_write, + .reg_defaults = tas642x_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas642x_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + +static const DECLARE_TLV_DB_SCALE(tas642x_volume_tlv, -10350, 50, 1); + +static const struct snd_kcontrol_new tas6424_controls[] = { + SOC_SINGLE("Spk 1 Playback Switch", TAS642X_CH_STATE, 7, 1, 1), + SOC_SINGLE_TLV("Spk 1 Playback Volume", + TAS642X_CH1_VOL, + 0, 0xff, 0, tas642x_volume_tlv), + SOC_SINGLE("Spk 2 Playback Switch", TAS642X_CH_STATE, 5, 1, 1), + SOC_SINGLE_TLV("Spk 2 Playback Volume", + TAS642X_CH2_VOL, + 0, 0xff, 0, tas642x_volume_tlv), + SOC_SINGLE("Spk 3 Playback Switch", TAS642X_CH_STATE, 3, 1, 1), + SOC_SINGLE_TLV("Spk 3 Playback Volume", + TAS642X_CH3_VOL, + 0, 0xff, 0, tas642x_volume_tlv), + SOC_SINGLE("Spk 4 Playback Switch", TAS642X_CH_STATE, 1, 1, 1), + SOC_SINGLE_TLV("Spk 4 Playback Volume", + TAS642X_CH4_VOL, + 0, 0xff, 0, tas642x_volume_tlv), +}; + +static const struct snd_soc_dapm_route tas642x_1_dapm_routes[] = { + { "Speaker 1", NULL, "Playback" }, + { "Speaker 2", NULL, "Playback" }, + { "Speaker 3", NULL, "Playback" }, + { "Speaker 4", NULL, "Playback" }, + { "OUT_1", NULL, "Speaker 1" }, + { "OUT_2", NULL, "Speaker 2" }, + { "OUT_3", NULL, "Speaker 3" }, + { "OUT_4", NULL, "Speaker 4" }, +}; + + +static const struct snd_soc_dapm_widget tas6424_1_dapm_widgets[] = { + SND_SOC_DAPM_DAC("Speaker 1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("Speaker 2", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("Speaker 3", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("Speaker 4", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_OUTPUT("OUT_1"), + SND_SOC_DAPM_OUTPUT("OUT_2"), + SND_SOC_DAPM_OUTPUT("OUT_3"), + SND_SOC_DAPM_OUTPUT("OUT_4"), +}; + + +static int tas642x_codec_probe(struct snd_soc_codec *codec) +{ + pr_info("TAS642X: i2c codec Probe\n"); + /* Set All channels to mute */ + snd_soc_write(codec, TAS642X_CH_STATE, 0xAA); + return 0; +} + +static const struct tas642x_chip tas6424_chip = { + .codec_driver = { + .probe = tas642x_codec_probe, + .component_driver = { + .controls = tas6424_controls, + .num_controls = ARRAY_SIZE(tas6424_controls), + .dapm_widgets = tas6424_1_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas6424_1_dapm_widgets), + .dapm_routes = tas642x_1_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(tas642x_1_dapm_routes), + }, + }, + .regmap_config = &tas642x_regmap_config, + .channel_number = 4, +}; + +static struct snd_soc_dai_driver tas642x_dai = { + .name = "tas642x-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 4, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &tas6424_dai_ops, +}; + +static const struct of_device_id tas642x_of_match[]; + +#if defined(CONFIG_OF) +static void tas642x_pdata_from_of(struct tas642x_private *priv) +{ + struct device_node *np = priv->dev->of_node; + + priv->chan_swap = of_property_read_bool(np, "chan-swap"); + priv->chan_sel = of_property_read_bool(np, "chan-sel"); +} +#else /* CONFIG_OF */ +static void tas642x_pdata_from_of(struct tas642x_private *priv) +{ +} +#endif /* CONFIG_OF */ + +void tas642x_device_init(struct tas642x_private *priv) +{ + dev_set_drvdata(priv->dev, priv); + + if (priv->dev->of_node) + tas642x_pdata_from_of(priv); +} + +static int tas642x_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct tas642x_private *priv; + struct device *dev = &client->dev; + const struct of_device_id *of_id; + + dev_info(dev, "i2c Probe\n"); + + priv = (struct tas642x_private *)devm_kzalloc(dev, + sizeof(struct tas642x_private), + GFP_KERNEL); + if (!priv) + return -ENOMEM; + + of_id = of_match_device(tas642x_of_match, dev); + if (!of_id) { + dev_err(dev, "Unknown device type\n"); + return -EINVAL; + } + priv->chip = of_id->data; + priv->dev = dev; + i2c_set_clientdata(client, priv); + priv->regmap = devm_regmap_init_i2c(client, priv->chip->regmap_config); + if (IS_ERR(priv->regmap)) + return PTR_ERR(priv->regmap); + + + tas642x_device_init(priv); + + dev_info(dev, "i2c Probe Ends!\n"); + return snd_soc_register_codec(&client->dev, &tas6424_chip.codec_driver, + &tas642x_dai, 1); +} + +static int tas642x_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct of_device_id tas642x_of_match[] = { + { .compatible = "ti,tas6424", .data = &tas6424_chip, }, + { } +}; +MODULE_DEVICE_TABLE(of, tas642x_of_match); + +static const struct i2c_device_id tas642x_i2c_id[] = { + { "tas6424", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas642x_i2c_id); + +static struct i2c_driver tas642x_i2c_driver = { + .driver = { + .name = "tas642x", + .of_match_table = of_match_ptr(tas642x_of_match), + }, + .probe = tas642x_i2c_probe, + .remove = tas642x_i2c_remove, + .id_table = tas642x_i2c_id, +}; +module_i2c_driver(tas642x_i2c_driver); + +MODULE_DESCRIPTION("ASoC tas642x driver"); +MODULE_AUTHOR("Aurelien Chanot <chanot.a(a)gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas642x.h b/sound/soc/codecs/tas642x.h new file mode 100644 index 0000000..e0dec07 --- /dev/null +++ b/sound/soc/codecs/tas642x.h @@ -0,0 +1,32 @@ +/* + * TAS642X amplifier audio driver + * + * Copyright (C) 2016 Witekio + * + * 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. + */ + +#ifndef _TAS642X_H +#define _TAS642X_H + +/* device registers */ +#define TAS642X_MODE_CTRL 0x00 +#define TAS642X_MISC_CTRL1 0x01 +#define TAS642X_MISC_CTRL2 0x02 + +#define TAS642X_SAP_CTRL 0x03 +# define TAS642X_SAP_SAMP_MSK 0xC0 +# define TAS642X_SAP_TDM_SEL_MSK 0x20 +# define TAS642X_SAP_TDM_SIZ_MSK 0x10 +# define TAS642X_SAP_CH_SWP_MSK 0x08 +# define TAS642X_SAP_FMT_MSK 0x07 +#define TAS642X_CH_STATE 0x04 +#define TAS642X_CH1_VOL 0x05 +#define TAS642X_CH2_VOL 0x06 +#define TAS642X_CH3_VOL 0x07 +#define TAS642X_CH4_VOL 0x08 + +#endif /* _TAS642X_H */ -- 2.9.3

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Re: [alsa-devel] [PATCH] ASoC: Intel: Skylake: Always acquire runtime pm ref on unload
by Vinod Koul 05 Nov '16

05 Nov '16

On Thu, Oct 20, 2016 at 12:26:16PM +0200, Lukas Wunner wrote: > skl_probe() releases a runtime pm ref unconditionally wheras > skl_remove() acquires one only if the device is wakeup capable. > Thus if the device is not wakeup capable, unloading and reloading > the module will result in the refcount being decreased below 0. > Fix it. Hi Lukas, Thanks for the patch, have you tested this? I did some tests last week and I dont remember module load/unload having an issue.. I will test again next week and get back. Was this found on hardware or by code inspection.. -- ~Vinod

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[alsa-devel] [PATCH 0/6] topology: Updates for PCM object in ABI v5
by mengdong.lin＠linux.intel.com 05 Nov '16

05 Nov '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> These ABI updates for PCM have just been applied to ASoC kernel. And kernel can handle the PCM ABI updates in a backward compatible way. So we update the user space code for PCM now. The updates for physical DAI link and DAIs will come later after their kernel updates be applied at first. Mengdong Lin (6): topology: ABI - Update stream caps and PCM objects to ABI v5 topology: Parse sig_bits of stream caps topology: Parse flags for PCM topology: Define a function to build a single PCM element topology: Parse and build private data for PCM topology: tplg_elem_lookup() checks parameter before searching include/sound/asoc.h | 9 ++++ include/topology.h | 11 +++++ src/topology/elem.c | 3 ++ src/topology/parser.c | 2 +- src/topology/pcm.c | 120 ++++++++++++++++++++++++++++++++++++++++++++-- src/topology/tplg_local.h | 2 +- 6 files changed, 140 insertions(+), 7 deletions(-) -- 2.7.4

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Re: [alsa-devel] Intel Cherry Trail platform sound issue
by Zhihe Tu 05 Nov '16

05 Nov '16

Can anyone give some suggestions? Thanks Tu 发件人: 涂治河 [mailto:tuzhihe@pcasl.com] 发送时间: 2015年11月5日 19:37 收件人: 'alsa-devel(a)alsa-project.org' 主题: Intel Cherry Trail platform sound issue Hi All, I get cherry trail tablet, and install ubuntu14.04.3, Kernel verison is 3.19. The table using rt5672 codec with no sound output in ubuntu, but it works in win8. I get intel sst firmware fw_sst_22a8.bin from https://git.kernel.org/cgit/linux/kernel/git/firmware/linux-firmware.git/tr… The driver load the firmware successfully and I change some alsa mixer items: amixer -c0 sset 'codec_out0 mix 0 pcm0_in' on amixer -c0 sset 'media0_out mix 0 media1_in' on amixer -c0 sset 'media1_in Gain 0' 80% amixer -c0 sset 'media1_in Gain 0 Ramp Delay' 50 amixer -c0 sset 'media1_in Gain 0' off amixer -c0 sset 'pcm0_in Gain 0' 80% amixer -c0 sset 'pcm0_in Gain 0 Ramp Delay' 50 amixer -c0 sset 'pcm0_in Gain 0' off amixer -c0 sset 'codec_out0 Gain 0' 80% amixer -c0 sset 'codec_out0 Gain 0 Ramp Delay' 50 amixer -c0 sset 'codec_out0 Gain 0' off But there are some errors with dmesg: [ 16.359646] Audio Port: ASoC: no backend DAIs enabled for Audio Port … [ 16.382156] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 16.383984] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 16.541809] intel_sst_acpi 808622A8:00: Alloc for str 1 pipe 0x90 [ 17.545512] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x2 [ 17.555207] intel_sst_acpi 808622A8:00: fw returned err -16 [ 18.605859] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 18.615908] intel_sst_acpi 808622A8:00: fw returned err -16 [ 19.621943] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 19.631499] intel_sst_acpi 808622A8:00: fw returned err -16 [ 20.637527] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 20.647083] intel_sst_acpi 808622A8:00: fw returned err -16 [ 20.682979] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 21.681280] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 21.690836] intel_sst_acpi 808622A8:00: fw returned err -16 [ 21.697087] intel_sst_acpi 808622A8:00: free stream returned err -19 [ 22.757082] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 22.766640] intel_sst_acpi 808622A8:00: fw returned err -16 [ 23.772867] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 23.782432] intel_sst_acpi 808622A8:00: fw returned err -16 [ 23.857938] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 23.989254] intel_sst_acpi 808622A8:00: Alloc for str 1 pipe 0x90 [ 23.989491] intel_sst_acpi 808622A8:00: FW sent error response 0x40034 [ 23.996864] intel_sst_acpi 808622A8:00: FW alloc failed ret -4 [ 24.003399] intel_sst_acpi 808622A8:00: sst_get_stream returned err -5 [ 24.010737] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5 [ 24.018652] Audio Port: ASoC: prepare FE Audio Port failed [ 25.024622] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 25.034180] intel_sst_acpi 808622A8:00: fw returned err -16 [ 26.040399] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 26.049955] intel_sst_acpi 808622A8:00: fw returned err -16 [ 27.056333] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 27.065890] intel_sst_acpi 808622A8:00: fw returned err -16 [ 28.072032] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 28.081591] intel_sst_acpi 808622A8:00: fw returned err -16 [ 28.117427] Audio Port: sst: error code = -22 [ 28.122401] snd_pcm_update_hw_ptr0: 120 callbacks suppressed [ 28.122409] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 28.132784] Audio Port: sst: error code = -22 [ 28.137749] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 28.372967] Audio Port: sst: error code = -22 [ 28.377946] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 28.388317] Audio Port: sst: error code = -22 [ 28.393289] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 28.403687] Audio Port: sst: error code = -22 lsmod info: snd_soc_sst_cht_bsw_rt5672 16384 0 snd_intel_sst_acpi 16384 1 snd_intel_sst_core 73728 1 snd_intel_sst_acpi snd_soc_sst_mfld_platform 77824 3 snd_intel_sst_core snd_soc_rt5670 106496 1 snd_soc_rl6231 16384 1 snd_soc_rt5670 snd_soc_core 196608 4 snd_soc_rt5670,snd_soc_sst_cht_bsw_rt5672,snd_soc_sst_mfld_platform snd_seq_midi 16384 0 snd_seq_midi_event 16384 1 snd_seq_midi snd_compress 20480 1 snd_soc_core snd_pcm_dmaengine 16384 1 snd_soc_core snd_rawmidi 32768 1 snd_seq_midi snd_pcm 106496 5 snd_soc_rt5670,snd_soc_core,snd_soc_sst_cht_bsw_rt5672,snd_soc_sst_mfld_platform,snd_pcm_dmaengine snd_seq 65536 2 snd_seq_midi_event,snd_seq_midi snd_seq_device 16384 3 snd_seq,snd_rawmidi,snd_seq_midi snd_timer 32768 2 snd_pcm,snd_seq snd 86016 10 snd_soc_core,snd_timer,snd_pcm,snd_seq,snd_rawmidi,snd_soc_sst_mfld_platform,snd_seq_device,snd_compress soundcore 16384 1 snd I have no idea with the next step, so can anyone give a look? Thanks Tu __________ Information from ESET Smart Security, version of virus signature database 4468 (20090929) __________ The message was checked by ESET Smart Security. http://www.eset.com __________ Information from ESET Smart Security, version of virus signature database 4468 (20090929) __________ The message was checked by ESET Smart Security. http://www.eset.com

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[alsa-devel] [PATCH v7 00/11] ASoC: topology: Remaining kernel patches
by mengdong.lin＠linux.intel.com 05 Nov '16

05 Nov '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series can support old topology ABI v4 in a backward compatible way, assuming users start to use topology from ABI v4 with alsa-lib v1.1.0. This series contains all remaining kernel patches of topology of ABI v5, including some ABI update to PCM (FrontEnds) and link (BackEnds) objects. Kernel can support topology files generated by ABI v4 without these updates. User will be able to config existing physical DAI links, configure more for FE links. Code are verified and can cover reqeust of Intel pre-release platforms for next year, so ABI should be stable. Current kernel topology code does not really touch Codec-Codec links since there is no user requst atm. We can add support for CC links later by reusing code and data structures for physical links, and no need to revise ABI. History: v2: Add the reason for creating BE DAI & DAI links by topology to commit message. Drop support for configuring DPCM trigger ordering in topology. v3: Topology no longer creates BE DAI or BE DAI links, but only configure existing ones. The API to find a DAI link can also check the name and stream name in case a soc card doesn't use unique ID for DAI links. The user space code is also ready. v4: Support ABI udpate and be backward compatible for topology ABI v4. Code refactoring to configure physical DAIs. v5: Minor code refactoring and modification on commit messages. v6: Change the variable of physical DAI count in manifest from "be_dai_elems" to "dai_elems", backward compatible. v7: Remove 2 v6 patches that have been merged. Split ABI update and non-ABI fix for PCM to seperate patches. No use asoc specific flags "ignore_suspend" & "ignore_powerdown_time" for links, but add a flag for voice wake up (the last patch). Mengdong Lin (11): ASoC: topology: Make PCM backward compatible from ABI v4 ASoC: topology: Only use valid names of PCM for the kernel DAI & DAI link ASoC: topology: Support topology file of ABI v4 ASoC: topology: ABI - Add flags and private data to PCM ASoC: topology: ABI - Update physical DAI link configuration for version 5 ASoC: Define API to find a dai link ASoC: topology: Add support to configure existing physical DAI links ASoC: topology: Rename the function to create a FE link ASoC: topology: ABI - Rename struct and type for physical DAIs ASoC: topology: Rename functions & variables for physical DAIs ASoC: topology: ABI - Add voice wake up flag for DAI links include/sound/soc.h | 3 + include/uapi/sound/asoc.h | 73 +++++++- sound/soc/soc-core.c | 42 +++++ sound/soc/soc-topology.c | 463 +++++++++++++++++++++++++++++++++++++++++----- 4 files changed, 522 insertions(+), 59 deletions(-) -- 2.7.4

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[alsa-devel] [GIT PULL] ASoC fixes for v4.9
by Mark Brown 04 Nov '16

04 Nov '16

The following changes since commit a909d3e636995ba7c349e2ca5dbb528154d4ac30: Linux 4.9-rc3 (2016-10-29 13:52:02 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.9-rc3 for you to fetch changes up to 0b203d699e08424288b5f3eab1347143dd5d1e68: Merge remote-tracking branch 'asoc/fix/topology-abi' into asoc-linus (2016-11-04 12:34:05 -0600) ---------------------------------------------------------------- ASoC: Fixes for v4.9 The most important fix in here is a change which removes the #error making the topology API unusable as-is since we have recently discovered some production uses on Chromebooks so need to acknowledge that what we've got there now is an ABI. There's also a very big batch of driver specific fixes here which have kept on being delayed due to more arriving so the update is another of these bigger than I would like ones. There is one especially big one in there, for the Qualcomm code which fixes simultaneous playback and capture which was broken during the merge window. The diff for that is large because it moves blocks of code to different functions but it's functionally fairly simple and if it breaks it should have been very obvious in testing. ---------------------------------------------------------------- Arnaud Pouliquen (2): ASoC: sti: fix channel status update after playback start ASoC: sti-sas: enable fast io for regmap Arnd Bergmann (2): ASoC: Intel: haswell depends on sst-firmware ASoC: PXA: Brownstone needs I2C Bard Liao (1): ASoC: rt298: fix jack type detect error Chen-Yu Tsai (2): ASoC: sun4i-codec: return error code instead of NULL when create_card fails ASoC: sun4i-codec: Enable bus clock after getting GPIO Dan Carpenter (1): ASoC: rt5663: fix a debug statement Jon Medhurst (Tixy) (1): ASoC: hdmi-codec: Fix hdmi_of_xlate_dai_name when #sound-dai-cells = <0> Lukas Wunner (1): ASoC: Intel: Skylake: Always acquire runtime pm ref on unload Marek Szyprowski (1): ASoC: samsung: get access to DMA engine early to defer probe properly Mark Brown (5): ASoC: topology: Reenable use from userspace Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branches 'asoc/fix/cs4270', 'asoc/fix/da7219', 'asoc/fix/hdmi-codec', 'asoc/fix/pxa', 'asoc/fix/qcom' and 'asoc/fix/rt298' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5663', 'asoc/fix/samsung', 'asoc/fix/sti', 'asoc/fix/sti-codec', 'asoc/fix/sunxi' and 'asoc/fix/tas571x' into asoc-linus Merge remote-tracking branch 'asoc/fix/topology-abi' into asoc-linus Petr Kulhavy (4): ASoC: tas571x: wait 50ms after oscillator trim ASoC: tas571x: move mixer volume controls from TAS5711 to TAS5717 ASoC: tas571x: extend the t_i2c time to comply with TAS5721 ASoC: tas571x: remove improper PDN signal usage in set_bias_level Sathyanarayana Nujella (1): ASoC: Intel: report JACK_LINEOUT event Sodhi, VunnyX (1): ASoC: Intel: Skylake: Fix to turn off hdmi power on probe failure Srinivas Kandagatla (2): ASoC: lpass-platform: Fix broken pcm data usage ASoC: lpass-cpu: add module licence and description Stephen Barber (1): ASoC: da7219: Connect output enable register to DAIOUT Sylwester Nawrocki (1): ASoC: samsung: spdif: Fix DMA filter initialization Wei Yongjun (1): ASoC: Intel: Atom: add terminate entry for dmi_system_id tables murray foster (1): ASoC: cs4270: fix DAPM stream name mismatch include/uapi/sound/asoc.h | 6 - sound/soc/codecs/cs4270.c | 8 +- sound/soc/codecs/da7219.c | 3 +- sound/soc/codecs/hdmi-codec.c | 7 +- sound/soc/codecs/rt298.c | 5 + sound/soc/codecs/rt5663.c | 4 +- sound/soc/codecs/sti-sas.c | 2 +- sound/soc/codecs/tas571x.c | 37 ++---- sound/soc/intel/Kconfig | 3 +- sound/soc/intel/atom/sst/sst_acpi.c | 1 + sound/soc/intel/boards/bxt_da7219_max98357a.c | 4 +- sound/soc/intel/skylake/skl.c | 8 +- sound/soc/pxa/Kconfig | 2 +- sound/soc/qcom/lpass-cpu.c | 3 + sound/soc/qcom/lpass-platform.c | 165 ++++++++++++-------------- sound/soc/qcom/lpass.h | 1 - sound/soc/samsung/ac97.c | 10 +- sound/soc/samsung/i2s.c | 19 +-- sound/soc/samsung/pcm.c | 19 +-- sound/soc/samsung/s3c2412-i2s.c | 16 +-- sound/soc/samsung/s3c24xx-i2s.c | 14 +-- sound/soc/samsung/spdif.c | 19 ++- sound/soc/sti/uniperif_player.c | 6 +- sound/soc/sunxi/sun4i-codec.c | 19 +-- 24 files changed, 188 insertions(+), 193 deletions(-)

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[alsa-devel] Applied "ASoC: stac9766: Remove unused DAI ID defines" to the asoc tree
by Mark Brown 04 Nov '16

04 Nov '16

The patch ASoC: stac9766: Remove unused DAI ID defines has been applied to the asoc tree at git://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 355602eb5af91ff8ddda435f0f9e910f3b18c438 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen <lars(a)metafoo.de> Date: Fri, 4 Nov 2016 18:26:53 +0100 Subject: [PATCH] ASoC: stac9766: Remove unused DAI ID defines The DAI ID defines are back from the time when DAIs were referenced by a numerical ID. These days a string is used instead and the defines are unused. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/stac9766.h | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h index c726f907e2c0..cb0d5505d571 100644 --- a/sound/soc/codecs/stac9766.h +++ b/sound/soc/codecs/stac9766.h @@ -10,8 +10,4 @@ #define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E) #define AC97_STAC_STEREO_MIC 0x78 -/* STAC9766 DAI ID's */ -#define STAC9766_DAI_AC97_ANALOG 0 -#define STAC9766_DAI_AC97_DIGITAL 1 - #endif -- 2.10.1

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[alsa-devel] Applied "ASoC: stac9766: Remove register paging support" to the asoc tree
by Mark Brown 04 Nov '16

04 Nov '16

The patch ASoC: stac9766: Remove register paging support has been applied to the asoc tree at git://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 2bea8f97d4c3aded4c71d72e8702aa7dbe9894cf Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen <lars(a)metafoo.de> Date: Fri, 4 Nov 2016 18:26:54 +0100 Subject: [PATCH] ASoC: stac9766: Remove register paging support The AC'97 standard defines paging support for the register range 0x60-0x6f. Meaning registers in this window are mapped to different physical registers depending on the setting of the page select register (0x24). The stac9766 implements support for switching between page 0 and page 1 depending on the addressed register. But the driver never accesses any registers from page 1, in addition page 0 is the page selected by default. Considering the development history it is unlikely that the driver will see any new features that require paging support. Removing the paging support makes transitioning the driver to regmap a bit more straight forward. The default register value table is update to contain the values from page 0, rather than page 1. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/stac9766.c | 16 ++-------------- sound/soc/codecs/stac9766.h | 5 ++--- 2 files changed, 4 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 27f30d352867..e54e4a4ce296 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -47,8 +47,8 @@ static const u16 stac9766_reg[] = { 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */ 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */ - 0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ + 0x1201, 0x0000, 0x0000, 0x0000, /* 66 */ + 0x0000, 0x0000, 0x0000, 0x1000, /* 6e */ 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */ }; @@ -145,12 +145,6 @@ static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); u16 *cache = codec->reg_cache; - if (reg > AC97_STAC_PAGE0) { - stac9766_ac97_write(codec, AC97_INT_PAGING, 0); - soc_ac97_ops->write(ac97, reg, val); - stac9766_ac97_write(codec, AC97_INT_PAGING, 1); - return 0; - } if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) return -EIO; @@ -165,12 +159,6 @@ static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); u16 val = 0, *cache = codec->reg_cache; - if (reg > AC97_STAC_PAGE0) { - stac9766_ac97_write(codec, AC97_INT_PAGING, 0); - val = soc_ac97_ops->read(ac97, reg - AC97_STAC_PAGE0); - stac9766_ac97_write(codec, AC97_INT_PAGING, 1); - return val; - } if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) return -EIO; diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h index cb0d5505d571..e35cee82f416 100644 --- a/sound/soc/codecs/stac9766.h +++ b/sound/soc/codecs/stac9766.h @@ -5,9 +5,8 @@ #ifndef _STAC9766_H #define _STAC9766_H -#define AC97_STAC_PAGE0 0x1000 -#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A) -#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E) +#define AC97_STAC_DA_CONTROL 0x6A +#define AC97_STAC_ANALOG_SPECIAL 0x6E #define AC97_STAC_STEREO_MIC 0x78 #endif -- 2.10.1

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