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[alsa-devel] [PATCH v3 0/2] reducing memory footprint of ALSA
by Jie Yang 27 May '15

27 May '15

We are planning to work on reducing memory footprint of ALSA, targeting on small memory embedded devices(e.g. IoT), this series focuses on disabling procfs, hw/sw params refinement, mmap, dpcm, dapm, compressed API, compat code APIs... We plan to enable those reducing items only for EXPERT, and add knobs in sound/core/Kconfig. These first two patches is a start, adding SND_PROC_FS to config disable/enable ALSA proc FS, this disabling can save about 9KB code size on x86_64 platform. Changes in v3: 1. drop SND_REDUCED_MEMORY_FOOTPRINT and make configuration available only for EXPERT; 2. move the configure into sound/core/Kconfig; 3. Add missing Signed-off-by for PATCH 2/2. Changes in v2: 1. Prompt "Sound Proc FS Support" appears only when SND_REDUCED_MEMORY_FOOTPRINT is true. Jie Yang (2): ALSA: Kconfig: add config item SND_PROC_FS for expert ALSA: replace CONFIG_PROC_FS with CONFIG_SND_PROC_FS include/sound/info.h | 8 ++++---- sound/core/Kconfig | 9 +++++++++ sound/core/Makefile | 2 +- sound/core/hwdep.c | 6 +++--- sound/core/init.c | 8 ++++---- sound/core/oss/mixer_oss.c | 6 +++--- sound/core/pcm.c | 6 +++--- sound/core/seq/oss/seq_oss.c | 6 +++--- sound/core/seq/oss/seq_oss_init.c | 5 ++--- sound/core/seq/oss/seq_oss_midi.c | 4 ++-- sound/core/seq/oss/seq_oss_readq.c | 4 ++-- sound/core/seq/oss/seq_oss_synth.c | 4 ++-- sound/core/seq/seq_clientmgr.c | 4 ++-- sound/core/seq/seq_device.c | 2 +- sound/core/seq/seq_info.h | 2 +- sound/core/seq/seq_queue.c | 4 ++-- sound/core/seq/seq_timer.c | 4 ++-- sound/core/sound.c | 4 ++-- sound/core/sound_oss.c | 4 ++-- sound/core/timer.c | 4 ++-- sound/pci/hda/Makefile | 3 ++- sound/pci/hda/hda_eld.c | 4 ++-- sound/pci/hda/hda_local.h | 4 ++-- sound/pci/hda/patch_hdmi.c | 4 ++-- sound/pci/hda/patch_sigmatel.c | 2 +- 25 files changed, 61 insertions(+), 52 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/2] reducing memory footprint of ALSA
by Jie Yang 27 May '15

27 May '15

We are planning to work on reducing memory footprint of ALSA, targeting on small memory embedded devices(e.g. IoT), this series focuses on disabling procfs, hw/sw params refinement, mmap, dpcm, dapm, compressed API, compat code APIs... These first two patches is a start, add config menu in Kconfig, and disable ALSA proc FS once reducing memory footprint is selected, which can save about 9KB code size on x86_64 platform. Changes in v2: Prompt "Sound Proc FS Support" appears only when SND_REDUCED_MEMORY_FOOTPRINT is true. Jie Yang (2): ALSA: Kconfig: add config menu for reduced memory footprint ALSA: set no sound proc fs for reduced memory footprint include/sound/info.h | 8 ++++---- sound/Kconfig | 16 ++++++++++++++++ sound/core/Makefile | 2 +- sound/core/hwdep.c | 6 +++--- sound/core/init.c | 8 ++++---- sound/core/oss/mixer_oss.c | 6 +++--- sound/core/pcm.c | 6 +++--- sound/core/seq/oss/seq_oss.c | 6 +++--- sound/core/seq/oss/seq_oss_init.c | 5 ++--- sound/core/seq/oss/seq_oss_midi.c | 4 ++-- sound/core/seq/oss/seq_oss_readq.c | 4 ++-- sound/core/seq/oss/seq_oss_synth.c | 4 ++-- sound/core/seq/seq_clientmgr.c | 4 ++-- sound/core/seq/seq_device.c | 2 +- sound/core/seq/seq_info.h | 2 +- sound/core/seq/seq_queue.c | 4 ++-- sound/core/seq/seq_timer.c | 4 ++-- sound/core/sound.c | 4 ++-- sound/core/sound_oss.c | 4 ++-- sound/core/timer.c | 4 ++-- sound/pci/hda/Makefile | 3 ++- sound/pci/hda/hda_eld.c | 4 ++-- sound/pci/hda/hda_local.h | 4 ++-- sound/pci/hda/patch_hdmi.c | 4 ++-- sound/pci/hda/patch_sigmatel.c | 2 +- 25 files changed, 68 insertions(+), 52 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda - Set stream_pm ops automatically by generic parser
by Takashi Iwai 27 May '15

27 May '15

This allows user to test power_save_node feature via sysfs or patch firmware even on the codecs that don't specify it. It'll also save a few lines. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 5 ++++- sound/pci/hda/patch_realtek.c | 1 - sound/pci/hda/patch_sigmatel.c | 1 - sound/pci/hda/patch_via.c | 1 - 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 1c8678775f40..ac0db1679f09 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4926,9 +4926,12 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, dig_only: parse_digital(codec); - if (spec->power_down_unused || codec->power_save_node) + if (spec->power_down_unused || codec->power_save_node) { if (!codec->power_filter) codec->power_filter = snd_hda_gen_path_power_filter; + if (!codec->patch_ops.stream_pm) + codec->patch_ops.stream_pm = snd_hda_gen_stream_pm; + } if (!spec->no_analog && spec->beep_nid) { err = snd_hda_attach_beep_device(codec, spec->beep_nid); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fc0ccc78bdc7..e3bf72bb278c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5724,7 +5724,6 @@ static int patch_alc269(struct hda_codec *codec) set_beep_amp(spec, spec->gen.mixer_nid, 0x04, HDA_INPUT); codec->patch_ops = alc_patch_ops; - codec->patch_ops.stream_pm = snd_hda_gen_stream_pm; #ifdef CONFIG_PM codec->patch_ops.suspend = alc269_suspend; codec->patch_ops.resume = alc269_resume; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 43c99ce4a520..0de0fd95144a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4403,7 +4403,6 @@ static const struct hda_codec_ops stac_patch_ops = { #ifdef CONFIG_PM .suspend = stac_suspend, #endif - .stream_pm = snd_hda_gen_stream_pm, .reboot_notify = stac_shutup, }; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 31a95cca015d..742087ef378f 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -472,7 +472,6 @@ static const struct hda_codec_ops via_patch_ops = { .init = via_init, .free = via_free, .unsol_event = snd_hda_jack_unsol_event, - .stream_pm = snd_hda_gen_stream_pm, #ifdef CONFIG_PM .suspend = via_suspend, .check_power_status = via_check_power_status, -- 2.4.1

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[alsa-devel] [PATCH] ASoC: fsl_spdif: Don't try to round-up for clock divisor calculation
by Nicolin Chen 27 May '15

27 May '15

As commit 6c8ca30eec7b ("ASoC: fsl_ssi: Don't try to round-up for PM divisor calculation") mentioned that there's no more need to use a round up work around to get a better divisor since the clk-divider driver has been refined a lot. So this patch applies the same modification to fsl_spdif driver. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Fabio Estevam <fabio.estevam(a)freescale.com> Cc: Zidan Wang <zidan.wang(a)freescale.com> --- Fabio and Zidan, Theoretically, it should have the same problem as fsl_ssi driver had. But I don't have an S/PDIF test environment. So I need your helps to confirm it. Thank you. sound/soc/fsl/fsl_spdif.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 91eb3ae..8e93221 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -417,11 +417,9 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, if (clk != STC_TXCLK_SPDIF_ROOT) goto clk_set_bypass; - /* - * The S/PDIF block needs a clock of 64 * fs * txclk_df. - * So request 64 * fs * (txclk_df + 1) to get rounded. - */ - ret = clk_set_rate(spdif_priv->txclk[rate], 64 * sample_rate * (txclk_df + 1)); + /* The S/PDIF block needs a clock of 64 * fs * txclk_df */ + ret = clk_set_rate(spdif_priv->txclk[rate], + 64 * sample_rate * txclk_df); if (ret) { dev_err(&pdev->dev, "failed to set tx clock rate\n"); return ret; @@ -1060,7 +1058,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, for (sysclk_df = sysclk_dfmin; sysclk_df <= sysclk_dfmax; sysclk_df++) { for (txclk_df = 1; txclk_df <= 128; txclk_df++) { - rate_ideal = rate[index] * (txclk_df + 1) * 64; + rate_ideal = rate[index] * txclk_df * 64; if (round) rate_actual = clk_round_rate(clk, rate_ideal); else -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: Add GPIO based jack device
by Dylan Reid 27 May '15

27 May '15

Add a jack device that allows for separate headphone and mic detect GPIOs. This device will be used as an aux device and will registered the jack with the card at init time. The gpios are each optional allowing for a headphone, microphone, or combination devices to be created depending on the board configuration. A board will be able to have several of these jacks, one for each physical connection. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Tested on Acer Chromebook 13 Tegra device. I'm travelling this week, I'll try to get the rockchip team to integrate this with simple card for the 3288. Changes since v1: Static component_driver Coding style Improve error handling Get GPIOs in device probe to better handle EPROBE_DEFER --- .../devicetree/bindings/sound/gpio-audio-jack.txt | 39 ++++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/gpio-audio-jack.c | 201 +++++++++++++++++++++ 4 files changed, 247 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/gpio-audio-jack.txt create mode 100644 sound/soc/codecs/gpio-audio-jack.c diff --git a/Documentation/devicetree/bindings/sound/gpio-audio-jack.txt b/Documentation/devicetree/bindings/sound/gpio-audio-jack.txt new file mode 100644 index 0000000..5505a49 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/gpio-audio-jack.txt @@ -0,0 +1,39 @@ +Gpio-Audio-Jack: + +GPIO based audio jacks. This can represent a headphone, mic, or combo jack. + +Required properties: + +- compatible : "linux,gpio-audio-jack" + +Optional properties: + +- gpio-audio-jack,jack-name : Name of the jack. Normally + 'Headphone Jack', 'Mic Jack', or + 'Headset Jack' +- gpio-audio-jack,sw-det-gpios : Reference to GPIOs that signals when + a jack is plugged. +- gpio-audio-jack,gpio-names : Names of the GPIOs. Must provide one + per sw-det-gpio entry. +- gpio-audio-jack,report-masks : Jack report mask from + dt-bindings/sound/audio-jack-events.h. + Must have one per sw-det-gpio entry. +- gpio-audio-jack,debounce-times : Debounce time in milliseconds for each + sw-det-gpio entry. + +Example of Headphone/Mic combo jack: + +audio_jack: gpio-audio-jack { + compatible = "linux,gpio-audio-jack"; + gpio-audio-jack,jack-name = "Headset Jack"; + gpio-audio-jack,sw-det-gpios = + <&gpio TEGRA_GPIO(I, 7) GPIO_ACTIVE_HIGH>, + <&gpio TEGRA_GPIO(R, 7) GPIO_ACTIVE_LOW>; + gpio-audio-jack,gpio-names = "Headphones", "Mic Jack"; + gpio-audio-jack,report-masks = <JACK_HEADPHONE>, <JACK_MICROPHONE>; + gpio-audio-jack,debounce-times = <150>, <150>; +}; + +sound { + nvidia,headset-dev = <&audio_jack>; + ... diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..4147593 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -62,6 +62,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C + select SND_SOC_GPIO_AUDIO_JACK select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -444,6 +445,10 @@ config SND_SOC_ES8328_SPI tristate select SND_SOC_ES8328 +config SND_SOC_GPIO_AUDIO_JACK + tristate "GPIO based audio jack detection" + depends on GPIOLIB || COMPILE_TEST + config SND_SOC_ISABELLE tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index abe2d7e..ae570f5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-gpio-audio-jack-objs := gpio-audio-jack.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -240,6 +241,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_GPIO_AUDIO_JACK) += snd-soc-gpio-audio-jack.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/gpio-audio-jack.c b/sound/soc/codecs/gpio-audio-jack.c new file mode 100644 index 0000000..29fdd8a --- /dev/null +++ b/sound/soc/codecs/gpio-audio-jack.c @@ -0,0 +1,201 @@ +/* + * GPIO based audio jack detection + * + * Copyright (C) 2015 Google, 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. + */ + +#include <linux/gpio.h> +#include <linux/init.h> +#include <linux/module.h> +#include <dt-bindings/sound/audio-jack-events.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/soc.h> + +#define JACK_NAME_CON_ID "gpio-audio-jack,jack-name" +#define GPIO_CON_ID "gpio-audio-jack,sw-det" +#define NAME_CON_ID "gpio-audio-jack,gpio-names" +#define REPORT_MASK_CON_ID "gpio-audio-jack,report-masks" +#define DEBOUNCE_CON_ID "gpio-audio-jack,debounce-times" + +struct gpio_audio_jack { + struct snd_soc_jack jack; + struct snd_soc_jack_gpio *gpios; + const char *jack_name; + int report_mask; + int gpio_count; +}; + +static int dt_jack_to_alsa_report(int dt_jack) +{ + switch (dt_jack) { + case JACK_HEADPHONE: + return SND_JACK_HEADPHONE; + case JACK_MICROPHONE: + return SND_JACK_MICROPHONE; + case JACK_LINEOUT: + return SND_JACK_LINEOUT; + case JACK_LINEIN: + return SND_JACK_LINEIN; + default: + return 0; + } +} + +static int gpio_audio_component_probe(struct snd_soc_component *component) +{ + struct device *dev = component->dev; + struct gpio_audio_jack *priv = dev_get_drvdata(dev); + int ret; + + ret = snd_soc_card_jack_new(component->card, priv->jack_name, + priv->report_mask, &priv->jack, NULL, 0); + if (ret) + return ret; + + return snd_soc_jack_add_gpiods(dev, &priv->jack, + priv->gpio_count, priv->gpios); +} + +static const struct snd_soc_component_driver gpio_jack_drv = { + .name = "gpio-audio-jack-dev", + .probe = gpio_audio_component_probe, +}; + +static int gpio_audio_jack_probe(struct platform_device *pdev) +{ + struct gpio_audio_jack *priv; + struct device *dev = &pdev->dev; + const char **gpio_names; + u32 *debounce; + u32 *gpio_report; + int ret; + int i; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + dev_set_drvdata(&pdev->dev, priv); + + ret = device_property_read_string(dev, JACK_NAME_CON_ID, + &priv->jack_name); + if (ret < 0) { + dev_err(dev, "Failed to read jack name %d\n", ret); + return ret; + } + + priv->gpio_count = gpiod_count(dev, GPIO_CON_ID); + if (priv->gpio_count <= 0) + return priv->gpio_count; + + gpio_names = devm_kcalloc(dev, priv->gpio_count, sizeof(*gpio_names), + GFP_KERNEL); + if (!gpio_names) + return -ENOMEM; + + gpio_report = devm_kcalloc(dev, priv->gpio_count, sizeof(*gpio_report), + GFP_KERNEL); + if (!gpio_report) + return -ENOMEM; + + debounce = devm_kcalloc(dev, priv->gpio_count, sizeof(*debounce), + GFP_KERNEL); + if (!debounce) + return -ENOMEM; + + priv->gpios = devm_kcalloc(dev, priv->gpio_count, sizeof(*priv->gpios), + GFP_KERNEL); + if (!priv->gpios) + return -ENOMEM; + + ret = device_property_read_string_array(dev, NAME_CON_ID, gpio_names, + priv->gpio_count); + if (ret < 0) { + dev_err(dev, "Failed to read gpio names %d\n", ret); + return ret; + } + + ret = device_property_read_u32_array(dev, REPORT_MASK_CON_ID, + gpio_report, priv->gpio_count); + if (ret < 0) { + dev_err(dev, "Failed to read gpio masks %d\n", ret); + return ret; + } + + ret = device_property_read_u32_array(dev, DEBOUNCE_CON_ID, + debounce, priv->gpio_count); + if (ret < 0) { + dev_err(dev, "Failed to read gpio debounce times %d\n", ret); + return ret; + } + + for (i = 0; i < priv->gpio_count; i++) { + priv->gpios[i].desc = gpiod_get_index(dev, GPIO_CON_ID, + i, GPIOD_IN); + if (IS_ERR(priv->gpios[i].desc)) { + ret = PTR_ERR(priv->gpios[i].desc); + dev_err(priv->gpios[i].gpiod_dev, + "Cannot get jack gpio at index %d: %d", + i, ret); + goto gpio_err; + } + + priv->gpios[i].name = gpio_names[i]; + priv->gpios[i].report = dt_jack_to_alsa_report(gpio_report[i]); + priv->gpios[i].debounce_time = debounce[i]; + priv->report_mask |= gpio_report[i]; + } + + devm_kfree(dev, gpio_names); + devm_kfree(dev, gpio_report); + devm_kfree(dev, debounce); + + return devm_snd_soc_register_component(&pdev->dev, &gpio_jack_drv, + NULL, 0); + +gpio_err: + for (i = 0; i < priv->gpio_count; i++) { + if (!IS_ERR(priv->gpios[i].desc)) + gpiod_put(priv->gpios[i].desc); + } + return ret; +} + +static int gpio_audio_jack_remove(struct platform_device *pdev) +{ + struct gpio_audio_jack *priv = dev_get_drvdata(&pdev->dev); + int i; + + for (i = 0; i < priv->gpio_count; i++) { + if (!IS_ERR(priv->gpios[i].desc)) + gpiod_put(priv->gpios[i].desc); + } + + return 0; +} + +static const struct of_device_id gpio_audio_of_match[] = { + { .compatible = "linux,gpio-audio-jack", }, + { } +}; +MODULE_DEVICE_TABLE(of, gpio_audio_of_match); + +static struct platform_driver gpio_audio_jack = { + .driver = { + .name = "gpio-audio-jack", + .of_match_table = gpio_audio_of_match, + }, + .probe = gpio_audio_jack_probe, + .remove = gpio_audio_jack_remove, +}; +module_platform_driver(gpio_audio_jack); + +MODULE_DESCRIPTION("ASoC GPIO audio jack driver"); +MODULE_AUTHOR("Dylan Reid <dgreid(a)chromium.org>"); +MODULE_LICENSE("GPL v2"); -- 2.2.1.62.g3f15098

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[alsa-devel] [PATCH] ASoC: max98925: Fix mask for setting DAI invert mode
by Axel Lin 26 May '15

26 May '15

The M98925_DAI_WCI_MASK bit is not updated with current code. To properly set the DAI invert mode, the mask should be M98925_DAI_BCI_MASK | M98925_DAI_WCI_MASK. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98925.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index 9b5a17d..aad6642 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -346,7 +346,7 @@ static int max98925_dai_set_fmt(struct snd_soc_dai *codec_dai, } regmap_update_bits(max98925->regmap, MAX98925_FORMAT, - M98925_DAI_BCI_MASK, invert); + M98925_DAI_BCI_MASK | M98925_DAI_WCI_MASK, invert); return 0; } -- 2.1.0

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[alsa-devel] [PATCH] ASoC: Simplify format_register_str() without stack usages
by Takashi Iwai 26 May '15

26 May '15

Instead of allocating two string buffers on stack and copying them back, manipulate directly the target string buffer. This simplifies the code well. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-core.c | 27 +++++++++------------------ 1 file changed, 9 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 23732523f87c..ee86f84b7ccc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -92,30 +92,21 @@ static int format_register_str(struct snd_soc_codec *codec, int wordsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; int regsize = codec->driver->reg_word_size * 2; int ret; - char tmpbuf[len + 1]; - char regbuf[regsize + 1]; - - /* since tmpbuf is allocated on the stack, warn the callers if they - * try to abuse this function */ - WARN_ON(len > 63); /* +2 for ': ' and + 1 for '\n' */ if (wordsize + regsize + 2 + 1 != len) return -EINVAL; - ret = snd_soc_read(codec, reg); - if (ret < 0) { - memset(regbuf, 'X', regsize); - regbuf[regsize] = '\0'; - } else { - snprintf(regbuf, regsize + 1, "%.*x", regsize, ret); - } - - /* prepare the buffer */ - snprintf(tmpbuf, len + 1, "%.*x: %s\n", wordsize, reg, regbuf); - /* copy it back to the caller without the '\0' */ - memcpy(buf, tmpbuf, len); + sprintf(buf, "%.*x: ", wordsize, reg); + buf += wordsize + 2; + ret = snd_soc_read(codec, reg); + if (ret < 0) + memset(buf, 'X', regsize); + else + sprintf(buf, "%.*x", regsize, ret); + buf[regsize] = '\n'; + /* no NUL-termination needed */ return 0; } -- 2.4.1

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Re: [alsa-devel] ALSA: firewire-lib: pass the number of data blocks in incoming packets to outgoing packets
by Dan Carpenter 26 May '15

26 May '15

Hello Takashi Sakamoto, The patch 6fc6b9ce41c6: "ALSA: firewire-lib: pass the number of data blocks in incoming packets to outgoing packets" from May 22, 2015, leads to the following static checker warning: sound/firewire/amdtp.c:857 in_stream_callback() warn: unsigned 'data_blocks' is never less than zero. sound/firewire/amdtp.c 855 856 data_blocks = handle_in_packet(s, payload_quadlets, buffer); 857 if (data_blocks < 0) { ^^^^^^^^^^^^^^^ Never true. But handle_in_packet() does return negative error codes. 858 s->packet_index = -1; 859 break; 860 } 861 regards, dan carpenter

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[alsa-devel] [PATCH 0/2] reducing memory footprint of ALSA
by Jie Yang 26 May '15

26 May '15

We are planning to work on reducing memory footprint of ALSA, targeting on small memory embedded devices(e.g. IoT), this series focuses on disabling procfs, hw/sw params refinement, mmap, dpcm, dapm, compressed API, compat code APIs... These first two patches is a start, add config menu in Kconfig, and disable ALSA proc FS once reducing memory footprint is selected, which can save about 9KB code size on x86_64 platform. Comments are welcomed. ~Keyon Jie Yang (2): ALSA: Kconfig: add config menu for reduced memory footprint ALSA: set no sound proc fs for reduced memory footprint include/sound/info.h | 8 ++++---- sound/Kconfig | 16 ++++++++++++++++ sound/core/Makefile | 2 +- sound/core/hwdep.c | 6 +++--- sound/core/init.c | 8 ++++---- sound/core/oss/mixer_oss.c | 6 +++--- sound/core/pcm.c | 6 +++--- sound/core/seq/oss/seq_oss.c | 6 +++--- sound/core/seq/oss/seq_oss_init.c | 5 ++--- sound/core/seq/oss/seq_oss_midi.c | 4 ++-- sound/core/seq/oss/seq_oss_readq.c | 4 ++-- sound/core/seq/oss/seq_oss_synth.c | 4 ++-- sound/core/seq/seq_clientmgr.c | 4 ++-- sound/core/seq/seq_device.c | 2 +- sound/core/seq/seq_info.h | 2 +- sound/core/seq/seq_queue.c | 4 ++-- sound/core/seq/seq_timer.c | 4 ++-- sound/core/sound.c | 4 ++-- sound/core/sound_oss.c | 4 ++-- sound/core/timer.c | 4 ++-- sound/pci/hda/Makefile | 3 ++- sound/pci/hda/hda_eld.c | 4 ++-- sound/pci/hda/hda_local.h | 4 ++-- sound/pci/hda/patch_hdmi.c | 4 ++-- sound/pci/hda/patch_sigmatel.c | 2 +- 25 files changed, 68 insertions(+), 52 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/5] Trivial cleanups
by Takashi Iwai 26 May '15

26 May '15

Hi, here is a series of patches to clean up / fix the issues spotted by sparse. Takashi === Takashi Iwai (5): ALSA: dummy: make local data static ALSA: lx6464es: Fix duplicated SSID entries ALSA: lx6464es: Use NULL for pointers ALSA: bcd2000: Make local data static sound: sound_firmware: Fix invalid use of vfs_read() sound/drivers/dummy.c | 14 +++++++------- sound/pci/lx6464es/lx6464es.c | 16 ++++++++-------- sound/sound_firmware.c | 4 +--- sound/usb/bcd2000/bcd2000.c | 2 +- 4 files changed, 17 insertions(+), 19 deletions(-) -- 2.4.1

1 5

Re: [alsa-devel] ALC256 alias name for Dell
by Takashi Iwai 26 May '15

26 May '15

At Mon, 25 May 2015 09:25:07 +0000, Kailang wrote: > > Hi Takashi, > > Dell has new platform for ALC256 codec. > It calls ALC3246 for codec. Applied, thanks. Takashi > > BR, > Kailang > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 0000-alc256-alias-name-for-dell.patch <application/octet-stream (base64)>] >

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Re: [alsa-devel] Bug 95261 - [SLK][Audio][HDA Analog] Audio capture is noisy if device is in RTD3 mode
by Takashi Iwai 26 May '15

26 May '15

At Mon, 25 May 2015 06:57:09 +0000, Pshou wrote: > > Hi Takashi, > This is when you shut up, occurred about five seconds record noise problems. > The problem, likely to occur in alc286 and sky board. I suppose the COEFs are specific to ALC286? If so, please either create alc286_shutup() or call COEF setups conditionally only for ALC286. thanks, Takashi > > Best regards > pshou > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 patch_realtek.diff <application/octet-stream (base64)>] >

2 1

[alsa-devel] [PATCH v2 0/3] ASoC: topology
by Liam Girdwood 25 May '15

25 May '15

ASoC topology (formerly called dynamic firmware) allows audio DSP topology to be defined as part of the firmware package and not hard coded into the driver. This allows DSP vendors to ship multiple different firmwares or firmwares with programmable topologies for different devices (i.e. phones, tablets, TVs) using a single DSP driver. The DSP topology can be contained as part of the firmware file or exist as separate files. The topology core and file format is generic text and not tied to any single vendor or device. The topology core consists of kernel and userspace components :- Kernel. The kernel core reads in the topology data and builds the driver capabilities and audio topology based on this data. This includes building the DAPM graph, setting PCM and DAI capabilities, registering kcontrols and registering DAPM widgets. The userspace tool allows the creation of binary DSP FW topology files based on a topology text files. The tool is temporarily available on github and its intended to be merged into the alsa-utils package after a clean up. https://github.com/lgirdwood/topology.git Changes since RFC :- o Fixed patch series to survive bisection builds. o Removed widget ID enum from UAPI. o Added channel mappings to controls. o Added TLV macro patch to UAPI (should this be in asound.h ?). o Fixed locking for runtime and init time topology loading. o Moved the UAPI group index field to the block header. Changes since V1 (back in 2012) :- o Name changed to ASoC Topology o Structures from soc.h and soc-dapm.h no longer exposed as UAPIs o Moved to component model from codec/platform model. o Added reserved fields in each UAPI structure. o UAPI structures packed and endianess defined for each structure. o Added support for PCM, DAI and codec <-> codec link capabilities. o Added support for client drivers manifests. o Added support for private data for each dynamic object type. o Single list of dynamic objects rather than list of each object type. Liam

1 0

[alsa-devel] [PATCH 0/5 v2] ALSA: firewire-lib: purge restriction of synchronization for non-blocking mode
by Takashi Sakamoto 25 May '15

25 May '15

This patchset renews my previous one. [alsa-devel] [PATCH 0/4] firewire-lib: purge restriction of synchronization for non-blocking mode http://mailman.alsa-project.org/pipermail/alsa-devel/2015-May/092086.html Changes: * Use stack to keep calculation result per callback of isochronous receive context. * Fix my stupid mistake to compare byte count to quadlet count. (oh...) * Rename flag to proper name. * Add 4th patch newly for better reading. Takashi Sakamoto (5): ALSA: firewire-lib: add buffer-over-run protection at receiving more data blocks than expected ALSA: firewire-lib: simplify function to calculate the number of data blocks ALSA: firewire-lib: pass the number of data blocks in incoming packets to outgoing packets ALSA: firewire-lib: set streaming error outside of packetization ALSA: firewire-lib: remove restriction for non-blocking mode sound/firewire/amdtp.c | 151 ++++++++++++++++++++++++-------------- sound/firewire/amdtp.h | 4 + sound/firewire/oxfw/oxfw-stream.c | 10 ++- 3 files changed, 106 insertions(+), 59 deletions(-) -- 2.1.4

2 7

[alsa-devel] [RESEND PATCH v1 0/2] Add dw_hdmi i2s audio driver support
by Yakir Yang 25 May '15

25 May '15

As Russell have send an series patches for dw_hdmi audio, it make us upload dw_hdmi audio patch more easier, and most important it works well on rockchip platform, so I make this patch that try to add another audio interface support in dw_hdmi driver. Here are the list that I have make in my series: - rename dw_hdmi-ahb-audio.h to dw_hdmi-audio.h - add dw_hdmi i2s audio driver Here are the series patches that I depends on: * drm: bridge/dw_hdmi: fixed codec style [Yakir Yang] * drm: bridge/dw_hdmi_rockchip: improve hdmi eye-diagram test * drm: bridge/dw_hdmi: improve hdmi single-end test * drm: imx/dw_hdmi: move phy comments [Russell King] * drm: bridge/dw_hdmi: clean up phy configuration * drm: bridge/dw_hdmi: combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() * drm: bridge/dw_hdmi: protect n/cts setting with a mutex * drm: bridge/dw_hdmi: adjust n/cts setting order * drm: bridge/dw_hdmi: clean up hdmi_set_clk_regenerator() * drm: bridge/dw_hdmi: use drm_hdmi_avi_infoframe_from_display_mode() * drm: bridge/dw_hdmi: simplify hdmi_config_AVI() a little * drm: bridge/dw_hdmi: remove mhsyncpolarity/mvsyncpolarity/minterlaced * drm: bridge/dw_hdmi: introduce interface to setting sample rate * drm: bridge/dw_hdmi: introduce interfaces to enable and disable audio * drm/edid: add function to help find SADs * sound/core: add DRM ELD helper * sound/core: add IEC958 channel status helper * drm: bridge/dw_hdmi-ahb-audio: add audio driver * drm: bridge/dw_hdmi-ahb-audio: parse ELD from HDMI driver Best regards, Yakir Yang Yakir Yang (2): drm: bridge/dw_hdmi: rename dw_hdmi-ahb-audio.h to dw_hdmi-audio.h drm: bridge/dw_hdmi-i2s-audio: add audio driver drivers/gpu/drm/bridge/Kconfig | 9 + drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw_hdmi-ahb-audio.c | 2 +- .../{dw_hdmi-ahb-audio.h => dw_hdmi-audio.h} | 0 drivers/gpu/drm/bridge/dw_hdmi-i2s-audio.c | 412 +++++++++++++++++++++ drivers/gpu/drm/bridge/dw_hdmi.c | 26 +- drivers/gpu/drm/bridge/dw_hdmi.h | 3 + 7 files changed, 442 insertions(+), 11 deletions(-) rename drivers/gpu/drm/bridge/{dw_hdmi-ahb-audio.h => dw_hdmi-audio.h} (100%) create mode 100644 drivers/gpu/drm/bridge/dw_hdmi-i2s-audio.c -- 2.1.2

3 4

[alsa-devel] ASoC: How to mix multi input ?
by Kuninori Morimoto 25 May '15

25 May '15

Hi Mark, ALSA SoC I would like to ask you how to mix multi input in ASoC. Renesas sound device(s) can mix multi input to 1 output. +-+ sound A --> |M| sound B --> |I| --> [ codec ] -> speaker sound C --> |X| +-+ Here, sound A/B/C will be used randomly. Does ALSA SoC already has this kind of system ? My image about these are like this. sound A: aplay -D hw:0 ... sound B: aplay -D hw:1 ... sound C: aplay -D hw:2 ... ... codec will be started if one of A/B/C are started, and it will be stopped if all of A/B/C are stopped.

4 8

[alsa-devel] [PATCH] ALSA: bebob: add Digidesign Mbox 2 Pro support
by Takashi Sakamoto 24 May '15

24 May '15

This device is based on DM1000E and BeBoB version 1 is installed. $ cat /proc/asound/cards 0 [Pro ]: BeBoB - Mbox 2 Pro DIGIDESIGN Mbox 2 Pro (id:1, rev:1), GUID 00a07e0100a90000 at fw1.0, S400 $ cat /proc/asound/Pro/firewire/firmware Manufacturer: bridgeCo Protocol Ver: 1 Build Ver: 0 GUID: 0x00A07E0100A90000 Model ID: 0x01 Model Rev: 1 Firmware Date: 20071031 Firmware Time: 034402 Firmware ID: 0xA9 Firmware Ver: 16777215 Base Addr: 0x20080000 Max Size: 1572864 Loader Date: 20051207 Loader Time: 205554 Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 1 + sound/firewire/bebob/bebob.c | 3 +++ 2 files changed, 4 insertions(+) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index ecec547..2a0a9aa 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -114,6 +114,7 @@ config SND_BEBOB * M-Audio FireWire410/AudioPhile/Solo * M-Audio Ozonic/NRV10/ProfireLightBridge * M-Audio FireWire 1814/ProjectMix IO + * Digidesign Mbox 2 Pro To compile this driver as a module, choose M here: the module will be called snd-bebob. diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 611b7da..e71bd35 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -57,6 +57,7 @@ static DECLARE_BITMAP(devices_used, SNDRV_CARDS); #define VEN_FOCUSRITE 0x0000130e #define VEN_MAUDIO1 0x00000d6c #define VEN_MAUDIO2 0x000007f5 +#define VEN_DIGIDESIGN 0x00a07e #define MODEL_FOCUSRITE_SAFFIRE_BOTH 0x00000000 #define MODEL_MAUDIO_AUDIOPHILE_BOTH 0x00010060 @@ -433,6 +434,8 @@ static const struct ieee1394_device_id bebob_id_table[] = { /* M-Audio ProjectMix */ SND_BEBOB_DEV_ENTRY(VEN_MAUDIO1, MODEL_MAUDIO_PROJECTMIX, &maudio_special_spec), + /* Digidesign Mbox 2 Pro */ + SND_BEBOB_DEV_ENTRY(VEN_DIGIDESIGN, 0x0000a9, &spec_normal), /* IDs are unknown but able to be supported */ /* Apogee, Mini-ME Firewire */ /* Apogee, Mini-DAC Firewire */ -- 2.1.4

2 4

[alsa-devel] [PATCH 0/4] ALSA: firewire-lib: code cleanup
by Takashi Sakamoto 24 May '15

24 May '15

Current implementation includes several macros and functions with improper name, hard-coded numerical value and so on. This patchset is for code cleanup. Takashi Sakamoto (4): ALSA: firewire-lib: rename local functions for code cleanup ALSA: firewire-lib: macro arrangement for code cleanup ALSA: firewire-lib: use dev_err() when detecting incoming streaming error ALSA: firewire-lib: use protocol error when detecting wrong value in CIP header sound/firewire/amdtp.c | 104 ++++++++++++++++++++++++++----------------------- 1 file changed, 55 insertions(+), 49 deletions(-) -- 2.1.4

3 11

Re: [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 24 May '15

24 May '15

Yet another experiment. Orpheus operation manual says there are six modes concerning Sync Source. -------------------------------------- Sync Source Master/Slave Description -------------------------------------- Local Master System is sync'ed to Orpheus' local clock Wordclock Master System is sync'ed to external wordclock applied to Orpheus DI Master System is sync'ed to external S/PDIF applied to Orpheus ADAT Master System is sync'ed to external ADAT (ADAT input must be enabled) PC DAW Slave Orpheus is slaved to system clock from host PC CSP Master System is sync'ed to Orpheus' 1394 clock (not recommended) -------------------------------------- With each of the five modes mentioned above (ADAT input not available), I tried to play music on Windows 7 using Audacity. The results are: ------------------- Local OK Wordclock NG DI NG PC DAW OK CSP OK ------------------- So it seems that if there is no external clock applied, when one is expected, Orpheus stops playing rather than trying to continue operation by free-running its clock. *** Takashi Sakamoto wrote: > Not funny, but headache. It's a customization by hardware vendor, I > mentioned. (in this case, by PrismSound Ltd.) > BridgeCo AG (currently ArchWave AG) produced DM1000/DM1100/DM1500 > chipset with BeBoB firmware. PrismSound Ltd. utilized the chipset and > firmware for streaming functionality of your Orpheus. > BeBoB firmware gives common ways to control itself, while the hadware > vendor added their own cusomization against it. I don't know exactly the > reason because there're few information about your device. I do not think I understand what you said here due to the lack of technical background, but what do you suggest at this moment to a casual applications programmer who wish to use snd-bebob with Orpheus? ------------ Answer 1 ------------ It should work. I received many successful reports for my final call for testing last year. It must be a hardware problem with your PC, or Firewire Cable, or Orpheus, or all of them. ------------ Answer 2 ------------ It should work. Check your software environment. Freshly installed Ubuntu 14.10 or later with the kernel 3.16 or later may not work. Always download the latest sources from GitHub, etc. ------------ Answer 3 ------------ Forget about snd-bebob, and fallback to libffado+jackd. The combination is loosing many packets, but if you say you do not notice any glitches, you should be completely happy with it. *** Regards, Mike Ood

2 2

[alsa-devel] [PATCH 1/3] ASoC: fsl_sai: add sai master mode support
by Zidan Wang 24 May '15

24 May '15

When sai works on master mode, set its bit clock and frame clock. SAI has 4 MCLK source, bus clock, MCLK1, MCLK2 and MCLK3. fsl_sai_set_bclk will select proper MCLK source, then calculate and set the bit clock divider. After fsl_sai_set_bclk, enable the selected mclk in hw_params(), and add hw_free() to disable the mclk. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 117 ++++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 9 +++- 2 files changed, 121 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ee2671b..1ccc10d1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1,7 +1,7 @@ /* * Freescale ALSA SoC Digital Audio Interface (SAI) driver. * - * Copyright 2012-2013 Freescale Semiconductor, Inc. + * Copyright 2012-2015 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 as published by the @@ -251,12 +251,14 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, val_cr4 |= FSL_SAI_CR4_FSD_MSTR; break; case SND_SOC_DAIFMT_CBM_CFM: + sai->is_slave_mode = true; break; case SND_SOC_DAIFMT_CBS_CFM: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; break; case SND_SOC_DAIFMT_CBM_CFS: val_cr4 |= FSL_SAI_CR4_FSD_MSTR; + sai->is_slave_mode = true; break; default: return -EINVAL; @@ -288,6 +290,79 @@ static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) return ret; } +static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(dai); + unsigned long clk_rate; + u32 savediv = 0, ratio, savesub = freq; + u32 id; + int ret = 0; + + /* Don't apply to slave mode */ + if (sai->is_slave_mode) + return 0; + + for (id = 0; id < FSL_SAI_MCLK_MAX; id++) { + clk_rate = clk_get_rate(sai->mclk_clk[id]); + if (!clk_rate) + continue; + + ratio = clk_rate / freq; + + ret = clk_rate - ratio * freq; + + /* + * Drop the source that can not be + * divided into the required rate. + */ + if (ret != 0 && clk_rate / ret < 1000) + continue; + + dev_dbg(dai->dev, + "ratio %d for freq %dHz based on clock %ldHz\n", + ratio, freq, clk_rate); + + if (ratio % 2 == 0 && ratio >= 2 && ratio <= 512) + ratio /= 2; + else + continue; + + if (ret < savesub) { + savediv = ratio; + sai->mclk_id[tx] = id; + savesub = ret; + } + + if (ret == 0) + break; + } + + if (savediv == 0) { + dev_err(dai->dev, "failed to derive required %cx rate: %d\n", + tx ? 'T' : 'R', freq); + return -EINVAL; + } + + if ((tx && sai->synchronous[TX]) || (!tx && !sai->synchronous[RX])) { + regmap_update_bits(sai->regmap, FSL_SAI_RCR2, + FSL_SAI_CR2_MSEL_MASK, + FSL_SAI_CR2_MSEL(sai->mclk_id[tx])); + regmap_update_bits(sai->regmap, FSL_SAI_RCR2, + FSL_SAI_CR2_DIV_MASK, savediv - 1); + } else { + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, + FSL_SAI_CR2_MSEL_MASK, + FSL_SAI_CR2_MSEL(sai->mclk_id[tx])); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, + FSL_SAI_CR2_DIV_MASK, savediv - 1); + } + + dev_dbg(dai->dev, "best fit: clock id=%d, div=%d, deviation =%d\n", + sai->mclk_id[tx], savediv, savesub); + + return 0; +} + static int fsl_sai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -297,6 +372,24 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, unsigned int channels = params_channels(params); u32 word_width = snd_pcm_format_width(params_format(params)); u32 val_cr4 = 0, val_cr5 = 0; + int ret; + + if (!sai->is_slave_mode) { + ret = fsl_sai_set_bclk(cpu_dai, tx, + 2 * word_width * params_rate(params)); + if (ret) + return ret; + + /* Do not enable the clock if it is already enabled */ + if (!(sai->mclk_streams & BIT(substream->stream))) { + ret = clk_prepare_enable(sai->mclk_clk[sai->mclk_id[tx]]); + if (ret) + return ret; + + sai->mclk_streams |= BIT(substream->stream); + } + + } if (!sai->is_dsp_mode) val_cr4 |= FSL_SAI_CR4_SYWD(word_width); @@ -322,6 +415,22 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, return 0; } +static int fsl_sai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + if (!sai->is_slave_mode && + sai->mclk_streams & BIT(substream->stream)) { + clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[tx]]); + sai->mclk_streams &= ~BIT(substream->stream); + } + + return 0; +} + + static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { @@ -428,6 +537,7 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { .set_sysclk = fsl_sai_set_dai_sysclk, .set_fmt = fsl_sai_set_dai_fmt, .hw_params = fsl_sai_hw_params, + .hw_free = fsl_sai_hw_free, .trigger = fsl_sai_trigger, .startup = fsl_sai_startup, .shutdown = fsl_sai_shutdown, @@ -600,8 +710,9 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->bus_clk = NULL; } - for (i = 0; i < FSL_SAI_MCLK_MAX; i++) { - sprintf(tmp, "mclk%d", i + 1); + sai->mclk_clk[0] = sai->bus_clk; + for (i = 1; i < FSL_SAI_MCLK_MAX; i++) { + sprintf(tmp, "mclk%d", i); sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 3466720..0662809 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -72,13 +72,15 @@ /* SAI Transmit and Recieve Configuration 2 Register */ #define FSL_SAI_CR2_SYNC BIT(30) -#define FSL_SAI_CR2_MSEL_MASK (0xff << 26) +#define FSL_SAI_CR2_MSEL_MASK (0x3 << 26) #define FSL_SAI_CR2_MSEL_BUS 0 #define FSL_SAI_CR2_MSEL_MCLK1 BIT(26) #define FSL_SAI_CR2_MSEL_MCLK2 BIT(27) #define FSL_SAI_CR2_MSEL_MCLK3 (BIT(26) | BIT(27)) +#define FSL_SAI_CR2_MSEL(ID) ((ID) << 26) #define FSL_SAI_CR2_BCP BIT(25) #define FSL_SAI_CR2_BCD_MSTR BIT(24) +#define FSL_SAI_CR2_DIV_MASK 0xff /* SAI Transmit and Recieve Configuration 3 Register */ #define FSL_SAI_CR3_TRCE BIT(16) @@ -120,7 +122,7 @@ #define FSL_SAI_CLK_MAST2 2 #define FSL_SAI_CLK_MAST3 3 -#define FSL_SAI_MCLK_MAX 3 +#define FSL_SAI_MCLK_MAX 4 /* SAI data transfer numbers per DMA request */ #define FSL_SAI_MAXBURST_TX 6 @@ -132,11 +134,14 @@ struct fsl_sai { struct clk *bus_clk; struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; + bool is_slave_mode; bool is_lsb_first; bool is_dsp_mode; bool sai_on_imx; bool synchronous[2]; + unsigned int mclk_id[2]; + unsigned int mclk_streams; struct snd_dmaengine_dai_dma_data dma_params_rx; struct snd_dmaengine_dai_dma_data dma_params_tx; }; -- 1.9.1

2 3

[alsa-devel] [PATCH trivial] ASoC: fsl_esai: Spelling s/specifially/specifically/
by Geert Uytterhoeven 24 May '15

24 May '15

Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/fsl/fsl_esai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 5c7597191e3ffc68..45eee131c47d43cd 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -517,7 +517,7 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, u32 bclk, mask, val; int ret; - /* Override slot_width if being specifially set */ + /* Override slot_width if being specifically set */ if (esai_priv->slot_width) slot_width = esai_priv->slot_width; -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Add quirk for MS LifeCam HD-3000
by Vittorio G (VittGam) 24 May '15

24 May '15

Microsoft LifeCam HD-3000 (045e:0779) needs a similar quirk for suppressing the unsupported sample rate inquiry. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98481 Cc: <stable(a)vger.kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Vittorio Gambaletta <linuxbugs(a)vittgam.net> --- The dmesg is showing this every time the ALSA audio is opened by the application: [3759907.370000] 2:3:1: cannot get freq at ep 0x82 And in fact I had to initialize ALSA and v4l2 in a specific order in my application in order for them to both work. diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 46facfc..2917534 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1118,6 +1118,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x075D): /* MS Lifecam Cinema */ case USB_ID(0x045E, 0x076D): /* MS Lifecam HD-5000 */ case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ + case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ return true; }

2 1

Re: [alsa-devel] [PATCH v2 1/3] ASoC: zx: Add ZTE zx296702 pcm support
by Jun Nie 23 May '15

23 May '15

2015-05-06 20:38 GMT+08:00 Jun Nie <jun.nie(a)linaro.org>: > This patch adds zx296702 ASoC DMA audio support. It > handle audio dma based on soc-dmaengine > Mark, Do you have any comments on this patch set? Or you suggest to add HDMI API calling in the same serial patch? For the HDMI calling, I have no good idea except direct call currently. Because I need dts hook to audio driver, while no HDMI call back data can be shared from video side to audio side in this independent initialization case. Jun > Signed-off-by: Jun Nie <jun.nie(a)linaro.org> > --- > sound/soc/Kconfig | 1 + > sound/soc/Makefile | 1 + > sound/soc/zte/Kconfig | 8 ++++++++ > sound/soc/zte/Makefile | 1 + > sound/soc/zte/zx296702-pcm.c | 46 ++++++++++++++++++++++++++++++++++++++++++++ > sound/soc/zte/zx296702-pcm.h | 6 ++++++ > 6 files changed, 63 insertions(+) > create mode 100644 sound/soc/zte/Kconfig > create mode 100644 sound/soc/zte/Makefile > create mode 100644 sound/soc/zte/zx296702-pcm.c > create mode 100644 sound/soc/zte/zx296702-pcm.h > > diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig > index 3ba52da..e2828e1 100644 > --- a/sound/soc/Kconfig > +++ b/sound/soc/Kconfig > @@ -57,6 +57,7 @@ source "sound/soc/tegra/Kconfig" > source "sound/soc/txx9/Kconfig" > source "sound/soc/ux500/Kconfig" > source "sound/soc/xtensa/Kconfig" > +source "sound/soc/zte/Kconfig" > > # Supported codecs > source "sound/soc/codecs/Kconfig" > diff --git a/sound/soc/Makefile b/sound/soc/Makefile > index 974ba70..57bf32d 100644 > --- a/sound/soc/Makefile > +++ b/sound/soc/Makefile > @@ -38,3 +38,4 @@ obj-$(CONFIG_SND_SOC) += tegra/ > obj-$(CONFIG_SND_SOC) += txx9/ > obj-$(CONFIG_SND_SOC) += ux500/ > obj-$(CONFIG_SND_SOC) += xtensa/ > +obj-$(CONFIG_SND_SOC) += zte/ > diff --git a/sound/soc/zte/Kconfig b/sound/soc/zte/Kconfig > new file mode 100644 > index 0000000..392ffa6 > --- /dev/null > +++ b/sound/soc/zte/Kconfig > @@ -0,0 +1,8 @@ > +config SND_SOC_ZX296702 > + tristate "SoC Audio for the ZX296702 System-on-Chip" > + depends on SOC_ZX296702 || COMPILE_TEST > + depends on COMMON_CLK > + select SND_SOC_GENERIC_DMAENGINE_PCM > + help > + Say Y or M here if you want support for SoC audio on ZX296702 > + chip. > diff --git a/sound/soc/zte/Makefile b/sound/soc/zte/Makefile > new file mode 100644 > index 0000000..f2a6af8 > --- /dev/null > +++ b/sound/soc/zte/Makefile > @@ -0,0 +1 @@ > +obj-$(CONFIG_SND_SOC_ZX296702) += zx296702-pcm.o > diff --git a/sound/soc/zte/zx296702-pcm.c b/sound/soc/zte/zx296702-pcm.c > new file mode 100644 > index 0000000..6e1a61a > --- /dev/null > +++ b/sound/soc/zte/zx296702-pcm.c > @@ -0,0 +1,46 @@ > +/* > + * Copyright (C) 2015 Linaro > + * > + * Author: Jun Nie <jun.nie(a)linaro.org> > + * > + * License terms: GNU General Public License (GPL) version 2 > + */ > + > +#include <linux/device.h> > +#include <linux/init.h> > +#include <linux/module.h> > +#include <sound/core.h> > +#include <sound/pcm.h> > +#include <sound/soc.h> > +#include <sound/dmaengine_pcm.h> > + > +#include "zx296702-pcm.h" > + > +static const struct snd_pcm_hardware snd_zx_hardware = { > + .info = SNDRV_PCM_INFO_MMAP | > + SNDRV_PCM_INFO_MMAP_VALID | > + SNDRV_PCM_INFO_INTERLEAVED | > + SNDRV_PCM_INFO_PAUSE | > + SNDRV_PCM_INFO_RESUME | > + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, > + .period_bytes_min = 32, > + .period_bytes_max = 16 * 1024, > + .periods_min = 2, > + .periods_max = 32, > + .buffer_bytes_max = 64 * 1024, > +}; > + > +static const struct snd_dmaengine_pcm_config zx_dmaengine_pcm_config = { > + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, > + .pcm_hardware = &snd_zx_hardware, > + .prealloc_buffer_size = 64 * 1024, > +}; > + > +int zx_pcm_platform_register(struct device *dev) > +{ > + return devm_snd_dmaengine_pcm_register(dev, &zx_dmaengine_pcm_config, > + 0); > +} > +EXPORT_SYMBOL_GPL(zx_pcm_platform_register); > + > +MODULE_LICENSE("GPL"); > diff --git a/sound/soc/zte/zx296702-pcm.h b/sound/soc/zte/zx296702-pcm.h > new file mode 100644 > index 0000000..4809a38 > --- /dev/null > +++ b/sound/soc/zte/zx296702-pcm.h > @@ -0,0 +1,6 @@ > +#ifndef _ZX_PCM_H > +#define _ZX_PCM_H > + > +int zx_pcm_platform_register(struct device *dev); > + > +#endif > -- > 1.9.1 >

2 4

Re: [alsa-devel] [PATCH v2 3/3] dt: Add documentation for the ZTE SPDIF controller
by Mark Brown 23 May '15

23 May '15

On Wed, May 06, 2015 at 08:38:05PM +0800, Jun Nie wrote: > This patch adds the devicetree documentation for the ZTE > zx296702 SPDIF audio controller. ...and here it is! Sorry - the DT binding is normally before the code in the patch series. This is fine, it's just patch 1 that needs to be resolved then.

2 1

[alsa-devel] [PATCH 1/2] ASoC: zx: Add ZTE zx296702 I2S DAI driver
by Jun Nie 23 May '15

23 May '15

Add ZTE zx296702 I2S DAI driver Signed-off-by: Jun Nie <jun.nie(a)linaro.org> --- sound/soc/zte/Kconfig | 7 + sound/soc/zte/Makefile | 1 + sound/soc/zte/zx296702-i2s.c | 437 +++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 445 insertions(+) create mode 100644 sound/soc/zte/zx296702-i2s.c diff --git a/sound/soc/zte/Kconfig b/sound/soc/zte/Kconfig index a08f94c..b3fffe8 100644 --- a/sound/soc/zte/Kconfig +++ b/sound/soc/zte/Kconfig @@ -13,3 +13,10 @@ config ZX296702_SPDIF help Say Y or M if you want to add support for codecs attached to the zx296702 spdif interface + +config ZX296702_I2S + bool "ZX296702 i2s" + select ZX_PCM_DMA + help + Say Y or M if you want to add support for codecs attached to the + zx296702 i2s interface diff --git a/sound/soc/zte/Makefile b/sound/soc/zte/Makefile index 6eafade..6791aeb 100644 --- a/sound/soc/zte/Makefile +++ b/sound/soc/zte/Makefile @@ -1,2 +1,3 @@ obj-$(CONFIG_SND_SOC_ZX296702) += zx296702-pcm.o obj-$(CONFIG_ZX296702_SPDIF) += zx296702-spdif.o +obj-$(CONFIG_ZX296702_I2S) += zx296702-i2s.o diff --git a/sound/soc/zte/zx296702-i2s.c b/sound/soc/zte/zx296702-i2s.c new file mode 100644 index 0000000..0688256 --- /dev/null +++ b/sound/soc/zte/zx296702-i2s.c @@ -0,0 +1,437 @@ +/* + * Copyright (C) 2015 Linaro + * + * Author: Jun Nie <jun.nie(a)linaro.org> + * + * License terms: GNU General Public License (GPL) version 2 + */ + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include <sound/core.h> +#include <sound/dmaengine_pcm.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "zx296702-pcm.h" + +#define ZX_I2S_PROCESS_CTRL 0x04 +#define ZX_I2S_TIMING_CTRL 0x08 +#define ZX_I2S_FIFO_CTRL 0x0C +#define ZX_I2S_FIFO_STATUS 0x10 +#define ZX_I2S_INT_EN 0x14 +#define ZX_I2S_INT_STATUS 0x18 +#define ZX_I2S_DATA 0x1C +#define ZX_I2S_FRAME_CNTR 0x20 + +#define I2S_DEAGULT_FIFO_THRES (0x10) +#define I2S_MAX_FIFO_THRES (0x20) + +#define ZX_I2S_PROCESS_TX_EN (1 << 0) +#define ZX_I2S_PROCESS_TX_DIS (0 << 0) +#define ZX_I2S_PROCESS_RX_EN (1 << 1) +#define ZX_I2S_PROCESS_RX_DIS (0 << 1) +#define ZX_I2S_PROCESS_I2S_EN (1 << 2) +#define ZX_I2S_PROCESS_I2S_DIS (0 << 2) + +#define ZX_I2S_TIMING_MAST (1 << 0) +#define ZX_I2S_TIMING_SLAVE (0 << 0) +#define ZX_I2S_TIMING_MS_MASK (1 << 0) +#define ZX_I2S_TIMING_LOOP (1 << 1) +#define ZX_I2S_TIMING_NOR (0 << 1) +#define ZX_I2S_TIMING_LOOP_MASK (1 << 1) +#define ZX_I2S_TIMING_PTNR (1 << 2) +#define ZX_I2S_TIMING_NTPR (0 << 2) +#define ZX_I2S_TIMING_PHASE_MASK (1 << 2) +#define ZX_I2S_TIMING_TDM (1 << 3) +#define ZX_I2S_TIMING_I2S (0 << 3) +#define ZX_I2S_TIMING_TIMING_MASK (1 << 3) +#define ZX_I2S_TIMING_LONG_SYNC (1 << 4) +#define ZX_I2S_TIMING_SHORT_SYNC (0 << 4) +#define ZX_I2S_TIMING_SYNC_MASK (1 << 4) +#define ZX_I2S_TIMING_TEAK_EN (1 << 5) +#define ZX_I2S_TIMING_TEAK_DIS (0 << 5) +#define ZX_I2S_TIMING_TEAK_MASK (1 << 5) +#define ZX_I2S_TIMING_STD_I2S (0 << 6) +#define ZX_I2S_TIMING_MSB_JUSTIF (1 << 6) +#define ZX_I2S_TIMING_LSB_JUSTIF (2 << 6) +#define ZX_I2S_TIMING_ALIGN_MASK (3 << 6) +#define ZX_I2S_TIMING_CHN_MASK (7 << 8) +#define ZX_I2S_TIMING_CHN(x) ((x - 1) << 8) +#define ZX_I2S_TIMING_LANE_MASK (3 << 11) +#define ZX_I2S_TIMING_LANE(x) ((x - 1) << 11) +#define ZX_I2S_TIMING_TSCFG_MASK (7 << 13) +#define ZX_I2S_TIMING_TSCFG(x) (x << 13) +#define ZX_I2S_TIMING_TS_WIDTH_MASK (0x1f << 16) +#define ZX_I2S_TIMING_TS_WIDTH(x) ((x - 1) << 16) +#define ZX_I2S_TIMING_DATA_SIZE_MASK (0x1f << 21) +#define ZX_I2S_TIMING_DATA_SIZE(x) ((x - 1) << 21) +#define ZX_I2S_TIMING_CFG_ERR_MASK (1 << 31) + +#define ZX_I2S_FIFO_CTRL_TX_RST (1 << 0) +#define ZX_I2S_FIFO_CTRL_TX_RST_MASK (1 << 0) +#define ZX_I2S_FIFO_CTRL_RX_RST (1 << 1) +#define ZX_I2S_FIFO_CTRL_RX_RST_MASK (1 << 1) +#define ZX_I2S_FIFO_CTRL_TX_DMA_EN (1 << 4) +#define ZX_I2S_FIFO_CTRL_TX_DMA_DIS (0 << 4) +#define ZX_I2S_FIFO_CTRL_TX_DMA_MASK (1 << 4) +#define ZX_I2S_FIFO_CTRL_RX_DMA_EN (1 << 5) +#define ZX_I2S_FIFO_CTRL_RX_DMA_DIS (0 << 5) +#define ZX_I2S_FIFO_CTRL_RX_DMA_MASK (1 << 5) +#define ZX_I2S_FIFO_CTRL_TX_THRES_MASK (0x1F << 8) +#define ZX_I2S_FIFO_CTRL_RX_THRES_MASK (0x1F << 16) + +#define CLK_RAT (32 * 4) + +struct zx_i2s_info { + struct snd_dmaengine_dai_dma_data dma_playback; + struct snd_dmaengine_dai_dma_data dma_capture; + struct clk *dai_clk; + void __iomem *reg_base; + int master; + resource_size_t mapbase; +}; + +static void zx_i2s_tx_en(void __iomem *base, bool on) +{ + unsigned long con; + + con = readl_relaxed(base + ZX_I2S_PROCESS_CTRL); + if (on) + con |= ZX_I2S_PROCESS_TX_EN | ZX_I2S_PROCESS_I2S_EN; + else + con &= ~(ZX_I2S_PROCESS_TX_EN | ZX_I2S_PROCESS_I2S_EN); + writel_relaxed(con, base + ZX_I2S_PROCESS_CTRL); +} + +static void zx_i2s_rx_en(void __iomem *base, bool on) +{ + unsigned long con; + + con = readl_relaxed(base + ZX_I2S_PROCESS_CTRL); + if (on) + con |= ZX_I2S_PROCESS_RX_EN | ZX_I2S_PROCESS_I2S_EN; + else + con &= ~(ZX_I2S_PROCESS_RX_EN | ZX_I2S_PROCESS_I2S_EN); + writel_relaxed(con, base + ZX_I2S_PROCESS_CTRL); +} + +static void zx_i2s_tx_dma_en(void __iomem *base, bool on) +{ + unsigned long con; + + con = readl_relaxed(base + ZX_I2S_FIFO_CTRL); + con |= ZX_I2S_FIFO_CTRL_TX_RST | (I2S_DEAGULT_FIFO_THRES << 8); + if (on) + con |= ZX_I2S_FIFO_CTRL_TX_DMA_EN; + else + con &= ~ZX_I2S_FIFO_CTRL_TX_DMA_EN; + writel_relaxed(con, base + ZX_I2S_FIFO_CTRL); +} + +static void zx_i2s_rx_dma_en(void __iomem *base, bool on) +{ + unsigned long con; + + con = readl_relaxed(base + ZX_I2S_FIFO_CTRL); + con |= ZX_I2S_FIFO_CTRL_RX_RST | (I2S_DEAGULT_FIFO_THRES << 16); + if (on) + con |= ZX_I2S_FIFO_CTRL_RX_DMA_EN; + else + con &= ~ZX_I2S_FIFO_CTRL_RX_DMA_EN; + writel_relaxed(con, base + ZX_I2S_FIFO_CTRL); +} + +#define ZX_I2S_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + 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) + +#define ZX_I2S_FMTBIT \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static int zx_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct zx_i2s_info *zx_i2s = dev_get_drvdata(dai->dev); + + snd_soc_dai_set_drvdata(dai, zx_i2s); + zx_i2s->dma_playback.addr = zx_i2s->mapbase + ZX_I2S_DATA; + zx_i2s->dma_playback.maxburst = 16; + zx_i2s->dma_capture.addr = zx_i2s->mapbase + ZX_I2S_DATA; + zx_i2s->dma_capture.maxburst = 16; + snd_soc_dai_init_dma_data(dai, &zx_i2s->dma_playback, + &zx_i2s->dma_capture); + return 0; +} + +static int zx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct zx_i2s_info *i2s = snd_soc_dai_get_drvdata(cpu_dai); + unsigned long con; + + con = readl_relaxed(i2s->reg_base + ZX_I2S_TIMING_CTRL); + con &= ~(ZX_I2S_TIMING_TIMING_MASK | ZX_I2S_TIMING_ALIGN_MASK | + ZX_I2S_TIMING_TEAK_MASK | ZX_I2S_TIMING_SYNC_MASK | + ZX_I2S_TIMING_MS_MASK); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + con |= (ZX_I2S_TIMING_I2S | ZX_I2S_TIMING_STD_I2S); + break; + case SND_SOC_DAIFMT_LEFT_J: + con |= (ZX_I2S_TIMING_I2S | ZX_I2S_TIMING_MSB_JUSTIF); + break; + case SND_SOC_DAIFMT_RIGHT_J: + con |= (ZX_I2S_TIMING_I2S | ZX_I2S_TIMING_LSB_JUSTIF); + break; + default: + dev_err(cpu_dai->dev, "Unknwon i2s timeing\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + i2s->master = 1; + con |= ZX_I2S_TIMING_MAST; + break; + case SND_SOC_DAIFMT_CBS_CFS: + i2s->master = 0; + con |= ZX_I2S_TIMING_SLAVE; + break; + default: + dev_err(cpu_dai->dev, "Unknwon master/slave format\n"); + return -EINVAL; + } + + writel_relaxed(con, i2s->reg_base + ZX_I2S_TIMING_CTRL); + return 0; +} + +static int zx_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *socdai) +{ + struct zx_i2s_info *i2s = snd_soc_dai_get_drvdata(socdai); + struct snd_dmaengine_dai_dma_data *dma_data; + unsigned int lane, ch_num, len, ret = 0; + unsigned long con, format; + unsigned long chn_cfg; + + dma_data = snd_soc_dai_get_dma_data(socdai, substream); + dma_data->addr_width = params_width(params) >> 3; + + con = readl_relaxed(i2s->reg_base + ZX_I2S_TIMING_CTRL); + con &= ~(ZX_I2S_TIMING_TS_WIDTH_MASK | ZX_I2S_TIMING_DATA_SIZE_MASK | + ZX_I2S_TIMING_LANE_MASK | ZX_I2S_TIMING_CHN_MASK | + ZX_I2S_TIMING_TSCFG_MASK); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + format = 0; + len = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + format = 1; + len = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + format = 2; + len = 32; + break; + default: + dev_err(socdai->dev, "Unknwon data format\n"); + return -EINVAL; + } + con |= ZX_I2S_TIMING_TS_WIDTH(len) | ZX_I2S_TIMING_DATA_SIZE(len); + + ch_num = params_channels(params); + switch (ch_num) { + case 1: + lane = 1; + chn_cfg = 2; + break; + case 2: + case 4: + case 6: + case 8: + lane = ch_num / 2; + chn_cfg = 3; + break; + default: + dev_err(socdai->dev, "Not support channel num %d\n", ch_num); + return -EINVAL; + } + con |= ZX_I2S_TIMING_LANE(lane); + con |= ZX_I2S_TIMING_TSCFG(chn_cfg); + con |= ZX_I2S_TIMING_CHN(ch_num); + writel_relaxed(con, i2s->reg_base + ZX_I2S_TIMING_CTRL); + + if (i2s->master) + ret = clk_set_rate(i2s->dai_clk, + params_rate(params) * ch_num * CLK_RAT); + return ret; +} + +static int zx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct zx_i2s_info *zx_i2s = dev_get_drvdata(dai->dev); + int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (capture) + zx_i2s_rx_dma_en(zx_i2s->reg_base, true); + else + zx_i2s_tx_dma_en(zx_i2s->reg_base, true); + /* fall thru */ + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (capture) + zx_i2s_rx_en(zx_i2s->reg_base, true); + else + zx_i2s_tx_en(zx_i2s->reg_base, true); + break; + case SNDRV_PCM_TRIGGER_STOP: + if (capture) + zx_i2s_rx_dma_en(zx_i2s->reg_base, false); + else + zx_i2s_tx_dma_en(zx_i2s->reg_base, false); + /* fall thru */ + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (capture) + zx_i2s_rx_en(zx_i2s->reg_base, false); + else + zx_i2s_tx_en(zx_i2s->reg_base, false); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int zx_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct zx_i2s_info *zx_i2s = dev_get_drvdata(dai->dev); + + return clk_prepare_enable(zx_i2s->dai_clk); +} + +static void zx_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct zx_i2s_info *zx_i2s = dev_get_drvdata(dai->dev); + + clk_disable_unprepare(zx_i2s->dai_clk); +} + +static struct snd_soc_dai_ops zx_i2s_dai_ops = { + .trigger = zx_i2s_trigger, + .hw_params = zx_i2s_hw_params, + .set_fmt = zx_i2s_set_fmt, + .startup = zx_i2s_startup, + .shutdown = zx_i2s_shutdown, +}; + +static const struct snd_soc_component_driver zx_i2s_component = { + .name = "zx-i2s", +}; + +struct snd_soc_dai_driver zx_i2s_dai = { + .name = "zx-i2s-dai", + .id = 0, + .probe = zx_i2s_dai_probe, + .playback = { + .channels_min = 1, + .channels_max = 8, + .rates = ZX_I2S_RATES, + .formats = ZX_I2S_FMTBIT, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ZX_I2S_RATES, + .formats = ZX_I2S_FMTBIT, + }, + .ops = &zx_i2s_dai_ops, +}; + +static int zx_i2s_probe(struct platform_device *pdev) +{ + struct resource *res; + struct zx_i2s_info *zx_i2s; + int ret; + + zx_i2s = kzalloc(sizeof(*zx_i2s), GFP_KERNEL); + if (!zx_i2s) + return -ENOMEM; + + zx_i2s->dai_clk = devm_clk_get(&pdev->dev, "tx"); + if (IS_ERR(zx_i2s->dai_clk)) { + dev_err(&pdev->dev, "Fail to get clk\n"); + return PTR_ERR(zx_i2s->dai_clk); + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + zx_i2s->mapbase = res->start; + zx_i2s->reg_base = devm_ioremap_resource(&pdev->dev, res); + if (!zx_i2s->reg_base) { + dev_err(&pdev->dev, "ioremap failed!\n"); + return -EIO; + } + + writel_relaxed(0, zx_i2s->reg_base + ZX_I2S_FIFO_CTRL); + platform_set_drvdata(pdev, zx_i2s); + + ret = snd_soc_register_component(&pdev->dev, &zx_i2s_component, + &zx_i2s_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Register DAI failed: %d\n", ret); + return ret; + } + + ret = zx_pcm_platform_register(&pdev->dev); + if (ret) + dev_err(&pdev->dev, "Register platform PCM failed: %d\n", ret); + + return ret; +} + +static const struct of_device_id zx_i2s_dt_ids[] = { + { .compatible = "zte,zx296702-i2s", }, + {} +}; +MODULE_DEVICE_TABLE(of, zx_i2s_dt_ids); + +static struct platform_driver i2s_driver = { + .probe = zx_i2s_probe, + .driver = { + .name = "zx-i2s", + .owner = THIS_MODULE, + .of_match_table = zx_i2s_dt_ids, + }, +}; + +module_platform_driver(i2s_driver); + +MODULE_AUTHOR("Jun Nie <jun.nie(a)linaro.org>"); +MODULE_DESCRIPTION("ZTE I2S SoC DAI"); +MODULE_LICENSE("GPL"); -- 1.9.1

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Re: [alsa-devel] getting the sound to work on my asus m6s00v
by Raymond Yau 23 May '15

23 May '15

> > I have done a couple of things > First I left the laptop off with no power connected; > This made the laptop awake in the non functional state regarding the soundcard. > I then used this opportunity to update to the latest kernel. > > With the latest kernel, the sound is still not working. > Unmuting the S/PDIF in alsamixer turn on red light in the headphone jack. > > also, using 3stack-digout option doesn't bring back sound. > > This is the output from the alsa information tool > http://www.alsa-project.org/db/?f=df46a9ee01e04dde11b194e0278c9f22594540d0 > > after updating this is the output from various commands > > [ > Nothing plugged in: > ./hdajacksensetest -a > Pin 0x14 (Green Headphone, Rear side): present = No > Pin 0x15 (Not connected): present = No > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Internal Mic): present = Yes > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = Yes > Pin 0x1b (Not connected): present = No > Pin 0x1c (Not connected): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Internal SPDIF Out, Mobile-In): present = No > Pin 0x1f (Not connected): present = No > > ./hda-verb /dev/snd/hwC0D0 0x1{4,5,8} SET_PIN_SENSE 0 nid > nid = 0x14, verb = 0x709, param = 0x0 > value = 0x0 > > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > > nid = 0x18, verb = 0x709, param = 0x0 > value = 0x0 > > ./hda-verb /dev/snd/hwC0D0 0x1{4,5,8} GET_PIN_SENSE 0 nid > nid = 0x14, verb = 0xf09, param = 0x0 > value = 0x80001f41 > > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0x80001f41 > > nid = 0x18, verb = 0xf09, param = 0x0 > value = 0x7fffffff > ] > > [ > With headphone plugged in > ./hdajacksensetest/hdajacksensetest -a > Pin 0x14 (Green Headphone, Rear side): present = No > Pin 0x15 (Not connected): present = Yes > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Internal Mic): present = Yes > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = Yes > Pin 0x1b (Not connected): present = No > Pin 0x1c (Not connected): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Internal SPDIF Out, Mobile-In): present = No > Pin 0x1f (Not connected): present = No > > // and also ecact same output for hda-verb as with nothing plugged in > ] > > [ > with Mic plugged in, the following items changed > > ./hdajacksensetest/hdajacksensetest -a > Pin 0x14 (Green Headphone, Rear side): present = Yes > Pin 0x15 (Not connected): present = Yes > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Internal Mic): present = Yes > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = Yes > Pin 0x1b (Not connected): present = No > Pin 0x1c (Not connected): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Internal SPDIF Out, Mobile-In): present = No > Pin 0x1f (Not connected): present = No > > ./hda-verb /dev/snd/hwC0D0 0x15 GET_PIN_SENSE 0 nid > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0x7fffffff > > ./hda-verb /dev/snd/hwC0D0 0x18 GET_PIN_SENSE 0 nid > nid = 0x18, verb = 0xf09, param = 0x0 > value = 0x80001f41 Seem impedance measurement is not correct, mic should have higher impedance than headphone https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… You can disable it by hint trigger_sense=0 - trigger_sense (bool): indicates that the jack detection needs the explicit call of AC_VERB_SET_PIN_SENSE verb Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Master Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Mic Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x0121411f: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Misc = NO_PRESENCE Pin-ctls: 0xc0: OUT HP Unsolicited: tag=00, enabled=0 Connection: 1 0x0c To get back headphone phantom jack if HP is really shared with speaker Disable hp mic by hint hp_mic_detect=0 - hp_mic_detect (bool): enable/disable the hp/mic shared input for a single built-in mic case; default true

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[alsa-devel] [PATCH v2 0/8] ASoC: Intel: refactor common SST IPC handling
by Vinod Koul 22 May '15

22 May '15

This series refactors the common SST IPC code to remove some hardcoded assumption which are no longer valid for future platforms like Skylake, These are made configurable and right configuration applied for respective drivers. The changes are mailbox sizes and dsp busy checks. changes in v2: - fix the mem allocation for IPC mailbox Subhransu S. Prusty (8): ASoC: Intel: Create an ops to check for DSP busy ASoC: Intel: Move the busy check to ops for Baytrail ASoC: Intel: Move the busy check to ops for HSW ASoC: Intel: Remove the direct register reference from common ipc ASoC: Intel: Allow to configure max size for mailbox data ASoC: Intel: Initialize max mailbox size for baytrail ASoC: Intel: Initialize max mailbox size for haswell ASoC: Intel: Allocate for the mailbox with max size sound/soc/intel/baytrail/sst-baytrail-ipc.c | 11 ++++++++++ sound/soc/intel/common/sst-ipc.c | 34 +++++++++++++++++++++++++---- sound/soc/intel/common/sst-ipc.h | 7 ++++-- sound/soc/intel/haswell/sst-haswell-ipc.c | 12 ++++++++++ 4 files changed, 58 insertions(+), 6 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v3 00/10] ASoC: qcom: add support to apq8016 audio
by Srinivas Kandagatla 22 May '15

22 May '15

Hi Mark, Am resending the the patches with rebased on top of topic/qcom branch. All these patches are tested and Acked. Thankyou for reviewing the v2 patches, here is the v3 patchset after incorporating review comments and testing on Storm Board. This patchset adds apq8016 audio support into lpass driver. Existing Lpass driver can not be used as-it-is for apq8016 as it contains code specific to ipq806x. Also the driver only supports single i2s port, single dma channel and single bitclk control. APQ8016 has 4 MI2S( Primary, Secondary, Tertiary, Quaternary) which can be routed to internal wcd codec or external codecs. This routing is controlled by 2 mux registers. This patch series firstly re-organizes the lpass driver such that the SOC specific bits are moved away from the driver. And secondly the SOC specifics are now passed as lpass variant data which would include various register offsets, dma channel allocations and SOC specific clock handling. Finally the last few patchs add apq8016 lpass and machine driver. All these patches are tested for HDMI audio via adv7533 bridge and Analog audio on APQ8016-SBC, msm8916-mtp boards and Kenneth tested this patchset on ipq806x Storm board too. Changes since v2(https://lkml.org/lkml/2015/5/16/236) - dropped 3 patches which are already applied by Mark. - rebased on top of topic/qcom branch. - changed dev_err to dev_warn spotted by Kenneth. - fixed typo spotted by kenneth. Changes since v1(https://lwn.net/Articles/644266/) - Fixed Kconfig dependencies spotted by Kenneth - Fixed compilation error on ipq806x spotted by Kenneth - Fixed clk error messages spotted by Kenneth Changes since RFC(https://lwn.net/Articles/642661/) - droped cleanup patches, as they are already applied by Mark. - prefixed shared functions with asoc_qcom_* as suggested by Kenneth - made LPASS driver depended on OF. - removed ipq806x specific bits from ipaif_reg.h - removed no_osr clk flag as suggested by Mark and Kenneth - fixed irq return values spotted by Kenneth - renamed the machine file to apq8016-sbc.c as suggested by Kenneth. - use dai->driver->id instead of dai->id as it breaks on Storm board. --srini Srinivas Kandagatla (10): ASoC: qcom: make lpass driver depend on OF ASoC: qcom: support bitclk and osrclk per i2s port ASoC: qcom: make osr clock optional ASoC: qcom: add dma channel control offset to variant data ASoC: qcom: Add ability to handle interrupts per dma channel ASoC: qcom: add bit map to track static dma channel allocations ASoC: qcom: Add apq8016 lpass driver support ASoC: qcom: add apq8016 sound card support ASoC: qcom: Document apq8016 bindings. ASoC: qcom: document apq8016 sbc machine driver bindings .../devicetree/bindings/sound/qcom,apq8016-sbc.txt | 61 ++++++ .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 13 +- include/dt-bindings/sound/apq8016-lpass.h | 9 + sound/soc/qcom/Kconfig | 17 ++ sound/soc/qcom/Makefile | 4 + sound/soc/qcom/apq8016_sbc.c | 215 ++++++++++++++++++ sound/soc/qcom/lpass-apq8016.c | 242 +++++++++++++++++++++ sound/soc/qcom/lpass-cpu.c | 77 +++++-- sound/soc/qcom/lpass-platform.c | 96 +++++--- sound/soc/qcom/lpass.h | 21 +- 10 files changed, 692 insertions(+), 63 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8016-sbc.txt create mode 100644 include/dt-bindings/sound/apq8016-lpass.h create mode 100644 sound/soc/qcom/apq8016_sbc.c create mode 100644 sound/soc/qcom/lpass-apq8016.c -- 1.9.1

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[alsa-devel] [PATCH 0/4 v2] ASoC: rsnd: fixup interrupt / spin lock issue
by Kuninori Morimoto 22 May '15

22 May '15

Hi Mark, Vinod Cc Laurent, Hiep-san These are v2 of fixup unknown interrupt issue patch which was reported by Hiep-san in v4.1-rcX kernel. It was not happened in v4.0 because ARM platform side couldn't use DMA transfer. We tested these same as v1 patches, and confirmed these solved problem Hiep-san, can you please re-test these patches ? Kuninori Morimoto (4): 1) dmaengine: rcar-dmac: fixup spinlock in rcar-dmac 2) ASoC: rsnd: indicate unknown HW start 3) ASoC: rsnd: add rsnd_dai_stream_quit() 4) ASoC: rsnd: spin lock for interrupt handler drivers/dma/sh/rcar-dmac.c | 55 +++++++++++++++++++++++++++++++------------------------ sound/soc/sh/rcar/core.c | 25 ++++++++++++++++++------- sound/soc/sh/rcar/rsnd.h | 3 +-- sound/soc/sh/rcar/src.c | 11 ++++++++--- sound/soc/sh/rcar/ssi.c | 18 ++++++++++++++---- 5 files changed, 72 insertions(+), 40 deletions(-) -- To unsubscribe from this list: send the line "unsubscribe linux-sh" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH 0/2 v2] ASoC: DPCM can care FE/BE merged format
by Kuninori Morimoto 22 May '15

22 May '15

Hi Mark These are v2 patches of new .dpcm_merged_foramt for merging FE/BE formats. It will be no sound if DPCM were below style, and user selects S24_LE. FE: S16_LE/S24_LE BE: S16_LE DPCM will selects FE/BE merged formats (= S16_LE here) if FE has .dpcm_merged_foramt. Kuninori Morimoto (2): ASoC: soc-pcm: DPCM cares BE format ASoC: rsnd: rsrc-card uses FE/BE merged format when DPCM include/sound/soc.h | 3 +++ sound/soc/sh/rcar/rsrc-card.c | 1 + sound/soc/soc-pcm.c | 47 ++++++++++++++++++++++++++++++++++++++++++----- 3 files changed, 46 insertions(+), 5 deletions(-) Best regards --- Kuninori Morimoto _______________________________________________ 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] [PATCH] ASoC: codecs-ac97: make selectable in config
by Maciej S. Szmigiero 22 May '15

22 May '15

Make generic ASoC AC'97 CODEC selectable in config. This way this driver can be used for platforms which don't need specialized AC'97 CODEC drivers but which are not directly selectable in config themselves (for example DT based ones). Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,7 +16,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 select SND_SOC_AB8500_CODEC if ABX500_CORE - select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS + select SND_SOC_AC97_CODEC select SND_SOC_AD1836 if SPI_MASTER select SND_SOC_AD193X_SPI if SPI_MASTER select SND_SOC_AD193X_I2C if I2C @@ -212,8 +212,9 @@ config SND_SOC_AB8500_CODEC tristate config SND_SOC_AC97_CODEC - tristate + tristate "Build generic ASoC AC97 CODEC driver" select SND_AC97_CODEC + select SND_SOC_AC97_BUS config SND_SOC_AD1836 tristate

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Re: [alsa-devel] [PATCH v2 2/3] ASoC: zx: Add zx296702 SPDIF support
by Mark Brown 22 May '15

22 May '15

On Wed, May 06, 2015 at 08:38:04PM +0800, Jun Nie wrote: > Add driver for zx296702 SPDIF controller This all seems fine apart from the fact that you're adding a new DT binding with no binding documentation - all new DT binndings require binnding documentation.

1 0

Re: [alsa-devel] [PATCH v2 1/3] ASoC: zx: Add ZTE zx296702 pcm support
by Mark Brown 22 May '15

22 May '15

On Wed, May 06, 2015 at 08:38:03PM +0800, Jun Nie wrote: > +static const struct snd_pcm_hardware snd_zx_hardware = { > + .info = SNDRV_PCM_INFO_MMAP | > + SNDRV_PCM_INFO_MMAP_VALID | > + SNDRV_PCM_INFO_INTERLEAVED | > + SNDRV_PCM_INFO_PAUSE | > + SNDRV_PCM_INFO_RESUME | > + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, > + .period_bytes_min = 32, > + .period_bytes_max = 16 * 1024, > + .periods_min = 2, > + .periods_max = 32, > + .buffer_bytes_max = 64 * 1024, > +}; As I said last time I'm not sure why you are manually defining these - the capabilities of the DMA controller should be being enumerated from the DMA controller driver. See dmaengine_pcm_set_runtime_hwparams().

1 0

[alsa-devel] how to ask you to write a new ALSA driver
by jean-noël chardron 22 May '15

22 May '15

hello, Last week I got a Komplet Audio 6 from Native intruments. The driver exist in ALSA driver for this king of interface audio (snd-usb-audio) (thanks for the team who write this) but for the model where Id Vendor is 0x17cc and Id product is 0x1000. The KA6 that I got has not the same ID product (0x1001) therefore I don't have any device capture even with the driver snd-usb-audio. The challenge is to write the modification for this new model. Who could do that ? If necessary I could send log, compile kernel source and so on. thanks, Jean-noel

2 3

[alsa-devel] [PATCH] ASoC: qcom: remove incorrect dependencies
by Arnd Bergmann 21 May '15

21 May '15

Compile-tests show a warning for the newly added SND_SOC_STORM symbol: warning: (SND_SOC_STORM) selects SND_SOC_LPASS_CPU which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && SND_SOC_QCOM) The problem is that it can be selected for COMPILE_TEST on non-QCOM builds, but the symbols it selects have a dependency. Dropping the dependencies makes it work without warnings and no other side-effects, because these are not user-visible. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: f380dd3f3cd ("ASoC: qcom: Add ability to build QCOM drivers") diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 5f58e4f1bca9..b07f183fc47f 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -6,12 +6,10 @@ config SND_SOC_QCOM config SND_SOC_LPASS_CPU tristate - depends on SND_SOC_QCOM select REGMAP_MMIO config SND_SOC_LPASS_PLATFORM tristate - depends on SND_SOC_QCOM select REGMAP_MMIO config SND_SOC_STORM

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[alsa-devel] [PATCH] ASoC: skip legacy dai naming if dai driver has all the information
by Srinivas Kandagatla 21 May '15

21 May '15

Original issue is that the id field in the dai is not same as the id in dai_driver when dai driver count == 1. This is due to the legacy dai naming check, which could possibly cause issues if the audio drivers written in assumption that dai->id would be always equal to dai_driver->id. This assumption is true only if the dai driver count is greater than 1, and false if dai driver count is 1. On Qcom Lpass driver we hit such issue while adding support to apq8016. The code path which falls back to legacy naming for cases where num_dai == 1 does not check if there is any valid information in the dai_driver. This patch fixes that by checking if the dai_driver has valid id and name before falling back to legacy dai naming Although the drivers can work around this issue by only using dai->driver->id, but this patch attempts to fix the actual issue. Suggested-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/soc-core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2373252..7d028e8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2599,7 +2599,8 @@ static int snd_soc_register_dais(struct snd_soc_component *component, * the same naming style even though those DAIs are not * component-less anymore. */ - if (count == 1 && legacy_dai_naming) { + if (count == 1 && legacy_dai_naming && + (dai_drv[i].id == 0 || dai_drv[i].name == NULL)) { dai->name = fmt_single_name(dev, &dai->id); } else { dai->name = fmt_multiple_name(dev, &dai_drv[i]); -- 1.9.1

3 2

[alsa-devel] Assert in pcm_params.c
by Alan Horstmann 21 May '15

21 May '15

Hi, The following assert is being hit in pcm_params.c: "pcm_params.c:2249: snd1_pcm_hw_params_slave: Assertion `err >= 0' failed." This is from a developer using Portaudio in his own application; I have yet to ascertain full details, but it is trying successive sample rates to see which are valid. Does this point to a flaw in the configuration or possibly a bug in Alsa-lib or can it occur just as a result of misuse of the Alsa API? Regards Alan

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[alsa-devel] [RFC PATCH 00/14] ASoC: qcom: add support to apq8016 audio
by Srinivas Kandagatla 21 May '15

21 May '15

Hi All, This patchset adds apq8016 audio support into lpass driver. Existing Lpass driver can not be used as-it-is for apq8016 as it contains code specific to ipq806x. Also the driver only supports single i2s port, single dma channel and single bitclk control. APQ8016 has 4 MI2S( Primary, Secondary, Tertiary, Quaternary) which can be routed to internal wcd codec or external codecs. This routing is controlled by 2 mux registers. This patch series firstly re-organizes the lpass driver such that the SOC specific bits are moved away from the driver. And secondly the SOC specifics are now passed as lpass variant data which would include various register offsets, dma channel allocations and SOC specific clock handling. Finally the patchset add apq8016 lpass and machine driver. This patchset also has two trivial cleanup patches which are to do with redundant checks and removing unnecessary header files. All these patches are tested for HDMI audio via adv7533 bridge and Analog audio on APQ8016-SBC and msm8916-mtp boards. I dont have access to ipq806x boards to test these patches. This is very first version of the patches which was developed with very mimimal/no access to IP documentation. I would like to get your opinon on the over all approch. Kenneth/Patrick, Could you please try these patches on storm board? Thanks, srini Srinivas Kandagatla (14): ASoC: qcom: Remove redundant error check. ASoC: qcom: remove unnecessary header files ASoC: qcom: move ipq806x specific bits out of lpass driver. ASoC: qcom: remove hardcoded i2s port number ASoC: qcom: remove hardcoded dma channel ASoC: qcom: support bitclk and osrclk per i2s port ASoC: qcom: add no osr clk flag to lpass variant ASoC: qcom: add dma channel control offset to variant data ASoC: qcom: Add ability to handle interrupts per dma channel ASoC: qcom: add bit map to track static dma channel allocations ASoC: qcom: Add apq8016 lpass driver support ASoC: qcom: add apq8016 sound card support ASoC: qcom: Document apq8016 bindings. ASoC: qcom: document apq8016 machine driver bindings .../bindings/sound/qcom,apq8016-machine.txt | 61 ++++++ .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 13 +- sound/soc/qcom/Kconfig | 25 ++- sound/soc/qcom/Makefile | 5 + sound/soc/qcom/apq8016.c | 214 ++++++++++++++++++ sound/soc/qcom/lpass-apq8016.c | 243 +++++++++++++++++++++ sound/soc/qcom/lpass-cpu.c | 234 ++++++++++---------- sound/soc/qcom/lpass-ipq806x.c | 95 ++++++++ sound/soc/qcom/lpass-lpaif-ipq806x.h | 172 --------------- sound/soc/qcom/lpass-lpaif-reg.h | 167 ++++++++++++++ sound/soc/qcom/lpass-platform.c | 197 +++++++++++------ sound/soc/qcom/lpass.h | 54 ++++- 12 files changed, 1121 insertions(+), 359 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8016-machine.txt create mode 100644 sound/soc/qcom/apq8016.c create mode 100644 sound/soc/qcom/lpass-apq8016.c create mode 100644 sound/soc/qcom/lpass-ipq806x.c delete mode 100644 sound/soc/qcom/lpass-lpaif-ipq806x.h create mode 100644 sound/soc/qcom/lpass-lpaif-reg.h -- 1.9.1

4 87

[alsa-devel] prealloc_buffer_size and buffer_bytes_max
by Peter Rosin 21 May '15

21 May '15

Hi! I got my hopes up when I read the commit message for c14e2591bf54 ASoC: atmel-pcm-dma: increase buffer_bytes_max. Quoting it: atmel-pcm-dma is not limited to a buffer size of 64kB like atmel-pcm-pdc. Increase buffer_bytes_max to 512kB to allow for higher bit rates (i.e. 32bps at 192kHz) to work correctly. By default, keep the prealloc at 64kB. However, as I (think I) request a bigger buffer it still caps out at 64kB. I'm using the latency argument of snd_pcm_set_params to control the buffer: snd_pcm_set_params(pcm, SND_PCM_FORMAT_S32_LE, SND_PCM_ACCESS_RW_INTERLEAVED, 2, /* channels */ 250000, /* rate */ 0, /* do not resample */ 90000); /* latency in us */ But snd_pcm_hw_params_get_buffer_size "only" returns 8192 frames (64kB) even if I request 9us * 250kHz * 2 * 4 = 180kB. If I change the prealloc from 64kB to 256kB I get a bigger buffer (and it works better too!). Admittedly I backported this patch on top of the linux-3.18-at91 branch from the Atmel git repo, so there might be some support missing that has gone in after 3.18? Or have I completely misunderstood, and these are unrelated buffers? Any insight in how I can get a big enough buffer without hacking the prealloc is appreciated! Cheers, Peter

2 3

[alsa-devel] snd-hda-intel : issues with mic boost on ALC3236
by Simon Derr 21 May '15

21 May '15

Hello, I'm having issues with the internal microphone on an Asus R553 (or S551) laptop. With the boost set to zero the microphone works but the level of the recording is very low. If I try another boost setting I don't get any signal anymore. I have uploaded alsa-info data here : http://www.alsa-project.org/db/?f=55b7fb0b4998757d099addae0a411e6a7cebc079 This is on a Linux 4.0 kernel. Any help is welcome, Regards, Simon

4 13

[alsa-devel] [REGRESSION] 4.1-rc4: Sound muted after boot
by Martin Steigerwald 21 May '15

21 May '15

Hello! On ThinkPad T520 (Intel Sandybridge) with 4.14-rc4 + two BTRFS patches and one Intel P-State debug patch I have no sound output on internal Intel HD audio codec. According to ThinkPad T520 LED with striked-through speaker the loud speaker is in disabled state. Also KDE´s Phonon says a soundcard is missing. In alsa-mixer I see nothing as MM for muted. If I press the key next to the LED I get a notification on the desktop that loud speaker has been enabled / disabled, but the LED still is on for it being disabled. I do not hear anything in Amarok or with aplay. alsa-mixer sound volumes are 100 for master and PCM. I also tried with disabling "Auto-Mute Mod", but still no sound. When I reboot and in BIOS or GRUB press the key then the LED toggles state. When I reboot into Linux 4.0.1 sound is working fine and loud speaker disabled LED is toggled off, so I bet this is a regression 4.1-rc4 compared to 4.0.1. The following is from non-working 4.1-rc4 kernel: merkaba:~> lspci -s 00:1b.0 -nnv 00:1b.0 Audio device [0403]: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller [8086:1c20] (rev 04) Subsystem: Lenovo Device [17aa:21cf] Flags: bus master, fast devsel, latency 0, IRQ 29 Memory at f2520000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel Capabilities: [130] Root Complex Link Kernel driver in use: snd_hda_intel merkaba:~> cat /proc/asound/cards 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xf2520000 irq 29 merkaba:~> cat /proc/asound/version Advanced Linux Sound Architecture Driver Version k4.1.0-rc4-tp520-btrfstrim- pstatetrace+. merkaba:~> cat /proc/asound/devices 2: [ 0] : control 3: [ 0- 0]: digital audio playback 4: [ 0- 0]: digital audio capture 5: [ 0- 3]: digital audio playback 6: [ 0- 7]: digital audio playback 7: [ 0- 8]: digital audio playback 8: [ 0- 0]: hardware dependent 9: [ 0- 1]: hardware dependent 10: [ 0- 3]: hardware dependent 33: : timer merkaba:~> merkaba:~> aplay -L null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server phonon HDA Intel PCH mit Mixer default:CARD=PCH HDA Intel PCH, CX20590 Analog Default Audio Device sysdefault:CARD=PCH HDA Intel PCH, CX20590 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog Front speakers surround21:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=PCH,DEV=0 HDA Intel PCH, HDMI 0 HDMI Audio Output hdmi:CARD=PCH,DEV=1 HDA Intel PCH, HDMI 1 HDMI Audio Output hdmi:CARD=PCH,DEV=2 HDA Intel PCH, HDMI 2 HDMI Audio Output dmix:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog Direct sample mixing device dmix:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Direct sample mixing device dmix:CARD=PCH,DEV=7 HDA Intel PCH, HDMI 1 Direct sample mixing device dmix:CARD=PCH,DEV=8 HDA Intel PCH, HDMI 2 Direct sample mixing device dsnoop:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog Direct sample snooping device dsnoop:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Direct sample snooping device dsnoop:CARD=PCH,DEV=7 HDA Intel PCH, HDMI 1 Direct sample snooping device dsnoop:CARD=PCH,DEV=8 HDA Intel PCH, HDMI 2 Direct sample snooping device hw:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog Direct hardware device without any conversions hw:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Direct hardware device without any conversions hw:CARD=PCH,DEV=7 HDA Intel PCH, HDMI 1 Direct hardware device without any conversions hw:CARD=PCH,DEV=8 HDA Intel PCH, HDMI 2 Direct hardware device without any conversions plughw:CARD=PCH,DEV=0 HDA Intel PCH, CX20590 Analog Hardware device with all software conversions plughw:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Hardware device with all software conversions plughw:CARD=PCH,DEV=7 HDA Intel PCH, HDMI 1 Hardware device with all software conversions plughw:CARD=PCH,DEV=8 HDA Intel PCH, HDMI 2 Hardware device with all software conversions Ciao, -- Martin 'Helios' Steigerwald - http://www.Lichtvoll.de GPG: 03B0 0D6C 0040 0710 4AFA B82F 991B EAAC A599 84C7

2 9

Re: [alsa-devel] [PATCH 1/4] ASoC: rockchip: add rockchip machine driver
by Mark Brown 21 May '15

21 May '15

On Wed, May 13, 2015 at 09:23:01PM +0800, zhengxing wrote: > On 2015年05月13日 03:22, Mark Brown wrote: > >Is it not possible to extend simple card to handle your use cases? > >Given the very generic naming and the fact that things like jack > >detection and so on should depend on the CODEC and board usually rather > >than on the SoC it doesn't sound like this is Rockchip specific. > >This also looks like you're reimplementing some device model enumeration > >stuff which probably shouldn't be happening but let's understand the > >problem you're trying to solve here before going too far into the code. > Because we are trying to bring rt5650 in the project, so we intend to > describe supported codecs with DTS via only a rockchip machine driver file, > others remain pre-implement(like max98090 / rt5645 that vendor machine > driver). I don't undertand what you're saying here, sorry - why is this not just a case of writing multiple machine drivers? > I communicated with Dylan, and he told me that the jack detection is an > issue in the simple-card, and suggested we are better to send them at > present. But what are these issues?

3 6

[alsa-devel] [PATCH] ASoC: mop500_ab8500: Add a NULL pointer check in mop500_ab8500_machine_init()
by Rajan Vaja 21 May '15

21 May '15

Avoid possible crash (NULL pointer dereference) by making sure that dem_kzalloc() is successful. Signed-off-by: Rajan Vaja <rajan.vaja(a)gmail.com> --- sound/soc/ux500/mop500_ab8500.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index b81a7a4c..85d810d 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -372,6 +372,10 @@ int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd) /* Create driver private-data struct */ drvdata = devm_kzalloc(dev, sizeof(struct mop500_ab8500_drvdata), GFP_KERNEL); + + if (!drvdata) + return -ENOMEM; + snd_soc_card_set_drvdata(rtd->card, drvdata); /* Setup clocks */ -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: jack: Remove MODULE_*() macros
by Takashi Iwai 21 May '15

21 May '15

The jack interface is statically included in sound core. Having doubly module information is rather confusing. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/jack.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/core/jack.c b/sound/core/jack.c index eb66327b5f1c..7237acbdcbbc 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -372,7 +372,3 @@ void snd_jack_report(struct snd_jack *jack, int status) } EXPORT_SYMBOL(snd_jack_report); - -MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); -MODULE_DESCRIPTION("Jack detection support for ALSA"); -MODULE_LICENSE("GPL"); -- 2.4.1

1 0

[alsa-devel] [PATCH] ASoC: rx51: use flags argument of devm_gpiod_get to set direction
by Uwe Kleine-König 20 May '15

20 May '15

Since 39b2bbe3d715 (gpio: add flags argument to gpiod_get*() functions) which appeared in v3.17-rc1, the gpiod_get* functions take an additional parameter that allows to specify direction and initial value for output. Use this to simplify the driver. Furthermore this is one caller less that stops us making the flags argument to gpiod_get*() mandatory. Fixes: 386e81ab3b4d ("ASoC: omap: rx51: get GPIO numbers via gpiod API") Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/omap/rx51.c | 30 ++++++++---------------------- 1 file changed, 8 insertions(+), 22 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index c2ddf0fbfa28..fded99362d39 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -455,50 +455,36 @@ static int rx51_soc_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, pdata); pdata->tvout_selection_gpio = devm_gpiod_get(card->dev, - "tvout-selection"); + "tvout-selection", + GPIOD_OUT_LOW); if (IS_ERR(pdata->tvout_selection_gpio)) { dev_err(card->dev, "could not get tvout selection gpio\n"); return PTR_ERR(pdata->tvout_selection_gpio); } - err = gpiod_direction_output(pdata->tvout_selection_gpio, 0); - if (err) { - dev_err(card->dev, "could not setup tvout selection gpio\n"); - return err; - } - pdata->jack_detection_gpio = devm_gpiod_get(card->dev, - "jack-detection"); + "jack-detection", + GPIOD_ASIS); if (IS_ERR(pdata->jack_detection_gpio)) { dev_err(card->dev, "could not get jack detection gpio\n"); return PTR_ERR(pdata->jack_detection_gpio); } - pdata->eci_sw_gpio = devm_gpiod_get(card->dev, "eci-switch"); + pdata->eci_sw_gpio = devm_gpiod_get(card->dev, "eci-switch", + GPIOD_OUT_HIGH); if (IS_ERR(pdata->eci_sw_gpio)) { dev_err(card->dev, "could not get eci switch gpio\n"); return PTR_ERR(pdata->eci_sw_gpio); } - err = gpiod_direction_output(pdata->eci_sw_gpio, 1); - if (err) { - dev_err(card->dev, "could not setup eci switch gpio\n"); - return err; - } - pdata->speaker_amp_gpio = devm_gpiod_get(card->dev, - "speaker-amplifier"); + "speaker-amplifier", + GPIOD_OUT_LOW); if (IS_ERR(pdata->speaker_amp_gpio)) { dev_err(card->dev, "could not get speaker enable gpio\n"); return PTR_ERR(pdata->speaker_amp_gpio); } - err = gpiod_direction_output(pdata->speaker_amp_gpio, 0); - if (err) { - dev_err(card->dev, "could not setup speaker enable gpio\n"); - return err; - } - err = devm_snd_soc_register_card(card->dev, card); if (err) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", err); -- 2.1.4

3 4

[alsa-devel] [PATCH] SoC: max98357a: use flags argument of devm_gpiod_get to set direction
by Uwe Kleine-König 20 May '15

20 May '15

Since 39b2bbe3d715 (gpio: add flags argument to gpiod_get*() functions) which appeared in v3.17-rc1, the gpiod_get* functions take an additional parameter that allows to specify direction and initial value for output. Use this to simplify the driver. Furthermore this is one caller less that stops us making the flags argument to gpiod_get*() mandatory. Fixes: af5adf129369 ("ASoC: max98357a: Add MAX98357A codec driver") Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/max98357a.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index bf3e933ee895..3a2fda08a893 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -60,13 +60,12 @@ static int max98357a_codec_probe(struct snd_soc_codec *codec) { struct gpio_desc *sdmode; - sdmode = devm_gpiod_get(codec->dev, "sdmode"); + sdmode = devm_gpiod_get(codec->dev, "sdmode", GPIOD_OUT_LOW); if (IS_ERR(sdmode)) { dev_err(codec->dev, "%s() unable to get sdmode GPIO: %ld\n", __func__, PTR_ERR(sdmode)); return PTR_ERR(sdmode); } - gpiod_direction_output(sdmode, 0); snd_soc_codec_set_drvdata(codec, sdmode); return 0; -- 2.1.4

3 2

[alsa-devel] [PATCH] SoC: sta32x: use devm_gpiod_get_optional for optional reset gpio
by Uwe Kleine-König 20 May '15

20 May '15

Since 39b2bbe3d715 (gpio: add flags argument to gpiod_get*() functions) which appeared in v3.17-rc1, the gpiod_get* functions take an additional parameter that allows to specify direction and initial value for output. Also there is a variant to find optional gpios that returns NULL if there is no gpio instead of -ENOENT. Make use of both features to simplify the driver. This changes behaviour if gpiod_get returns -ENOSYS which is the case if CONFIG_GPIOLIB is not enabled. This is a good change because without GPIOLIB there is no way to determine if the reset gpio is specified in the device tree. And if it is it must be handled, so erroring out is the right thing to do. Furthermore this is one caller less that stops us making the flags argument to gpiod_get*() mandatory. Fixes: b66a29808e1f ("ASoC: sta32x: make sta32x a gpio consumer for the reset GPIO") Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/sta32x.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 007a0e3bc273..0111baf9a5d4 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -1096,16 +1096,10 @@ static int sta32x_i2c_probe(struct i2c_client *i2c, #endif /* GPIOs */ - sta32x->gpiod_nreset = devm_gpiod_get(dev, "reset"); - if (IS_ERR(sta32x->gpiod_nreset)) { - ret = PTR_ERR(sta32x->gpiod_nreset); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta32x->gpiod_nreset = NULL; - } else { - gpiod_direction_output(sta32x->gpiod_nreset, 0); - } + sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(sta32x->gpiod_nreset)) + return PTR_ERR(sta32x->gpiod_nreset); /* regulators */ for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: omap: fix up SND_OMAP_SOC_OMAP_ABE_TWL6040 dependency, again
by Arnd Bergmann 20 May '15

20 May '15

I tried to fix this before and submitted a working patch, but after some discussion we came up with what seemed to be a nicer solution, resulting in commit 3d4cf65e2d ("ASoC: omap: fix up SND_OMAP_SOC_OMAP_ABE_TWL6040 dependency"). Unfortunately, that version was incomplete, and we still get this build error: drivers/clk/clk-palmas.c:46:16: error: field 'hw' has incomplete type drivers/clk/clk-palmas.c: In function 'to_palmas_clks_info': drivers/clk/clk-palmas.c:54:74: warning: initialization from incompatible pointer type [-Winc This happens only in randconfig builds that turn on MFD_PALMAS on a platform other than OMAP2+ when COMPILE_TEST is set but COMMON_CLK is not. The new approach is only 'select COMMON_CLK_PALMAS' if we know that we are on an OMAP5 platform and MFD_PALMAS is already set. This patch has survived thousands of randconfig builds and I don't see a remaining hole in the logic. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 3d4cf65e2d ("ASoC: omap: fix up SND_OMAP_SOC_OMAP_ABE_TWL6040 dependency") diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 6768e4f7d7d0..30d0109703a9 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -100,12 +100,13 @@ config SND_OMAP_SOC_OMAP_TWL4030 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || SOC_OMAP5 || COMPILE_TEST) + depends on TWL6040_CORE && SND_OMAP_SOC + depends on ARCH_OMAP4 || (SOC_OMAP5 && MFD_PALMAS) || COMPILE_TEST select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 select SND_SOC_DMIC - select COMMON_CLK_PALMAS if MFD_PALMAS + select COMMON_CLK_PALMAS if (SOC_OMAP5 && MFD_PALMAS) help Say Y if you want to add support for SoC audio on OMAP boards using ABE and twl6040 codec. This driver currently supports:

3 2

Re: [alsa-devel] [PATCH v3 06/17] video: lcd: add LoCoMo LCD driver
by Jingoo Han 20 May '15

20 May '15

> LoCoMo has some special handling for TFT screens attached to Collie and > Poodle. Implement that as a separate driver. > > Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> Acked-by: Jingoo Han <jingoohan1(a)gmail.com> Best regards, Jingoo Han > --- > drivers/video/backlight/Kconfig | 10 ++ > drivers/video/backlight/Makefile | 1 + > drivers/video/backlight/locomo_lcd.c | 285 +++++++++++++++++++++++++++++++++++ > 3 files changed, 296 insertions(+) > create mode 100644 drivers/video/backlight/locomo_lcd.c

1 0

[alsa-devel] [PATCH 1/7] ASoC: alc5623: Replace direct snd_soc_codec dapm field access
by Lars-Peter Clausen 20 May '15

20 May '15

The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm with snd_soc_codec_get_dapm(). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Full series depends on topic/dapm --- sound/soc/codecs/alc5623.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index e92b5ae..0fc24e0 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -893,7 +893,7 @@ static int alc5623_resume(struct snd_soc_codec *codec) static int alc5623_probe(struct snd_soc_codec *codec) { struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); alc5623_reset(codec); -- 1.8.0

2 7

[alsa-devel] [patch] ASoC: wm8741: check for error returns from wm8741_set_pdata()
by Dan Carpenter 20 May '15

20 May '15

Static checkers complain that "ret" is always zero so the conditions are never true. The intention here was clearly to check for errors from wm8741_set_pdata(). Although, since wm8741_set_pdata() never returns errors, it doesn't affect runtime. Fixes: c354b54cfdf6 ('ASoC: wm8741: Add differential mono mode support') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index c065ea1..09ff01f 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -613,7 +613,7 @@ static int wm8741_i2c_probe(struct i2c_client *i2c, return ret; } - wm8741_set_pdata(&i2c->dev, wm8741); + ret = wm8741_set_pdata(&i2c->dev, wm8741); if (ret != 0) { dev_err(&i2c->dev, "Failed to set pdata: %d\n", ret); return ret; @@ -679,7 +679,7 @@ static int wm8741_spi_probe(struct spi_device *spi) return ret; } - wm8741_set_pdata(&spi->dev, wm8741); + ret = wm8741_set_pdata(&spi->dev, wm8741); if (ret != 0) { dev_err(&spi->dev, "Failed to set pdata: %d\n", ret); return ret;

3 2

[alsa-devel] ALSA: dmix: pcm status incomplete
by Dan Hordern 20 May '15

20 May '15

I require timestamp information from the soundcard and host for our application. However, I noticed that when using "dmix:0,0" instead of "hw:0,0", the audio timestamp and delay frames are both zero. Looking in to the code, it seems that the 'snd_pcm_dmix_status' function does not populate either of these. In order to expose these, I have modified this function to get this information from the slave pcm; patch below. I was hoping for some feedback on this change. I am not sure if the delay value makes sense as yet, and whether there are consequences of using the hardware trigger timestamp etc. I am guessing the dsnoop and dshare plugins may also have this issue, though I have not checked. Thanks, Dan diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index babde6a..a4d7fc0 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -475,12 +475,12 @@ static int snd_pcm_dmix_status(snd_pcm_t *pcm, snd_pcm_status_t * status) break; } memset(status, 0, sizeof(*status)); + snd_pcm_status(dmix->spcm, status); status->state = snd_pcm_dmix_state(pcm); - status->trigger_tstamp = dmix->trigger_tstamp; - gettimestamp(&status->tstamp, pcm->tstamp_type); status->avail = snd_pcm_mmap_playback_avail(pcm); status->avail_max = status->avail > dmix->avail_max ? status->avail : dmix->avail_max; dmix->avail_max = 0; + status->delay = snd_pcm_mmap_playback_delay(pcm); return 0; }

2 1

[alsa-devel] [PATCH 0/4] ASoC: rsnd: fixup interrupt / spin lock issue
by Kuninori Morimoto 20 May '15

20 May '15

Hi Mark, Vinod Cc Laurent These patches fixup unknown interrupt issue which was reported by Hiep-san in v4.1-rcX kernel. It was not happened in v4.0 because ARM platform side couldn't use DMA transfer. --------- Unable to handle kernel NULL pointer dereference at virtual address 00000048 pgd = edfac000 [00000048] *pgd=6e0f0831, *pte=00000000, *ppte=00000000 Internal error: Oops: 17 [#1] SMP ARM CPU: 0 PID: 2009 Comm: aplay Not tainted 4.1.0-rc2-dirty #4 Hardware name: Generic R8A7790 (Flattened Device Tree) ... -------- Renesas sound driver had 2 issues. [1] spin lock was unlocked un-expected timing [2] HW restart (when error timing) is called without spin lock In our environment, it was difficult to reproduce this error, because it happens in very rare timing. But I noticed it is easy to reproduce it if I added this setting. [Kernel hacking] - [Lock Debugging] - all ON And we decided this error has gone by these tests. Test1: reproduce error 1. start / stop sound playback in 100 times 2. count error 3. test above in 10 sets result: 100% error happen. In average, this error happens once during 5 playbacks. Test2: apply these fixup patces, and retry Test1 result: 0 error happen. Kuninori Morimoto (4): 1) dmaengine: rcar-dmac: fixup spinlock in rcar-dmac 2) ASoC: rsnd: indicate unknown HW start 3) ASoC: rsnd: add rsnd_dai_stream_quit() 4) ASoC: rsnd: spin lock for interrupt handler drivers/dma/sh/rcar-dmac.c | 55 +++++++++++++++++++++++++++++++------------------------ sound/soc/sh/rcar/core.c | 25 ++++++++++++++++++------- sound/soc/sh/rcar/rsnd.h | 3 +-- sound/soc/sh/rcar/src.c | 11 ++++++++--- sound/soc/sh/rcar/ssi.c | 18 ++++++++++++++---- 5 files changed, 72 insertions(+), 40 deletions(-)

3 6

[alsa-devel] [PATCH] ALSA: hda - Reduce verbs by node power-saves
by Takashi Iwai 20 May '15

20 May '15

The widget (node) power-saves restore the widget states at each transition from D3 to D0 on each node. This was added in the commit [d545a57c5f84:ALSA: hda - Sync node attributes at resume from widget power saving]. However, the test was rater false-positive; this wasn't needed for any codecs. Since the resync may take significant number of additional verbs to be executed, it's better to reduce it. Let's disable it for now again. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 788f969b1a68..1c8678775f40 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -844,8 +844,16 @@ static hda_nid_t path_power_update(struct hda_codec *codec, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, state); changed = nid; + /* all known codecs seem to be capable to handl + * widgets state even in D3, so far. + * if any new codecs need to restore the widget + * states after D0 transition, call the function + * below. + */ +#if 0 /* disabled */ if (state == AC_PWRST_D0) snd_hdac_regmap_sync_node(&codec->core, nid); +#endif } } return changed; -- 2.4.1

1 0

[alsa-devel] [PATCH] ALSA: sound/atmel/ac97c.c: remove unused variable
by Arnd Bergmann 20 May '15

20 May '15

The recently added DT support for the ac97 driver is causing a gcc warning: sound/atmel/ac97c.c: In function 'atmel_ac97c_probe_dt': sound/atmel/ac97c.c:919:29: warning: unused variable 'match' [-Wunused-variable] const struct of_device_id *match; The variable is clearly unused, so we can remove it. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index cf4cedf2b420..6dad042630d8 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -916,7 +916,6 @@ static struct ac97c_platform_data *atmel_ac97c_probe_dt(struct device *dev) { struct ac97c_platform_data *pdata; struct device_node *node = dev->of_node; - const struct of_device_id *match; if (!node) { dev_err(dev, "Device does not have associated DT data\n");

3 2

[alsa-devel] [PATCH v2] ALSA: hda - Move hda_i915.c from sound/pci/hda to sound/hda
by mengdong.lin＠intel.com 20 May '15

20 May '15

From: Mengdong Lin <mengdong.lin(a)intel.com> The file is moved to hda core and renamed to hdac_i915.c, so can be used by both legacy HDA driver and new Skylake audio driver. - Add snd_hdac_ prefix to the public APIs. - The i915 audio component is moved to core bus and dynamically allocated. - A static pointer hdac_acomp is used to help bind/unbind callbacks to get this component, because the sound card's private_data is used by the azx chip pointer, which is a legacy structure. It could be removed if private _data changes to some core structure which can be extended to find the bus. - snd_hdac_get_display_clk() is added to get the display core clock for HSW/BDW. - haswell_set_bclk() is moved to hda_intel.c because it needs to write the controller registers EM4/EM5, and only legacy HD-A needs it for HSW/BDW. - Move definition of HSW/BDW-specific registers EM4/EM5 to hda_register.h and rename them to HSW_EM4/HSW_EM5, because other HD-A controllers have different layout for the extended mode registers. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h new file mode 100644 index 0000000..adb5ba5 --- /dev/null +++ b/include/sound/hda_i915.h @@ -0,0 +1,36 @@ +/* + * HD-Audio helpers to sync with i915 driver + */ +#ifndef __SOUND_HDA_I915_H +#define __SOUND_HDA_I915_H + +#ifdef CONFIG_SND_HDA_I915 +int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable); +int snd_hdac_display_power(struct hdac_bus *bus, bool enable); +int snd_hdac_get_display_clk(struct hdac_bus *bus); +int snd_hdac_i915_init(struct hdac_bus *bus); +int snd_hdac_i915_exit(struct hdac_bus *bus); +#else +static int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) +{ + return 0; +} +static inline int snd_hdac_display_power(struct hdac_bus *bus, bool enable) +{ + return 0; +} +static inline int snd_hdac_get_display_clk(struct hdac_bus *bus) +{ + return 0; +} +static inline int snd_hdac_i915_init(struct hdac_bus *bus) +{ + return -ENODEV; +} +static inline int snd_hdac_i915_exit(struct hdac_bus *bus) +{ + return 0; +} +#endif + +#endif /* __SOUND_HDA_I915_H */ diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index 4f6d3fc..0c7536e 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -84,6 +84,10 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_SD_BDLPL 0x18 #define AZX_REG_SD_BDLPU 0x1c +/* Haswell/Broadwell display HD-A controller Extended Mode registers */ +#define AZX_REG_HSW_EM4 0x100c +#define AZX_REG_HSW_EM5 0x1010 + /* PCI space */ #define AZX_PCIREG_TCSEL 0x44 diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index b97c59e..64fff4d 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -11,6 +11,7 @@ #include <sound/core.h> #include <sound/memalloc.h> #include <sound/hda_verbs.h> +#include <drm/i915_component.h> /* codec node id */ typedef u16 hda_nid_t; @@ -285,6 +286,10 @@ struct hdac_bus { /* locks */ spinlock_t reg_lock; struct mutex cmd_mutex; + + /* i915 component interface */ + struct i915_audio_component *audio_component; + int i915_power_refcount; }; int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev, diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 7a17fca..ac5ffac 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -4,3 +4,9 @@ config SND_HDA_CORE config SND_HDA_DSP_LOADER bool + +config SND_HDA_I915 + bool + default y + depends on DRM_I915 + depends on SND_HDA_CORE diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 5b4bb47..55dd465 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -4,4 +4,7 @@ snd-hda-core-objs := hda_bus_type.o hdac_bus.o hdac_device.o hdac_sysfs.o \ snd-hda-core-objs += trace.o CFLAGS_trace.o := -I$(src) +# for sync with i915 gfx driver +snd-hda-core-$(CONFIG_SND_HDA_I915) += hdac_i915.o + obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c new file mode 100644 index 0000000..cb78c25 --- /dev/null +++ b/sound/hda/hdac_i915.c @@ -0,0 +1,193 @@ +/* + * hdac_i915.c - routines for sync between HD-A core and i915 display driver + * + * 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. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pci.h> +#include <linux/component.h> +#include <drm/i915_component.h> +#include <sound/core.h> +#include <sound/hdaudio.h> +#include <sound/hda_i915.h> + +static struct i915_audio_component *hdac_acomp; + +int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) +{ + struct i915_audio_component *acomp = bus->audio_component; + + if (!acomp->ops) + return -ENODEV; + + if (!acomp->ops->codec_wake_override) { + dev_warn(bus->dev, + "Invalid codec wake callback\n"); + return 0; + } + + dev_dbg(bus->dev, "%s codec wakeup\n", + enable ? "enable" : "disable"); + + acomp->ops->codec_wake_override(acomp->dev, enable); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_set_codec_wakeup); + +int snd_hdac_display_power(struct hdac_bus *bus, bool enable) +{ + struct i915_audio_component *acomp = bus->audio_component; + + if (!acomp->ops) + return -ENODEV; + + dev_dbg(bus->dev, "display power %s\n", + enable ? "enable" : "disable"); + + if (enable) { + if (!bus->i915_power_refcount++) + acomp->ops->get_power(acomp->dev); + } else { + WARN_ON(!bus->i915_power_refcount); + if (!--bus->i915_power_refcount) + acomp->ops->put_power(acomp->dev); + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_display_power); + +int snd_hdac_get_display_clk(struct hdac_bus *bus) +{ + struct i915_audio_component *acomp = bus->audio_component; + + if (!acomp->ops) + return -ENODEV; + + return acomp->ops->get_cdclk_freq(acomp->dev); +} +EXPORT_SYMBOL_GPL(snd_hdac_get_display_clk); + +static int hdac_component_master_bind(struct device *dev) +{ + struct i915_audio_component *acomp = hdac_acomp; + int ret; + + ret = component_bind_all(dev, acomp); + if (ret < 0) + return ret; + + if (WARN_ON(!(acomp->dev && acomp->ops && acomp->ops->get_power && + acomp->ops->put_power && acomp->ops->get_cdclk_freq))) { + ret = -EINVAL; + goto out_unbind; + } + + /* + * Atm, we don't support dynamic unbinding initiated by the child + * component, so pin its containing module until we unbind. + */ + if (!try_module_get(acomp->ops->owner)) { + ret = -ENODEV; + goto out_unbind; + } + + return 0; + +out_unbind: + component_unbind_all(dev, acomp); + + return ret; +} + +static void hdac_component_master_unbind(struct device *dev) +{ + struct i915_audio_component *acomp = hdac_acomp; + + module_put(acomp->ops->owner); + component_unbind_all(dev, acomp); + WARN_ON(acomp->ops || acomp->dev); +} + +static const struct component_master_ops hdac_component_master_ops = { + .bind = hdac_component_master_bind, + .unbind = hdac_component_master_unbind, +}; + +static int hdac_component_master_match(struct device *dev, void *data) +{ + /* i915 is the only supported component */ + return !strcmp(dev->driver->name, "i915"); +} + +int snd_hdac_i915_init(struct hdac_bus *bus) +{ + struct component_match *match = NULL; + struct device *dev = bus->dev; + struct i915_audio_component *acomp; + int ret; + + acomp = kzalloc(sizeof(*acomp), GFP_KERNEL); + if (!acomp) + return -ENOMEM; + bus->audio_component = acomp; + hdac_acomp = acomp; + + component_match_add(dev, &match, hdac_component_master_match, bus); + ret = component_master_add_with_match(dev, &hdac_component_master_ops, + match); + if (ret < 0) + goto out_err; + + /* + * Atm, we don't support deferring the component binding, so make sure + * i915 is loaded and that the binding successfully completes. + */ + request_module("i915"); + + if (!acomp->ops) { + ret = -ENODEV; + goto out_master_del; + } + dev_dbg(dev, "bound to i915 component master\n"); + + return 0; +out_master_del: + component_master_del(dev, &hdac_component_master_ops); +out_err: + kfree(acomp); + bus->audio_component = NULL; + dev_err(dev, "failed to add i915 component master (%d)\n", ret); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_hdac_i915_init); + +int snd_hdac_i915_exit(struct hdac_bus *bus) +{ + struct device *dev = bus->dev; + struct i915_audio_component *acomp = bus->audio_component; + + WARN_ON(bus->i915_power_refcount); + if (bus->i915_power_refcount > 0 && acomp && acomp->ops) + acomp->ops->put_power(acomp->dev); + + component_master_del(dev, &hdac_component_master_ops); + + kfree(acomp); + bus->audio_component = NULL; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_i915_exit); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 117bf5c..98ced97 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -145,11 +145,6 @@ config SND_HDA_CODEC_HDMI comment "Set to Y if you want auto-loading the codec driver" depends on SND_HDA=y && SND_HDA_CODEC_HDMI=m -config SND_HDA_I915 - bool - default y - depends on DRM_I915 - config SND_HDA_CODEC_CIRRUS tristate "Build Cirrus Logic codec support" select SND_HDA_GENERIC diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 9c259ce..90e69b2 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,7 +1,5 @@ snd-hda-intel-objs := hda_intel.o snd-hda-tegra-objs := hda_tegra.o -# for haswell power well -snd-hda-intel-$(CONFIG_SND_HDA_I915) += hda_i915.o snd-hda-codec-y := hda_bind.o hda_codec.o hda_jack.o hda_auto_parser.o hda_sysfs.o snd-hda-codec-y += hda_controller.o diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c deleted file mode 100644 index 8521702..0000000 --- a/sound/pci/hda/hda_i915.c +++ /dev/null @@ -1,227 +0,0 @@ -/* - * hda_i915.c - routines for Haswell HDA controller power well support - * - * 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. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/pci.h> -#include <linux/component.h> -#include <drm/i915_component.h> -#include <sound/core.h> -#include "hda_controller.h" -#include "hda_intel.h" - -/* Intel HSW/BDW display HDA controller Extended Mode registers. - * EM4 (M value) and EM5 (N Value) are used to convert CDClk (Core Display - * Clock) to 24MHz BCLK: BCLK = CDCLK * M / N - * The values will be lost when the display power well is disabled. - */ -#define AZX_REG_EM4 0x100c -#define AZX_REG_EM5 0x1010 - -int hda_set_codec_wakeup(struct hda_intel *hda, bool enable) -{ - struct i915_audio_component *acomp = &hda->audio_component; - - if (!acomp->ops) - return -ENODEV; - - if (!acomp->ops->codec_wake_override) { - dev_warn(&hda->chip.pci->dev, - "Invalid codec wake callback\n"); - return 0; - } - - dev_dbg(&hda->chip.pci->dev, "%s codec wakeup\n", - enable ? "enable" : "disable"); - - acomp->ops->codec_wake_override(acomp->dev, enable); - - return 0; -} - -int hda_display_power(struct hda_intel *hda, bool enable) -{ - struct i915_audio_component *acomp = &hda->audio_component; - - if (!acomp->ops) - return -ENODEV; - - dev_dbg(&hda->chip.pci->dev, "display power %s\n", - enable ? "enable" : "disable"); - - if (enable) { - if (!hda->i915_power_refcount++) - acomp->ops->get_power(acomp->dev); - } else { - WARN_ON(!hda->i915_power_refcount); - if (!--hda->i915_power_refcount) - acomp->ops->put_power(acomp->dev); - } - - return 0; -} - -void haswell_set_bclk(struct hda_intel *hda) -{ - int cdclk_freq; - unsigned int bclk_m, bclk_n; - struct i915_audio_component *acomp = &hda->audio_component; - struct pci_dev *pci = hda->chip.pci; - - /* Only Haswell/Broadwell need set BCLK */ - if (pci->device != 0x0a0c && pci->device != 0x0c0c - && pci->device != 0x0d0c && pci->device != 0x160c) - return; - - if (!acomp->ops) - return; - - cdclk_freq = acomp->ops->get_cdclk_freq(acomp->dev); - switch (cdclk_freq) { - case 337500: - bclk_m = 16; - bclk_n = 225; - break; - - case 450000: - default: /* default CDCLK 450MHz */ - bclk_m = 4; - bclk_n = 75; - break; - - case 540000: - bclk_m = 4; - bclk_n = 90; - break; - - case 675000: - bclk_m = 8; - bclk_n = 225; - break; - } - - azx_writew(&hda->chip, EM4, bclk_m); - azx_writew(&hda->chip, EM5, bclk_n); -} - -static int hda_component_master_bind(struct device *dev) -{ - struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); - struct i915_audio_component *acomp = &hda->audio_component; - int ret; - - ret = component_bind_all(dev, acomp); - if (ret < 0) - return ret; - - if (WARN_ON(!(acomp->dev && acomp->ops && acomp->ops->get_power && - acomp->ops->put_power && acomp->ops->get_cdclk_freq))) { - ret = -EINVAL; - goto out_unbind; - } - - /* - * Atm, we don't support dynamic unbinding initiated by the child - * component, so pin its containing module until we unbind. - */ - if (!try_module_get(acomp->ops->owner)) { - ret = -ENODEV; - goto out_unbind; - } - - return 0; - -out_unbind: - component_unbind_all(dev, acomp); - - return ret; -} - -static void hda_component_master_unbind(struct device *dev) -{ - struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); - struct i915_audio_component *acomp = &hda->audio_component; - - module_put(acomp->ops->owner); - component_unbind_all(dev, acomp); - WARN_ON(acomp->ops || acomp->dev); -} - -static const struct component_master_ops hda_component_master_ops = { - .bind = hda_component_master_bind, - .unbind = hda_component_master_unbind, -}; - -static int hda_component_master_match(struct device *dev, void *data) -{ - /* i915 is the only supported component */ - return !strcmp(dev->driver->name, "i915"); -} - -int hda_i915_init(struct hda_intel *hda) -{ - struct component_match *match = NULL; - struct device *dev = &hda->chip.pci->dev; - struct i915_audio_component *acomp = &hda->audio_component; - int ret; - - component_match_add(dev, &match, hda_component_master_match, hda); - ret = component_master_add_with_match(dev, &hda_component_master_ops, - match); - if (ret < 0) - goto out_err; - - /* - * Atm, we don't support deferring the component binding, so make sure - * i915 is loaded and that the binding successfully completes. - */ - request_module("i915"); - - if (!acomp->ops) { - ret = -ENODEV; - goto out_master_del; - } - - dev_dbg(dev, "bound to i915 component master\n"); - - return 0; -out_master_del: - component_master_del(dev, &hda_component_master_ops); -out_err: - dev_err(dev, "failed to add i915 component master (%d)\n", ret); - - return ret; -} - -int hda_i915_exit(struct hda_intel *hda) -{ - struct device *dev = &hda->chip.pci->dev; - struct i915_audio_component *acomp = &hda->audio_component; - - WARN_ON(hda->i915_power_refcount); - if (hda->i915_power_refcount > 0 && acomp->ops) - acomp->ops->put_power(acomp->dev); - - component_master_del(dev, &hda_component_master_ops); - - return 0; -} diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5c84d40..391e4f8 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -57,6 +57,8 @@ #endif #include <sound/core.h> #include <sound/initval.h> +#include <sound/hdaudio.h> +#include <sound/hda_i915.h> #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> @@ -496,13 +498,13 @@ static void azx_init_pci(struct azx *chip) static void hda_intel_init_chip(struct azx *chip, bool full_reset) { - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct hdac_bus *bus = azx_bus(chip); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) - hda_set_codec_wakeup(hda, true); + snd_hdac_set_codec_wakeup(bus, true); azx_init_chip(chip, full_reset); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) - hda_set_codec_wakeup(hda, false); + snd_hdac_set_codec_wakeup(bus, false); } /* calculate runtime delay from LPIB */ @@ -560,9 +562,9 @@ static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev) /* Enable/disable i915 display power for the link */ static int azx_intel_link_power(struct azx *chip, bool enable) { - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct hdac_bus *bus = azx_bus(chip); - return hda_display_power(hda, enable); + return snd_hdac_display_power(bus, enable); } /* @@ -800,6 +802,50 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) #define azx_del_card_list(chip) /* NOP */ #endif /* CONFIG_PM */ +/* Intel HSW/BDW display HDA controller is in GPU. Both its power and link BCLK + * depends on GPU. Two Extended Mode registers EM4 (M value) and EM5 (N Value) + * are used to convert CDClk (Core Display Clock) to 24MHz BCLK: + * BCLK = CDCLK * M / N + * The values will be lost when the display power well is disabled and need to + * be restored to avoid abnormal playback speed. + */ +static void haswell_set_bclk(struct hda_intel *hda) +{ + struct azx *chip = &hda->chip; + int cdclk_freq; + unsigned int bclk_m, bclk_n; + + if (!hda->need_i915_power) + return; + + cdclk_freq = snd_hdac_get_display_clk(azx_bus(chip)); + switch (cdclk_freq) { + case 337500: + bclk_m = 16; + bclk_n = 225; + break; + + case 450000: + default: /* default CDCLK 450MHz */ + bclk_m = 4; + bclk_n = 75; + break; + + case 540000: + bclk_m = 4; + bclk_n = 90; + break; + + case 675000: + bclk_m = 8; + bclk_n = 225; + break; + } + + azx_writew(chip, HSW_EM4, bclk_m); + azx_writew(chip, HSW_EM5, bclk_n); +} + #if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) /* * power management @@ -833,7 +879,7 @@ static int azx_suspend(struct device *dev) pci_disable_msi(chip->pci); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) - hda_display_power(hda, false); + snd_hdac_display_power(bus, false); trace_azx_suspend(chip); return 0; @@ -856,7 +902,7 @@ static int azx_resume(struct device *dev) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) { - hda_display_power(hda, true); + snd_hdac_display_power(azx_bus(chip), true); haswell_set_bclk(hda); } if (chip->msi) @@ -902,7 +948,7 @@ static int azx_runtime_suspend(struct device *dev) azx_clear_irq_pending(chip); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) - hda_display_power(hda, false); + snd_hdac_display_power(azx_bus(chip), false); trace_azx_runtime_suspend(chip); return 0; @@ -913,6 +959,7 @@ static int azx_runtime_resume(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; struct hda_intel *hda; + struct hdac_bus *bus; struct hda_codec *codec; int status; @@ -929,11 +976,12 @@ static int azx_runtime_resume(struct device *dev) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) { - hda_display_power(hda, true); + bus = azx_bus(chip); + snd_hdac_display_power(bus, true); haswell_set_bclk(hda); /* toggle codec wakeup bit for STATESTS read */ - hda_set_codec_wakeup(hda, true); - hda_set_codec_wakeup(hda, false); + snd_hdac_set_codec_wakeup(bus, true); + snd_hdac_set_codec_wakeup(bus, false); } /* Read STATESTS before controller reset */ @@ -1152,10 +1200,11 @@ static int azx_free(struct azx *chip) #ifdef CONFIG_SND_HDA_PATCH_LOADER release_firmware(chip->fw); #endif + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { if (hda->need_i915_power) - hda_display_power(hda, false); - hda_i915_exit(hda); + snd_hdac_display_power(bus, false); + snd_hdac_i915_exit(bus); } kfree(hda); @@ -1914,6 +1963,7 @@ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] = { static int azx_probe_continue(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct hdac_bus *bus = azx_bus(chip); struct pci_dev *pci = chip->pci; int dev = chip->dev_index; int err; @@ -1930,11 +1980,11 @@ static int azx_probe_continue(struct azx *chip) if (pci->device != 0x0f04 && pci->device != 0x2284) hda->need_i915_power = 1; - err = hda_i915_init(hda); + err = snd_hdac_i915_init(bus); if (err < 0) goto i915_power_fail; - err = hda_display_power(hda, true); + err = snd_hdac_display_power(bus, true); if (err < 0) { dev_err(chip->card->dev, "Cannot turn on display power on i915\n"); @@ -1986,7 +2036,7 @@ static int azx_probe_continue(struct azx *chip) out_free: if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && !hda->need_i915_power) - hda_display_power(hda, false); + snd_hdac_display_power(bus, false); i915_power_fail: if (err < 0) diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index 7fd3254..354f0bb 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -16,7 +16,6 @@ #ifndef __SOUND_HDA_INTEL_H #define __SOUND_HDA_INTEL_H -#include <drm/i915_component.h> #include "hda_controller.h" struct hda_intel { @@ -44,36 +43,7 @@ struct hda_intel { /* secondary power domain for hdmi audio under vga device */ struct dev_pm_domain hdmi_pm_domain; - /* i915 component interface */ bool need_i915_power:1; /* the hda controller needs i915 power */ - struct i915_audio_component audio_component; - int i915_power_refcount; }; -#ifdef CONFIG_SND_HDA_I915 -int hda_set_codec_wakeup(struct hda_intel *hda, bool enable); -int hda_display_power(struct hda_intel *hda, bool enable); -void haswell_set_bclk(struct hda_intel *hda); -int hda_i915_init(struct hda_intel *hda); -int hda_i915_exit(struct hda_intel *hda); -#else -static inline int hda_set_codec_wakeup(struct hda_intel *hda, bool enable) -{ - return 0; -} -static inline int hda_display_power(struct hda_intel *hda, bool enable) -{ - return 0; -} -static inline void haswell_set_bclk(struct hda_intel *hda) { return; } -static inline int hda_i915_init(struct hda_intel *hda) -{ - return 0; -} -static inline int hda_i915_exit(struct hda_intel *hda) -{ - return 0; -} -#endif - #endif -- 1.9.1

2 1

[alsa-devel] Focusrite Forte
by e-rouge 19 May '15

19 May '15

Hello, I just bought a Focusrite Forte, and yeah I've fooled by the class-compliant thingy and the touch controls. For some reasons the card controls require a deamon that only runs on window and mac. Well, OK, I'm a programmer, I can fix that. There must be a way to access all the parameters (namely gain and phantom switches). Can you provide me some basic commands to know what this card has under the hood ? What tools do you use to do that ? Thanks a lot in advance T. P.S. : well if someone has a solution out-of-the-box, I can take it too :)

1 0

Re: [alsa-devel] linux-next: Tree for May 19 (sound/soc/atmel/)
by Randy Dunlap 19 May '15

19 May '15

On 05/19/15 00:30, Stephen Rothwell wrote: > Hi all, > > Changes since 20150518: > on x86_64: sound/built-in.o: In function `atmel_ssc_set_audio': (.text+0x87d90): undefined reference to `atmel_pcm_pdc_platform_register' sound/built-in.o: In function `atmel_ssc_put_audio': (.text+0x8879a): undefined reference to `atmel_pcm_pdc_platform_unregister' Full randconfig file is attached. -- ~Randy

1 0

Re: [alsa-devel] USB headset cannot get freq
by Greg KH 19 May '15

19 May '15

On Sat, May 16, 2015 at 04:25:23PM +0200, Heinz Diehl wrote: > Hi, > > this is the Jabra Evolve 40 MS headset: > Bus 001 Device 008: ID 0b0e:0305 GN Netcom > > When plugged in, dmesg says: > > [ 247.215998] usb 1-5: new full-speed USB device number 8 using xhci_hcd > [ 247.384000] usb 1-5: New USB device found, idVendor=0b0e, idProduct=0305 > [ 247.384008] usb 1-5: New USB device strings: Mfr=1, Product=2, SerialNumber=3 > [ 247.384012] usb 1-5: Product: Jabra EVOLVE LINK MS > [ 247.384016] usb 1-5: Manufacturer: GN Netcom A/S > [ 247.384019] usb 1-5: SerialNumber: 0001A3E08CFC09 > [ 247.386542] usb 1-5: 1:1: cannot get freq at ep 0x83 > [ 247.388490] usb 1-5: 2:1: cannot get freq at ep 0x4 > [ 247.406532] input: GN Netcom A/S Jabra EVOLVE LINK MS as /devices/pci0000:00/0000:00:14.0/usb1/1-5/1-5:1.3/0003:0B0E:0305.0009/input/input12 > [ 247.457203] hid-generic 0003:0B0E:0305.0009: input,hiddev0,hidraw1: USB HID v1.00 Device [GN Netcom A/S Jabra EVOLVE LINK MS] on usb-0000:00:14.0-5/input3 > [ 247.467683] usb 1-5: 1:1: cannot get freq at ep 0x83 > [ 247.467912] usb 1-5: 1:1: cannot get freq at ep 0x83 > [ 247.473505] usb 1-5: 2:1: cannot get freq at ep 0x4 > [ 247.473718] usb 1-5: 2:1: cannot get freq at ep 0x4 > [ 247.475488] usb 1-5: 1:1: cannot get freq at ep 0x83 > [ 247.475721] usb 1-5: 1:1: cannot get freq at ep 0x83 > [ 247.481899] usb 1-5: 2:1: cannot get freq at ep 0x4 > [ 247.482097] usb 1-5: 2:1: cannot get freq at ep 0x4 > [ 247.483351] usb 1-5: 1:1: cannot get freq at ep 0x83 > [ 247.483566] usb 1-5: 1:1: cannot get freq at ep 0x83 > [ 247.487396] usb 1-5: 2:1: cannot get freq at ep 0x4 > [ 247.487586] usb 1-5: 2:1: cannot get freq at ep 0x4 > [ 247.494730] usb 1-5: 1:1: cannot get freq at ep 0x83 > [ 247.495018] usb 1-5: 1:1: cannot get freq at ep 0x83 > > The micro/playback devices show up, but there's no sound and the mic is also > dead. > > I've spent the last two days reading on the net, tried a lot of different > options, but whithout any success. In a lot of similar bug reports, people > point out that this is a bug in the Linux kernel USB subsystem.. > > What can I do to debug this further, in order to get a working headset? The sound developers should be able to help you (added to the cc:) greg k-h

3 2

[alsa-devel] [PATCH v3 00/17] new LoCoMo driver set
by Dmitry Eremin-Solenikov 19 May '15

19 May '15

Sharp Zaurus SL-5500 and SL-5600 use special companion Gate Array. With this patchset I tried to modernise and restructure the LoCoMo drivers. * Use platform bus and mfd core code * Use GPIO API instead of custom locomo GPIO code * Use irqdomains to manage IRQs * Split mixed locomolcd driver to lcd and backlight parts * Implement IRQ handling in GPIO driver * Implement SPI driver used as a host for mmc_spi device to control SD cards Changes since V2: * Implemented most of review findings (lots of small changes) * Added comments where the review feedback was ignored The following changes since commit b787f68c36d49bb1d9236f403813641efa74a031: Linux 4.1-rc1 (2015-04-26 17:59:10 -0700) are available in the git repository at: git://git.infradead.org/users/dbaryshkov/zaurus.git locomo-v3 for you to fetch changes up to c7405f110836863287c505650ac2133fecabea7b: ARM: drop old LoCoMo driver (2015-05-17 19:23:20 +0300) ---------------------------------------------------------------- Dmitry Eremin-Solenikov (17): mfd: add new driver for Sharp LoCoMo leds: port locomo leds driver to new locomo core input: convert LoCoMo keyboard driver to use new locomo core input: make LoCoMo keyboard driver support both poodle and collie video: backlight: add new locomo backlight driver video: lcd: add LoCoMo LCD driver gpio: port LoCoMo gpio support from old driver gpio: locomo: implement per-pin irq handling spi: add locomo SPI driver i2c: add locomo i2c driver ARM: sa1100: make collie use new locomo drivers ARM: sa1100: don't preallocate IRQ space for locomo ASoC: pxa: poodle: make use of new locomo GPIO interface ARM: pxa: poodle: use new LoCoMo driver ARM: pxa: poodle: don't preallocate IRQ space for locomo video: backlight: drop old locomo bl/lcd driver ARM: drop old LoCoMo driver arch/arm/common/Kconfig | 3 - arch/arm/common/Makefile | 1 - arch/arm/common/locomo.c | 914 ----------------------------- arch/arm/include/asm/hardware/locomo.h | 221 ------- arch/arm/mach-pxa/Kconfig | 1 - arch/arm/mach-pxa/include/mach/poodle.h | 8 +- arch/arm/mach-pxa/poodle.c | 58 +- arch/arm/mach-sa1100/Kconfig | 1 - arch/arm/mach-sa1100/collie.c | 213 +++++-- arch/arm/mach-sa1100/include/mach/collie.h | 16 +- arch/arm/mach-sa1100/include/mach/irqs.h | 19 +- drivers/gpio/Kconfig | 13 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-locomo.c | 285 +++++++++ drivers/i2c/busses/Kconfig | 12 + drivers/i2c/busses/Makefile | 1 + drivers/i2c/busses/i2c-locomo.c | 135 +++++ drivers/input/keyboard/Kconfig | 2 +- drivers/input/keyboard/locomokbd.c | 264 ++++----- drivers/leds/Kconfig | 2 +- drivers/leds/leds-locomo.c | 117 ++-- drivers/mfd/Kconfig | 10 + drivers/mfd/Makefile | 1 + drivers/mfd/locomo.c | 346 +++++++++++ drivers/spi/Kconfig | 10 + drivers/spi/Makefile | 1 + drivers/spi/spi-locomo.c | 332 +++++++++++ drivers/video/backlight/Kconfig | 16 +- drivers/video/backlight/Makefile | 3 +- drivers/video/backlight/locomo_bl.c | 153 +++++ drivers/video/backlight/locomo_lcd.c | 285 +++++++++ drivers/video/backlight/locomolcd.c | 255 -------- include/linux/mfd/locomo.h | 167 ++++++ sound/soc/pxa/poodle.c | 52 +- 34 files changed, 2216 insertions(+), 1702 deletions(-) delete mode 100644 arch/arm/common/locomo.c delete mode 100644 arch/arm/include/asm/hardware/locomo.h create mode 100644 drivers/gpio/gpio-locomo.c create mode 100644 drivers/i2c/busses/i2c-locomo.c create mode 100644 drivers/mfd/locomo.c create mode 100644 drivers/spi/spi-locomo.c create mode 100644 drivers/video/backlight/locomo_bl.c create mode 100644 drivers/video/backlight/locomo_lcd.c delete mode 100644 drivers/video/backlight/locomolcd.c create mode 100644 include/linux/mfd/locomo.h -- With best wishes Dmitry

5 26

[alsa-devel] [PATCH] ASoC: rt5645: Rename HP control to Headphone
by Nicolas Boichat 19 May '15

19 May '15

Use the standard name "Headphone" instead of "HP" for better userspace integration. Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> --- sound/soc/codecs/rt5645.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index be4d741..bc925f0 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -464,9 +464,9 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = { RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv), /* Headphone Output Volume */ - SOC_DOUBLE("HP Channel Switch", RT5645_HP_VOL, + SOC_DOUBLE("Headphone Channel Switch", RT5645_HP_VOL, RT5645_VOL_L_SFT, RT5645_VOL_R_SFT, 1, 1), - SOC_DOUBLE_TLV("HP Playback Volume", RT5645_HP_VOL, + SOC_DOUBLE_TLV("Headphone Playback Volume", RT5645_HP_VOL, RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv), /* OUTPUT Control */ -- 2.2.0.rc0.207.ga3a616c

4 4

[alsa-devel] [PATCH] ASoC: rt5645: fix kernel hang when call rt5645_set_jack_detect()
by John Lin 19 May '15

19 May '15

rt5645_set_jack_detect() is usually called from snd_soc_dai_link.init() and it calls snd_soc_jack_report() from rt5645_irq_detection() if jack is inserted. snd_soc_jack_report() results in kernel hang if it is called from a context which cannot sleep. This patch makes sure snd_soc_jack_report() is called from workqueue. It can fix the kernel hang issue. Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5645.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 14b12c5..aaede45 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2876,6 +2876,7 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) } static int rt5645_irq_detection(struct rt5645_priv *rt5645); +static irqreturn_t rt5645_irq(int irq, void *data); int rt5645_set_jack_detect(struct snd_soc_codec *codec, struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack, @@ -2895,7 +2896,7 @@ int rt5645_set_jack_detect(struct snd_soc_codec *codec, regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1, RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); } - rt5645_irq_detection(rt5645); + rt5645_irq(0, rt5645); return 0; } -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix NULL dereference from CA0132 DSP loader
by Takashi Iwai 19 May '15

19 May '15

The CA0132 DSP loader leads to NULL deference since the recent transition to HDA core code, as it unconditionally accesses hdac_stream->substream->runtime. For DSP loading, the substream shouldn't be assigned. This patch addresses the NULL dereference above in addition to assure the substream is cleared while DSP loading. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98151 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_stream.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index 52a894fe478f..4c15d0accc9e 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -139,9 +139,13 @@ EXPORT_SYMBOL_GPL(snd_hdac_stream_reset); int snd_hdac_stream_setup(struct hdac_stream *azx_dev) { struct hdac_bus *bus = azx_dev->bus; - struct snd_pcm_runtime *runtime = azx_dev->substream->runtime; + struct snd_pcm_runtime *runtime; unsigned int val; + if (azx_dev->substream) + runtime = azx_dev->substream->runtime; + else + runtime = NULL; /* make sure the run bit is zero for SD */ snd_hdac_stream_clear(azx_dev); /* program the stream_tag */ @@ -189,14 +193,15 @@ int snd_hdac_stream_setup(struct hdac_stream *azx_dev) * we ignore it; currently set the threshold statically to * 64 frames */ - if (runtime->period_size > 64) + if (runtime && runtime->period_size > 64) azx_dev->delay_negative_threshold = -frames_to_bytes(runtime, 64); else azx_dev->delay_negative_threshold = 0; /* wallclk has 24Mhz clock source */ - azx_dev->period_wallclk = (((runtime->period_size * 24000) / + if (runtime) + azx_dev->period_wallclk = (((runtime->period_size * 24000) / runtime->rate) * 1000); return 0; @@ -611,6 +616,7 @@ int snd_hdac_dsp_prepare(struct hdac_stream *azx_dev, unsigned int format, if (err < 0) goto err_alloc; + azx_dev->substream = NULL; azx_dev->bufsize = byte_size; azx_dev->period_bytes = byte_size; azx_dev->format_val = format; -- 2.4.1

1 0

[alsa-devel] [PATCH 0/8] ASoC: Intel: refactor common SST IPC handling
by Vinod Koul 19 May '15

19 May '15

This series refactors the common SST IPC code to remove some hardcoded assumption which are no longer valid for future platforms like Skylake, These are made configurable and right configuration applied for respective drivers. The changes are mailbox sizes and dsp busy checks. Subhransu S. Prusty (8): ASoC: Intel: Create an ops to check for DSP busy ASoC: Intel: Move the busy check to ops for Baytrail ASoC: Intel: Move the busy check to ops for HSW ASoC: Intel: Remove the direct register reference from common ipc ASoC: Intel: Allow to configure max size for mailbox data ASoC: Intel: Initialize max mailbox size for baytrail ASoC: Intel: Initialize max mailbox size for haswell ASoC: Intel: Allocate for the mailbox with max size sound/soc/intel/baytrail/sst-baytrail-ipc.c | 11 +++++++++++ sound/soc/intel/common/sst-ipc.c | 20 +++++++++++++++++--- sound/soc/intel/common/sst-ipc.h | 7 +++++-- sound/soc/intel/haswell/sst-haswell-ipc.c | 12 ++++++++++++ 4 files changed, 45 insertions(+), 5 deletions(-) -- 1.9.1

1 9

[alsa-devel] [PATCH] ALSA: usb-audio: Add quirk for MS LifeCam Studio
by Takashi Iwai 19 May '15

19 May '15

Microsoft LifeCam Studio (045e:0772) needs a similar quirk for suppressing the wrong sample rate inquiry. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98481 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 7c5a70139278..46facfc9aec1 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1117,6 +1117,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) switch (chip->usb_id) { case USB_ID(0x045E, 0x075D): /* MS Lifecam Cinema */ case USB_ID(0x045E, 0x076D): /* MS Lifecam HD-5000 */ + case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ return true; } -- 2.4.1

1 0

[alsa-devel] [PATCH 1/2] ALSA: hda - Add headset-mode* model options for ALC269 & co
by Takashi Iwai 19 May '15

19 May '15

Modern machines tend to have only one headset jack nowadays, and they often need these quirks. Let's allow them applicable via model option for ease of debugging. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa/HD-Audio-Models.txt | 2 ++ sound/pci/hda/patch_realtek.c | 2 ++ 2 files changed, 4 insertions(+) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 5a3163cac6c3..d92b85d38adc 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -29,6 +29,8 @@ ALC269/270/275/276/28x/29x alc271-dmic Enable ALC271X digital mic workaround inv-dmic Inverted internal mic workaround headset-mic Indicates a combined headset (headphone+mic) jack + headset-mode More comprehensive headset support for ALC269 & co + headset-mode-no-hp-mic Headset mode support without headphone mic lenovo-dock Enables docking station I/O for some Lenovos dell-headset-multi Headset jack, which can also be used as mic-in dell-headset-dock Headset jack (without mic-in), and also dock I/O diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6e1c8c34e841..4d11fbfca5e2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5242,6 +5242,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC271_FIXUP_DMIC, .name = "alc271-dmic"}, {.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"}, {.id = ALC269_FIXUP_HEADSET_MIC, .name = "headset-mic"}, + {.id = ALC269_FIXUP_HEADSET_MODE, .name = "headset-mode"}, + {.id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, .name = "headset-mode-no-hp-mic"}, {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"}, {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, -- 2.4.1

1 1

[alsa-devel] [RFC PATCH] ALSA: pcm: Modify double acknowledged interrupts check condition
by Koro Chen 19 May '15

19 May '15

Currently in snd_pcm_update_hw_ptr0 during interrupt, we consider there were double acknowledged interrupts when: 1. HW reported pointer is smaller than expected, and 2. Time from last update time (hdelta) is over half a buffer time. However, when HW reported pointer is only a few bytes smaller than expected, and when hdelta is just a little larger than half a buffer time (e.g. ping-pong buffer), it wrongly treats this IRQ as double acknowledged. The condition #2 uses jiffies, but jiffies is not high resolution since it is integer. We should consider jiffies inaccuracy. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index ac6b33f..7d45645 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -339,7 +339,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (delta > new_hw_ptr) { /* check for double acknowledged interrupts */ hdelta = curr_jiffies - runtime->hw_ptr_jiffies; - if (hdelta > runtime->hw_ptr_buffer_jiffies/2) { + if (hdelta > runtime->hw_ptr_buffer_jiffies/2 + 1) { hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) { hw_base = 0; -- 1.8.1.1.dirty

2 1

[alsa-devel] Intel HDA / ca0132: support for Alienware 15 Creative Sound Core3D-EX
by Gabriele Martino 19 May '15

19 May '15

Hi, the new Alienware 15 laptop (2015, not the M15x) has a Creative Sound Core3D-EX Audio which seems to work with ca0132. I'm using a 4.0.0 kernel with gentoo patchset, but with the default configuration: - the line out/headphone jack doesn't work, it only outputs some white noise - alsamixer shows only "Master" and "PCM" channel, no "Front", "Line Out" or similar - alsamixer has a toggle switch for the "HP/Speaker" but no headphone/line out switch - the jack detection is enabled on the "line in" jack instead of the "line out" (if I plug something on the "line in", the speakers are muted) - if I play too much with these switches, the speakers stop working until suspend/reboot To fix the first issue I run "hdajacksensetest" to get the headphone jack pin: hdajacksensetest -c 1 -a Pin 0x0b (Internal Speaker): present = Yes Pin 0x0c (Not connected): present = No Pin 0x0d (Not connected): present = No Pin 0x0e (Not connected): present = No Pin 0x0f (Not connected): present = Yes Pin 0x10 (Not connected): present = No Pin 0x11 (Black Line In, Left side): present = No Pin 0x12 (Internal Mic, Mobile-In): present = No Pin 0x13 (Not connected): present = No Pin 0x18 (Not connected): present = No and replaced the pin number: --- /usr/src/linux-4.0.0-gentoo/sound/pci/hda/patch_ca0132.c.old 2015-04-13 00:12:50.000000000 +0200 +++ /usr/src/linux-4.0.0-gentoo/sound/pci/hda/patch_ca0132.c 2015-04-25 01:43:09.607531550 +0200 @@ -4625,7 +4625,7 @@ spec->num_outputs = 2; spec->out_pins[0] = 0x0b; /* speaker out */ - spec->out_pins[1] = 0x10; /* headphone out */ + spec->out_pins[1] = 0x0f; /* default: 0x10 - headphone out -0x0f on Alienware 15 2015 */ spec->shared_out_nid = 0x2; spec->num_inputs = 3; Now the "HP/Speaker Auto Detect" is ignored on both jacks, but I can switch between headphones and speakers manually with the "HP/Speaker" item. This is a dirty hack (and probably won't work on different cards), is there a better way to fix this and get at least the jack detection working? Regards, Gabriele

2 20

[alsa-devel] [PATCH 0/4] firewire-lib: purge restriction of synchronization for non-blocking mode
by Takashi Sakamoto 19 May '15

19 May '15

This patchset allows non-blocking streams to use timestamp synchronization. In full duplex synchronization mode, current implementation pass the value of 'syt' field of CIP header in incoming packets to outgoing packet processing, while the number of data blocks is not passed. Therefore, this mode is allowed just to blocking streams, because of a lack of validator for the number which actual devices transfers. As long as the number is suspicious, it's not reused to outgoing packet processing. To allow the synchronization mode to non-blocking streams, this patchset: * adds a validator for the number of data blocks in incoming packets * adds a flag still to support illegal models (i.e. Behringer FCA202) * pass the evaluated number from incoming packet processing to outgoing * remove the restriction Takashi Sakamoto (4): ALSA: firewire-lib: add buffer-over-run protection at receiving more data blocks than expected ALSA: firewire-lib: simplify function to calculate the number of data blocks ALSA: firewire-lib: pass the number of data blocks in incoming packets to outgoing packets ALSA: firewire-lib: remove restriction for non-blocking mode sound/firewire/amdtp.c | 108 ++++++++++++++++++++++++-------------- sound/firewire/amdtp.h | 4 ++ sound/firewire/oxfw/oxfw-stream.c | 10 +++- 3 files changed, 81 insertions(+), 41 deletions(-) -- 2.1.4

2 8

[alsa-devel] [PATCH RESEND] sound: ppc: keywest: drop using attach adapter
by Wolfram Sang 18 May '15

18 May '15

As we now have deferred probing, we can use a custom mechanism and finally get rid of this legacy interface from the i2c core. Signed-off-by: Wolfram Sang <wsa(a)the-dreams.de> Tested-by: Dan DeVoto <dand1972(a)yahoo.com> Tested-by: Mark Elliott <txlitebeer(a)gmail.com> --- Changes since last time: Added two test tags from the Debian PPC list. I'd love to have that in 4.2, so I can get rid of that legacy interface. I can take it via I2C but ALSA looks like the proper path to me. sound/ppc/keywest.c | 23 +++++++++++++++++++---- 1 file changed, 19 insertions(+), 4 deletions(-) diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index 0d1c27e911b8dd..d7627bae08362e 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -52,7 +52,7 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter) return -EINVAL; if (strncmp(adapter->name, "mac-io", 6)) - return 0; /* ignored */ + return -EINVAL; /* ignored */ memset(&info, 0, sizeof(struct i2c_board_info)); strlcpy(info.type, "keywest", I2C_NAME_SIZE); @@ -100,7 +100,6 @@ static struct i2c_driver keywest_driver = { .driver = { .name = "PMac Keywest Audio", }, - .attach_adapter = keywest_attach_adapter, .probe = keywest_probe, .remove = keywest_remove, .id_table = keywest_i2c_id, @@ -132,16 +131,32 @@ int snd_pmac_tumbler_post_init(void) /* exported */ int snd_pmac_keywest_init(struct pmac_keywest *i2c) { - int err; + struct i2c_adapter *adap; + int err, i = 0; if (keywest_ctx) return -EBUSY; + adap = i2c_get_adapter(0); + if (!adap) + return -EPROBE_DEFER; + keywest_ctx = i2c; if ((err = i2c_add_driver(&keywest_driver))) { snd_printk(KERN_ERR "cannot register keywest i2c driver\n"); + i2c_put_adapter(adap); return err; } - return 0; + + /* We assume Macs have consecutive I2C bus numbers starting at 0 */ + while (adap) { + err = keywest_attach_adapter(adap); + if (!err) + return 0; + i2c_put_adapter(adap); + adap = i2c_get_adapter(++i); + } + + return -ENODEV; } -- 2.1.4

2 1

[alsa-devel] Left/Right Mic mute independently
by Arijit De 18 May '15

18 May '15

Hi, I want to do Left/Right Mic mute independently on a Audio capture device(it has two audio channels), but I dont see that kind of control is provided in alsamixer(GUI) for capture devices. Is Left/Right Mic mute is supported in alsa mixer ? Thanks Arijit

2 1

[alsa-devel] [PATCH - alsa-plugins/pulse 1/1] Add 24 bit data formats to Pulse plugin.
by alex.d.wiggins＠gmail.com 18 May '15

18 May '15

From: Alex Wiggins <alex.d.wiggins(a)gmail.com> Signed-off-by: Alex Wiggins <alex.d.wiggins(a)gmail.com> diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index 0d3704b..5cb3452 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -994,7 +994,11 @@ static int pulse_hw_constraint(snd_pcm_pulse_t * pcm) SND_PCM_FORMAT_FLOAT_LE, SND_PCM_FORMAT_FLOAT_BE, SND_PCM_FORMAT_S32_LE, - SND_PCM_FORMAT_S32_BE + SND_PCM_FORMAT_S32_BE, + SND_PCM_FORMAT_S24_3LE, + SND_PCM_FORMAT_S24_3BE, + SND_PCM_FORMAT_S24_LE, + SND_PCM_FORMAT_S24_BE }; int err; -- 2.3.6

2 1

[alsa-devel] [PATCH] conf/ucm: broadwell-rt286: change to set capture volume and capture device
by Jie Yang 18 May '15

18 May '15

Set 'Mic Capture Volume' in capture device EnableSequence, to fix capture no volume by default issue. Also add JackHWMute Value item to mute onboard dmic while headset mic is plugged in. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- src/conf/ucm/broadwell-rt286/HiFi | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/src/conf/ucm/broadwell-rt286/HiFi b/src/conf/ucm/broadwell-rt286/HiFi index 58983ea..c09a01c 100644 --- a/src/conf/ucm/broadwell-rt286/HiFi +++ b/src/conf/ucm/broadwell-rt286/HiFi @@ -86,6 +86,7 @@ SectionDevice."Handset" { EnableSequence [ cdev "hw:broadwellrt286" + cset "name='Mic Capture Volume' 28" cset "name='ADC 0 Mux' 0" cset "name='ADC0 Capture Switch' on" cset "name='ADC0 Capture Volume' 127" @@ -99,6 +100,9 @@ SectionDevice."Handset" { Value { CaptureChannels "2" + JackDev "rt286-jack" + JackControl "Mic Jack" + JackHWMute "Mainmic" } } @@ -112,6 +116,7 @@ SectionDevice."Mainmic" { EnableSequence [ cdev "hw:broadwellrt286" + cset "name='Mic Capture Volume' 30" cset "name='ADC 0 Mux' 2" cset "name='ADC0 Capture Switch' on" cset "name='ADC0 Capture Volume' 127" -- 1.9.1

3 2

[alsa-devel] [PATCH] ucm: allow multiple devices in JackHWMute
by Tanu Kaskinen 18 May '15

18 May '15

One jack may mute multiple devices, so let's make JackHWMute a list of device names instead of just a single device name. Signed-off-by: Tanu Kaskinen <tanu.kaskinen(a)linux.intel.com> Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Acked-by: Mark Brown <broonie(a)kernel.org> --- include/use-case.h | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) Changes in v2: - Fixed the typo that Arun pointed out. - Send to alsa-devel (v1 was accidentally not sent to the list). diff --git a/include/use-case.h b/include/use-case.h index e3308b1..c7789c0 100644 --- a/include/use-case.h +++ b/include/use-case.h @@ -311,14 +311,15 @@ int snd_use_case_get_list(snd_use_case_mgr_t *uc_mgr, * applications are likely to support only one or the other. * * If **JackHWMute** is set, it indicates that when the jack is plugged - * in, the hardware automatically mutes some other device. The - * JackHWMute value is the name of the muted device. Note that - * JackHWMute should be used only when the hardware enforces the - * automatic muting. If the hardware doesn't enforce any muting, it may - * still be tempting to set JackHWMute to trick upper software layers to - * e.g. automatically mute speakers when headphones are plugged in, but - * that's application policy configuration that doesn't belong to UCM - * configuration files. + * in, the hardware automatically mutes some other device(s). The + * JackHWMute value is a space-separated list of device names (this + * isn't compatible with device names with spaces in them, so don't use + * such device names!). Note that JackHWMute should be used only when + * the hardware enforces the automatic muting. If the hardware doesn't + * enforce any muting, it may still be tempting to set JackHWMute to + * trick upper software layers to e.g. automatically mute speakers when + * headphones are plugged in, but that's application policy + * configuration that doesn't belong to UCM configuration files. */ int snd_use_case_get(snd_use_case_mgr_t *uc_mgr, const char *identifier, -- 1.9.3

2 1

[alsa-devel] [PATCH] sound: oss: fix build warning
by Sudip Mukherjee 18 May '15

18 May '15

while building with allyesconfig it was giving a build warning about unused variable. declare the variable only if the driver is built as a module. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- build tested with allyesconfig and allmodconfig sound/oss/ad1848.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index ec1ee07..10c8de1 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -2860,6 +2860,7 @@ static struct { {NULL} }; +#ifdef MODULE static struct isapnp_device_id id_table[] = { { ISAPNP_VENDOR('C','M','I'), ISAPNP_DEVICE(0x0001), ISAPNP_VENDOR('@','@','@'), ISAPNP_FUNCTION(0x0001), 0 }, @@ -2877,6 +2878,7 @@ static struct isapnp_device_id id_table[] = { }; MODULE_DEVICE_TABLE(isapnp, id_table); +#endif static struct pnp_dev *activate_dev(char *devname, char *resname, struct pnp_dev *dev) { -- 1.8.1.2

2 1

[alsa-devel] [RFC PATCH] ALSA: hda - Move hda_i915.c from sound/pci/hda to sound/hda
by mengdong.lin＠intel.com 18 May '15

18 May '15

From: Mengdong Lin <mengdong.lin(a)intel.com> The file is moved to hda core, so can be used by both legacy HDA driver and new Skylake audio driver. - The i915 audio component is moved to core bus and dynamically allocated. - A static pointer hda_acomp is used to help bind/unbind callbacks to get this component, because the sound card's private_data is used by the azx chip pointer, which is a legacy structure. It could be removed if private _data changes to some core structure which can be extended to find the bus. - hda_get_display_clk() is added to get the display core clock for HSW/BDW. - haswell_set_bclk() is moved to hda_intel.c because it needs to write the controller registers EM4/EM5. - Move definition of HSW/BDW-specific registers EM4/EM5 to hda_register.h and rename them to HSW_EM4/HSW_EM5, because other HD-A controllers have different layout for the extended mode registers. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h new file mode 100644 index 0000000..440e4c0 --- /dev/null +++ b/include/sound/hda_i915.h @@ -0,0 +1,36 @@ +/* + * HD-Audio helpers to sync with i915 driver + */ +#ifndef __SOUND_HDA_I915_H +#define __SOUND_HDA_I915_H + +#ifdef CONFIG_SND_HDA_I915 +int hda_set_codec_wakeup(struct hdac_bus *bus, bool enable); +int hda_display_power(struct hdac_bus *bus, bool enable); +int hda_get_display_clk(struct hdac_bus *bus); +int hda_i915_init(struct hdac_bus *bus); +int hda_i915_exit(struct hdac_bus *bus); +#else +static int hda_set_codec_wakeup(struct hdac_bus *bus, bool enable) +{ + return 0; +} +static inline int hda_display_power(struct hdac_bus *bus, bool enable) +{ + return 0; +} +static inline int hda_get_display_clk(struct hdac_bus *bus) +{ + return 0; +} +static inline int hda_i915_init(struct hdac_bus *bus) +{ + return -ENODEV; +} +static inline int hda_i915_exit(struct hdac_bus *bus) +{ + return 0; +} +#endif + +#endif /* __SOUND_HDA_I915_H */ diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index 4f6d3fc..0c7536e 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -84,6 +84,10 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_SD_BDLPL 0x18 #define AZX_REG_SD_BDLPU 0x1c +/* Haswell/Broadwell display HD-A controller Extended Mode registers */ +#define AZX_REG_HSW_EM4 0x100c +#define AZX_REG_HSW_EM5 0x1010 + /* PCI space */ #define AZX_PCIREG_TCSEL 0x44 diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index b97c59e..64fff4d 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -11,6 +11,7 @@ #include <sound/core.h> #include <sound/memalloc.h> #include <sound/hda_verbs.h> +#include <drm/i915_component.h> /* codec node id */ typedef u16 hda_nid_t; @@ -285,6 +286,10 @@ struct hdac_bus { /* locks */ spinlock_t reg_lock; struct mutex cmd_mutex; + + /* i915 component interface */ + struct i915_audio_component *audio_component; + int i915_power_refcount; }; int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev, diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 7a17fca..ac5ffac 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -4,3 +4,9 @@ config SND_HDA_CORE config SND_HDA_DSP_LOADER bool + +config SND_HDA_I915 + bool + default y + depends on DRM_I915 + depends on SND_HDA_CORE diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 5b4bb47..1ce78c6 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -4,4 +4,7 @@ snd-hda-core-objs := hda_bus_type.o hdac_bus.o hdac_device.o hdac_sysfs.o \ snd-hda-core-objs += trace.o CFLAGS_trace.o := -I$(src) +# for sync with i915 gfx driver +snd-hda-core-$(CONFIG_SND_HDA_I915) += hda_i915.o + obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o diff --git a/sound/hda/hda_i915.c b/sound/hda/hda_i915.c new file mode 100644 index 0000000..5c87938 --- /dev/null +++ b/sound/hda/hda_i915.c @@ -0,0 +1,193 @@ +/* + * hda_i915.c - routines for Haswell HDA controller power well support + * + * 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. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pci.h> +#include <linux/component.h> +#include <drm/i915_component.h> +#include <sound/core.h> +#include <sound/hdaudio.h> +#include <sound/hda_i915.h> + +static struct i915_audio_component *hda_acomp; + +int hda_set_codec_wakeup(struct hdac_bus *bus, bool enable) +{ + struct i915_audio_component *acomp = bus->audio_component; + + if (!acomp->ops) + return -ENODEV; + + if (!acomp->ops->codec_wake_override) { + dev_warn(bus->dev, + "Invalid codec wake callback\n"); + return 0; + } + + dev_dbg(bus->dev, "%s codec wakeup\n", + enable ? "enable" : "disable"); + + acomp->ops->codec_wake_override(acomp->dev, enable); + + return 0; +} +EXPORT_SYMBOL_GPL(hda_set_codec_wakeup); + +int hda_display_power(struct hdac_bus *bus, bool enable) +{ + struct i915_audio_component *acomp = bus->audio_component; + + if (!acomp->ops) + return -ENODEV; + + dev_dbg(bus->dev, "display power %s\n", + enable ? "enable" : "disable"); + + if (enable) { + if (!bus->i915_power_refcount++) + acomp->ops->get_power(acomp->dev); + } else { + WARN_ON(!bus->i915_power_refcount); + if (!--bus->i915_power_refcount) + acomp->ops->put_power(acomp->dev); + } + + return 0; +} +EXPORT_SYMBOL_GPL(hda_display_power); + +int hda_get_display_clk(struct hdac_bus *bus) +{ + struct i915_audio_component *acomp = bus->audio_component; + + if (!acomp->ops) + return -ENODEV; + + return acomp->ops->get_cdclk_freq(acomp->dev); +} +EXPORT_SYMBOL_GPL(hda_get_display_clk); + +static int hda_component_master_bind(struct device *dev) +{ + struct i915_audio_component *acomp = hda_acomp; + int ret; + + ret = component_bind_all(dev, acomp); + if (ret < 0) + return ret; + + if (WARN_ON(!(acomp->dev && acomp->ops && acomp->ops->get_power && + acomp->ops->put_power && acomp->ops->get_cdclk_freq))) { + ret = -EINVAL; + goto out_unbind; + } + + /* + * Atm, we don't support dynamic unbinding initiated by the child + * component, so pin its containing module until we unbind. + */ + if (!try_module_get(acomp->ops->owner)) { + ret = -ENODEV; + goto out_unbind; + } + + return 0; + +out_unbind: + component_unbind_all(dev, acomp); + + return ret; +} + +static void hda_component_master_unbind(struct device *dev) +{ + struct i915_audio_component *acomp = hda_acomp; + + module_put(acomp->ops->owner); + component_unbind_all(dev, acomp); + WARN_ON(acomp->ops || acomp->dev); +} + +static const struct component_master_ops hda_component_master_ops = { + .bind = hda_component_master_bind, + .unbind = hda_component_master_unbind, +}; + +static int hda_component_master_match(struct device *dev, void *data) +{ + /* i915 is the only supported component */ + return !strcmp(dev->driver->name, "i915"); +} + +int hda_i915_init(struct hdac_bus *bus) +{ + struct component_match *match = NULL; + struct device *dev = bus->dev; + struct i915_audio_component *acomp; + int ret; + + acomp = kzalloc(sizeof(*acomp), GFP_KERNEL); + if (!acomp) + return -ENOMEM; + bus->audio_component = acomp; + hda_acomp = acomp; + + component_match_add(dev, &match, hda_component_master_match, bus); + ret = component_master_add_with_match(dev, &hda_component_master_ops, + match); + if (ret < 0) + goto out_err; + + /* + * Atm, we don't support deferring the component binding, so make sure + * i915 is loaded and that the binding successfully completes. + */ + request_module("i915"); + + if (!acomp->ops) { + ret = -ENODEV; + goto out_master_del; + } + dev_dbg(dev, "bound to i915 component master\n"); + + return 0; +out_master_del: + component_master_del(dev, &hda_component_master_ops); +out_err: + kfree(acomp); + bus->audio_component = NULL; + dev_err(dev, "failed to add i915 component master (%d)\n", ret); + + return ret; +} +EXPORT_SYMBOL_GPL(hda_i915_init); + +int hda_i915_exit(struct hdac_bus *bus) +{ + struct device *dev = bus->dev; + struct i915_audio_component *acomp = bus->audio_component; + + WARN_ON(bus->i915_power_refcount); + if (bus->i915_power_refcount > 0 && acomp && acomp->ops) + acomp->ops->put_power(acomp->dev); + + component_master_del(dev, &hda_component_master_ops); + + kfree(acomp); + bus->audio_component = NULL; + + return 0; +} +EXPORT_SYMBOL_GPL(hda_i915_exit); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 117bf5c..98ced97 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -145,11 +145,6 @@ config SND_HDA_CODEC_HDMI comment "Set to Y if you want auto-loading the codec driver" depends on SND_HDA=y && SND_HDA_CODEC_HDMI=m -config SND_HDA_I915 - bool - default y - depends on DRM_I915 - config SND_HDA_CODEC_CIRRUS tristate "Build Cirrus Logic codec support" select SND_HDA_GENERIC diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index c5e6651..a32ab83 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,7 +1,5 @@ snd-hda-intel-objs := hda_intel.o snd-hda-tegra-objs := hda_tegra.o -# for haswell power well -snd-hda-intel-$(CONFIG_SND_HDA_I915) += hda_i915.o snd-hda-codec-y := hda_bind.o hda_codec.o hda_jack.o hda_auto_parser.o hda_sysfs.o snd-hda-codec-y += hda_controller.o diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c deleted file mode 100644 index 8521702..0000000 --- a/sound/pci/hda/hda_i915.c +++ /dev/null @@ -1,227 +0,0 @@ -/* - * hda_i915.c - routines for Haswell HDA controller power well support - * - * 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. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/pci.h> -#include <linux/component.h> -#include <drm/i915_component.h> -#include <sound/core.h> -#include "hda_controller.h" -#include "hda_intel.h" - -/* Intel HSW/BDW display HDA controller Extended Mode registers. - * EM4 (M value) and EM5 (N Value) are used to convert CDClk (Core Display - * Clock) to 24MHz BCLK: BCLK = CDCLK * M / N - * The values will be lost when the display power well is disabled. - */ -#define AZX_REG_EM4 0x100c -#define AZX_REG_EM5 0x1010 - -int hda_set_codec_wakeup(struct hda_intel *hda, bool enable) -{ - struct i915_audio_component *acomp = &hda->audio_component; - - if (!acomp->ops) - return -ENODEV; - - if (!acomp->ops->codec_wake_override) { - dev_warn(&hda->chip.pci->dev, - "Invalid codec wake callback\n"); - return 0; - } - - dev_dbg(&hda->chip.pci->dev, "%s codec wakeup\n", - enable ? "enable" : "disable"); - - acomp->ops->codec_wake_override(acomp->dev, enable); - - return 0; -} - -int hda_display_power(struct hda_intel *hda, bool enable) -{ - struct i915_audio_component *acomp = &hda->audio_component; - - if (!acomp->ops) - return -ENODEV; - - dev_dbg(&hda->chip.pci->dev, "display power %s\n", - enable ? "enable" : "disable"); - - if (enable) { - if (!hda->i915_power_refcount++) - acomp->ops->get_power(acomp->dev); - } else { - WARN_ON(!hda->i915_power_refcount); - if (!--hda->i915_power_refcount) - acomp->ops->put_power(acomp->dev); - } - - return 0; -} - -void haswell_set_bclk(struct hda_intel *hda) -{ - int cdclk_freq; - unsigned int bclk_m, bclk_n; - struct i915_audio_component *acomp = &hda->audio_component; - struct pci_dev *pci = hda->chip.pci; - - /* Only Haswell/Broadwell need set BCLK */ - if (pci->device != 0x0a0c && pci->device != 0x0c0c - && pci->device != 0x0d0c && pci->device != 0x160c) - return; - - if (!acomp->ops) - return; - - cdclk_freq = acomp->ops->get_cdclk_freq(acomp->dev); - switch (cdclk_freq) { - case 337500: - bclk_m = 16; - bclk_n = 225; - break; - - case 450000: - default: /* default CDCLK 450MHz */ - bclk_m = 4; - bclk_n = 75; - break; - - case 540000: - bclk_m = 4; - bclk_n = 90; - break; - - case 675000: - bclk_m = 8; - bclk_n = 225; - break; - } - - azx_writew(&hda->chip, EM4, bclk_m); - azx_writew(&hda->chip, EM5, bclk_n); -} - -static int hda_component_master_bind(struct device *dev) -{ - struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); - struct i915_audio_component *acomp = &hda->audio_component; - int ret; - - ret = component_bind_all(dev, acomp); - if (ret < 0) - return ret; - - if (WARN_ON(!(acomp->dev && acomp->ops && acomp->ops->get_power && - acomp->ops->put_power && acomp->ops->get_cdclk_freq))) { - ret = -EINVAL; - goto out_unbind; - } - - /* - * Atm, we don't support dynamic unbinding initiated by the child - * component, so pin its containing module until we unbind. - */ - if (!try_module_get(acomp->ops->owner)) { - ret = -ENODEV; - goto out_unbind; - } - - return 0; - -out_unbind: - component_unbind_all(dev, acomp); - - return ret; -} - -static void hda_component_master_unbind(struct device *dev) -{ - struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); - struct i915_audio_component *acomp = &hda->audio_component; - - module_put(acomp->ops->owner); - component_unbind_all(dev, acomp); - WARN_ON(acomp->ops || acomp->dev); -} - -static const struct component_master_ops hda_component_master_ops = { - .bind = hda_component_master_bind, - .unbind = hda_component_master_unbind, -}; - -static int hda_component_master_match(struct device *dev, void *data) -{ - /* i915 is the only supported component */ - return !strcmp(dev->driver->name, "i915"); -} - -int hda_i915_init(struct hda_intel *hda) -{ - struct component_match *match = NULL; - struct device *dev = &hda->chip.pci->dev; - struct i915_audio_component *acomp = &hda->audio_component; - int ret; - - component_match_add(dev, &match, hda_component_master_match, hda); - ret = component_master_add_with_match(dev, &hda_component_master_ops, - match); - if (ret < 0) - goto out_err; - - /* - * Atm, we don't support deferring the component binding, so make sure - * i915 is loaded and that the binding successfully completes. - */ - request_module("i915"); - - if (!acomp->ops) { - ret = -ENODEV; - goto out_master_del; - } - - dev_dbg(dev, "bound to i915 component master\n"); - - return 0; -out_master_del: - component_master_del(dev, &hda_component_master_ops); -out_err: - dev_err(dev, "failed to add i915 component master (%d)\n", ret); - - return ret; -} - -int hda_i915_exit(struct hda_intel *hda) -{ - struct device *dev = &hda->chip.pci->dev; - struct i915_audio_component *acomp = &hda->audio_component; - - WARN_ON(hda->i915_power_refcount); - if (hda->i915_power_refcount > 0 && acomp->ops) - acomp->ops->put_power(acomp->dev); - - component_master_del(dev, &hda_component_master_ops); - - return 0; -} diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 706879a..229a1e4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -57,6 +57,8 @@ #endif #include <sound/core.h> #include <sound/initval.h> +#include <sound/hdaudio.h> +#include <sound/hda_i915.h> #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> @@ -493,13 +495,13 @@ static void azx_init_pci(struct azx *chip) static void hda_intel_init_chip(struct azx *chip, bool full_reset) { - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct hdac_bus *bus = azx_bus(chip); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) - hda_set_codec_wakeup(hda, true); + hda_set_codec_wakeup(bus, true); azx_init_chip(chip, full_reset); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) - hda_set_codec_wakeup(hda, false); + hda_set_codec_wakeup(bus, false); } /* calculate runtime delay from LPIB */ @@ -557,9 +559,9 @@ static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev) /* Enable/disable i915 display power for the link */ static int azx_intel_link_power(struct azx *chip, bool enable) { - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct hdac_bus *bus = azx_bus(chip); - return hda_display_power(hda, enable); + return hda_display_power(bus, enable); } /* @@ -797,6 +799,50 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) #define azx_del_card_list(chip) /* NOP */ #endif /* CONFIG_PM */ +/* Intel HSW/BDW display HDA controller is in GPU. Both its power and link BCLK + * depends on GPU. Two Extended Mode registers EM4 (M value) and EM5 (N Value) + * are used to convert CDClk (Core Display Clock) to 24MHz BCLK: + * BCLK = CDCLK * M / N + * The values will be lost when the display power well is disabled and need to + * be restored to avoid abnormal playback speed. + */ +static void haswell_set_bclk(struct hda_intel *hda) +{ + struct azx *chip = &hda->chip; + int cdclk_freq; + unsigned int bclk_m, bclk_n; + + if (!hda->need_i915_power) + return; + + cdclk_freq = hda_get_display_clk(azx_bus(chip)); + switch (cdclk_freq) { + case 337500: + bclk_m = 16; + bclk_n = 225; + break; + + case 450000: + default: /* default CDCLK 450MHz */ + bclk_m = 4; + bclk_n = 75; + break; + + case 540000: + bclk_m = 4; + bclk_n = 90; + break; + + case 675000: + bclk_m = 8; + bclk_n = 225; + break; + } + + azx_writew(chip, HSW_EM4, bclk_m); + azx_writew(chip, HSW_EM5, bclk_n); +} + #if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) /* * power management @@ -830,7 +876,7 @@ static int azx_suspend(struct device *dev) pci_disable_msi(chip->pci); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) - hda_display_power(hda, false); + hda_display_power(bus, false); return 0; } @@ -851,7 +897,7 @@ static int azx_resume(struct device *dev) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) { - hda_display_power(hda, true); + hda_display_power(azx_bus(chip), true); haswell_set_bclk(hda); } if (chip->msi) @@ -895,7 +941,7 @@ static int azx_runtime_suspend(struct device *dev) azx_clear_irq_pending(chip); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) - hda_display_power(hda, false); + hda_display_power(azx_bus(chip), false); return 0; } @@ -905,6 +951,7 @@ static int azx_runtime_resume(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; struct hda_intel *hda; + struct hdac_bus *bus; struct hda_codec *codec; int status; @@ -921,11 +968,12 @@ static int azx_runtime_resume(struct device *dev) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) { - hda_display_power(hda, true); + bus = azx_bus(chip); + hda_display_power(bus, true); haswell_set_bclk(hda); /* toggle codec wakeup bit for STATESTS read */ - hda_set_codec_wakeup(hda, true); - hda_set_codec_wakeup(hda, false); + hda_set_codec_wakeup(bus, true); + hda_set_codec_wakeup(bus, false); } /* Read STATESTS before controller reset */ @@ -1143,10 +1191,11 @@ static int azx_free(struct azx *chip) #ifdef CONFIG_SND_HDA_PATCH_LOADER release_firmware(chip->fw); #endif + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { if (hda->need_i915_power) - hda_display_power(hda, false); - hda_i915_exit(hda); + hda_display_power(bus, false); + hda_i915_exit(bus); } kfree(hda); @@ -1905,6 +1954,7 @@ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] = { static int azx_probe_continue(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct hdac_bus *bus = azx_bus(chip); struct pci_dev *pci = chip->pci; int dev = chip->dev_index; int err; @@ -1921,11 +1971,11 @@ static int azx_probe_continue(struct azx *chip) if (pci->device != 0x0f04 && pci->device != 0x2284) hda->need_i915_power = 1; - err = hda_i915_init(hda); + err = hda_i915_init(bus); if (err < 0) goto i915_power_fail; - err = hda_display_power(hda, true); + err = hda_display_power(bus, true); if (err < 0) { dev_err(chip->card->dev, "Cannot turn on display power on i915\n"); @@ -1977,7 +2027,7 @@ static int azx_probe_continue(struct azx *chip) out_free: if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && !hda->need_i915_power) - hda_display_power(hda, false); + hda_display_power(bus, false); i915_power_fail: if (err < 0) diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index 7fd3254..354f0bb 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -16,7 +16,6 @@ #ifndef __SOUND_HDA_INTEL_H #define __SOUND_HDA_INTEL_H -#include <drm/i915_component.h> #include "hda_controller.h" struct hda_intel { @@ -44,36 +43,7 @@ struct hda_intel { /* secondary power domain for hdmi audio under vga device */ struct dev_pm_domain hdmi_pm_domain; - /* i915 component interface */ bool need_i915_power:1; /* the hda controller needs i915 power */ - struct i915_audio_component audio_component; - int i915_power_refcount; }; -#ifdef CONFIG_SND_HDA_I915 -int hda_set_codec_wakeup(struct hda_intel *hda, bool enable); -int hda_display_power(struct hda_intel *hda, bool enable); -void haswell_set_bclk(struct hda_intel *hda); -int hda_i915_init(struct hda_intel *hda); -int hda_i915_exit(struct hda_intel *hda); -#else -static inline int hda_set_codec_wakeup(struct hda_intel *hda, bool enable) -{ - return 0; -} -static inline int hda_display_power(struct hda_intel *hda, bool enable) -{ - return 0; -} -static inline void haswell_set_bclk(struct hda_intel *hda) { return; } -static inline int hda_i915_init(struct hda_intel *hda) -{ - return 0; -} -static inline int hda_i915_exit(struct hda_intel *hda) -{ - return 0; -} -#endif - #endif -- 1.9.1

3 2

[alsa-devel] Low volume since linux 3.19
by Jan Hinnerk Stosch 18 May '15

18 May '15

Hi, I hope this is the right place to report my issue!? Since linux 3.19 I have very low volume on my Fujitsu E753 headphone even on highest volume level. As I don't have much experience with audio configuration and it worked before I did a kernel bisect and found out, that commit "03ad6a8c93b6df2d65c305b5b5f9474068b45bfb" is "responsible" for this. Is there something that I have to change in my configuration? Or do you need some further information? Here is some output from dmesg: [ 5.853009] sound hdaudioC0D0: ALC269VB: SKU not ready 0x909701f0 [ 5.853857] sound hdaudioC0D0: autoconfig: line_outs=1 (0x21/0x0/0x0/0x0/0x0) type:line [ 5.853860] sound hdaudioC0D0: speaker_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 5.853862] sound hdaudioC0D0: hp_outs=1 (0x1a/0x0/0x0/0x0/0x0) [ 5.853864] sound hdaudioC0D0: mono: mono_out=0x0 [ 5.853866] sound hdaudioC0D0: inputs: [ 5.853869] sound hdaudioC0D0: Mic=0x18 [ 5.853871] sound hdaudioC0D0: Dock Mic=0x1b [ 5.853873] sound hdaudioC0D0: Internal Mic=0x12 Thanks in advance, jhs

2 6

[alsa-devel] [PATCH 1/3] ALSA: hda - add hdac stream trace
by libin.yang＠intel.com 18 May '15

18 May '15

From: Libin Yang <libin.yang(a)intel.com> Add the trace of snd_hdac_stream_start and snd_hdac_stream_stop. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/hda/hdac_stream.c | 5 +++++ sound/hda/trace.h | 27 +++++++++++++++++++++++++++ 2 files changed, 32 insertions(+) diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index 1ba0462..52a894f 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -10,6 +10,7 @@ #include <sound/pcm.h> #include <sound/hdaudio.h> #include <sound/hda_register.h> +#include "trace.h" /** * snd_hdac_stream_init - initialize each stream (aka device) @@ -48,6 +49,8 @@ void snd_hdac_stream_start(struct hdac_stream *azx_dev, bool fresh_start) { struct hdac_bus *bus = azx_dev->bus; + trace_snd_hdac_stream_start(bus, azx_dev); + azx_dev->start_wallclk = snd_hdac_chip_readl(bus, WALLCLK); if (!fresh_start) azx_dev->start_wallclk -= azx_dev->period_wallclk; @@ -82,6 +85,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_stream_clear); */ void snd_hdac_stream_stop(struct hdac_stream *azx_dev) { + trace_snd_hdac_stream_stop(azx_dev->bus, azx_dev); + snd_hdac_stream_clear(azx_dev); /* disable SIE */ snd_hdac_chip_updatel(azx_dev->bus, INTCTL, 1 << azx_dev->index, 0); diff --git a/sound/hda/trace.h b/sound/hda/trace.h index 33a7eb5..e27e2c0 100644 --- a/sound/hda/trace.h +++ b/sound/hda/trace.h @@ -50,6 +50,33 @@ TRACE_EVENT(hda_unsol_event, ), TP_printk("%s", __get_str(msg)) ); + +DECLARE_EVENT_CLASS(hdac_stream, + TP_PROTO(struct hdac_bus *bus, struct hdac_stream *azx_dev), + + TP_ARGS(bus, azx_dev), + + TP_STRUCT__entry( + __field(unsigned char, stream_tag) + ), + + TP_fast_assign( + __entry->stream_tag = (azx_dev)->stream_tag; + ), + + TP_printk("stream_tag: %d", __entry->stream_tag) +); + +DEFINE_EVENT(hdac_stream, snd_hdac_stream_start, + TP_PROTO(struct hdac_bus *bus, struct hdac_stream *azx_dev), + TP_ARGS(bus, azx_dev) +); + +DEFINE_EVENT(hdac_stream, snd_hdac_stream_stop, + TP_PROTO(struct hdac_bus *bus, struct hdac_stream *azx_dev), + TP_ARGS(bus, azx_dev) +); + #endif /* __HDAC_TRACE_H */ /* This part must be outside protection */ -- 1.9.1

2 3

[alsa-devel] [PATCH 1/6] ALSA: hda/tegra - Set CORBRP self-clear flag
by Thierry Reding 18 May '15

18 May '15

From: Thierry Reding <treding(a)nvidia.com> This is set for the MCP variants of the NVIDIA HDA controller, which the Tegra variant was derived from. This fixes the following warning at boot time: [ 2.486610] tegra-hda 70030000.hda: CORB reset timeout#1, CORBRP = 0 Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index db0bb50fa5b9..7003677f7473 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -460,11 +460,12 @@ MODULE_DEVICE_TABLE(of, hda_tegra_match); static int hda_tegra_probe(struct platform_device *pdev) { + const unsigned int driver_flags = AZX_DCAPS_RIRB_DELAY | + AZX_DCAPS_CORBRP_SELF_CLEAR; struct snd_card *card; struct azx *chip; struct hda_tegra *hda; int err; - const unsigned int driver_flags = AZX_DCAPS_RIRB_DELAY; hda = devm_kzalloc(&pdev->dev, sizeof(*hda), GFP_KERNEL); if (!hda) -- 2.3.5

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Re: [alsa-devel] Support Dell headset mode for ALC298 platform
by Takashi Iwai 18 May '15

18 May '15

At Mon, 18 May 2015 07:38:53 +0000, Kailang wrote: > > Hi Takashi, > > Dell has new platform for ALC298. > Attach patch was enable headset mode for ALC298 machine. Applied, thanks. Takashi > > BR, > Kailang > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 0000-support-dell-headset-mode-for-alc298.patch <application/octet-stream (base64)>] >

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[alsa-devel] [PATCH] ALSA: hda/realtek - ALC292 dock fix for Thinkpad L450
by Ansgar Hegerfeld 18 May '15

18 May '15

The Lenovo ThinkPad L450 requires the ALC292_FIXUP_TPT440_DOCK fix in order to get sound output on the docking stations audio port. This patch was tested using a ThinkPad L450 (20DSS00B00) using kernel 4.0.3 and a ThinkPad Pro Dock. Signed-off-by: Ansgar Hegerfeld <linux(a)hegerfeld.org> --- 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 e2afd53..a9f41a4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5148,6 +5148,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5034, "Thinkpad T450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),

2 1

[alsa-devel] [PATCH] ALSA: hda - Add Conexant codecs CX20721, CX20722, CX20723 and CX20724
by David Henningsson 18 May '15

18 May '15

This patch adds support for Conexant HD Audio codecs CX20721, CX20722, CX20723 and CX20724. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1454656 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f8f0dfb..78b719b 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -968,6 +968,14 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_conexant_auto }, { .id = 0x14f150b9, .name = "CX20665", .patch = patch_conexant_auto }, + { .id = 0x14f150f1, .name = "CX20721", + .patch = patch_conexant_auto }, + { .id = 0x14f150f2, .name = "CX20722", + .patch = patch_conexant_auto }, + { .id = 0x14f150f3, .name = "CX20723", + .patch = patch_conexant_auto }, + { .id = 0x14f150f4, .name = "CX20724", + .patch = patch_conexant_auto }, { .id = 0x14f1510f, .name = "CX20751/2", .patch = patch_conexant_auto }, { .id = 0x14f15110, .name = "CX20751/2", @@ -1002,6 +1010,10 @@ MODULE_ALIAS("snd-hda-codec-id:14f150ab"); MODULE_ALIAS("snd-hda-codec-id:14f150ac"); MODULE_ALIAS("snd-hda-codec-id:14f150b8"); MODULE_ALIAS("snd-hda-codec-id:14f150b9"); +MODULE_ALIAS("snd-hda-codec-id:14f150f1"); +MODULE_ALIAS("snd-hda-codec-id:14f150f2"); +MODULE_ALIAS("snd-hda-codec-id:14f150f3"); +MODULE_ALIAS("snd-hda-codec-id:14f150f4"); MODULE_ALIAS("snd-hda-codec-id:14f1510f"); MODULE_ALIAS("snd-hda-codec-id:14f15110"); MODULE_ALIAS("snd-hda-codec-id:14f15111"); -- 1.9.1

2 1

[alsa-devel] [PATCH 0/4] Fixes and cleanups for the recent procfs changes
by Takashi Iwai 18 May '15

18 May '15

Hi, here is a series of patches to fix and clean up the recent changes for ALSA core procfs support code. Takashi === Takashi Iwai (4): ALSA: info: Fix leaks of child entries at snd_info_free_entry() ALSA: info: Register proc entries recursively, too ALSA: info: Move list addition to snd_info_create_entry() ALSA: info: Drop kerneldoc comment from snd_info_create_entry() sound/core/info.c | 62 +++++++++++++++++++++++++++++++++++++++++-------------- sound/core/init.c | 18 ++++++---------- 2 files changed, 52 insertions(+), 28 deletions(-) -- 2.3.6

1 4

[alsa-devel] [PATCH 1/1] ALSA: core: Fix regression for ELD/codec information files
by han.lu＠intel.com 18 May '15

18 May '15

From: "Lu, Han" <han.lu(a)intel.com> The commit c560a6797e3b ("ALSA: core: Remove child proc file elements recursively") introduced a regression that ELD/codec information files can not be found on /proc/asound/cardX/. This patch corrects it. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/include/sound/info.h b/include/sound/info.h index 1626995..ae8d30a 100644 --- a/include/sound/info.h +++ b/include/sound/info.h @@ -149,6 +149,10 @@ static inline int snd_card_proc_new(struct snd_card *card, const char *name, struct snd_info_entry **entryp) { *entryp = snd_info_create_card_entry(card, name, card->proc_root); + if (snd_info_register(*entryp) < 0) { + snd_info_free_entry(*entryp); + *entryp = NULL; + } return *entryp ? 0 : -ENOMEM; } -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda - Fix headset mic and mic-in for a Dell desktop
by David Henningsson 18 May '15

18 May '15

ALC662 does not need any special verbs to change the jack functionality, and enables mic in through the headphone jack mode by changing the direction of the headphone pin node. BugLink: https://bugs.launchpad.net/bugs/1454235 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 46 ++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 45 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ce41271..30ec30f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3674,6 +3674,10 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, alc_process_coef_fw(codec, coef0293); snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); break; + case 0x10ec0662: + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); + snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); + break; case 0x10ec0668: alc_write_coef_idx(codec, 0x11, 0x0001); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); @@ -4012,7 +4016,7 @@ static void alc_update_headset_mode(struct hda_codec *codec) if (new_headset_mode != ALC_HEADSET_MODE_MIC) { snd_hda_set_pin_ctl_cache(codec, hp_pin, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); - if (spec->headphone_mic_pin) + if (spec->headphone_mic_pin && spec->headphone_mic_pin != hp_pin) snd_hda_set_pin_ctl_cache(codec, spec->headphone_mic_pin, PIN_VREFHIZ); } @@ -4215,6 +4219,18 @@ static void alc_fixup_dell_xps13(struct hda_codec *codec, } } +static void alc_fixup_headset_mode_alc662(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + spec->gen.hp_mic = 1; /* Mic-in is same pin as headphone */ + } else + alc_fixup_headset_mode(codec, fix, action); +} + static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6080,7 +6096,9 @@ enum { ALC662_FIXUP_NO_JACK_DETECT, ALC662_FIXUP_ZOTAC_Z68, ALC662_FIXUP_INV_DMIC, + ALC662_FIXUP_DELL_MIC_NO_PRESENCE, ALC668_FIXUP_DELL_MIC_NO_PRESENCE, + ALC662_FIXUP_HEADSET_MODE, ALC668_FIXUP_HEADSET_MODE, ALC662_FIXUP_BASS_MODE4_CHMAP, ALC662_FIXUP_BASS_16, @@ -6273,6 +6291,20 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE }, + [ALC662_FIXUP_DELL_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a1113c }, /* use as headset mic, without its own jack detect */ + /* headphone mic by setting pin control of 0x1b (headphone out) to in + vref_50 */ + { } + }, + .chained = true, + .chain_id = ALC662_FIXUP_HEADSET_MODE + }, + [ALC662_FIXUP_HEADSET_MODE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode_alc662, + }, [ALC668_FIXUP_DELL_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -6432,6 +6464,18 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {0x1f, 0x411111f0} static const struct snd_hda_pin_quirk alc662_pin_fixup_tbl[] = { + SND_HDA_PIN_QUIRK(0x10ec0662, 0x1028, "Dell", ALC662_FIXUP_DELL_MIC_NO_PRESENCE, + {0x12, 0x4004c000}, + {0x14, 0x01014010}, + {0x15, 0x411111f0}, + {0x16, 0x411111f0}, + {0x18, 0x01a19020}, + {0x19, 0x411111f0}, + {0x1a, 0x0181302f}, + {0x1b, 0x0221401f}, + {0x1c, 0x411111f0}, + {0x1d, 0x4054c601}, + {0x1e, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0668, 0x1028, "Dell", ALC668_FIXUP_AUTO_MUTE, ALC668_STANDARD_PINS, {0x12, 0x99a30130}, -- 1.9.1

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[alsa-devel] [GIT PULL] ASoC fixes for v4.1
by Mark Brown 18 May '15

18 May '15

The following changes since commit 030bbdbf4c833bc69f502eae58498bc5572db736: Linux 4.1-rc3 (2015-05-10 15:12:29 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.1-rc3 for you to fetch changes up to 7730c0b55043f6ff1c27e9cb45e13679995b2361: Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/mc13783', 'asoc/fix/uda1380', 'asoc/fix/wm8960' and 'asoc/fix/wm8994' into asoc-linus (2015-05-12 20:31:33 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.1 A few more fixes for v4.1, some driver fixes plus one core fix which fixes registration of DAI links when adding prefixes to CODECs to deuplicate in multi-CODEC systems. ---------------------------------------------------------------- Axel Lin (2): ASoC: mc13783: Fix wrong mask value used in mc13xxx_reg_rmw() calls ASoC: rt5645: Fix mask for setting RT5645_DMIC_2_DP_GPIO12 bit Fang, Yang A (1): ASoC: rt5645: Add ACPI match ID Koro Chen (1): ASoC: dapm: Modify widget stream name according to prefix Mark Brown (3): Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/mc13783', 'asoc/fix/uda1380', 'asoc/fix/wm8960' and 'asoc/fix/wm8994' into asoc-linus Peter Ujfalusi (1): ASoC: davinci-mcasp: Correct pm status check in suspend callback Vasily Khoruzhick (1): ASoC: uda1380: Avoid accessing i2c bus when codec is disabled Zidan Wang (2): ASoC: wm8960: fix "RINPUT3" audio route error ASoC: wm8994: correct BCLK DIV 348 to 384 sound/soc/codecs/mc13783.c | 4 ++-- sound/soc/codecs/uda1380.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/davinci/davinci-mcasp.c | 2 +- sound/soc/soc-dapm.c | 11 ++++++++--- 6 files changed, 14 insertions(+), 9 deletions(-)

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Re: [alsa-devel] Fwd: Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 17 May '15

17 May '15

> > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/patch/sound/pci > > /hda/patch_realtek.c?id=c3e837bbcc03e39991da5dd1f791c06cde9f7a45 > > > > When the static model were used, the jack was configured as line out > > instead of headphone jack and there is no headphone playback volume > > That's rather "relevant change". Good job in finding it. I would like such > configuration (or better) back. > I don't see any side effect neither with your changes to hda_generic nor > with > vanilla kernel, but I don't know what problems should I expect. > Do headphone get same volume level when the jack is configured as Line Out (without enable Headphone Amplifier) ? The driver use different automute method when the jack is configured as HP and Line Out https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… if you look at speaker.conf , lineout.conf and headphone.conf, it seem pulseaudio is unlikely mute all internal speakers when one pair of the playback volume control need to be shared by internal speaker and external speakers you need dirver 's automute instead of pulseaudio 's automute ==> Best config: lo_type=2, wired=1, mio=1 multi_outs = 15/0/0/0 : 2/3/4/0 (type HP) out path: depth=3 '02:0c:15' multi_ios(2) = 1a/18 : 3/4 mio path: depth=3 '03:0d:1a' mio path: depth=3 '04:0e:18' spk_outs = 14/17/16/0 : 5/25/4/0 spk path: depth=3 '05:0f:14' spk path: depth=3 '25:26:17' spk path: depth=3 '04:0e:16' https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… when line out type == AUTO_PIN_HP_OUT should first do_mute() use speaker_path instead of out_path <https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p…> >>> I started MS Windows and without externals speakers plugged in I >> >> could test >> > 5.1 configuration and it seemed that sound was coming from >> different directions. >> >> > >> > What I managed to determine: >> > I've got 3 stacks: >> > 1 ST: Headphone, Line-out (FL and FR), SPDIF >> > 2 ST: Microphone, Line-out (CLFE) >> > 3 ST: Line-in, Line-out (RL and RR) >> > >> > I've got 5 responsive output pins >> > PIN 0x14: Internal FL FR >> > PIN 0x15: External FL FR >> > PIN 0x17: Internal LFE >> > PIN 0x18: External CLFE >> > PIN 0x1a: External RL RR >> > >> > How can we make it work with these information? speaker-test -Dhw:0,0 -c6 -t wav >> >> >> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/plain/Document… >> [1] >> >> seem chnmap type _FIXED , _VAR and _PAIRED does not fit when LFE and >> FC can be swapped for internal speaker and external speaker > > > Right, you can't do it with a single map of one of those types. http://git.alsa-project.org/?p=alsa-utils.git&a=search&h=HEAD&st=commit&s=c… do chmap of internal speakers need {FL FR RL RR LFE NA} ? http://git.alsa-project.org/?p=alsa-lib.git&a=search&h=HEAD&st=commit&s=chm… as set_chmap() still not yet implemented in pcm_route.c do speaker-test need to specify channel map when using those odd number channels ? speaker-test -D surround21 -c3 -t wav speaker-test -D surround41 -c5 -t wav -mFL,FR,RL,RR,LFE speaker-test -D surround50 -c5 -t wav http://git.alsa-project.org/?p=alsa-utils.git;a=commitdiff;h=7b14c00b56784f… this example seem not working speaker-test -c2 -t wav -Dplug:front -mFR,FL > > >> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… >> [2] >> >> How do the driver implement channel map when codec has swap >> center/lfe switch ? >> >> Do chmap contain both {FL FR RL RR FC LFE} and {FL FR RL RR LFE FC} ? > > > If you want to provide support for userspace doing it both ways, then yes, you need those two _FIXED maps. Can chmap contain two _FIXED maps of same number of channels ?

2 2

[alsa-devel] Microphone issues with Microsoft LifeCam Studio
by Dainius Masiliūnas 16 May '15

16 May '15

I'm having an issue with the Microsoft LifeCam Studio in that the camera does not work if the integrated microphone in the camera is activated at the same time. When that happens, the kernel log outputs a variety of warnings such as: cannot get freq at ep 0x82 ALSA lib pcm_dsnoop.c:614:(snd_pcm_dsnoop_open) unable to open slave uvcvideo: Failed to query (GET_DEF) UVC control 2 on unit 4: -32 (exp. 2). uvcvideo: Failed to query (GET_DEF) UVC control 2 on unit 4: 0 (exp. 2). ALSA lib pcm.c:7980:(snd_pcm_set_params) Channels count (1) not available for CAPTURE: Incorrect argument ALSA lib pcm.c:7956:(snd_pcm_set_params) Broken configuration for CAPTURE: no configurations available ALSA lib pcm_route.c:867:(find_matching_chmap) Found no matching channel map uvcvideo: Failed to set UVC probe control : -32 (exp. 26). There is a workaround in that disabling the built-in microphone by blacklisting snd_usb_audio does make the camera work, but obviously it's not ideal at all, as that way half the camera's functionality is not available and no other USB audio peripherals can be used. I'm assuming that this is some sort of a quirk in the camera itself; the UVC page ( http://www.ideasonboard.org/uvc/#footnote-13 ) does mention that there is an issue with the bandwidth requests, but setting the 0x80 quirk does not change the situation with the microphone in any way, unfortunately. I am running the latest mainline kernel (kernel-ml from elrepo) on CentOS 7.1. Can anything be done about this issue, and is there anything I could do to help debug it? Also, this issue has been noted a few years back, but it was left unsolved: https://bugs.launchpad.net/ubuntu/+source/guvcview/+bug/1033146 https://sourceforge.net/p/linux-uvc/mailman/message/29645043/ The original reporter said it was a Flash bug, but that is in no way the case because my machine does not have Flash installed to begin with (I'm testing this with WebRTC). -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

Re: [alsa-devel] [pulseaudio-discuss] ThinkPad T-510 audio output mute LED non-workingness
by Takashi Iwai 16 May '15

16 May '15

[Switched Cc to alsa-devel ML, as it's irrelevant from PA but rather a driver issue.] At Mon, 27 Apr 2015 07:58:59 -0600, Glenn Golden wrote: > > Takashi Iwai <tiwai(a)suse.de> [2015-04-27 08:50:34 +0200]: > > > > Gah, I hate reports without alsa-info.sh output (not a link to > > somewhere else). > > > > Glenn, please attach alsa-info.sh outputs (run it with --no-upload > > option) and the whole dmesg output after boot. 3.19 should already > > contain the hook to thinkpad_acpi.c. If anything is missing, you > > should have a warning indicating it. > > > > Apologies for the omissions. Both are attached. Thanks! > Fwiw -- perhaps nothing at this point -- a good deal of info has also been > posted to the kernel bugtracker ticket: > > https://bugzilla.kernel.org/show_bug.cgi?id=96171 > > Re: "...you should have a warning indicating it": I'm an unskilled laboror > here w.r.t. kernel affairs, and not even sure what sort of warnings to be > looking for, so may have missed something important without realizing it. > Just let me know what add'l info you need, I'll be happy to post it. I do > appreciate your attention to the issue. I see no such warning in the log you attached, so it's not the issue, as it seems. Below is a debug patch, and now you should see messages like "XXX ..." in kernel log at loading the driver. If there is no such a message, it means that the thinkpad hook isn't built in at all. Please check the kernel log again with the patch. Takashi --- diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 0a4ad5feb82e..48ce61e46599 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -34,6 +34,7 @@ static void update_tpacpi_mute_led(void *private_data, int enabled) if (old_vmaster_hook) old_vmaster_hook(private_data, enabled); + pr_info("XXX mute led updated %d\n", enabled); if (led_set_func) led_set_func(TPACPI_LED_MUTE, !enabled); } @@ -58,8 +59,10 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, bool removefunc = false; if (action == HDA_FIXUP_ACT_PROBE) { + codec_info(codec, "XXX hda_fixup_thinkpad_acpi init...\n"); if (!is_thinkpad(codec)) return; + codec_info(codec, "XXX ok, this is thinkpad\n"); if (!led_set_func) led_set_func = symbol_request(tpacpi_led_set); if (!led_set_func) { @@ -70,18 +73,24 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, removefunc = true; if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { + codec_info(codec, "XXX ok, mute LED is hooked\n"); old_vmaster_hook = spec->vmaster_mute.hook; spec->vmaster_mute.hook = update_tpacpi_mute_led; removefunc = false; + } else { + codec_err(codec, "XXX mute LED NOT WORKING!\n"); } if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { if (spec->num_adc_nids > 1) codec_dbg(codec, "Skipping micmute LED control due to several ADCs"); else { + codec_info(codec, "XXX ok, mic LED is hooked\n"); spec->cap_sync_hook = update_tpacpi_micmute_led; removefunc = false; } + } else { + codec_err(codec, "XXX mic LED NOT WORKING!\n"); } }

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[alsa-devel] [PATCH] ASoC: wm8955: Fix setting wrong register for WM8955_K_8_0_MASK bits
by Axel Lin 16 May '15

16 May '15

WM8955_K_8_0_MASK bits is controlled by WM8955_PLL_CONTROL_3 rather than WM8955_PLL_CONTROL_2. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm8955.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index f400d5c..08063a6 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -298,7 +298,7 @@ static int wm8955_configure_clocking(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8955_PLL_CONTROL_2, WM8955_K_17_9_MASK, (pll.k >> 9) & WM8955_K_17_9_MASK); - snd_soc_update_bits(codec, WM8955_PLL_CONTROL_2, + snd_soc_update_bits(codec, WM8955_PLL_CONTROL_3, WM8955_K_8_0_MASK, pll.k & WM8955_K_8_0_MASK); if (pll.k) -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: Drop unnecessary bias level check on resume
by Lars-Peter Clausen 15 May '15

15 May '15

The suspended flag will only be set if the CODEC bias level was either STANDBY or OFF. This means we don't need to check for that on resume since the condition will always be true. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 19 +++---------------- 1 file changed, 3 insertions(+), 16 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2373252..95b5f03 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -750,23 +750,10 @@ static void soc_resume_deferred(struct work_struct *work) } list_for_each_entry(codec, &card->codec_dev_list, card_list) { - /* If the CODEC was idle over suspend then it will have been - * left with bias OFF or STANDBY and suspended so we must now - * resume. Otherwise the suspend was suppressed. - */ if (codec->suspended) { - switch (codec->dapm.bias_level) { - case SND_SOC_BIAS_STANDBY: - case SND_SOC_BIAS_OFF: - if (codec->driver->resume) - codec->driver->resume(codec); - codec->suspended = 0; - break; - default: - dev_dbg(codec->dev, - "ASoC: CODEC was on over suspend\n"); - break; - } + if (codec->driver->resume) + codec->driver->resume(codec); + codec->suspended = 0; } } -- 1.8.0

2 1

[alsa-devel] [PATCH 1/8] ASoC: sn95031: Replace direct snd_soc_codec dapm field access
by Lars-Peter Clausen 15 May '15

15 May '15

The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm.bias_level with snd_soc_codec_get_bias_level(). While we are at it also replace the if(x == A) ... else if(x == B) ... construct with a switch-case. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Complete series depends on topic/dapm --- sound/soc/codecs/sn95031.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index e474368..3a7de01 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -194,7 +194,7 @@ static int sn95031_set_vaud_bias(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_PREPARE: - if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY) { pr_debug("vaud_bias powering up pll\n"); /* power up the pll */ snd_soc_write(codec, SN95031_AUDPLLCTRL, BIT(5)); @@ -205,17 +205,22 @@ static int sn95031_set_vaud_bias(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + switch (snd_soc_codec_get_bias_level(codec)) { + case SND_SOC_BIAS_OFF: pr_debug("vaud_bias power up rail\n"); /* power up the rail */ snd_soc_write(codec, SN95031_VAUD, BIT(2)|BIT(1)|BIT(0)); msleep(1); - } else if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) { + break; + case SND_SOC_BIAS_PREPARE: /* turn off pcm */ pr_debug("vaud_bias power dn pcm\n"); snd_soc_update_bits(codec, SN95031_PCM2C2, BIT(0), 0); snd_soc_write(codec, SN95031_AUDPLLCTRL, 0); + break; + default: + break; } break; -- 1.8.0

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[alsa-devel] USB-Audio.conf routing and dmix
by sylvain.bertrand＠gmail.com 15 May '15

15 May '15

Hi, I have a bose USB audio 5.1 device, a generic USB audio device. I noticed that the proper channels routing was missing from the default generic USB pcm, but dmix was used. And for the surround pcms I noticed it was the other way around! I did modify the default pcm to include a routing plugin. I put 8 channels in the routing plugin. I guess the proper way would be to detect the hw card number of channels and select a route plugin configuration with the correct count of channels (maybe their is a function to get that info?). I linked the surround51 and surround71 pcms with the 'disable-able' dmix section from the default pcm. I did not touch the surround40* pcms, surround pcms which are specific to some cards and the iec958 ones. http://filebin.ca/214UTT8fBnUm/USB-Audio.conf regards, -- Sylvain BERTRAND

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[alsa-devel] [PATCH early RFC 0/2] Implement ASoC HDMI codec library
by Jyri Sarha 15 May '15

15 May '15

This is on early RFC and should not be merged yet. The idea is just to share my ideas early on as there has been a lot of development going on around HDMI audio. The I2S DAI of the ASoC side patch is usable already, the spdif support has not been tested and the EDID SADs handling should use Russel's DRM ELD helper when it is ready. The tda998x patch is just to demonstrate the usage of the HDMI-codec-lib. At least the audio related DT-bindings are missing completely and the configuration is hard coded to work on Beaglebone-Black. Jean-Francois, would you consider trying the generic ASoC patch with your HW, as I can not test the spdif functionality with mine? The library could also be implemented as a separate platform driver, but then adding a pointer for private data to struct snd_soc_dai_driver, snd_soc_codec, or to snd_soc_component would be of great help. These patches, my tilcdc refactoring[1], and my latest BCLK fixes for davinci-mcasp diver [2], can found in a branch that produces a working HDMI audio on Beaglebone-Black here: https://github.com/jsarha/linux.git linux-master-bbb-hdmi-20150512 [1] http://lists.freedesktop.org/archives/dri-devel/2015-May/082537.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-April/090974.html Jyri Sarha (2): ASoC: hdmi-codec-lib: Add hdmi-codec-lib for external HDMI-encoders drm/i2c: tda998x: HACK Implement primitive HDMI audio with ASoC hdmi-code-lib drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 238 +++++++++++++++++ include/sound/hdmi-codec-lib.h | 105 ++++++++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdmi-codec-lib.c | 536 ++++++++++++++++++++++++++++++++++++++ 6 files changed, 886 insertions(+) create mode 100644 include/sound/hdmi-codec-lib.h create mode 100644 sound/soc/codecs/hdmi-codec-lib.c -- 1.9.1

2 4

[alsa-devel] No sound from Mac Pro headphone port
by Todd Dailey 15 May '15

15 May '15

https://bugtrack.alsa-project.org/alsa-bug does not work so following instructions to post bug here. Linux 4. Mac Pro 2009. ACL889A codec. No sound from headphone port. Seems to detect plug / unplug OK. Works OK when machine booted to OS X or Windows 7. Have found several bug repors on similar issues with Mac laptops but no fixes. Have written e-mail with David Henningsson on problem but unable to find a solution. ALSA-info output: http://www.alsa-project.org/db/?f=1dc7146eb34af5ad307d029868f775fac5e2764e -- TD todd(a)todddailey.ca

2 5

[alsa-devel] [PATCH 01/11] ASoC: sta32x: Replace direct snd_soc_codec dapm field access
by Lars-Peter Clausen 14 May '15

14 May '15

The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm.bias_level with snd_soc_codec_get_bias_level(). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- All patches in this series depend on topic/dapm --- sound/soc/codecs/sta32x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 847fb20..60eff36 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -819,7 +819,7 @@ static int sta32x_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); if (ret != 0) { -- 1.8.0

2 11

Re: [alsa-devel] Intel DG45ID 5stack (No line in)
by Raymond Yau 14 May '15

14 May '15

> I have been using this MotherBoard for years (it is probably the most reliable one I have ever had): > http://www.intel.com/p/en_US/support/highlights/dsktpboards/dg45id > but from the beginning I had trouble with ALSA and the Line In port - documented here: > http://mailman.alsa-project.org/pipermail/alsa-devel/2009-February/014825.h… > Mostly this has not been a problem since I only use the Line In port on infrequent occasions but it is annoying that when I do need to use it, sometimes, the method I have found to activate the port does not work and now using it seems to have permanently messed up the configuration for Back Mic for normal use somehow. I have installed Fedora 22 on a new drive (currently using 21) and booting on the new disk cures the introduced problem so it appears not to be permanent sound card damage at all and just some sort of caching or other issue. My procedure for getting the Line In port used to be this (although it didn't work this time and just caused a problem with the Back Mic): > - uninstall PulseAudio (or rename /usr/bin/pulseaudio) > - stop all apps auto starting that use ALSA > - reboot > - systemctl stop alsa-state.service > - echo 0x0c 0x01813021 > /sys/class/sound/hwC0D2/user_pin_configs > - echo 1 > /sys/class/sound/hwC0D2/reconfig > - chmod -R 777 /dev/snd > - systemctl start alsa-state.service > Going through this procedure showed up new controls in xfce4-mixer and everything worked fine for what I wanted to do - a reboot restored the original setup (it could also be done with echoing the original values). It has been some years since I used this technique but now I can't get it to work and some of the new controls that show up in xfce4-mixer ("Capture 1", "Input Source 1") don't disappear with a reboot - even if I: > rm .config/xfce4/xfconf/xfce-perchannel-xml/xfce4-mixer.xml > !? Also, from Audacity I can now no longer select "Default" for the microphone and I can't get any of the other options to record . . > Is anybody able to tell me what is going on? I can just starting using F22 on the new drive and forget about using this procedure in the future but I don't like not knowing why things don't work reliably . . > Any useful information would be appreciated you have to post the output of alsa-info.sh https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=52d… pin fixup seem using [Jack] Speaker instead of [Jack] Line Out at Ext Rear Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange Line in not assigned correctly https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=4d2… static const struct hda_pintbl intel_dg45id_pin_configs[] = { { 0x0a, 0x02214230 }, { 0x0b, 0x02A19240 }, - { 0x0c, 0x01013214 }, + { 0x0c, 0x01813021 }, { 0x0d, 0x01014210 }, { 0x0e, 0x01A19250 }, { 0x0f, 0x01011212 }, { 0x10, 0x01016211 }, {} };

1 0

Re: [alsa-devel] [PATCH v2 02/17] leds: port locomo leds driver to new locomo core
by Dmitry Torokhov 14 May '15

14 May '15

On Wed, May 13, 2015 at 04:53:03PM +0200, Jacek Anaszewski wrote: > On 05/13/2015 04:14 PM, Dmitry Eremin-Solenikov wrote: > >2015-05-13 13:31 GMT+03:00 Jacek Anaszewski <j.anaszewski(a)samsung.com>: > >>On 05/12/2015 05:35 PM, Dmitry Eremin-Solenikov wrote: > >>> > >>>2015-05-06 18:05 GMT+03:00 Jacek Anaszewski <j.anaszewski(a)samsung.com>: > >>>> > >>>>On 04/28/2015 01:55 AM, Dmitry Eremin-Solenikov wrote: > >>>>> > >>>>> > >>>>>Adapt locomo leds driver to new locomo core setup. > >>>>> > >>>>>Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> > >>>>>--- > >>>>> drivers/leds/Kconfig | 1 - > >>>>> drivers/leds/leds-locomo.c | 119 > >>>>>+++++++++++++++++++++++---------------------- > >>>>> 2 files changed, 61 insertions(+), 59 deletions(-) > >>>>> > >>>>>diff --git a/drivers/leds/Kconfig b/drivers/leds/Kconfig > >>>>>index 966b960..4b4650b 100644 > >>>>>--- a/drivers/leds/Kconfig > >>>>>+++ b/drivers/leds/Kconfig > >>>>>@@ -79,7 +79,6 @@ config LEDS_LM3642 > >>>>> config LEDS_LOCOMO > >>>>> tristate "LED Support for Locomo device" > >>>>> depends on LEDS_CLASS > >>>>>- depends on SHARP_LOCOMO > >>>> > >>>> > >>>> > >>>>Why do you remove this dependency? > >>> > >>> > >>>Because SHARP_LOCOMO is a Kconfig symbol for the old driver. New driver > >>>uses MFD_LOCOMO Kconfig entry. Also the driver now uses generic platform > >>>device and regmap interfaces, so there is no direct dependency on main > >>>LoCoMo driver. And the policy (IIRC) was not to have such dependencies. > >> > >> > >>Ack. Shouldn't you also need "select REGMAP_MMIO" ? > > > >No. Maybe I should add "select REGMAP" instead. > > REGMAP is enabled by default if REGMAP_MMIO is enabled. Having That is unfortunately not how select works: it does not automatically select parents for the selected symbol. Thanks. -- Dmitry

2 1

[alsa-devel] commit b1ef29725865 (ACPI _REV=2) causes sound regression on Dell XPS 13 [Was: Discussion around quirking the _REV behavior for the XPS 13 (2015) until 4.2]
by Dominik Brodowski 13 May '15

13 May '15

Hi, (CC'ing sound experts) indeed, commit b1ef29725865 causes a severe regression -- actually, it causes sound to be totally unusuable on the Dell XPS 13 (2015) on Debian jessie, while it works fine in 4.0 / with b1ef29725865 reverted. According to an off-list discussion, the sound breakage (and not just some jack detection issue) seems to be caused due to alsa-lib being too old. The matter is further complicated by the issue that the driver asks for some firmware blob intel/IntcPP01.bin which (at least) I cannot find anywhere. Under the no-regression rule, this means that either b1ef29725865 needs to be reverted or we need to find another solution to this matter, such as an override. And I think it is needed for longer than just for 4.1, as it will continue to be cause regressions on quite recent userspace. Best, Dominik PS/OT: Probably I'm preaching to the choir, but @Mario: it's a pity the XPS (or some versions of it) with pre-installed Ubuntu actually ships with a Wifi adapter which seems to be unsupported by upstream Linux... On Mon, May 04, 2015 at 06:26:25PM -0500, Mario Limonciello wrote: > > On 04/28/2015 06:21 PM, Mario Limonciello wrote: > >Hi, > > > >Due to b1ef2972586577e0ca9675254ee141f65a8824e5 landing recently as a result of some previous ACPI community discussion, the Dell XPS 13 (2015) no longer has fully functional audio. In case you aren't aware, the XPS 13 (2015) switches audio modes based upon a combination of what's detected for the _OSI string and _REV values. When "Windows 2013" is found with _REV=2 the audio codec will be configured for I2S audio mode. When "Windows 2013" with _REV=5 is found it will be configured for HDA audio mode. > > > >I2S audio is improving, but not as mature as HDA audio mode is at this point. Currently jack detection does not work properly in 4.1 with I2S audio codecs. I've been notified that the support for jack detection will be landing in 4.2. I'd like to discuss the possibility of including a quirk for the _REV behavior when the DMI information corresponding to the Dell XPS 13 (2015) is detected for 4.1. This will let it continue to run in HDA mode until I2S mode is mature. > > > >I'd also like to plan for the quirk to be dropped when jack detection has landed for 4.2 (or if it gets pushed out further when it lands for 4.3). > > > >Would you guys be open to that? > > > >Thanks, > Hello, > > I just wanted to follow up on this, I would like to come up with a solution before it's too late for 4.1 on the XPS 13 (2015). > > Thanks,

4 11

Re: [alsa-devel] [PATCH 2/4] ASoC: rockchip: add rockchip machine driver for max98090
by Mark Brown 13 May '15

13 May '15

On Wed, May 13, 2015 at 09:23:20PM +0800, zhengxing wrote: > On 2015年05月13日 03:26, Mark Brown wrote: > >This looks more like a normal and reasonable machine driver but then why > >have you created the generic rockchip machine driver? It seems like > >this should just be a regular machine driver like other platforms have, > >were it not for that this would be mostly fine apart from a couple of > >nitpicks below. > We just use rockchip_machine_driver to describe the supported codecs base on > origianl way, vendor machine driver(rockchip_max98090) will achieve > functions. Like I said in reply to the other message I just don't understand what you're saying here - take a look at the various Tegra machine drivers for example, they also have per-CODEC handling but they just use normal machine drivers to do it.

1 0

[alsa-devel] [PATCH] ASoC: rt5677: Add reset-gpio dts option
by Anatol Pomozov 13 May '15

13 May '15

It allows to configure codec's RESET pin gpio Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- Documentation/devicetree/bindings/sound/rt5677.txt | 2 ++ sound/soc/codecs/rt5677.c | 32 ++++++++++++++++++++-- sound/soc/codecs/rt5677.h | 1 + 3 files changed, 33 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index 740ff77..f070789 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -18,6 +18,7 @@ Required properties: Optional properties: - realtek,pow-ldo2-gpio : The GPIO that controls the CODEC's POW_LDO2 pin. +- realtek,reset-gpio : The GPIO that controls the CODEC's RESET pin. - realtek,in1-differential - realtek,in2-differential @@ -70,6 +71,7 @@ rt5677 { realtek,pow-ldo2-gpio = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; + realtek,reset-gpio = <&gpio TEGRA_GPIO(BB, 3) GPIO_ACTIVE_LOW>; realtek,in1-differential = "true"; realtek,gpio-config = /bits/ 8 <0 0 0 0 0 2>; /* pull up GPIO6 */ realtek,jd2-gpio = <3>; /* Enables Jack detection for GPIO6 */ diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 8bb7be8..235064a 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4765,6 +4765,8 @@ static int rt5677_remove(struct snd_soc_codec *codec) regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec); if (gpio_is_valid(rt5677->pow_ldo2)) gpio_set_value_cansleep(rt5677->pow_ldo2, 0); + if (gpio_is_valid(rt5677->reset_pin)) + gpio_set_value_cansleep(rt5677->reset_pin, 0); return 0; } @@ -4780,6 +4782,8 @@ static int rt5677_suspend(struct snd_soc_codec *codec) if (gpio_is_valid(rt5677->pow_ldo2)) gpio_set_value_cansleep(rt5677->pow_ldo2, 0); + if (gpio_is_valid(rt5677->reset_pin)) + gpio_set_value_cansleep(rt5677->reset_pin, 0); } return 0; @@ -4790,10 +4794,13 @@ static int rt5677_resume(struct snd_soc_codec *codec) struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); if (!rt5677->dsp_vad_en) { - if (gpio_is_valid(rt5677->pow_ldo2)) { + if (gpio_is_valid(rt5677->pow_ldo2)) gpio_set_value_cansleep(rt5677->pow_ldo2, 1); + if (gpio_is_valid(rt5677->reset_pin)) + gpio_set_value_cansleep(rt5677->reset_pin, 1); + if (gpio_is_valid(rt5677->pow_ldo2) || + gpio_is_valid(rt5677->reset_pin)) msleep(10); - } regcache_cache_only(rt5677->regmap, false); regcache_sync(rt5677->regmap); @@ -5031,6 +5038,8 @@ static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np) rt5677->pow_ldo2 = of_get_named_gpio(np, "realtek,pow-ldo2-gpio", 0); + rt5677->reset_pin = of_get_named_gpio(np, + "realtek,reset-gpio", 0); /* * POW_LDO2 is optional (it may be statically tied on the board). @@ -5041,6 +5050,9 @@ static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np) if (!gpio_is_valid(rt5677->pow_ldo2) && (rt5677->pow_ldo2 != -ENOENT)) return rt5677->pow_ldo2; + if (!gpio_is_valid(rt5677->reset_pin) && + (rt5677->reset_pin != -ENOENT)) + return rt5677->reset_pin; of_property_read_u8_array(np, "realtek,gpio-config", rt5677->pdata.gpio_config, RT5677_GPIO_NUM); @@ -5142,6 +5154,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, } } else { rt5677->pow_ldo2 = -EINVAL; + rt5677->reset_pin = -EINVAL; } if (gpio_is_valid(rt5677->pow_ldo2)) { @@ -5153,6 +5166,21 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, rt5677->pow_ldo2, ret); return ret; } + } + + if (gpio_is_valid(rt5677->reset_pin)) { + ret = devm_gpio_request_one(&i2c->dev, rt5677->reset_pin, + GPIOF_OUT_INIT_HIGH, + "RT5677 RESET"); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request RESET %d: %d\n", + rt5677->reset_pin, ret); + return ret; + } + } + + if (gpio_is_valid(rt5677->pow_ldo2) || + gpio_is_valid(rt5677->reset_pin)) { /* Wait a while until I2C bus becomes available. The datasheet * does not specify the exact we should wait but startup * sequence mentiones at least a few milliseconds. diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 62571d0..7eca38a 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1776,6 +1776,7 @@ struct rt5677_priv { int pll_in; int pll_out; int pow_ldo2; /* POW_LDO2 pin */ + int reset_pin; /* RESET pin */ enum rt5677_type type; #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; -- 2.2.0.rc0.207.ga3a616c

1 0

[alsa-devel] [PATCH v3] ASoC: wm8741: Add differential mono mode support
by Sergej Sawazki 13 May '15

13 May '15

The WM8741 DAC supports several differential output modes (stereo, stereo reversed, mono left, mono right). Add platform data and DT bindings to configure it. Signed-off-by: Sergej Sawazki <ce3a(a)gmx.de> --- Mark, Charles, thanks for your comments. This patch has been tested on a board with two codecs in mono mode (left and right channel). Changes since v2: - Add controls for mono codecs with a channel neutral name. Note: Machine drivers can attach prefixes like left, right or center to this controls using the name_prefix field in struct snd_soc_codec_conf (Suggested by Charles Keepax). Changes since v1: - Remove controls for mono codecs. Note: The machine driver is responsible for controls if the codec is configured in mono mode. Many thanks in advance for your time and effort, Sergej Documentation/devicetree/bindings/sound/wm8741.txt | 11 ++ sound/soc/codecs/wm8741.c | 129 ++++++++++++++++++--- sound/soc/codecs/wm8741.h | 10 ++ 3 files changed, 137 insertions(+), 13 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/wm8741.txt b/Documentation/devicetree/bindings/sound/wm8741.txt index 74bda58..a133154 100644 --- a/Documentation/devicetree/bindings/sound/wm8741.txt +++ b/Documentation/devicetree/bindings/sound/wm8741.txt @@ -10,9 +10,20 @@ Required properties: - reg : the I2C address of the device for I2C, the chip select number for SPI. +Optional properties: + + - diff-mode: Differential output mode configuration. Default value for field + DIFF in register R8 (MODE_CONTROL_2). If absent, the default is 0, shall be: + 0 = stereo + 1 = mono left + 2 = stereo reversed + 3 = mono right + Example: codec: wm8741@1a { compatible = "wlf,wm8741"; reg = <0x1a>; + + diff-mode = <3>; }; diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 9e71c76..c065ea1 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -41,6 +41,7 @@ static const char *wm8741_supply_names[WM8741_NUM_SUPPLIES] = { /* codec private data */ struct wm8741_priv { + struct wm8741_platform_data pdata; struct regmap *regmap; struct regulator_bulk_data supplies[WM8741_NUM_SUPPLIES]; unsigned int sysclk; @@ -87,13 +88,27 @@ static int wm8741_reset(struct snd_soc_codec *codec) static const DECLARE_TLV_DB_SCALE(dac_tlv_fine, -12700, 13, 0); static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 400, 0); -static const struct snd_kcontrol_new wm8741_snd_controls[] = { +static const struct snd_kcontrol_new wm8741_snd_controls_stereo[] = { SOC_DOUBLE_R_TLV("Fine Playback Volume", WM8741_DACLLSB_ATTENUATION, WM8741_DACRLSB_ATTENUATION, 1, 255, 1, dac_tlv_fine), SOC_DOUBLE_R_TLV("Playback Volume", WM8741_DACLMSB_ATTENUATION, WM8741_DACRMSB_ATTENUATION, 0, 511, 1, dac_tlv), }; +static const struct snd_kcontrol_new wm8741_snd_controls_mono_left[] = { +SOC_SINGLE_TLV("Fine Playback Volume", WM8741_DACLLSB_ATTENUATION, + 1, 255, 1, dac_tlv_fine), +SOC_SINGLE_TLV("Playback Volume", WM8741_DACLMSB_ATTENUATION, + 0, 511, 1, dac_tlv), +}; + +static const struct snd_kcontrol_new wm8741_snd_controls_mono_right[] = { +SOC_SINGLE_TLV("Fine Playback Volume", WM8741_DACRLSB_ATTENUATION, + 1, 255, 1, dac_tlv_fine), +SOC_SINGLE_TLV("Playback Volume", WM8741_DACRMSB_ATTENUATION, + 0, 511, 1, dac_tlv), +}; + static const struct snd_soc_dapm_widget wm8741_dapm_widgets[] = { SND_SOC_DAPM_DAC("DACL", "Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DACR", "Playback", SND_SOC_NOPM, 0, 0), @@ -398,7 +413,7 @@ static struct snd_soc_dai_driver wm8741_dai = { .name = "wm8741", .playback = { .stream_name = "Playback", - .channels_min = 2, /* Mono modes not yet supported */ + .channels_min = 2, .channels_max = 2, .rates = WM8741_RATES, .formats = WM8741_FORMATS, @@ -416,6 +431,65 @@ static int wm8741_resume(struct snd_soc_codec *codec) #define wm8741_resume NULL #endif +static int wm8741_configure(struct snd_soc_codec *codec) +{ + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + + /* Configure differential mode */ + switch (wm8741->pdata.diff_mode) { + case WM8741_DIFF_MODE_STEREO: + case WM8741_DIFF_MODE_STEREO_REVERSED: + case WM8741_DIFF_MODE_MONO_LEFT: + case WM8741_DIFF_MODE_MONO_RIGHT: + snd_soc_update_bits(codec, WM8741_MODE_CONTROL_2, + WM8741_DIFF_MASK, + wm8741->pdata.diff_mode << WM8741_DIFF_SHIFT); + break; + default: + return -EINVAL; + } + + /* Change some default settings - latch VU */ + snd_soc_update_bits(codec, WM8741_DACLLSB_ATTENUATION, + WM8741_UPDATELL, WM8741_UPDATELL); + snd_soc_update_bits(codec, WM8741_DACLMSB_ATTENUATION, + WM8741_UPDATELM, WM8741_UPDATELM); + snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, + WM8741_UPDATERL, WM8741_UPDATERL); + snd_soc_update_bits(codec, WM8741_DACRMSB_ATTENUATION, + WM8741_UPDATERM, WM8741_UPDATERM); + + return 0; +} + +static int wm8741_add_controls(struct snd_soc_codec *codec) +{ + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + + switch (wm8741->pdata.diff_mode) { + case WM8741_DIFF_MODE_STEREO: + case WM8741_DIFF_MODE_STEREO_REVERSED: + snd_soc_add_codec_controls(codec, + wm8741_snd_controls_stereo, + ARRAY_SIZE(wm8741_snd_controls_stereo)); + break; + case WM8741_DIFF_MODE_MONO_LEFT: + snd_soc_add_codec_controls(codec, + wm8741_snd_controls_mono_left, + ARRAY_SIZE(wm8741_snd_controls_mono_left)); + break; + case WM8741_DIFF_MODE_MONO_RIGHT: + snd_soc_add_codec_controls(codec, + wm8741_snd_controls_mono_right, + ARRAY_SIZE(wm8741_snd_controls_mono_right)); + break; + default: + return -EINVAL; + } + + return 0; +} + static int wm8741_probe(struct snd_soc_codec *codec) { struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); @@ -434,15 +508,17 @@ static int wm8741_probe(struct snd_soc_codec *codec) goto err_enable; } - /* Change some default settings - latch VU */ - snd_soc_update_bits(codec, WM8741_DACLLSB_ATTENUATION, - WM8741_UPDATELL, WM8741_UPDATELL); - snd_soc_update_bits(codec, WM8741_DACLMSB_ATTENUATION, - WM8741_UPDATELM, WM8741_UPDATELM); - snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, - WM8741_UPDATERL, WM8741_UPDATERL); - snd_soc_update_bits(codec, WM8741_DACRMSB_ATTENUATION, - WM8741_UPDATERM, WM8741_UPDATERM); + ret = wm8741_configure(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to change default settings\n"); + goto err_enable; + } + + ret = wm8741_add_controls(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to add controls\n"); + goto err_enable; + } dev_dbg(codec->dev, "Successful registration\n"); return ret; @@ -467,8 +543,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8741 = { .remove = wm8741_remove, .resume = wm8741_resume, - .controls = wm8741_snd_controls, - .num_controls = ARRAY_SIZE(wm8741_snd_controls), .dapm_widgets = wm8741_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8741_dapm_widgets), .dapm_routes = wm8741_dapm_routes, @@ -493,6 +567,23 @@ static const struct regmap_config wm8741_regmap = { .readable_reg = wm8741_readable, }; +static int wm8741_set_pdata(struct device *dev, struct wm8741_priv *wm8741) +{ + const struct wm8741_platform_data *pdata = dev_get_platdata(dev); + u32 diff_mode; + + if (dev->of_node) { + if (of_property_read_u32(dev->of_node, "diff-mode", &diff_mode) + >= 0) + wm8741->pdata.diff_mode = diff_mode; + } else { + if (pdata != NULL) + memcpy(&wm8741->pdata, pdata, sizeof(wm8741->pdata)); + } + + return 0; +} + #if IS_ENABLED(CONFIG_I2C) static int wm8741_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -522,6 +613,12 @@ static int wm8741_i2c_probe(struct i2c_client *i2c, return ret; } + wm8741_set_pdata(&i2c->dev, wm8741); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to set pdata: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, wm8741); ret = snd_soc_register_codec(&i2c->dev, @@ -582,6 +679,12 @@ static int wm8741_spi_probe(struct spi_device *spi) return ret; } + wm8741_set_pdata(&spi->dev, wm8741); + if (ret != 0) { + dev_err(&spi->dev, "Failed to set pdata: %d\n", ret); + return ret; + } + spi_set_drvdata(spi, wm8741); ret = snd_soc_register_codec(&spi->dev, diff --git a/sound/soc/codecs/wm8741.h b/sound/soc/codecs/wm8741.h index 56c1b1d..c8835f6 100644 --- a/sound/soc/codecs/wm8741.h +++ b/sound/soc/codecs/wm8741.h @@ -194,6 +194,12 @@ #define WM8741_DITHER_SHIFT 0 /* DITHER - [1:0] */ #define WM8741_DITHER_WIDTH 2 /* DITHER - [1:0] */ +/* DIFF field values */ +#define WM8741_DIFF_MODE_STEREO 0 /* stereo normal */ +#define WM8741_DIFF_MODE_STEREO_REVERSED 2 /* stereo reversed */ +#define WM8741_DIFF_MODE_MONO_LEFT 1 /* mono left */ +#define WM8741_DIFF_MODE_MONO_RIGHT 3 /* mono right */ + /* * R32 (0x20) - ADDITONAL_CONTROL_1 */ @@ -208,4 +214,8 @@ #define WM8741_SYSCLK 0 +struct wm8741_platform_data { + u32 diff_mode; /* Differential Output Mode */ +}; + #endif -- 1.9.1

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Re: [alsa-devel] [PATCH v2 02/17] leds: port locomo leds driver to new locomo core
by Dmitry Eremin-Solenikov 13 May '15

13 May '15

2015-05-13 13:31 GMT+03:00 Jacek Anaszewski <j.anaszewski(a)samsung.com>: > On 05/12/2015 05:35 PM, Dmitry Eremin-Solenikov wrote: >> >> 2015-05-06 18:05 GMT+03:00 Jacek Anaszewski <j.anaszewski(a)samsung.com>: >>> >>> On 04/28/2015 01:55 AM, Dmitry Eremin-Solenikov wrote: >>>> >>>> >>>> Adapt locomo leds driver to new locomo core setup. >>>> >>>> Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> >>>> --- >>>> drivers/leds/Kconfig | 1 - >>>> drivers/leds/leds-locomo.c | 119 >>>> +++++++++++++++++++++++---------------------- >>>> 2 files changed, 61 insertions(+), 59 deletions(-) >>>> >>>> diff --git a/drivers/leds/Kconfig b/drivers/leds/Kconfig >>>> index 966b960..4b4650b 100644 >>>> --- a/drivers/leds/Kconfig >>>> +++ b/drivers/leds/Kconfig >>>> @@ -79,7 +79,6 @@ config LEDS_LM3642 >>>> config LEDS_LOCOMO >>>> tristate "LED Support for Locomo device" >>>> depends on LEDS_CLASS >>>> - depends on SHARP_LOCOMO >>> >>> >>> >>> Why do you remove this dependency? >> >> >> Because SHARP_LOCOMO is a Kconfig symbol for the old driver. New driver >> uses MFD_LOCOMO Kconfig entry. Also the driver now uses generic platform >> device and regmap interfaces, so there is no direct dependency on main >> LoCoMo driver. And the policy (IIRC) was not to have such dependencies. > > > Ack. Shouldn't you also need "select REGMAP_MMIO" ? No. Maybe I should add "select REGMAP" instead. >>>> static void locomoled_brightness_set(struct led_classdev *led_cdev, >>>> - enum led_brightness value, int offset) >>>> + enum led_brightness value) >>>> { >>>> - struct locomo_dev *locomo_dev = >>>> LOCOMO_DEV(led_cdev->dev->parent); >>>> - unsigned long flags; >>>> + struct locomo_led *led = container_of(led_cdev, struct >>>> locomo_led, >>>> led); >>>> >>>> - local_irq_save(flags); >>>> - if (value) >>>> - locomo_writel(LOCOMO_LPT_TOFH, locomo_dev->mapbase + >>>> offset); >>>> - else >>>> - locomo_writel(LOCOMO_LPT_TOFL, locomo_dev->mapbase + >>>> offset); >>>> - local_irq_restore(flags); >>>> + regmap_write(led->regmap, led->reg, >>>> + value ? LOCOMO_LPT_TOFH : LOCOMO_LPT_TOFL); >>>> } >>> >>> >>> >>> Please use work queue for setting brightness. This is required for the >>> driver to be compatible with led triggers. You can refer to the >>> existing LED drivers on how to implement this. >> >> >> Hmm. Why? The regmap here uses MMIO access, so it is atomic operation >> and doesn't need to be wrapped into work queue, does it? > > > Triggers can call brightness_set op in the interrupt context, so it > cannot sleep. I see "map->lock(map->lock_arg)" in regmap_write, thus > I am inferring that it can sleep. > > I am not sure if regmap implements its 'lock' op when using MMIO. > > The best way to figure this out is turning "LED Timer Trigger" on > in the config and execute: > > echo "timer" > trigger It works without any "might sleep/sleeping in atomic context" warnings. Technically I'd agree with you. If I'm using regmaps, ideally I should not depend on the exact backend and put working with it to the work queue. However as this is a driver for quite old IP block, it is not used with regmap backends other than MMIO, I'd deduce, it's ok to do things in a more relaxed way and to call regmap_write even from atomic contexts. -- With best wishes Dmitry

1 0

Re: [alsa-devel] commit b1ef29725865 (ACPI _REV=2) causes sound regression on Dell XPS 13 [Was: Discussion around quirking the _REV behavior for the XPS 13 (2015) until 4.2]
by Rafael J. Wysocki 13 May '15

13 May '15

On Tuesday, May 12, 2015 10:43:33 AM Matthew Garrett wrote: > On Tue, May 12, 2015 at 10:29 AM, Mario Limonciello > <mario_limonciello(a)dell.com> wrote: > > Just to be clear, from the Dell side only the XPS 13 9343 has this _REV > > behavior. > > The Inspiron 7437 queries _REV and uses it to modify its EC behaviour, > and apparently breaks on Linux without that. > > > What's wrong with a DMI quirk until I2S is mature on the kernel side and > > userspace is new enough in distros? Is it too early in boot for DMI quirks? > > That would enforce that no other machines can use _REV value of 5 to detect > > Linux without breaking the 9343. > > DMI quirking is fine, but it has to be behind a config option. We're > going to have to carry the quirk for several years because we have to > support old userspace for an arbitrarily long time, and if there's no > config option then Linux will be stuck in HDA mode for that entire > time. One problem with the DMI-based approach is that _REV is constant in ACPICA, so to be able to return different values from it for different systems, we'll need to add some ACPICA-ish code handling that specifically for Linux and we're a bit too late in the cycle for that. So here's what I'm going to do. I'll revert commit b1ef29725865 for 4.1, but *only* for 4.1 (I've actually queued up a revert of it already) and I'll revert that revert during the 4.2 merge window. In the meantime, we'll develop the code to implement the DMI-based quirks for the Dells and apply it for 4.2. Are there any volunteers for doing the last part? -- I speak only for myself. Rafael J. Wysocki, Intel Open Source Technology Center.

2 2

[alsa-devel] [PATCH v3 0/8] Add support for Wolfson WM8998 and WM1814 codecs
by Richard Fitzgerald 13 May '15

13 May '15

This set of patches adds support for the Wolfson Microelectronics WM8998 and WM1814 codecs. Richard Fitzgerald (8): mfd: arizona: Split INx_MODE into two fields ASoC: arizona: add defines for single-input gain control mfd: arizona: Add support for WM8998 and WM1814 gpio: arizona: add support for WM8998 and WM1814 regulator: arizona: add support for WM8998 and WM1814 extcon: arizona: Add support for WM8998 and WM1814 ASoC: wm8998: Initial WM8998 codec driver Documentation: Add WM8998/WM1814 device tree bindings Documentation/devicetree/bindings/mfd/arizona.txt | 13 +- drivers/extcon/extcon-arizona.c | 33 +- drivers/gpio/gpio-arizona.c | 2 + drivers/mfd/Kconfig | 6 + drivers/mfd/Makefile | 3 + drivers/mfd/arizona-core.c | 105 ++- drivers/mfd/arizona-i2c.c | 8 + drivers/mfd/arizona-irq.c | 9 + drivers/mfd/arizona.h | 5 + drivers/mfd/wm8998-tables.c | 1592 +++++++++++++++++++++ drivers/regulator/arizona-ldo1.c | 2 + include/linux/mfd/arizona/core.h | 3 + include/linux/mfd/arizona/pdata.h | 7 +- include/linux/mfd/arizona/registers.h | 247 +++- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 7 + sound/soc/codecs/arizona.h | 7 +- sound/soc/codecs/wm8998.c | 1406 ++++++++++++++++++ sound/soc/codecs/wm8998.h | 23 + 20 files changed, 3454 insertions(+), 32 deletions(-) create mode 100644 drivers/mfd/wm8998-tables.c create mode 100644 sound/soc/codecs/wm8998.c create mode 100644 sound/soc/codecs/wm8998.h -- 1.7.2.5

3 18

[alsa-devel] [PATCH] ASoC: wm8903: Fix define for WM8903_VMID_RES_250K
by Axel Lin 13 May '15

13 May '15

VMID Control 0 BIT[2:1] is VMID Divider Enable and Select 00 = VMID disabled (for OFF mode) 01 = 2 x 50kΩ divider (for normal operation) 10 = 2 x 250kΩ divider (for low power standby) 11 = 2 x 5kΩ divider (for fast start-up) So WM8903_VMID_RES_250K should be 2 << 1, which is 4. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm8903.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h index db94931..0bb4a64 100644 --- a/sound/soc/codecs/wm8903.h +++ b/sound/soc/codecs/wm8903.h @@ -172,7 +172,7 @@ extern int wm8903_mic_detect(struct snd_soc_codec *codec, #define WM8903_VMID_BUF_ENA_WIDTH 1 /* VMID_BUF_ENA */ #define WM8903_VMID_RES_50K 2 -#define WM8903_VMID_RES_250K 3 +#define WM8903_VMID_RES_250K 4 #define WM8903_VMID_RES_5K 6 /* -- 2.1.0

2 1

[alsa-devel] [PATCH RFT] ASoC: wm8737: Fixup setting VMID Impedance control register
by Axel Lin 13 May '15

13 May '15

According to the datasheet: R10 (0Ah) VMID Impedance Control BIT 3:2 VMIDSEL DEFAULT 00 DESCRIPTION: VMID impedance selection control 00: 75kΩ output 01: 300kΩ output 10: 2.5kΩ output WM8737_VMIDSEL_MASK is 0xC (VMIDSEL - [3:2]), so it needs to left shift WM8737_VMIDSEL_SHIFT bits for setting these bits. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm8737.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 4a9407d..d38f1a9 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -483,7 +483,8 @@ static int wm8737_set_bias_level(struct snd_soc_codec *codec, /* Fast VMID ramp at 2*2.5k */ snd_soc_update_bits(codec, WM8737_MISC_BIAS_CONTROL, - WM8737_VMIDSEL_MASK, 0x4); + WM8737_VMIDSEL_MASK, + 2 << WM8737_VMIDSEL_SHIFT); /* Bring VMID up */ snd_soc_update_bits(codec, WM8737_POWER_MANAGEMENT, @@ -497,7 +498,8 @@ static int wm8737_set_bias_level(struct snd_soc_codec *codec, /* VMID at 2*300k */ snd_soc_update_bits(codec, WM8737_MISC_BIAS_CONTROL, - WM8737_VMIDSEL_MASK, 2); + WM8737_VMIDSEL_MASK, + 1 << WM8737_VMIDSEL_SHIFT); break; -- 2.1.0

2 1

[alsa-devel] ASoC suspend resume question
by Ben Zhang 13 May '15

13 May '15

Hi Mark, I have a question about the design of ASoC suspend/resume path. Who should call snd_soc_suspend() and snd_soc_resume()? They are part of the soc-audio driver's dev_pm_ops, but some ASoC platform drivers also calls them. For example, hsw_pcm_prepare() in sound/soc/intel/haswell/sst-haswell-pcm.c sst_soc_prepare() in sound/soc/intel/atom/sst-mfld-platform-pcm.c Thanks, Ben

2 1

[alsa-devel] [PATCH v2 1/3] ASoC: fsl_sai: add sai master mode support
by Zidan Wang 13 May '15

13 May '15

When sai works on master mode, set its bit clock and frame clock. SAI has 4 MCLK source, bus clock, MCLK1, MCLK2 and MCLK3. fsl_sai_set_bclk will select proper MCLK source, then calculate and set the bit clock divider. After fsl_sai_set_bclk, enable the selected mclk in hw_params(), and add hw_free() to disable the mclk. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_sai.c | 117 ++++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 9 +++- 2 files changed, 121 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ee2671b..1ccc10d1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1,7 +1,7 @@ /* * Freescale ALSA SoC Digital Audio Interface (SAI) driver. * - * Copyright 2012-2013 Freescale Semiconductor, Inc. + * Copyright 2012-2015 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 as published by the @@ -251,12 +251,14 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, val_cr4 |= FSL_SAI_CR4_FSD_MSTR; break; case SND_SOC_DAIFMT_CBM_CFM: + sai->is_slave_mode = true; break; case SND_SOC_DAIFMT_CBS_CFM: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; break; case SND_SOC_DAIFMT_CBM_CFS: val_cr4 |= FSL_SAI_CR4_FSD_MSTR; + sai->is_slave_mode = true; break; default: return -EINVAL; @@ -288,6 +290,79 @@ static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) return ret; } +static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(dai); + unsigned long clk_rate; + u32 savediv = 0, ratio, savesub = freq; + u32 id; + int ret = 0; + + /* Don't apply to slave mode */ + if (sai->is_slave_mode) + return 0; + + for (id = 0; id < FSL_SAI_MCLK_MAX; id++) { + clk_rate = clk_get_rate(sai->mclk_clk[id]); + if (!clk_rate) + continue; + + ratio = clk_rate / freq; + + ret = clk_rate - ratio * freq; + + /* + * Drop the source that can not be + * divided into the required rate. + */ + if (ret != 0 && clk_rate / ret < 1000) + continue; + + dev_dbg(dai->dev, + "ratio %d for freq %dHz based on clock %ldHz\n", + ratio, freq, clk_rate); + + if (ratio % 2 == 0 && ratio >= 2 && ratio <= 512) + ratio /= 2; + else + continue; + + if (ret < savesub) { + savediv = ratio; + sai->mclk_id[tx] = id; + savesub = ret; + } + + if (ret == 0) + break; + } + + if (savediv == 0) { + dev_err(dai->dev, "failed to derive required %cx rate: %d\n", + tx ? 'T' : 'R', freq); + return -EINVAL; + } + + if ((tx && sai->synchronous[TX]) || (!tx && !sai->synchronous[RX])) { + regmap_update_bits(sai->regmap, FSL_SAI_RCR2, + FSL_SAI_CR2_MSEL_MASK, + FSL_SAI_CR2_MSEL(sai->mclk_id[tx])); + regmap_update_bits(sai->regmap, FSL_SAI_RCR2, + FSL_SAI_CR2_DIV_MASK, savediv - 1); + } else { + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, + FSL_SAI_CR2_MSEL_MASK, + FSL_SAI_CR2_MSEL(sai->mclk_id[tx])); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, + FSL_SAI_CR2_DIV_MASK, savediv - 1); + } + + dev_dbg(dai->dev, "best fit: clock id=%d, div=%d, deviation =%d\n", + sai->mclk_id[tx], savediv, savesub); + + return 0; +} + static int fsl_sai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -297,6 +372,24 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, unsigned int channels = params_channels(params); u32 word_width = snd_pcm_format_width(params_format(params)); u32 val_cr4 = 0, val_cr5 = 0; + int ret; + + if (!sai->is_slave_mode) { + ret = fsl_sai_set_bclk(cpu_dai, tx, + 2 * word_width * params_rate(params)); + if (ret) + return ret; + + /* Do not enable the clock if it is already enabled */ + if (!(sai->mclk_streams & BIT(substream->stream))) { + ret = clk_prepare_enable(sai->mclk_clk[sai->mclk_id[tx]]); + if (ret) + return ret; + + sai->mclk_streams |= BIT(substream->stream); + } + + } if (!sai->is_dsp_mode) val_cr4 |= FSL_SAI_CR4_SYWD(word_width); @@ -322,6 +415,22 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, return 0; } +static int fsl_sai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + if (!sai->is_slave_mode && + sai->mclk_streams & BIT(substream->stream)) { + clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[tx]]); + sai->mclk_streams &= ~BIT(substream->stream); + } + + return 0; +} + + static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { @@ -428,6 +537,7 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { .set_sysclk = fsl_sai_set_dai_sysclk, .set_fmt = fsl_sai_set_dai_fmt, .hw_params = fsl_sai_hw_params, + .hw_free = fsl_sai_hw_free, .trigger = fsl_sai_trigger, .startup = fsl_sai_startup, .shutdown = fsl_sai_shutdown, @@ -600,8 +710,9 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->bus_clk = NULL; } - for (i = 0; i < FSL_SAI_MCLK_MAX; i++) { - sprintf(tmp, "mclk%d", i + 1); + sai->mclk_clk[0] = sai->bus_clk; + for (i = 1; i < FSL_SAI_MCLK_MAX; i++) { + sprintf(tmp, "mclk%d", i); sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 3466720..0662809 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -72,13 +72,15 @@ /* SAI Transmit and Recieve Configuration 2 Register */ #define FSL_SAI_CR2_SYNC BIT(30) -#define FSL_SAI_CR2_MSEL_MASK (0xff << 26) +#define FSL_SAI_CR2_MSEL_MASK (0x3 << 26) #define FSL_SAI_CR2_MSEL_BUS 0 #define FSL_SAI_CR2_MSEL_MCLK1 BIT(26) #define FSL_SAI_CR2_MSEL_MCLK2 BIT(27) #define FSL_SAI_CR2_MSEL_MCLK3 (BIT(26) | BIT(27)) +#define FSL_SAI_CR2_MSEL(ID) ((ID) << 26) #define FSL_SAI_CR2_BCP BIT(25) #define FSL_SAI_CR2_BCD_MSTR BIT(24) +#define FSL_SAI_CR2_DIV_MASK 0xff /* SAI Transmit and Recieve Configuration 3 Register */ #define FSL_SAI_CR3_TRCE BIT(16) @@ -120,7 +122,7 @@ #define FSL_SAI_CLK_MAST2 2 #define FSL_SAI_CLK_MAST3 3 -#define FSL_SAI_MCLK_MAX 3 +#define FSL_SAI_MCLK_MAX 4 /* SAI data transfer numbers per DMA request */ #define FSL_SAI_MAXBURST_TX 6 @@ -132,11 +134,14 @@ struct fsl_sai { struct clk *bus_clk; struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; + bool is_slave_mode; bool is_lsb_first; bool is_dsp_mode; bool sai_on_imx; bool synchronous[2]; + unsigned int mclk_id[2]; + unsigned int mclk_streams; struct snd_dmaengine_dai_dma_data dma_params_rx; struct snd_dmaengine_dai_dma_data dma_params_tx; }; -- 1.9.1

2 3

[alsa-devel] [PATCH v2 00/17] new locomo driver set
by Dmitry Eremin-Solenikov 13 May '15

13 May '15

Sharp Zaurus SL-5500 and SL-5600 use special companion Gate Array. With this patchset I tried to modernise and restructure the LoCoMo drivers. * Use platform bus and mfd core code * Use GPIO API instead of custom locomo GPIO code * Use irqdomains to manage IRQs * Split mixed locomolcd driver to lcd and backlight parts * Implement IRQ handling in GPIO driver * Implement SPI driver used as a host for mmc_spi device to control SD cards V1 was submitted about a half of a year ago. Major changes: * Use gpiod_* in most of places * Use regmap for accessing registers instead of manual RMW cycles * Split dac driver from core module to i2c bus and separate IIO driver * Replaced mdelay() calls with usleep_range or msleep where it was suitable * Rewrote SPI driver to use core transfer_one() interface

7 38

Re: [alsa-devel] [PATCH 3/4] ASoC: rockchip: add rockchip machine driver for rt5650/rt5645
by Mark Brown 12 May '15

12 May '15

On Tue, May 12, 2015 at 05:26:36PM +0800, Xing Zheng wrote: > From: zhengxing <zhengxing(a)rock-chips.com> > > The driver is used for rockchip board using a rt5650/rt5645. > Test on RK3288 with rt5650. Similar comments to the Maxim driver plus... > +/* Jack detect via rt5645 driver. */ > +extern int rt5645_set_jack_detect(struct snd_soc_codec *codec, > + struct snd_soc_jack *hp_jack, > + struct snd_soc_jack *mic_jack, > + struct snd_soc_jack *btn_jack); > + > +/* For compatible. */ > +int __weak rt5645_set_jack_detect(struct snd_soc_codec *codec, > + struct snd_soc_jack *hp_jack, > + struct snd_soc_jack *mic_jack, > + struct snd_soc_jack *btn_jack) > +{ > + dev_warn(codec->dev, "Jack detect dummy, have no implement it?\n"); > + > + return 0; > +} Don't do this - if the functionality isn't there in the CODEC driver get it implemented before you use it. Make sure you're using the API the CODEC actually ends up exporting. > + /* Enable Headset and 4 Buttons Jack detection */ > + ret = snd_soc_card_jack_new(card, "Headphone Jack", > + SND_JACK_HEADPHONE | SND_JACK_MICROPHONE, > + ret = snd_soc_card_jack_new(card, "Mic Jack", > + > + ret = snd_soc_card_jack_new(card, "Button Jack", No, don't do this - as discussed in another Chromebook review (I'm guessing this is a Chromebook) the jacks should represent the physical jacks in the system. It's possible you have separate headphone and microphone jacks but I really expect microphone and button to be on the same socket.

1 0

Re: [alsa-devel] [PATCH 2/4] ASoC: rockchip: add rockchip machine driver for max98090
by Mark Brown 12 May '15

12 May '15

On Tue, May 12, 2015 at 05:26:35PM +0800, Xing Zheng wrote: > sound/soc/rockchip/Kconfig | 10 ++ > sound/soc/rockchip/Makefile | 2 + > sound/soc/rockchip/rockchip_machine_driver.c | 6 + > sound/soc/rockchip/rockchip_machine_driver.h | 6 + > sound/soc/rockchip/rockchip_max98090.c | 185 ++++++++++++++++++++++++++ This looks more like a normal and reasonable machine driver but then why have you created the generic rockchip machine driver? It seems like this should just be a regular machine driver like other platforms have, were it not for that this would be mostly fine apart from a couple of nitpicks below. > +config SND_SOC_ROCKCHIP_MAX98090 > + tristate "ASoC support for Rockchip boards using a MAX98090 codec" > + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB > + select SND_SOC_ROCKCHIP_I2S > + select SND_SOC_MAX98090 > + select SND_SOC_TS3A227E This looks like it's a driver specific to Chromebooks (possibly even specific Chromeooks) and so should be named as such. > + card->dapm.idle_bias_off = true; Just set this when declaring the card, don't do it at runtime. > + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); > + snd_soc_dapm_enable_pin(dapm, "Headphone"); > + snd_soc_dapm_enable_pin(dapm, "Speaker"); > + snd_soc_dapm_enable_pin(dapm, "Int Mic"); No need to do this, all pins are enabled by default. > + snd_soc_dapm_sync(dapm); This has no effect during initialization.

1 0

Re: [alsa-devel] [PATCH 1/4] ASoC: rockchip: add rockchip machine driver
by Mark Brown 12 May '15

12 May '15

On Tue, May 12, 2015 at 05:26:34PM +0800, Xing Zheng wrote: > From: zhengxing <zhengxing(a)rock-chips.com> > > In most cases, we maybe use simple-card as generic machine driver > on next kernel, but there are some issue that jack detection and > widgets extandable on simple-card. > > Own machine drvier can select one valid codec from supported codecs, > and add needful function from dts on rockchip platform. Is it not possible to extend simple card to handle your use cases? Given the very generic naming and the fact that things like jack detection and so on should depend on the CODEC and board usually rather than on the SoC it doesn't sound like this is Rockchip specific. This also looks like you're reimplementing some device model enumeration stuff which probably shouldn't be happening but let's understand the problem you're trying to solve here before going too far into the code.

1 0

[alsa-devel] [PATCH 1/4] ASoC: wm8960: Let wm8960 driver configure its bit clock and frame clock
by Zidan Wang 12 May '15

12 May '15

wm8960 codec driver missing configure its bit clock and frame clock for codec master mode, so add support for it. It will calculate a appropriate frequency dividing ratio according to the system clock, bit clock and frame clock, then set the corresponding registers. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 101 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 101 insertions(+) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 6fa832b..b072501 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -127,6 +127,8 @@ struct wm8960_priv { struct snd_soc_dapm_widget *out3; bool deemph; int playback_fs; + int bclk; + int sysclk; struct wm8960_data pdata; }; @@ -563,6 +565,72 @@ static struct { { 8000, 5 }, }; +/* Multiply 256 for internal 256 div */ +static const int dac_divs[] = { 256, 384, 512, 768, 1024, 1408, 1536 }; + +/* Multiply 10 to eliminate decimials */ +static const int bclk_divs[] = { + 10, 15, 20, 30, 40, 55, 60, 80, 110, + 120, 160, 220, 240, 320, 320, 320 +}; + +static void wm8960_configure_clocking(struct snd_soc_codec *codec, + bool tx, int lrclk) +{ + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + u16 iface1 = snd_soc_read(codec, WM8960_IFACE1); + u16 iface2 = snd_soc_read(codec, WM8960_IFACE2); + u32 sysclk; + int i, j; + + if (!(iface1 & (1<<6))) { + dev_dbg(codec->dev, + "Codec is slave mode, no need to configure clock\n"); + return; + } + + if (!wm8960->sysclk) { + dev_dbg(codec->dev, "No SYSCLK configured\n"); + return; + } + + if (!wm8960->bclk || !lrclk) { + dev_dbg(codec->dev, "No audio clocks configured\n"); + return; + } + + for (i = 0; i < ARRAY_SIZE(dac_divs); ++i) { + if (wm8960->sysclk == lrclk * dac_divs[i]) { + for (j = 0; j < ARRAY_SIZE(bclk_divs); ++j) { + sysclk = wm8960->bclk * bclk_divs[j] / 10; + if (wm8960->sysclk == sysclk) + break; + } + if(j != ARRAY_SIZE(bclk_divs)) + break; + } + } + + if (i == ARRAY_SIZE(dac_divs)) { + dev_err(codec->dev, "Unsupported sysclk %d\n", wm8960->sysclk); + return; + } + + /* + * configure frame clock. If ADCLRC configure as GPIO pin, DACLRC + * pin is used as a frame clock for ADCs and DACs. + */ + if (iface2 & (1<<6)) + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); + else if (tx) + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); + else if (!tx) + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, i << 6); + + /* configure bit clock */ + snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, j); +} + static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -570,8 +638,13 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int i; + wm8960->bclk = snd_soc_params_to_bclk(params); + if (params_channels(params) == 1) + wm8960->bclk *= 2; + /* bit size */ switch (params_width(params)) { case 16: @@ -602,6 +675,9 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, /* set iface */ snd_soc_write(codec, WM8960_IFACE1, iface); + + wm8960_configure_clocking(codec, tx, params_rate(params)); + return 0; } @@ -946,6 +1022,30 @@ static int wm8960_set_bias_level(struct snd_soc_codec *codec, return wm8960->set_bias_level(codec, level); } +static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case WM8960_SYSCLK_MCLK: + snd_soc_update_bits(codec, WM8960_CLOCK1, + 0x1, WM8960_SYSCLK_MCLK); + break; + case WM8960_SYSCLK_PLL: + snd_soc_update_bits(codec, WM8960_CLOCK1, + 0x1, WM8960_SYSCLK_PLL); + break; + default: + return -EINVAL; + } + + wm8960->sysclk = freq; + + return 0; +} + #define WM8960_RATES SNDRV_PCM_RATE_8000_48000 #define WM8960_FORMATS \ @@ -958,6 +1058,7 @@ static const struct snd_soc_dai_ops wm8960_dai_ops = { .set_fmt = wm8960_set_dai_fmt, .set_clkdiv = wm8960_set_dai_clkdiv, .set_pll = wm8960_set_dai_pll, + .set_sysclk = wm8960_set_dai_sysclk, }; static struct snd_soc_dai_driver wm8960_dai = { -- 1.9.1

3 7

Re: [alsa-devel] [PATCH v2] ASoC: bt-sco: Add devicetree support for bt-sco codec
by Mark Brown 12 May '15

12 May '15

On Fri, May 08, 2015 at 09:02:34PM +0200, Marek Belisko wrote: > Add devicetree support for bluetooth SCO link codec. Applied with a bit of effort, you seem to have sent this against some non-mainline tree - please check that in future.

2 1

Re: [alsa-devel] [PATCH 1/4] ASoC: rt5645: improve headphone depop function
by Mark Brown 12 May '15

12 May '15

On Tue, May 12, 2015 at 08:43:02PM +0800, John Lin wrote: > We add a calibration function and call it at the beginning of i2c_probe. > The calibration value will be kept until codec is shutdown. We will reset > the codec after the calibration is finished. So, we set cache_bypass in > the calibration function. The benefit is we can shorter the delay time > in headphone depop. Applied all, thanks.

1 0

[alsa-devel] [PATCH 1/5] ASoC: wm8996: Pass the IRQF_ONESHOT flag
by Fabio Estevam 12 May '15

12 May '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Since commit 1c6c69525b40eb76de8adf039409722015927dc3 ("genirq: Reject bogus threaded irq requests") threaded IRQs without a primary handler need to be requested with IRQF_ONESHOT, otherwise the request will fail. So pass the IRQF_ONESHOT flag in this case. The semantic patch that makes this change is available in scripts/coccinelle/misc/irqf_oneshot.cocci. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm8996.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 3dce507..232f234 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2644,10 +2644,12 @@ static int wm8996_probe(struct snd_soc_codec *codec) if (irq_flags & (IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING)) ret = request_threaded_irq(i2c->irq, NULL, wm8996_edge_irq, - irq_flags, "wm8996", codec); + irq_flags | IRQF_ONESHOT, + "wm8996", codec); else ret = request_threaded_irq(i2c->irq, NULL, wm8996_irq, - irq_flags, "wm8996", codec); + irq_flags | IRQF_ONESHOT, + "wm8996", codec); if (ret == 0) { /* Unmask the interrupt */ -- 1.9.1

2 6

[alsa-devel] [PATCH 0/3] Add sai master mode, tdm slot operation and add some sample rate support
by Zidan Wang 12 May '15

12 May '15

Add SAI master mode support. Add tdm slots operation for SAI master mode. Add 12kHz, 24kHz, 176.4kHz and 192kHz sample rate support. Zidan Wang (3): SoC: fsl_sai: add sai master mode support ASoC: fsl_sai: Add tdm slots operation for SAI master mode ASoC: fsl_sai: add 12kHz, 24kHz, 176.4kHz and 192kHz sample rate support sound/soc/fsl/fsl_sai.c | 165 +++++++++++++++++++++++++++++++++++++++++++++--- sound/soc/fsl/fsl_sai.h | 12 +++- 2 files changed, 165 insertions(+), 12 deletions(-) -- 1.9.1

2 6

[alsa-devel] [PATCH] ASoC: dapm: Modify widget stream name according to prefix
by Koro Chen 12 May '15

12 May '15

When there is prefix specified, currently we will add this prefix in widget->name, but not in widget->sname. it causes failure at snd_soc_dapm_link_dai_widgets: if (!w->sname || !strstr(w->sname, dai_w->name)) because dai_w->name has prefix added, but w->sname does not. We should also add prefix for stream name Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/soc-dapm.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index defe0f0..158204d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3100,11 +3100,16 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, } prefix = soc_dapm_prefix(dapm); - if (prefix) + if (prefix) { w->name = kasprintf(GFP_KERNEL, "%s %s", prefix, widget->name); - else + if (widget->sname) + w->sname = kasprintf(GFP_KERNEL, "%s %s", prefix, + widget->sname); + } else { w->name = kasprintf(GFP_KERNEL, "%s", widget->name); - + if (widget->sname) + w->sname = kasprintf(GFP_KERNEL, "%s", widget->sname); + } if (w->name == NULL) { kfree(w); return NULL; -- 1.8.1.1.dirty

2 1

[alsa-devel] [PATCH] ASoC: codecs-ac97: Remove rate constraints
by Maciej S. Szmigiero 12 May '15

12 May '15

Remove rate constraints from generic ASoC AC'97 CODEC. Supported rates should be detected and constrained anyway by AC'97 generic code - was tested with VT1613 CODEC and iMX6 SSI controller. This way this driver can be used for platforms which don't need specialized AC'97 CODEC drivers while at the same avoiding code duplication from implementing equivalent functionality in a controller driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -44,10 +44,6 @@ static int ac97_prepare(struct snd_pcm_substream *substream, return snd_ac97_set_rate(ac97, reg, substream->runtime->rate); } -#define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000) - static const struct snd_soc_dai_ops ac97_dai_ops = { .prepare = ac97_prepare, }; @@ -58,13 +54,13 @@ static struct snd_soc_dai_driver ac97_dai = { .stream_name = "AC97 Playback", .channels_min = 1, .channels_max = 2, - .rates = STD_AC97_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SND_SOC_STD_AC97_FMTS,}, .capture = { .stream_name = "AC97 Capture", .channels_min = 1, .channels_max = 2, - .rates = STD_AC97_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &ac97_dai_ops, };

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: sta32x: Use devm_gpiod_get_optional at appropriate place
by Axel Lin 12 May '15

12 May '15

devm_gpiod_get_optional() is equivalent to devm_gpiod_get(), except that when no GPIO was assigned to the requested function it will return NULL. This is convenient for drivers that need to handle optional GPIOs. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- v2: Fixup subject line sound/soc/codecs/sta32x.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 033b7d9..847fb20 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -1095,16 +1095,10 @@ static int sta32x_i2c_probe(struct i2c_client *i2c, #endif /* GPIOs */ - sta32x->gpiod_nreset = devm_gpiod_get(dev, "reset"); - if (IS_ERR(sta32x->gpiod_nreset)) { - ret = PTR_ERR(sta32x->gpiod_nreset); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta32x->gpiod_nreset = NULL; - } else { - gpiod_direction_output(sta32x->gpiod_nreset, 0); - } + sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(sta32x->gpiod_nreset)) + return PTR_ERR(sta32x->gpiod_nreset); /* regulators */ for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) -- 2.1.0

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Re: [alsa-devel] commit b1ef29725865 (ACPI _REV=2) causes sound regression on Dell XPS 13 [Was: Discussion around quirking the _REV behavior for the XPS 13 (2015) until 4.2]
by Mario Limonciello 12 May '15

12 May '15

On 05/12/2015 12:43 PM, Matthew Garrett wrote: > On Tue, May 12, 2015 at 10:29 AM, Mario Limonciello > <mario_limonciello(a)dell.com> wrote: >> Just to be clear, from the Dell side only the XPS 13 9343 has this _REV >> behavior. > The Inspiron 7437 queries _REV and uses it to modify its EC behaviour, > and apparently breaks on Linux without that. Thanks for bringing this to my attention. I'll check with my BIOS architecture team on this. XPS 13 9343 was supposed to be the only system doing this. >> What's wrong with a DMI quirk until I2S is mature on the kernel side and >> userspace is new enough in distros? Is it too early in boot for DMI quirks? >> That would enforce that no other machines can use _REV value of 5 to detect >> Linux without breaking the 9343. > DMI quirking is fine, but it has to be behind a config option. We're > going to have to carry the quirk for several years because we have to > support old userspace for an arbitrarily long time, and if there's no > config option then Linux will be stuck in HDA mode for that entire > time. > OK, that makes more sense to me. Thanks.

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Re: [alsa-devel] commit b1ef29725865 (ACPI _REV=2) causes sound regression on Dell XPS 13 [Was: Discussion around quirking the _REV behavior for the XPS 13 (2015) until 4.2]
by Mario Limonciello 12 May '15

12 May '15

On 05/12/2015 12:21 PM, Matthew Garrett wrote: > On Tue, May 12, 2015 at 9:18 AM, Mark Brown <broonie(a)kernel.org> wrote: >> You need PulseAudio 6.0 but otherwise yes, it works for me. > Since there's no way to get this working without using new userspace, > we're basically stuck with having to support _REV 5. And since we're > supposed to support old userspace basically forever, this isn't going > to be something that we can just remove in a few months. But since the > hardware is intended to be used in I2S mode, we don't want this quirk > to be there forever. So we're basically left with adding > CONFIG_ACPI_WORKAROUND_BROKEN_DELLS which enables the quirk, default > it to Y and then let distributions set it to N when they've got > sufficiently new userspace. Unless someone has a better plan? Just to be clear, from the Dell side only the XPS 13 9343 has this _REV behavior. What's wrong with a DMI quirk until I2S is mature on the kernel side and userspace is new enough in distros? Is it too early in boot for DMI quirks? That would enforce that no other machines can use _REV value of 5 to detect Linux without breaking the 9343.

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Re: [alsa-devel] commit b1ef29725865 (ACPI _REV=2) causes sound regression on Dell XPS 13 [Was: Discussion around quirking the _REV behavior for the XPS 13 (2015) until 4.2]
by Mark Brown 12 May '15

12 May '15

On Tue, May 12, 2015 at 10:21:41AM -0700, Matthew Garrett wrote: > Since there's no way to get this working without using new userspace, > we're basically stuck with having to support _REV 5. And since we're > supposed to support old userspace basically forever, this isn't going > to be something that we can just remove in a few months. But since the > hardware is intended to be used in I2S mode, we don't want this quirk > to be there forever. So we're basically left with adding > CONFIG_ACPI_WORKAROUND_BROKEN_DELLS which enables the quirk, default > it to Y and then let distributions set it to N when they've got > sufficiently new userspace. Unless someone has a better plan? That makes sense to me given that the firmware doesn't seem to do anything problematic in Linux mode - hopefully we won't have too many more of these systems to contend with.

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Re: [alsa-devel] [PATCH v2 02/17] leds: port locomo leds driver to new locomo core
by Dmitry Eremin-Solenikov 12 May '15

12 May '15

2015-05-06 18:05 GMT+03:00 Jacek Anaszewski <j.anaszewski(a)samsung.com>: > On 04/28/2015 01:55 AM, Dmitry Eremin-Solenikov wrote: >> >> Adapt locomo leds driver to new locomo core setup. >> >> Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> >> --- >> drivers/leds/Kconfig | 1 - >> drivers/leds/leds-locomo.c | 119 >> +++++++++++++++++++++++---------------------- >> 2 files changed, 61 insertions(+), 59 deletions(-) >> >> diff --git a/drivers/leds/Kconfig b/drivers/leds/Kconfig >> index 966b960..4b4650b 100644 >> --- a/drivers/leds/Kconfig >> +++ b/drivers/leds/Kconfig >> @@ -79,7 +79,6 @@ config LEDS_LM3642 >> config LEDS_LOCOMO >> tristate "LED Support for Locomo device" >> depends on LEDS_CLASS >> - depends on SHARP_LOCOMO > > > Why do you remove this dependency? Because SHARP_LOCOMO is a Kconfig symbol for the old driver. New driver uses MFD_LOCOMO Kconfig entry. Also the driver now uses generic platform device and regmap interfaces, so there is no direct dependency on main LoCoMo driver. And the policy (IIRC) was not to have such dependencies. > >> help >> This option enables support for the LEDs on Sharp Locomo. >> Zaurus models SL-5500 and SL-5600. >> diff --git a/drivers/leds/leds-locomo.c b/drivers/leds/leds-locomo.c >> index 80ba048..7fde812 100644 >> --- a/drivers/leds/leds-locomo.c >> +++ b/drivers/leds/leds-locomo.c >> @@ -9,89 +9,92 @@ >> */ >> >> #include <linux/kernel.h> >> -#include <linux/init.h> >> -#include <linux/module.h> >> -#include <linux/device.h> >> #include <linux/leds.h> >> - >> -#include <mach/hardware.h> >> -#include <asm/hardware/locomo.h> >> +#include <linux/module.h> >> +#include <linux/platform_device.h> >> +#include <linux/regmap.h> >> +#include <linux/mfd/locomo.h> > > > Please keep alphabetical order. Ack > >> + >> +struct locomo_led { >> + struct led_classdev led; >> + struct regmap *regmap; >> + unsigned int reg; >> +}; >> >> static void locomoled_brightness_set(struct led_classdev *led_cdev, >> - enum led_brightness value, int offset) >> + enum led_brightness value) >> { >> - struct locomo_dev *locomo_dev = LOCOMO_DEV(led_cdev->dev->parent); >> - unsigned long flags; >> + struct locomo_led *led = container_of(led_cdev, struct locomo_led, >> led); >> >> - local_irq_save(flags); >> - if (value) >> - locomo_writel(LOCOMO_LPT_TOFH, locomo_dev->mapbase + >> offset); >> - else >> - locomo_writel(LOCOMO_LPT_TOFL, locomo_dev->mapbase + >> offset); >> - local_irq_restore(flags); >> + regmap_write(led->regmap, led->reg, >> + value ? LOCOMO_LPT_TOFH : LOCOMO_LPT_TOFL); >> } > > > Please use work queue for setting brightness. This is required for the > driver to be compatible with led triggers. You can refer to the > existing LED drivers on how to implement this. Hmm. Why? The regmap here uses MMIO access, so it is atomic operation and doesn't need to be wrapped into work queue, does it? [skipped] -- With best wishes Dmitry

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[alsa-devel] ALC298 headset mode support
by Kailang 12 May '15

12 May '15

Hi Takashi, I finished the alc298 headset mode supported. But it only tested via our develop system. BR, Kailang

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[alsa-devel] [PATCH] Revert "ASoC: rsnd: make sure it uses lock when it calls rsnd_dai_call"
by Kuninori Morimoto 12 May '15

12 May '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This reverts commit 'e9c390df671f ("ASoC: rsnd: make sure it uses lock when it calls rsnd_dai_call)' The additional locks make 1") lock issue when boot 2) lock issue when unbind/rmmod. And there is no problem without these locks. This patch revert it. Reported-by: Geert Uytterhoeven <geert+renesas(a)glider.be> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 27 +++++---------------------- 1 file changed, 5 insertions(+), 22 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 99eb109..405cacd 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -731,15 +731,10 @@ static int rsnd_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_dai *dai = rsnd_substream_to_dai(substream); struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); - struct rsnd_priv *priv = rsnd_dai_to_priv(dai); struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); - unsigned long flags; int ret; - rsnd_lock(priv, flags); ret = rsnd_dai_call(hw_params, io, substream, hw_params); - rsnd_unlock(priv, flags); - if (ret) return ret; @@ -926,16 +921,14 @@ int rsnd_kctrl_new_e(struct rsnd_mod *mod, static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *dai = rtd->cpu_dai; - struct rsnd_priv *priv = rsnd_dai_to_priv(dai); struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); - unsigned long flags; - int ret = 0; + int ret; - rsnd_lock(priv, flags); - ret |= rsnd_dai_call(pcm_new, &rdai->playback, rtd); - ret |= rsnd_dai_call(pcm_new, &rdai->capture, rtd); - rsnd_unlock(priv, flags); + ret = rsnd_dai_call(pcm_new, &rdai->playback, rtd); + if (ret) + return ret; + ret = rsnd_dai_call(pcm_new, &rdai->capture, rtd); if (ret) return ret; @@ -958,11 +951,8 @@ static const struct snd_soc_component_driver rsnd_soc_component = { static int rsnd_rdai_continuance_probe(struct rsnd_priv *priv, struct rsnd_dai_stream *io) { - unsigned long flags; int ret; - rsnd_lock(priv, flags); - ret = rsnd_dai_call(probe, io, priv); if (ret == -EAGAIN) { /* @@ -995,7 +985,6 @@ static int rsnd_rdai_continuance_probe(struct rsnd_priv *priv, */ ret = rsnd_dai_call(probe, io, priv); } - rsnd_unlock(priv, flags); return ret; } @@ -1011,7 +1000,6 @@ static int rsnd_probe(struct platform_device *pdev) struct rsnd_dai *rdai; const struct of_device_id *of_id = of_match_device(rsnd_of_match, dev); const struct rsnd_of_data *of_data; - unsigned long flags; int (*probe_func[])(struct platform_device *pdev, const struct rsnd_of_data *of_data, struct rsnd_priv *priv) = { @@ -1098,12 +1086,10 @@ static int rsnd_probe(struct platform_device *pdev) exit_snd_soc: snd_soc_unregister_platform(dev); exit_snd_probe: - rsnd_lock(priv, flags); for_each_rsnd_dai(rdai, priv, i) { rsnd_dai_call(remove, &rdai->playback, priv); rsnd_dai_call(remove, &rdai->capture, priv); } - rsnd_unlock(priv, flags); return ret; } @@ -1112,7 +1098,6 @@ static int rsnd_remove(struct platform_device *pdev) { struct rsnd_priv *priv = dev_get_drvdata(&pdev->dev); struct rsnd_dai *rdai; - unsigned long flags; void (*remove_func[])(struct platform_device *pdev, struct rsnd_priv *priv) = { rsnd_ssi_remove, @@ -1123,12 +1108,10 @@ static int rsnd_remove(struct platform_device *pdev) pm_runtime_disable(&pdev->dev); - rsnd_lock(priv, flags); for_each_rsnd_dai(rdai, priv, i) { ret |= rsnd_dai_call(remove, &rdai->playback, priv); ret |= rsnd_dai_call(remove, &rdai->capture, priv); } - rsnd_unlock(priv, flags); for (i = 0; i < ARRAY_SIZE(remove_func); i++) remove_func[i](pdev, priv); -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: dapm: Add cache to speed up adding of routes
by Charles Keepax 12 May '15

12 May '15

Some CODECs have a significant number of DAPM routes and for each route, when it is added to the card, the entire card widget list must be searched. When adding routes it is very likely, however, that adjacent routes will require adjacent widgets. For example all the routes for a mux are likely added in a block and the sink widget will be the same each time and it is also quite likely that the source widgets are sequential located in the widget list. This patch adds a cache to the DAPM context, this cache will hold the source and sink widgets from the last call to snd_soc_dapm_add_route for that context. A small search of the widget list will be made from those points for both the sink and source. Currently this search only checks both the last widget and the one adjacent to it. On wm8280 which has approximately 500 widgets and 30000 routes (one of the largest CODECs in mainline), the number of paths that hit the cache is 24000, which significantly improves probe time. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v1: - Provide better encapsulation of the cache, to make swapping out the implementation simpler - Put the cache in the DAPM context instead of passing explicitly to snd_soc_dapm_add_route. I went with the context over the card as it is likely that requests on the same context will hit the cache in the case we have multiple context's interleaving requests. I hope you don't mind Lars but I didn't pull your reviewed-by forward as the code has changed quite a bit for what you reviewed. Thanks, Charles include/sound/soc-dapm.h | 7 +++++++ sound/soc/soc-dapm.c | 39 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 46 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index bfc415e..2f66e1c 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -593,6 +593,10 @@ struct snd_soc_dapm_update { int val; }; +struct snd_soc_dapm_wcache { + struct snd_soc_dapm_widget *widget; +}; + /* DAPM context */ struct snd_soc_dapm_context { enum snd_soc_bias_level bias_level; @@ -614,6 +618,9 @@ struct snd_soc_dapm_context { int (*set_bias_level)(struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level); + struct snd_soc_dapm_wcache path_sink_cache; + struct snd_soc_dapm_wcache path_source_cache; + #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dapm; #endif diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7654161..327f9357 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -572,6 +572,35 @@ static void soc_dapm_async_complete(struct snd_soc_dapm_context *dapm) snd_soc_component_async_complete(dapm->component); } +static struct snd_soc_dapm_widget * +dapm_wcache_lookup(struct snd_soc_dapm_wcache *wcache, const char *name) +{ + struct snd_soc_dapm_widget *w = wcache->widget; + struct list_head *wlist; + const int depth = 2; + int i = 0; + + if (w) { + wlist = &w->dapm->card->widgets; + + list_for_each_entry_from(w, wlist, list) { + if (!strcmp(name, w->name)) + return w; + + if (++i == depth) + break; + } + } + + return NULL; +} + +static inline void dapm_wcache_update(struct snd_soc_dapm_wcache *wcache, + struct snd_soc_dapm_widget *w) +{ + wcache->widget = w; +} + /** * snd_soc_dapm_force_bias_level() - Sets the DAPM bias level * @dapm: The DAPM context for which to set the level @@ -2610,6 +2639,12 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, source = route->source; } + wsource = dapm_wcache_lookup(&dapm->path_source_cache, source); + wsink = dapm_wcache_lookup(&dapm->path_sink_cache, sink); + + if (wsink && wsource) + goto skip_search; + /* * find src and dest widgets over all widgets but favor a widget from * current DAPM context @@ -2644,6 +2679,10 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, return -ENODEV; } +skip_search: + dapm_wcache_update(&dapm->path_sink_cache, wsink); + dapm_wcache_update(&dapm->path_source_cache, wsource); + ret = snd_soc_dapm_add_path(dapm, wsource, wsink, route->control, route->connected); if (ret) -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Remove unused call to "get_pin" in patch_hdmi.c
by David Henningsson 12 May '15

12 May '15

Janitorial patch (no functional change) Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e8d8478..fd11ad7 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2049,9 +2049,7 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hda_pcm *info; struct hda_pcm_stream *pstr; - struct hdmi_spec_per_pin *per_pin; - per_pin = get_pin(spec, pin_idx); info = snd_hda_codec_pcm_new(codec, "HDMI %d", pin_idx); if (!info) return -ENOMEM; -- 1.9.1

2 2

[alsa-devel] [PATCH 0/5] ASoC: rsnd: tidyup patches
by Kuninori Morimoto 12 May '15

12 May '15

Hi Mark These are not so very important, but tidyup patches for Renesas R-Car driver. Kuninori Morimoto (5): ASoC: rsnd: add rsnd_dai_to_priv() macro ASoC: rsnd: make sure it uses lock when it calls rsnd_dai_call ASoC: rsnd: tidyup SSI parent related function/macro names ASoC: rsnd: make sure SSI parent/child uses same number of sound channel. ASoC: rsnd: care snd_kcontrol's index sound/soc/sh/rcar/core.c | 34 +++++++++++++++++++++++++++------- sound/soc/sh/rcar/ssi.c | 52 +++++++++++++++++++++++++++++++++++++++++++++------- 2 files changed, 72 insertions(+), 14 deletions(-) Best regards --- Kuninori Morimoto

3 11

[alsa-devel] [PATCH v2] ASoC: wm8741: Add differential mono mode support
by Sergej Sawazki 11 May '15

11 May '15

The WM8741 DAC supports several differential output modes (stereo, stereo reversed, mono left, mono right). Add platform data and DT bindings to configure it. Signed-off-by: Sergej Sawazki <ce3a(a)gmx.de> --- Documentation/devicetree/bindings/sound/wm8741.txt | 11 +++ sound/soc/codecs/wm8741.c | 108 ++++++++++++++++++--- sound/soc/codecs/wm8741.h | 10 ++ 3 files changed, 117 insertions(+), 12 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/wm8741.txt b/Documentation/devicetree/bindings/sound/wm8741.txt index 74bda58..a133154 100644 --- a/Documentation/devicetree/bindings/sound/wm8741.txt +++ b/Documentation/devicetree/bindings/sound/wm8741.txt @@ -10,9 +10,20 @@ Required properties: - reg : the I2C address of the device for I2C, the chip select number for SPI. +Optional properties: + + - diff-mode: Differential output mode configuration. Default value for field + DIFF in register R8 (MODE_CONTROL_2). If absent, the default is 0, shall be: + 0 = stereo + 1 = mono left + 2 = stereo reversed + 3 = mono right + Example: codec: wm8741@1a { compatible = "wlf,wm8741"; reg = <0x1a>; + + diff-mode = <3>; }; diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 9e71c76..cbf90ab 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -41,6 +41,7 @@ static const char *wm8741_supply_names[WM8741_NUM_SUPPLIES] = { /* codec private data */ struct wm8741_priv { + struct wm8741_platform_data pdata; struct regmap *regmap; struct regulator_bulk_data supplies[WM8741_NUM_SUPPLIES]; unsigned int sysclk; @@ -398,7 +399,7 @@ static struct snd_soc_dai_driver wm8741_dai = { .name = "wm8741", .playback = { .stream_name = "Playback", - .channels_min = 2, /* Mono modes not yet supported */ + .channels_min = 2, .channels_max = 2, .rates = WM8741_RATES, .formats = WM8741_FORMATS, @@ -416,6 +417,60 @@ static int wm8741_resume(struct snd_soc_codec *codec) #define wm8741_resume NULL #endif +static int wm8741_configure(struct snd_soc_codec *codec) +{ + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + + /* Configure differential mode */ + switch (wm8741->pdata.diff_mode) { + case WM8741_DIFF_MODE_STEREO: + case WM8741_DIFF_MODE_STEREO_REVERSED: + case WM8741_DIFF_MODE_MONO_LEFT: + case WM8741_DIFF_MODE_MONO_RIGHT: + snd_soc_update_bits(codec, WM8741_MODE_CONTROL_2, + WM8741_DIFF_MASK, + wm8741->pdata.diff_mode << WM8741_DIFF_SHIFT); + break; + default: + return -EINVAL; + } + + /* Change some default settings - latch VU */ + snd_soc_update_bits(codec, WM8741_DACLLSB_ATTENUATION, + WM8741_UPDATELL, WM8741_UPDATELL); + snd_soc_update_bits(codec, WM8741_DACLMSB_ATTENUATION, + WM8741_UPDATELM, WM8741_UPDATELM); + snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, + WM8741_UPDATERL, WM8741_UPDATERL); + snd_soc_update_bits(codec, WM8741_DACRMSB_ATTENUATION, + WM8741_UPDATERM, WM8741_UPDATERM); + + return 0; +} + +static int wm8741_add_controls(struct snd_soc_codec *codec) +{ + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + + switch (wm8741->pdata.diff_mode) { + case WM8741_DIFF_MODE_STEREO: + case WM8741_DIFF_MODE_STEREO_REVERSED: + snd_soc_add_codec_controls(codec, wm8741_snd_controls, + ARRAY_SIZE(wm8741_snd_controls)); + break; + case WM8741_DIFF_MODE_MONO_LEFT: + case WM8741_DIFF_MODE_MONO_RIGHT: + /* The machine driver is responsible for mixer controls + * if the codec is configured in differential mono mode. + */ + break; + default: + return -EINVAL; + } + + return 0; +} + static int wm8741_probe(struct snd_soc_codec *codec) { struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); @@ -434,15 +489,17 @@ static int wm8741_probe(struct snd_soc_codec *codec) goto err_enable; } - /* Change some default settings - latch VU */ - snd_soc_update_bits(codec, WM8741_DACLLSB_ATTENUATION, - WM8741_UPDATELL, WM8741_UPDATELL); - snd_soc_update_bits(codec, WM8741_DACLMSB_ATTENUATION, - WM8741_UPDATELM, WM8741_UPDATELM); - snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, - WM8741_UPDATERL, WM8741_UPDATERL); - snd_soc_update_bits(codec, WM8741_DACRMSB_ATTENUATION, - WM8741_UPDATERM, WM8741_UPDATERM); + ret = wm8741_configure(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to change default settings\n"); + goto err_enable; + } + + ret = wm8741_add_controls(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to add controls\n"); + goto err_enable; + } dev_dbg(codec->dev, "Successful registration\n"); return ret; @@ -467,8 +524,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8741 = { .remove = wm8741_remove, .resume = wm8741_resume, - .controls = wm8741_snd_controls, - .num_controls = ARRAY_SIZE(wm8741_snd_controls), .dapm_widgets = wm8741_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8741_dapm_widgets), .dapm_routes = wm8741_dapm_routes, @@ -493,6 +548,23 @@ static const struct regmap_config wm8741_regmap = { .readable_reg = wm8741_readable, }; +static int wm8741_set_pdata(struct device *dev, struct wm8741_priv *wm8741) +{ + const struct wm8741_platform_data *pdata = dev_get_platdata(dev); + u32 diff_mode; + + if (dev->of_node) { + if (of_property_read_u32(dev->of_node, "diff-mode", &diff_mode) + >= 0) + wm8741->pdata.diff_mode = diff_mode; + } else { + if (pdata != NULL) + memcpy(&wm8741->pdata, pdata, sizeof(wm8741->pdata)); + } + + return 0; +} + #if IS_ENABLED(CONFIG_I2C) static int wm8741_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -522,6 +594,12 @@ static int wm8741_i2c_probe(struct i2c_client *i2c, return ret; } + wm8741_set_pdata(&i2c->dev, wm8741); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to set pdata: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, wm8741); ret = snd_soc_register_codec(&i2c->dev, @@ -582,6 +660,12 @@ static int wm8741_spi_probe(struct spi_device *spi) return ret; } + wm8741_set_pdata(&spi->dev, wm8741); + if (ret != 0) { + dev_err(&spi->dev, "Failed to set pdata: %d\n", ret); + return ret; + } + spi_set_drvdata(spi, wm8741); ret = snd_soc_register_codec(&spi->dev, diff --git a/sound/soc/codecs/wm8741.h b/sound/soc/codecs/wm8741.h index 56c1b1d..c8835f6 100644 --- a/sound/soc/codecs/wm8741.h +++ b/sound/soc/codecs/wm8741.h @@ -194,6 +194,12 @@ #define WM8741_DITHER_SHIFT 0 /* DITHER - [1:0] */ #define WM8741_DITHER_WIDTH 2 /* DITHER - [1:0] */ +/* DIFF field values */ +#define WM8741_DIFF_MODE_STEREO 0 /* stereo normal */ +#define WM8741_DIFF_MODE_STEREO_REVERSED 2 /* stereo reversed */ +#define WM8741_DIFF_MODE_MONO_LEFT 1 /* mono left */ +#define WM8741_DIFF_MODE_MONO_RIGHT 3 /* mono right */ + /* * R32 (0x20) - ADDITONAL_CONTROL_1 */ @@ -208,4 +214,8 @@ #define WM8741_SYSCLK 0 +struct wm8741_platform_data { + u32 diff_mode; /* Differential Output Mode */ +}; + #endif -- 1.9.1

3 6

[alsa-devel] Atmel_usba_udc: Kernel panic in usba_ep_enable
by Angelo Compagnucci 11 May '15

11 May '15

Hello List, I'm having troubles with usb audio on an SAMA5D3 on 3.19.8 with a strange kernel panic. The function f_audio_set_alt (f_uac1.c) is passing a NULL out_ep and so the usba_ep_enable crashes with a null pointer dereference as you can see from the debugging output: [ 24.210000] ep1: ep_enable: desc= (null) Unfortunately, there is nothing strange with my configuration, I'm trying to use the soundcard (which plays files flawlessly) as a usb soundcard, so the only thing I did was to compile g_audio as a module, enable UAC1 and did a modprode. Can you share any advice? Following kernel panic output: [ 23.500000] g_audio gadget: adding config #1 'Linux USB Audio Gadget'/bf0232c8 [ 23.510000] g_audio gadget: adding 'g_audio'/cf9c6400 to config 'Linux USB Audio Gadget'/bf0232c8 [ 23.600000] g_audio gadget: Hardware params: access 3, format 2, channels 2, rate 48000 [ 23.630000] g_audio gadget: cfg 1/bf0232c8 speeds: high full [ 23.630000] g_audio gadget: interface 0 = g_audio/cf9c6400 [ 23.640000] g_audio gadget: interface 1 = g_audio/cf9c6400 [ 23.640000] g_audio gadget: Linux USB Audio Gadget, version: Feb 2, 2012 [ 23.650000] g_audio gadget: g_audio ready [ 24.200000] g_audio gadget: high-speed config #1: Linux USB Audio Gadget [ 24.200000] g_audio gadget: intf 0, alt 0 [ 24.200000] g_audio gadget: intf 1, alt 0 [ 24.210000] g_audio gadget: intf 1, alt 1 [ 24.210000] ep1: ep_enable: desc= (null) [ 24.210000] Unable to handle kernel NULL pointer dereference at virtual address 00000002 [ 24.210000] pgd = c0004000 [ 24.210000] [00000002] *pgd=00000000 [ 24.210000] Internal error: Oops: 17 [#1] ARM [ 24.210000] Modules linked in: usb_f_uac1 g_audio libcomposite [ 24.210000] CPU: 0 PID: 0 Comm: swapper Not tainted 3.19.8+ #69 [ 24.210000] Hardware name: Atmel SAMA5 (Device Tree) [ 24.210000] task: c052c7b8 ti: c0522000 task.ti: c0522000 [ 24.210000] PC is at usba_ep_enable+0x28/0x200 [ 24.210000] LR is at usba_ep_enable+0x24/0x200 [ 24.210000] pc : [<c02815bc>] lr : [<c02815b8>] psr: 200f0193 [ 24.210000] sp : c0523e18 ip : 00000007 fp : 0000bb80 [ 24.210000] r10: 00000100 r9 : cfb17c00 r8 : 00000001 [ 24.210000] r7 : 000000c8 r6 : cf99aa10 r5 : 00000000 r4 : cf99d868 [ 24.210000] r3 : 00000006 r2 : 00000007 r1 : 00000000 r0 : 00000001 [ 24.210000] Flags: nzCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment kernel [ 24.210000] Control: 10c53c7d Table: 2fb28059 DAC: 00000015 [ 24.210000] Process swapper (pid: 0, stack limit = 0xc0522230) [ 24.210000] Stack: (0xc0523e18 to 0xc0524000) [ 24.210000] 3e00: 00000001 cf99d868 [ 24.210000] 3e20: cf9c6400 bf028694 00000001 c0530008 cfa13ac0 cfb17c00 cfb17a00 00000001 [ 24.210000] 3e40: cf9c6400 00000001 00000001 cf99aa18 00000000 bf015104 c0523e74 c0034d9c [ 24.210000] 3e60: cf9471c0 600f0193 cfb05f6c c0530704 c00398b8 c00398c4 c05306f8 c0039400 [ 24.210000] 3e80: 00000000 200f0193 00000007 00000006 00000000 cf99d810 0000000b cf99d848 [ 24.210000] 3ea0: cf99aa10 00000000 00000000 00000000 00000800 c0282f0c 0000001c ffffffff [ 24.210000] 3ec0: 7fffffff c0547d80 00000000 00000125 00000001 c054e23e 00010b01 00000001 [ 24.210000] 3ee0: 00000000 cf998240 cf802540 00000000 00000000 0000002d cf901dc0 c054e23e [ 24.210000] 3f00: 00000000 c0040af0 cf802200 c054eb00 cf901dc0 cf802540 00000000 00000000 [ 24.210000] 3f20: cf802200 c051ac50 cfffc480 c0040be0 cf901dc0 c0042e00 0000002d 0000002d [ 24.210000] 3f40: 00000000 c0040314 c0534ca0 c0040538 c000eec4 600f0013 ffffffff c0523f9c [ 24.210000] 3f60: c054e340 c0011b00 00000000 c0530918 00000000 c0017880 c0522000 c052a0c4 [ 24.210000] 3f80: c054e23c c054e23c c054e340 c051ac50 cfffc480 00000000 01000000 c0523fb0 [ 24.210000] 3fa0: c000eec0 c000eec4 600f0013 ffffffff c0522000 c0039b58 ffffffff c04f5c30 [ 24.210000] 3fc0: ffffffff ffffffff c04f5678 00000000 00000000 c051ac50 c054e514 c052a070 [ 24.210000] 3fe0: c051ac4c c052d860 20004059 410fc051 00000000 20008070 00000000 00000000 [ 24.210000] [<c02815bc>] (usba_ep_enable) from [<bf028694>] (f_audio_set_alt+0xc4/0x19c [usb_f_uac1]) [ 24.210000] [<bf028694>] (f_audio_set_alt [usb_f_uac1]) from [<bf015104>] (composite_setup+0x4e0/0x1640 [libcomposite]) [ 24.210000] [<bf015104>] (composite_setup [libcomposite]) from [<c0282f0c>] (usba_udc_irq+0x75c/0xc74) [ 24.210000] [<c0282f0c>] (usba_udc_irq) from [<c0040af0>] (handle_irq_event_percpu+0x78/0x140) [ 24.210000] [<c0040af0>] (handle_irq_event_percpu) from [<c0040be0>] (handle_irq_event+0x28/0x38) [ 24.210000] [<c0040be0>] (handle_irq_event) from [<c0042e00>] (handle_fasteoi_irq+0x9c/0x164) [ 24.210000] [<c0042e00>] (handle_fasteoi_irq) from [<c0040314>] (generic_handle_irq+0x2c/0x3c) [ 24.210000] [<c0040314>] (generic_handle_irq) from [<c0040538>] (__handle_domain_irq+0x54/0xa8) [ 24.210000] [<c0040538>] (__handle_domain_irq) from [<c0011b00>] (__irq_svc+0x40/0x54) [ 24.210000] [<c0011b00>] (__irq_svc) from [<c000eec4>] (arch_cpu_idle+0x38/0x3c) [ 24.210000] [<c000eec4>] (arch_cpu_idle) from [<c0039b58>] (cpu_startup_entry+0xa8/0xf4) [ 24.210000] [<c0039b58>] (cpu_startup_entry) from [<c04f5c30>] (start_kernel+0x374/0x380) [ 24.210000] Code: e5941004 e5946024 eb04d705 e5d40033 (e5d51002) [ 24.210000] ---[ end trace d5ac053970e7d2d1 ]--- [ 24.210000] Kernel panic - not syncing: Fatal exception in interrupt [ 24.210000] ---[ end Kernel panic - not syncing: Fatal exception in interrupt -- Profile: http://it.linkedin.com/in/compagnucciangelo

2 1

[alsa-devel] [PATCH 01/11] ASoC: 88pm860x: Replace direct snd_soc_codec dapm field access
by Lars-Peter Clausen 11 May '15

11 May '15

The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm.bias_level with snd_soc_codec_get_bias_level(). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- All patches in this series have a dependency on topic/dapm --- sound/soc/codecs/88pm860x-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index c0b2686..ee31fa7 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1140,7 +1140,7 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { /* Enable Audio PLL & Audio section */ data = AUDIO_PLL | AUDIO_SECTION_ON; pm860x_reg_write(pm860x->i2c, REG_MISC2, data); -- 1.8.0

2 11

[alsa-devel] [PATCH 0/3] ASoC: DPCM can care FE/BE merged format
by Kuninori Morimoto 11 May '15

11 May '15

Hi Mark These patches add new .dpcm_merged_foramt to merge FE/BE formats. It will be no sound if DPCM were below style, and user selects S24_LE. FE: S16_LE/S24_LE BE: S16_LE DPCM will selects FE/BE merged formats (= S16_LE here) if FE has .dpcm_merged_foramt. Kuninori Morimoto (3): ASoC: soc.h: tidyup struct snd_soc_dai_link definition order ASoC: soc-pcm: DPCM cares BE format ASoC: rsnd: rsrc-card uses FE/BE merged format when DPCM include/sound/soc.h | 39 +++++++++++++++++++++------------------ sound/soc/sh/rcar/rsrc-card.c | 1 + sound/soc/soc-pcm.c | 47 ++++++++++++++++++++++++++++++++++++++++++----- 3 files changed, 64 insertions(+), 23 deletions(-) Best regards --- Kuninori Morimoto

3 12

Re: [alsa-devel] [PATCH 2/2] sound/usb: Update ALSA driver to use media controller API
by Shuah Khan 11 May '15

11 May '15

On 05/08/2015 10:51 PM, Eric Wong wrote: > Shuah Khan <shuahkh(a)osg.samsung.com> wrote: >> @@ -541,13 +591,19 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, >> [QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk, >> [QUIRK_AUDIO_STANDARD_MIXER] = create_standard_mixer_quirk, >> }; >> + int ret; >> >> + if (quirk->media_device) { >> + /* don't want to fail when media_device_init() doesn't work */ >> + ret = media_device_init(iface); >> + } >> if (quirk->type < QUIRK_TYPE_COUNT) { >> - return quirk_funcs[quirk->type](chip, iface, driver, quirk); >> + ret = quirk_funcs[quirk->type](chip, iface, driver, quirk); >> } else { >> usb_audio_err(chip, "invalid quirk type %d\n", quirk->type); >> return -ENXIO; >> } >> + return ret; >> } > > What is the point of saving 'ret' of media_device_init if it'll > only be clobbered or ignored for ENXIO? > Agreed. There is no point in saving it. thanks, -- Shuah -- Shuah Khan Sr. Linux Kernel Developer Open Source Innovation Group Samsung Research America (Silicon Valley) shuahkh(a)osg.samsung.com | (970) 217-8978

1 0

Re: [alsa-devel] [PATCH 1/2] ASoC: rt5645: fix jack type detect error
by Mark Brown 11 May '15

11 May '15

On Mon, May 11, 2015 at 02:33:36AM +0000, John Lin wrote: > > -----Original Message----- > > From: Mark Brown [mailto:broonie@kernel.org] > > Sent: Saturday, May 09, 2015 1:14 AM > > To: John Lin > > Cc: lgirdwood(a)gmail.com; alsa-devel(a)alsa-project.org; Bard Liao; Flove; > > Oder Chiou; koro.chen(a)mediatek.com; yang.a.fang(a)intel.com; > > zhengxing(a)rock-chips.com > > Subject: Re: [PATCH 1/2] ASoC: rt5645: fix jack type detect error > > On Fri, May 08, 2015 at 01:53:33PM +0800, John Lin wrote: > > > rt5645_jack_detect doesn't report the correct jack type consistently. > > > This patch can make it more stable. > > Can you be more specific about what the problem is and how this fixes it > > pleased? It's a bit hard to review based on your changelog. > It mistakes OMTP type headset to CTIA type if "LDO2" power is disable in > jack detection function. So, we keep "LDO2" power enabled until jack out. > We also change the register sequence to make it more stable and remove > unnecessary power such as "micbias1" and "micbias2" for jack detection. This stuff needs to be in the changelog. It also sounds like there's at least two probably three patches here - one to keep the LDO power enabled for lonnger and one or more changes to the register sequence.

1 0

[alsa-devel] stereo uac device control over mixer
by Arijit De 11 May '15

11 May '15

Hi, I am working on a USB audio MIC device(capture device), implemented UAC1.0 spec inside the firmware for that, my device is a stereo uac capture? device. I am trying to enable left and write channel separate control(volume, mute etc.) over mixer interface. my understanding about how to enable channel specific control is in UAC Feature Unit Descriptor if I add bmaControls(0) and bmaControls(1) with correct bitmap for Mute & Volume support amixer is able to identify my device as a two MIC device, amixer output for my usb audio capture device ********************************** Simple mixer control 'Mic',0 Capabilities: cvolumecaps joined 780 cvolume-joined cswitch cswitch-joined Capture channels: Mono Limits: Capture 0 - 15 Mono: Capture 9 [60%] [27.00dB] [on] Simple mixer control 'Mic',1 Capabilities: cvolumecaps joined 780 cvolume-joined cswitch cswitch-joined Capture channels: Mono Limits: Capture 0 - 15 Mono: Capture 9 [60%] [27.00dB] [on] ********************************** is the correct way of achieving it OR ideally there should be a single Mic with "Front Left - Front Right" support ? Thanks Arijit

2 4

[alsa-devel] [PATCH 1/2] ASoC: fsl: fsl_dma: Use true/false for boolean init
by Fabio Estevam 11 May '15

11 May '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Bool initializations should use true and false. Bool tests don't need comparisons. Based on contributions from Joe Perches, Rusty Russell and Bruce W Allan. The semantic patch that makes this change is available in scripts/coccinelle/misc/boolinit.cocci. More information about semantic patching is available at http://coccinelle.lip6.fr/ Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 93d7e56..ccadefc 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -445,7 +445,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) return ret; } - dma->assigned = 1; + dma->assigned = true; snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware); @@ -814,7 +814,7 @@ static int fsl_dma_close(struct snd_pcm_substream *substream) substream->runtime->private_data = NULL; } - dma->assigned = 0; + dma->assigned = false; return 0; } -- 1.9.1

3 3

Re: [alsa-devel] Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 11 May '15

11 May '15

> I started MS Windows and without externals speakers plugged in I could test > 5.1 configuration and it seemed that sound was coming from different directions. > > What I managed to determine: > I've got 3 stacks: > 1 ST: Headphone, Line-out (FL and FR), SPDIF > 2 ST: Microphone, Line-out (CLFE) > 3 ST: Line-in, Line-out (RL and RR) > > I've got 5 responsive output pins > PIN 0x14: Internal FL FR > PIN 0x15: External FL FR > PIN 0x17: Internal LFE > PIN 0x18: External CLFE > PIN 0x1a: External RL RR > > How can we make it work with these information? https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/plain/Document… seem chnmap type _FIXED , _VAR and _PAIRED does not fit when LFE and FC can be swapped for internal speaker and external speaker https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… How do the driver implement channel map when codec has swap center/lfe switch ? Do chnmap contain both {FL FR RL RR FC LFE} and {FL FR RL RR LFE FC} ? https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… atihdmi_paired_swap_fc_lfe

1 0

[alsa-devel] [PATCH] ASoC: codecs-ac97: Remove rate constraints
by Maciej S. Szmigiero 10 May '15

10 May '15

Remove rate constraints from generic ASoC AC'97 CODEC and make it selectable in config. Supported rates should be detected and constrained anyway by AC'97 generic code - was tested with VT1613 CODEC and iMX6 SSI controller. This way this driver can be used for platforms which don't need specialized AC'97 CODEC drivers while at the same avoiding code duplication from implementing equivalent functionality in a controller driver. Resending due to no response received. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index d0ac723..5b3224c 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -44,10 +44,6 @@ static int ac97_prepare(struct snd_pcm_substream *substream, return snd_ac97_set_rate(ac97, reg, substream->runtime->rate); } -#define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000) - static const struct snd_soc_dai_ops ac97_dai_ops = { .prepare = ac97_prepare, }; @@ -58,13 +54,13 @@ static struct snd_soc_dai_driver ac97_dai = { .stream_name = "AC97 Playback", .channels_min = 1, .channels_max = 2, - .rates = STD_AC97_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SND_SOC_STD_AC97_FMTS,}, .capture = { .stream_name = "AC97 Capture", .channels_min = 1, .channels_max = 2, - .rates = STD_AC97_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &ac97_dai_ops, }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..84cad9a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,7 +16,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 select SND_SOC_AB8500_CODEC if ABX500_CORE - select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS + select SND_SOC_AC97_CODEC select SND_SOC_AD1836 if SPI_MASTER select SND_SOC_AD193X_SPI if SPI_MASTER select SND_SOC_AD193X_I2C if I2C @@ -211,8 +211,9 @@ config SND_SOC_AB8500_CODEC tristate config SND_SOC_AC97_CODEC - tristate + tristate "Build generic ASoC AC97 CODEC driver" select SND_AC97_CODEC + select SND_SOC_AC97_BUS config SND_SOC_AD1836 tristate

2 4

[alsa-devel] [PATCH 1/2] ASoC: sta32x: Use devm_gpiod_get_optional appropriate place
by Axel Lin 09 May '15

09 May '15

devm_gpiod_get_optional() is equivalent to devm_gpiod_get(), except that when no GPIO was assigned to the requested function it will return NULL. This is convenient for drivers that need to handle optional GPIOs. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/sta32x.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 033b7d9..847fb20 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -1095,16 +1095,10 @@ static int sta32x_i2c_probe(struct i2c_client *i2c, #endif /* GPIOs */ - sta32x->gpiod_nreset = devm_gpiod_get(dev, "reset"); - if (IS_ERR(sta32x->gpiod_nreset)) { - ret = PTR_ERR(sta32x->gpiod_nreset); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta32x->gpiod_nreset = NULL; - } else { - gpiod_direction_output(sta32x->gpiod_nreset, 0); - } + sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(sta32x->gpiod_nreset)) + return PTR_ERR(sta32x->gpiod_nreset); /* regulators */ for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) -- 2.1.0

1 1

[alsa-devel] PreSonus Information For Developers
by Karl Swisher 09 May '15

09 May '15

Hello all, I have what might be useful information to some developers but I do not sure who or how to contact the right people. I recently purchased a PreSonus StudioLive AI 32.4.2 mixer. Have it half way working under Linux. This mixer has a network connection in addition to the firewire interface. I've installed their software under VirtualBox and ran tcpflow capturing packet content between the computer and the mixer. I have started creating text files of the packets of which below is an example of their software performing functions like the mixer booting, backing up, restore, and selecting the digital output channels 33 through 48. 55 43 00 01 06 00 4b 41 65 00 64 00 55 43 00 01 06 00 4b 41 65 00 64 00 55 43 00 01 06 00 4b 41 65 00 64 00 55 43 00 01 06 00 4b 41 65 00 64 00 55 43 00 01 06 00 4b 41 65 00 64 00 55 43 00 01 06 00 4b 41 65 00 64 00 55 43 00 01 06 00 4b 41 65 00 64 00 55 43 00 01 1d 00 46 52 65 00 64 00 01 00 Listpresets/scene 00 00 55 43 00 01 1b 00 46 52 65 00 64 00 01 00 Listpresets/channel 00 00 55 43 00 01 19 00 46 52 65 00 64 00 01 00 Listpresets/geq 00 00 55 43 00 01 19 00 46 52 65 00 64 00 01 00 Listpresets/reverb 00 00 I'm trying to separate out the bin code from the text. Does anyone here know if there is value in this information or anyone that might want it? Have a few functions decoded in the format above. Thanks Karl

2 1

Re: [alsa-devel] Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 09 May '15

09 May '15

> > > > http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixe > > > > r/paths/analog-output-headphones.conf > > > > > > > > pulseaudio mute external center/lfe controls when headphone jack is > > > > plugged > > That would be unwanted. > This is common to all notebook using headphone jack and two other jacks to support surround51 or surround71 since pulseaudio mute the other retasked mio jack when headphone jack is plugged > > > > I used hdajackretask to override: > > > PIN ID: 0x15 to "Line out (front)" > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/patch/sound/pc… When the static model were used, the jack was configured as line out instead of headphone jack and there is no headphone playback volume -static const struct hda_verb alc888_acer_aspire_4930g_verbs[] = { -/* Enable unsolicited event for HP jack */ - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, -/* Connect HP out to front */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, -static const struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), The jack is configured as headphone jack after the auto parser is used > > Pulseaudio don't know node 0x15 is headphone / Line Out (External > speaker) > > > > select speaker profile mute the headphone playback switch > > Doesn't HP mean HeadPhone? Isn't solution to my problem obvious, namely add > Line out capability to 0x15? > > Are there any side effect if you change the headphone jack to Line Out jack as the jack no longer enable HP amp ? use add_out_jack_modes for create Headphone/Line Out jack mode control to change Headphone to Line Out and vice versa diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 3d2597b..f25cd23 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -180,10 +180,10 @@ static void parse_user_hints(struct hda_codec *codec) /* the following two are just for compatibility */ val = snd_hda_get_bool_hint(codec, "add_out_jack_modes"); if (val >= 0) - spec->add_jack_modes = !!val; + spec->add_out_jack_modes = !!val; val = snd_hda_get_bool_hint(codec, "add_in_jack_modes"); if (val >= 0) spec->add_jack_modes = !!val; @@ -2640,7 +2642,7 @@ static void get_jack_mode_name(struct hda_codec *codec, hda_nid_t pin, static int get_out_jack_num_items(struct hda_codec *codec, hda_nid_t pin) { struct hda_gen_spec *spec = codec->spec; - if (spec->add_jack_modes) { + if (spec->add_out_jack_modes || spec->add_jack_modes) { unsigned int pincap = snd_hda_query_pin_caps(codec, pin); if ((pincap & AC_PINCAP_OUT) && (pincap & AC_PINCAP_HP_DRV)) return 2; @@ -4892,7 +4895,7 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, return err; } - if (spec->add_jack_modes) { + if (spec->add_out_jack_modes || spec->add_jack_modes) { if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { err = create_out_jack_modes(codec, cfg->line_outs, cfg->line_out_pins); diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 56e4139..8d7568b 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -227,6 +227,8 @@ struct hda_gen_spec { unsigned int prefer_hp_amp:1; /* enable HP amp for speaker if any */ unsigned int add_stereo_mix_input:2; /* add aamix as a capture src */ unsigned int add_jack_modes:1; /* add i/o jack mode enum ctls */ + unsigned int add_out_jack_modes:1; unsigned int power_down_unused:1; /* power down unused widgets */ unsigned int dac_min_mute:1; /* minimal = mute for DACs */

1 0

[alsa-devel] [PATCH 1/3] ASoC: rt5645: fix add missing widget
by Bard Liao 08 May '15

08 May '15

"IF1 DAC0" and "IF1 DAC3" are used in rt5645_dapm_routes but missing in rt5645_dapm_widgets. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 7996c9c..a72d989 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -1697,8 +1697,10 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { /* Digital Interface */ SND_SOC_DAPM_SUPPLY("I2S1", RT5645_PWR_DIG1, RT5645_PWR_I2S1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC0", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 DAC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC3", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("RT5645 IF1 DAC1 L Mux", SND_SOC_NOPM, 0, 0, &rt5645_if1_dac0_tdm_sel_mux), SND_SOC_DAPM_MUX("RT5645 IF1 DAC1 R Mux", SND_SOC_NOPM, 0, 0, -- 1.8.1.1.439.g50a6b54

5 15

Re: [alsa-devel] [PATCH] ASoC: bt-sco: Add devicetree support for bt-sco codec
by Mark Brown 08 May '15

08 May '15

On Tue, May 05, 2015 at 10:15:41PM +0200, Marek Belisko wrote: > +static const struct of_device_id bt_sco_codec_of_match[] = { > + { .compatible = "delta,dfbmcs320", }, > + { .compatible = "linux,bt-sco", }, Is there any burning reason for adding the linux,bt-sco compatible?

2 2

Re: [alsa-devel] [PATCH 2/2] ASoC: rt5645: improve headphone depop function
by Mark Brown 08 May '15

08 May '15

On Fri, May 08, 2015 at 01:53:34PM +0800, John Lin wrote: > We add a calibration function and call it at the beginning of > i2c_probe. The calibration value will be kept until codec is > shutdown. We will reset the codec after the calibration is finished. > So, we set cache_bypass in the calibration function. The benefit is > we can shorter the delay time in headphone depop. This looks fine but seems to depend on the previous patch.

1 0

Re: [alsa-devel] [PATCH 1/2] ASoC: rt5645: fix jack type detect error
by Mark Brown 08 May '15

08 May '15

On Fri, May 08, 2015 at 01:53:33PM +0800, John Lin wrote: > rt5645_jack_detect doesn't report the correct jack type consistently. > This patch can make it more stable. Can you be more specific about what the problem is and how this fixes it pleased? It's a bit hard to review based on your changelog.

1 0

[alsa-devel] [PATCH] ASoC: rt5677: Add DMIC ASRC detect function
by Oder Chiou 08 May '15

08 May '15

The patch adds DMIC ASRC detect function to dominate whether the DMIC ASRC enable or not. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 81 +++++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 75 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 685e268..8bb7be8 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1230,6 +1230,75 @@ int rt5677_sel_asrc_clk_src(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(rt5677_sel_asrc_clk_src); +static int rt5677_dmic_use_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int asrc_setting; + + switch (source->shift) { + case 11: + regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); + asrc_setting = (asrc_setting & RT5677_AD_STO1_CLK_SEL_MASK) >> + RT5677_AD_STO1_CLK_SEL_SFT; + if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && + asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) + return 1; + break; + + case 10: + regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); + asrc_setting = (asrc_setting & RT5677_AD_STO2_CLK_SEL_MASK) >> + RT5677_AD_STO2_CLK_SEL_SFT; + if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && + asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) + return 1; + break; + + case 9: + regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); + asrc_setting = (asrc_setting & RT5677_AD_STO3_CLK_SEL_MASK) >> + RT5677_AD_STO3_CLK_SEL_SFT; + if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && + asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) + return 1; + break; + + case 8: + regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); + asrc_setting = (asrc_setting & RT5677_AD_STO4_CLK_SEL_MASK) >> + RT5677_AD_STO4_CLK_SEL_SFT; + if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && + asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) + return 1; + break; + + case 7: + regmap_read(rt5677->regmap, RT5677_ASRC_6, &asrc_setting); + asrc_setting = (asrc_setting & RT5677_AD_MONOL_CLK_SEL_MASK) >> + RT5677_AD_MONOL_CLK_SEL_SFT; + if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && + asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) + return 1; + break; + + case 6: + regmap_read(rt5677->regmap, RT5677_ASRC_6, &asrc_setting); + asrc_setting = (asrc_setting & RT5677_AD_MONOR_CLK_SEL_MASK) >> + RT5677_AD_MONOR_CLK_SEL_SFT; + if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && + asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) + return 1; + break; + + default: + break; + } + + return 0; +} + /* Digital Mixer */ static const struct snd_kcontrol_new rt5677_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5677_STO1_ADC_MIXER, @@ -3087,12 +3156,12 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { }; static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { - { "Stereo1 DMIC Mux", NULL, "DMIC STO1 ASRC", can_use_asrc }, - { "Stereo2 DMIC Mux", NULL, "DMIC STO2 ASRC", can_use_asrc }, - { "Stereo3 DMIC Mux", NULL, "DMIC STO3 ASRC", can_use_asrc }, - { "Stereo4 DMIC Mux", NULL, "DMIC STO4 ASRC", can_use_asrc }, - { "Mono DMIC L Mux", NULL, "DMIC MONO L ASRC", can_use_asrc }, - { "Mono DMIC R Mux", NULL, "DMIC MONO R ASRC", can_use_asrc }, + { "Stereo1 DMIC Mux", NULL, "DMIC STO1 ASRC", rt5677_dmic_use_asrc }, + { "Stereo2 DMIC Mux", NULL, "DMIC STO2 ASRC", rt5677_dmic_use_asrc }, + { "Stereo3 DMIC Mux", NULL, "DMIC STO3 ASRC", rt5677_dmic_use_asrc }, + { "Stereo4 DMIC Mux", NULL, "DMIC STO4 ASRC", rt5677_dmic_use_asrc }, + { "Mono DMIC L Mux", NULL, "DMIC MONO L ASRC", rt5677_dmic_use_asrc }, + { "Mono DMIC R Mux", NULL, "DMIC MONO R ASRC", rt5677_dmic_use_asrc }, { "I2S1", NULL, "I2S1 ASRC", can_use_asrc}, { "I2S2", NULL, "I2S2 ASRC", can_use_asrc}, { "I2S3", NULL, "I2S3 ASRC", can_use_asrc}, -- 1.8.1.1.439.g50a6b54

2 1

Re: [alsa-devel] [PATCH RFC v2 10/13] sound/core: add DRM ELD helper
by Russell King - ARM Linux 08 May '15

08 May '15

On Fri, May 08, 2015 at 04:16:08PM +0300, Jyri Sarha wrote: > On 2015-04-02 12:22, Russell King wrote: > >+static const unsigned int eld_rates[] = { > >+ 32000, > >+ 44100, > >+ 48000, > >+ 88200, > >+ 96000, > >+ 176400, > >+ 192000, > >+}; > >+ > > For what I can see, you are setting the cross dependency between > the sample rate and channels incorrectly. There is a dependency between sample rate and channels. > You should look for the > currently tested channel count directly from hw params. I'm not so sure that's right after looking at snd_ac97_pcm_double_rate_rules(). That's pretty much the same problem - AC'97 codecs can operate at a sample rate of either the bus frame frequency or twice the bus frame frequency. They achieve twice the bus frame frequency by reallocating a couple of the PCM data slots which would otherwise be used for >2 channels to the first two channels, thus allowing two front channel samples per frame. AC'97 limits the number of channels to two if: if (rate->min > 48000) { and limits the sample rate to 1-48kHz if: if (channels->min > 2) { > From this side the dependency is missing all together. Yes, it I initially couldn't work out how to do that bit... but I have a solution now which seems to sort-of work. Testing with aplay, I find that it seems to mostly work, and I guess that: aplay -D hw:0,0 -v -f S24_LE -c 6 -r 192000 /dev/zero resulting in: Playing raw data '/dev/zero' : Signed 24 bit Little Endian, Rate 192000 Hz, Channels 6 Warning: rate is not accurate (requested = 192000Hz, got = 48000Hz) please, try the plug plugin Hardware PCM card 0 'DW-HDMI' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S24_LE subformat : STD channels : 6 rate : 48000 exact rate : 48000 (48000/1) is acceptable. Also looking at AC'97, I don't need to list the same parameter as a dependent of the rule... something I'll try when I'm back from the hospital later this afternoon... -- FTTC broadband for 0.8mile line: currently at 10.5Mbps down 400kbps up according to speedtest.net.

2 1

Re: [alsa-devel] [PATCH RFC v2 10/13] sound/core: add DRM ELD helper
by Russell King - ARM Linux 08 May '15

08 May '15

On Fri, May 08, 2015 at 01:56:02PM +0300, Jyri Sarha wrote: > On 2015-04-05 20:26, Russell King - ARM Linux wrote: > >On Sun, Apr 05, 2015 at 06:46:13PM +0200, Takashi Iwai wrote: > >>At Sun, 5 Apr 2015 17:20:34 +0100, > >>Russell King - ARM Linux wrote: > >>> Since (afaik) ALSA has a lack of support for dynamic reconfiguration > >>> according to the attached device changing, the best we can do without > >>> a huge amount of re-work of HDMI support across all adapters is to > >>> process the capabilities of the attached device at prepare time > >>> according to the current capabilities. > >> > >>Yeah, reconfiguration is tricky. BTW, how is the HDMI unplug handled > >>during playback? > > > >We don't handle it right now - and we don't have any notification to > >the audio drivers that that has happened. Even if we did have such a > >notification, I'm not sure what the audio driver could do with it at > >the moment. > > > >>> Implementing dynamic reconfiguration in ALSA is not something I want to > >>> get involved with, and as we /already/ have HDMI support merged in the > >>> kernel, this is not a blocker for at least trying to get some semblence > >>> of sanity, rather than having every driver re-implementing stuff like > >>> this. > >> > >>Well, I didn't mean about the dynamic reconfiguration. I thought of > >>rather min/max pairs, but it was just a wrong assumption. Scratch > >>it. > >> > >>One another question: don't we need to deal with the sample bits in > >>sad[2]? > > > >It should, but I'm very wary about doing that without seeing more > >examples of real SADs. > > If the same constraint setting helpers are to be used also with > external HDMI encoders with i2s interface, there should be an option > to leave out the constraints for the sample bits. There is no direct > connection between the number of bits on I2S and HDMI link. The CPU > DAI may apply its own constraints on the available sample formats and > too strict constraints at the encoder end may lead to zero available > formats. Possibly, but then how do we know which SAD to apply to limit the number of channels? I suspect for the use case you're suggesting above, it's better left to the user rather than generic code to work it out otherwise things get far too complex. We'd need to have some way for the driver to report back to this generic code the actual number of bits on the HDMI link. However, as we currently know, ALSA appears to have a problem here which pretty much makes properly restricting the channels and sample rates impossible to do. That issue is the one I'm much more keen to solve right now, because without that being solved, this set of patches is pretty much dead in the water. -- FTTC broadband for 0.8mile line: currently at 10.5Mbps down 400kbps up according to speedtest.net.

1 0

[alsa-devel] [PATCH v12 0/6] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 08 May '15

08 May '15

Based on 4.1-rc1 As Russell King's DRM ELD helper is not yet in any repository, it is included here with a little patch. v12: - use Russell King's DRM ELD helper (Mark Brown) v11: - reduce the patch series to adding the tda998x codec only v10: - add the generic dt-card - define the audio ports from a DT graph of ports (Russell King) - reuse HDMI constants (Andrew Jackson - Jyri Sarha) - alloc rate_constraints in codec (Jyri Sarha) - fix bad number of channels (Jyri Sarha) - correct codec generation from config (Russell King - Jyri Sarha) - no module init/exit (Russell King) v9: - back to a TDA998x specific CODEC - more comments - change magic values to constants v8: - change some comments about the patches v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (6): drm/edid: add function to help find SADs sound/core: add DRM ELD helper sound/core: remove the trace from the DRM ELD helper drm/i2c: tda998x: Add support of a DT graph of ports drm/i2c: tda998x: Change drvdata for audio extension ASoC: tda998x: add a codec to the HDMI transmitter .../devicetree/bindings/drm/i2c/tda998x.txt | 51 ++++++ drivers/gpu/drm/i2c/tda998x_drv.c | 178 +++++++++++++++++++-- include/drm/drm_edid.h | 19 +++ include/sound/pcm_drm_eld.h | 6 + include/sound/tda998x.h | 23 +++ sound/core/Kconfig | 3 + sound/core/Makefile | 1 + sound/core/pcm_drm_eld.c | 91 +++++++++++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 132 +++++++++++++++ 11 files changed, 501 insertions(+), 11 deletions(-) create mode 100644 include/sound/pcm_drm_eld.h create mode 100644 include/sound/tda998x.h create mode 100644 sound/core/pcm_drm_eld.c create mode 100644 sound/soc/codecs/tda998x.c -- 2.1.4

2 8

[alsa-devel] [PATCH] ASoC: fsl_ssi: No need call of_device_is_available()
by Fabio Estevam 08 May '15

08 May '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> The comment and the call to of_device_is_available() are not really needed. It is the expected behaviour to probe only the ssi nodes that are enabled in the device tree. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0d48804..c7647e0 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1292,13 +1292,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) void __iomem *iomem; char name[64]; - /* SSIs that are not connected on the board should have a - * status = "disabled" - * property in their device tree nodes. - */ - if (!of_device_is_available(np)) - return -ENODEV; - of_id = of_match_device(fsl_ssi_ids, &pdev->dev); if (!of_id || !of_id->data) return -EINVAL; -- 1.9.1

3 2

[alsa-devel] [PATCH 1/4] ASoC: brownstone: Automatically disconnect non-connected pins
by Lars-Peter Clausen 08 May '15

08 May '15

Most DAPM input and output pins of the wm8994 are either used in the card's DAPM routing table or are marked as not connected. The only exception is DMIC2DAT input. Given that DMIC1DAT is explicitly mentioned in the DAPM routes lets assume that DMIC2DAT simply has been overlooked and should be marked as not connected. Set the fully_routed flag of the card instead of manually marking the unused inputs and outputs as not connected. This makes the code a bit shorter and cleaner. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/pxa/brownstone.c | 25 +------------------------ 1 file changed, 1 insertion(+), 24 deletions(-) diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 79936e3..2b26318 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -45,29 +45,6 @@ static const struct snd_soc_dapm_route brownstone_audio_map[] = { {"MICBIAS1", NULL, "Main Mic"}, }; -static int brownstone_wm8994_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* set endpoints to not connected */ - snd_soc_dapm_nc_pin(dapm, "HPOUT2P"); - snd_soc_dapm_nc_pin(dapm, "HPOUT2N"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT1N"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT1P"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT2N"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT2P"); - snd_soc_dapm_nc_pin(dapm, "IN1LN"); - snd_soc_dapm_nc_pin(dapm, "IN1LP"); - snd_soc_dapm_nc_pin(dapm, "IN1RP"); - snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN"); - snd_soc_dapm_nc_pin(dapm, "IN2RN"); - snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); - snd_soc_dapm_nc_pin(dapm, "IN2LN"); - - return 0; -} - static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -115,7 +92,6 @@ static struct snd_soc_dai_link brownstone_wm8994_dai[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ops = &brownstone_ops, - .init = brownstone_wm8994_init, }, }; @@ -132,6 +108,7 @@ static struct snd_soc_card brownstone = { .num_dapm_widgets = ARRAY_SIZE(brownstone_dapm_widgets), .dapm_routes = brownstone_audio_map, .num_dapm_routes = ARRAY_SIZE(brownstone_audio_map), + .fully_routed = true, }; static int brownstone_probe(struct platform_device *pdev) -- 1.8.0

2 4

[alsa-devel] [TINYCOMPRESS][PATCH v3 0/7] Convert build system to autotools
by Qais Yousef 08 May '15

08 May '15

This series converts the build system to use autotools and splits the files into lib and utils. Also patch 5 fixes error handling code. I can't test Android builds but both cross and native compiling worked for me. I left AUTHORS, ChangeLog and NEWS files empty. I used the license header to fill COPYING. Changes in v3: - fix a typo in PATCH 1 where we compile cplay.c instead of crec.c for crecord app. - Add mising SOB lines. Qais Yousef (7): Convert the build system to autotools add .gitignore file makefile.linux: delete as no longer necessary/used Android.mk: Update to use the new location of source files compress.c: fix check for errors from poll(), read() and write() src/lib/utils.c: remove this empty file tinycompress.h: explicitly include required headers .gitignore | 39 ++++ AUTHORS | 0 Android.mk | 6 +- COPYING | 51 +++++ ChangeLog | 0 INSTALL | 365 ++++++++++++++++++++++++++++++++++++ Makefile.am | 3 + NEWS | 0 autogen.sh | 6 + configure.ac | 32 ++++ include/Makefile.am | 6 + include/tinycompress/tinycompress.h | 3 + m4/.place_holder | 0 makefile.linux | 41 ---- src/Makefile.am | 1 + src/lib/Makefile.am | 5 + compress.c => src/lib/compress.c | 22 ++- src/utils/Makefile.am | 11 ++ cplay.c => src/utils/cplay.c | 0 crec.c => src/utils/crec.c | 0 utils.c | 62 ------ 21 files changed, 537 insertions(+), 116 deletions(-) create mode 100644 .gitignore create mode 100644 AUTHORS create mode 100644 COPYING create mode 100644 ChangeLog create mode 100644 INSTALL create mode 100644 Makefile.am create mode 100644 NEWS create mode 100755 autogen.sh create mode 100644 configure.ac create mode 100644 include/Makefile.am create mode 100644 m4/.place_holder delete mode 100644 makefile.linux create mode 100644 src/Makefile.am create mode 100644 src/lib/Makefile.am rename compress.c => src/lib/compress.c (98%) create mode 100644 src/utils/Makefile.am rename cplay.c => src/utils/cplay.c (100%) rename crec.c => src/utils/crec.c (100%) delete mode 100644 utils.c -- 2.1.0

2 8

Re: [alsa-devel] [PATCH 3/3] ASoC: rt5645: add device tree support
by Mark Brown 08 May '15

08 May '15

On Fri, May 08, 2015 at 11:13:39AM +0300, Mika Westerberg wrote: > Looks like some of the context is lost but if I decode the above > right, you want to know if there is anything that corresponds to > of_get_named_gpio() in the generic property API? > You can use fwnode_get_named_gpiod() if you need to extract a GPIO > directly from firmware node. Otherwise you should be using > devm_gpiod_get() and friends. Right, almost certainly devm_gpiod_get() in this case as I've said several times now :(

1 0

[alsa-devel] [PATCH] ASoC: dapm: Add missing mutex unlock
by Charles Keepax 08 May '15

08 May '15

The is a missing mutex unlock on the error path in snd_soc_dapm_get_enum_double. This was introduced in commit 561ed680b764 ("ASoC: dapm: Add support for autodisable mux controls"). This patch adds the missing unlock. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ea3348e..6374852 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3075,8 +3075,10 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); if (e->reg != SND_SOC_NOPM && dapm_kcontrol_is_powered(kcontrol)) { int ret = soc_dapm_read(dapm, e->reg, &reg_val); - if (ret) + if (ret) { + mutex_unlock(&card->dapm_mutex); return ret; + } } else { reg_val = dapm_kcontrol_get_value(kcontrol); } -- 1.7.2.5

2 1

Re: [alsa-devel] [PATCH v3 3/8] mfd: arizona: Add support for WM8998 and WM1814
by Richard Fitzgerald 08 May '15

08 May '15

On Thu, May 07, 2015 at 09:52:38AM +0100, Lee Jones wrote: > On Fri, 01 May 2015, Richard Fitzgerald wrote: > > > Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> > > --- > > drivers/mfd/Kconfig | 6 + > > drivers/mfd/Makefile | 3 + > > drivers/mfd/arizona-core.c | 105 ++- > > drivers/mfd/arizona-i2c.c | 8 + > > drivers/mfd/arizona-irq.c | 9 + > > drivers/mfd/arizona.h | 5 + > > drivers/mfd/wm8998-tables.c | 1592 +++++++++++++++++++++++++++++++++ > > include/linux/mfd/arizona/core.h | 3 + > > include/linux/mfd/arizona/pdata.h | 2 + > > include/linux/mfd/arizona/registers.h | 220 +++++ > > 10 files changed, 1945 insertions(+), 8 deletions(-) > > create mode 100644 drivers/mfd/wm8998-tables.c > > How orthogonal is this patch? Can I just take it? > > For my own reference: > Acked-by: Lee Jones <lee.jones(a)linaro.org> This patches code added by patch 1 "mfd: arizona: Split INx_MODE into two fields" so you'll need to take that one as well. But if you are ok with that change too then yes it's ok to take both of them into mfd.

1 0

Re: [alsa-devel] Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 08 May '15

08 May '15

> > > My laptop has built-in front left, front right, rear left, rear right and > > > subwoofer and something is strange. > > > When I set channel output to 5.1 or 4.1 and external speakers are > > > disconnected, then front left and front right work as expected but > > > > built-in > > > > > subwoofer is heard only if I test rear left channel (wrong channel > > > > mapping?) > > > > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pc > > i/hda/hda_codec.c?id=cd4035e814b04c009ece9939396d06370c53ed44 > > > > Seem only two pairs of stereo speakers and internal subwoofer at left > > channel before this patch > > > > ALSA: hda - Enable surround speakers (when line out is also present) > > On my laptop case "Virtual SurroundSound" is written . When I test front left > and rear left, I hear distinct sound, so it would mean it's from two different > sources of sound. The driver use more than two channels only after David 's patch But all internal speakers of the notebook are at front of the user, The 3D sound effect is quite different from your external 5.1 speakers Do you mean your 5930g need champ {FL FR RL RR LFE }? This can be easily implemented by using "swap center/lfe" switch for those codec support R/L swap http://git.alsa-project.org/?p=alsa-lib.git;a=blob_plain;f=test/chmap.c;hb=… http://git.alsa-project.org/?p=alsa-lib.git;a=tree;f=src/conf/pcm;hb=HEAD If codec does not support R/L swap, your 4.1 swap chmap need 5.0 routing Do you expect pulseaudio show 4.1 swap(5.0) profile for internal speakers and 4.1 and 5.1 profiles for external speakers ?

1 0

[alsa-devel] Focusrite 6i6 invalid PCM device "front"
by Ramon Marco Navarro 08 May '15

08 May '15

# Background I can't seem to use any of the PCM device list defined automatically for my [Focusrite 6i6 device][1]. I tested this using `speaker-test`. # Tests: ## `speaker-test` output comparing between using [`sysdefault`][2] and [`front:USB`][3]. The `sysdefault` device emits sound for all of the formats, except for FLOAT_LE 44100Hz and 48000Hz. While front:USB doesn't emit anything at all. [1]: http://www.alsa-project.org/db/?f=b1c10accc89cd34dfdeb0d61b087a3cf413ea1a0 [2]: https://gist.githubusercontent.com/ramonmaruko/deb1918efa27eb860aa1/raw/f7a… [3]: https://gist.githubusercontent.com/ramonmaruko/1377846e39a7a6acd72f/raw/4d6…

1 0

Re: [alsa-devel] [PATCH v2 1/4] ASoC: Add gtm601 codec driver
by Mark Brown 07 May '15

07 May '15

On Wed, May 06, 2015 at 10:38:47PM +0200, Marek Belisko wrote: > This driver add PCM interface to a GTM601 UMTS modem chip. > There is no configuration interface. Applied, thanks. Please remember to CC maintainers on patches, you've missed Liam for some reason.

2 1

Re: [alsa-devel] [PATCH v2 2/4] Documentation: devicetree: sound: Add documentation for gtm601 codec
by Mark Brown 07 May '15

07 May '15

On Wed, May 06, 2015 at 10:38:48PM +0200, Marek Belisko wrote: > Add small documentation for gtm601 UMTS modem audio interface. git am complains that this patch is corrupt so I can't apply it. Please also use subject lines reflecting the style for the subsystem.

1 0

[alsa-devel] [PATCH 1/2] ASoC: dapm: Break out of widget search when source and sink are located
by Charles Keepax 07 May '15

07 May '15

Currently snd_soc_dapm_add_route will continue to search the widget list even after both the source and sink for the route have been located. This patch breaks out of the search when both are located giving a small improvement in probe time for drivers. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Note these patches depend on: commit 92fa12426741d52b39ec92ad77c9843d3fc2b3d6 Author: Lars-Peter Clausen <lars(a)metafoo.de> Date: Fri May 1 18:02:42 2015 +0200 ASoC: dapm: Add new widgets to the end of the widget list Thanks, Charles sound/soc/soc-dapm.c | 10 ++++++++-- 1 files changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a2e5f22..ea3348e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2617,14 +2617,20 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, list_for_each_entry(w, &dapm->card->widgets, list) { if (!wsink && !(strcmp(w->name, sink))) { wtsink = w; - if (w->dapm == dapm) + if (w->dapm == dapm) { wsink = w; + if (wsource) + break; + } continue; } if (!wsource && !(strcmp(w->name, source))) { wtsource = w; - if (w->dapm == dapm) + if (w->dapm == dapm) { wsource = w; + if (wsink) + break; + } } } /* use widget from another DAPM context if not found from this */ -- 1.7.2.5

4 12

[alsa-devel] Alsa triggering methods
by Andy Ng 07 May '15

07 May '15

Hi, I am working on a ALSA card driver that handles a DSP instance. The problem I have is that the triggering mechanism in ALSA to start a PCM has the interrupts disabled, and my dsp framework is actually acknowledging the start trigger with an event after 50 microseconds time that I can't handle because the IRQs are disabled. Is there any way when the trigger mechanism goes down to the card to be able to handle any acknowledgements what will happen within a sort period of time. I have seen bespoke triggering method but not sure if everyone is using it. Any ideas would be helpful. Best regards, Andreas

3 2

Re: [alsa-devel] [PATCH v2 RESEND] ARM: dts: Support audio on Exynos5422-odroidxu3 using simple-audio-card
by Inha Song 07 May '15

07 May '15

Hi Kukjin, Any comments about this patch? Best Regards, Inha Song. On Fri, 10 Apr 2015 14:48:13 +0530 Anand Moon <linux.amoon(a)gmail.com> wrote: > Hi Inha, > > Tested-by: Anand Moon <linux.amoon(a)gmail.com> > > -Anand Moon > > On 10 April 2015 at 14:32, Markus Reichl <m.reichl(a)fivetechno.de> wrote: > > Hi Inha, > > > > I have tested this on XU3 (Headphone output). > > > > Tested-by: Markus Reichl <m.reichl(a)fivetechno.de> > > > > Best Regards > > -- > > Markus Reichl > > > > Am 10.04.2015 um 09:32 schrieb Inha Song: > >> Add MAX98090 audio codec, I2S interface and the sound nodes to support > >> audio on Exynos5422 SoC Based Odroid-XU3 board. Now we can support audio > >> in Odroid-XU3 board using simple-audio-card DT binding. > >> > >> Signed-off-by: Inha Song <ideal.song(a)samsung.com> > >> Reviewed-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> > >> --- > >> arch/arm/boot/dts/exynos5420.dtsi | 9 +++++ > >> arch/arm/boot/dts/exynos5422-odroidxu3.dts | 59 ++++++++++++++++++++++++++++++ > >> 2 files changed, 68 insertions(+) > >> > >> diff --git a/arch/arm/boot/dts/exynos5420.dtsi b/arch/arm/boot/dts/exynos5420.dtsi > >> index ac6f860..73886d6 100644 > >> --- a/arch/arm/boot/dts/exynos5420.dtsi > >> +++ b/arch/arm/boot/dts/exynos5420.dtsi > >> @@ -415,6 +415,9 @@ > >> <&clock_audss EXYNOS_I2S_BUS>, > >> <&clock_audss EXYNOS_SCLK_I2S>; > >> clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; > >> + #clock-cells = <1>; > >> + clock-output-names = "i2s_cdclk0"; > >> + #sound-dai-cells = <1>; > >> samsung,idma-addr = <0x03000000>; > >> pinctrl-names = "default"; > >> pinctrl-0 = <&i2s0_bus>; > >> @@ -429,6 +432,9 @@ > >> dma-names = "tx", "rx"; > >> clocks = <&clock CLK_I2S1>, <&clock CLK_SCLK_I2S1>; > >> clock-names = "iis", "i2s_opclk0"; > >> + #clock-cells = <1>; > >> + clock-output-names = "i2s_cdclk1"; > >> + #sound-dai-cells = <1>; > >> pinctrl-names = "default"; > >> pinctrl-0 = <&i2s1_bus>; > >> status = "disabled"; > >> @@ -442,6 +448,9 @@ > >> dma-names = "tx", "rx"; > >> clocks = <&clock CLK_I2S2>, <&clock CLK_SCLK_I2S2>; > >> clock-names = "iis", "i2s_opclk0"; > >> + #clock-cells = <1>; > >> + clock-output-names = "i2s_cdclk2"; > >> + #sound-dai-cells = <1>; > >> pinctrl-names = "default"; > >> pinctrl-0 = <&i2s2_bus>; > >> status = "disabled"; > >> diff --git a/arch/arm/boot/dts/exynos5422-odroidxu3.dts b/arch/arm/boot/dts/exynos5422-odroidxu3.dts > >> index edc25cf..c038f0d 100644 > >> --- a/arch/arm/boot/dts/exynos5422-odroidxu3.dts > >> +++ b/arch/arm/boot/dts/exynos5422-odroidxu3.dts > >> @@ -11,6 +11,7 @@ > >> */ > >> > >> /dts-v1/; > >> +#include <dt-bindings/sound/samsung-i2s.h> > >> #include "exynos5800.dtsi" > >> > >> / { > >> @@ -285,6 +286,64 @@ > >> rtc@101E0000 { > >> status = "okay"; > >> }; > >> + > >> + sound: sound { > >> + compatible = "simple-audio-card"; > >> + > >> + simple-audio-card,name = "Odroid-XU3"; > >> + simple-audio-card,widgets = > >> + "Headphone", "Headphone Jack", > >> + "Speakers", "Speakers"; > >> + simple-audio-card,routing = > >> + "Headphone Jack", "HPL", > >> + "Headphone Jack", "HPR", > >> + "Headphone Jack", "MICBIAS", > >> + "IN1", "Headphone Jack", > >> + "Speakers", "SPKL", > >> + "Speakers", "SPKR"; > >> + > >> + simple-audio-card,format = "i2s"; > >> + simple-audio-card,bitclock-master = <&link0_codec>; > >> + simple-audio-card,frame-master = <&link0_codec>; > >> + > >> + simple-audio-card,cpu { > >> + sound-dai = <&i2s0 0>; > >> + system-clock-frequency = <19200000>; > >> + }; > >> + > >> + link0_codec: simple-audio-card,codec { > >> + sound-dai = <&max98090>; > >> + clocks = <&i2s0 CLK_I2S_CDCLK>; > >> + }; > >> + }; > >> +}; > >> + > >> +&clock_audss { > >> + assigned-clocks = <&clock_audss EXYNOS_MOUT_AUDSS>, > >> + <&clock_audss EXYNOS_MOUT_I2S>, > >> + <&clock_audss EXYNOS_DOUT_AUD_BUS>; > >> + assigned-clock-parents = <&clock CLK_FIN_PLL>, > >> + <&clock_audss EXYNOS_MOUT_AUDSS>; > >> + assigned-clock-rates = <0>, > >> + <0>, > >> + <19200000>; > >> +}; > >> + > >> +&hsi2c_5 { > >> + status = "okay"; > >> + max98090: max98090@10 { > >> + compatible = "maxim,max98090"; > >> + reg = <0x10>; > >> + interrupt-parent = <&gpx3>; > >> + interrupts = <2 0>; > >> + clocks = <&i2s0 CLK_I2S_CDCLK>; > >> + clock-names = "mclk"; > >> + #sound-dai-cells = <0>; > >> + }; > >> +}; > >> + > >> +&i2s0 { > >> + status = "okay"; > >> }; > >> > >> &hdmi { > >> > > > -- > To unsubscribe from this list: send the line "unsubscribe linux-samsung-soc" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html

3 2

[alsa-devel] [PATCH 0/5] ASoC: Intel: let machines configure SSP configuration
by Vinod Koul 06 May '15

06 May '15

This series first add frame and data polarity to ssp config and then starts to use local values for ssp config and then eventually lets that be loaded by hw_params and updates by pcm format and tdm calls to allow machine t specify these parameters Vinod Koul (5): ASoC: Intel: add frame and data polarity to ssp config ASoC: Intel: use local values for ssp configuration ASoC: Intel: load hw_defaults in hw_params of ssp be ASoC: intel: add support for specifying PCM format ASoC: Intel: add support for configuring TDM slots for SSP sound/soc/intel/atom/sst-atom-controls.c | 167 +++++++++++++++++++++++---- sound/soc/intel/atom/sst-atom-controls.h | 7 ++ sound/soc/intel/atom/sst-mfld-platform-pcm.c | 47 +++++++- sound/soc/intel/atom/sst-mfld-platform.h | 2 + 4 files changed, 197 insertions(+), 26 deletions(-) -- 1.9.1

2 6

Re: [alsa-devel] Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 06 May '15

06 May '15

> > > > > > My problem is that when I set channel output to "Analog output > > > > surround > > > > > > > 5.1" > > > > > (or 2.1 or 4.1), I hear no sound in front speakers, but in case of > > > > 5.1 or > > > > > > > 4.1 > > > > You need pulseaudio lfe-filter for 2.1 > > > > http://cgit.freedesktop.org/pulseaudio/pulseaudio/log/?qt=grep&q=lfe-filter > > External 5.1 and 4.1 worked fine without lfe-filter. Fedora doesn't package lfe- > filter and I'm more interested to get my 5.1 configuration back. > > > do you mean surround21 need 6 channel instead of channels with 5.1 internal > > /external speaker ? > > > > http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=48f1b308cc66152eb > > 6db66742dd0d08d888cda8d;hp=5c4cd46810cef8850b037fca9e38ffd43b0bff22 > > No. When in KDE channel output is set to 2.1 it makes no difference what is set > in alsamixer. When my external speakers are disconnected built-in front left > and front right works on my laptop, but built-in subwoofer doesn't work There > is no difference what channel output is set in alsamixer. > > My laptop has built-in front left, front right, rear left, rear right and > subwoofer and something is strange. > When I set channel output to 5.1 or 4.1 and external speakers are > disconnected, then front left and front right work as expected but built-in > subwoofer is heard only if I test rear left channel (wrong channel mapping?) https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… Seem only two pairs of stereo speakers and internal subwoofer at left channel before this patch ALSA: hda - Enable surround speakers (when line out is also present)

1 0

[alsa-devel] [PATCH] ASoC: omap-twl4030: Use card DAPM context to access widgets
by Lars-Peter Clausen 06 May '15

06 May '15

Use the card DAPM context instead of the CODEC DAPM context since only card level widgets are accessed here. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/omap/omap-twl4030.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index 3673ada..7431314 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -159,9 +159,8 @@ static inline void twl4030_disconnect_pin(struct snd_soc_dapm_context *dapm, static int omap_twl4030_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = rtd->card; - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_context *dapm = &card->dapm; struct omap_tw4030_pdata *pdata = dev_get_platdata(card->dev); struct omap_twl4030 *priv = snd_soc_card_get_drvdata(card); int ret = 0; -- 1.8.0

2 1

[alsa-devel] [PATCH v3 0/7] ASoC: intel - add skylake PCM driver
by Vinod Koul 06 May '15

06 May '15

This patch series adds the hda codec wrapper first followed by asoc hda controller library, then SKL PCM driver and last decouples the controller for splitting the links Jeeja KP (7): ASoC: hda - add soc hda codec driver wrapper ALSA: hda - add new HDA registers ASoC: hda - adds SoC controller and stream operations ASoC: intel - add Skylake HDA platform driver ASoC: intel - add Skylake HDA audio driver ASoC: intel - add makefile support for SKL driver ASoC: intel - adds support for decoupled mode in skl driver include/sound/hda_register.h | 87 +++ include/sound/soc-hdaudio.h | 343 ++++++++++++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/hda/Kconfig | 3 + sound/soc/hda/Makefile | 3 + sound/soc/hda/soc-hda-codec.c | 85 +++ sound/soc/hda/soc-hda-codec.h | 44 ++ sound/soc/hda/soc-hdac-controller.c | 289 ++++++++++ sound/soc/hda/soc-hdac-stream.c | 390 ++++++++++++++ sound/soc/intel/Kconfig | 20 + sound/soc/intel/Makefile | 1 + sound/soc/intel/skylake/Makefile | 3 + sound/soc/intel/skylake/hda-skl-pcm.c | 933 +++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/hda-skl.c | 743 ++++++++++++++++++++++++++ sound/soc/intel/skylake/hda-skl.h | 74 +++ 16 files changed, 3020 insertions(+) create mode 100644 include/sound/soc-hdaudio.h create mode 100644 sound/soc/hda/Kconfig create mode 100644 sound/soc/hda/Makefile create mode 100644 sound/soc/hda/soc-hda-codec.c create mode 100644 sound/soc/hda/soc-hda-codec.h create mode 100644 sound/soc/hda/soc-hdac-controller.c create mode 100644 sound/soc/hda/soc-hdac-stream.c create mode 100644 sound/soc/intel/skylake/Makefile create mode 100644 sound/soc/intel/skylake/hda-skl-pcm.c create mode 100644 sound/soc/intel/skylake/hda-skl.c create mode 100644 sound/soc/intel/skylake/hda-skl.h -- 1.7.9.5

3 25

[alsa-devel] [PATCH 1/4] ASoC: dapm: Add new widgets to the end of the widget list
by Lars-Peter Clausen 06 May '15

06 May '15

Currently new widgets are appended to the beginning of the cards widget list. This has the effect that widgets that are created while iterating over the widget list in snd_soc_dapm_new_widgets() (like e.g. the auto-disable widgets) are not covered during that invocation of the function. If no further invocations of snd_soc_dapm_new_widgets() happen these widgets will not be fully initialized and e.g. no debugfs entries are created for them. By adding new widgets to the end of the widget list we make sure that widgets that are created in snd_soc_dapm_new_widgets() will still be handled during the same snd_soc_dapm_new_widgets() invocation and are always fully initialized. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- This series depends on Charles' mux auto-disable series. --- 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 79e6cf4..5c159f4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3252,7 +3252,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); INIT_LIST_HEAD(&w->dirty); - list_add(&w->list, &dapm->card->widgets); + list_add_tail(&w->list, &dapm->card->widgets); w->inputs = -1; w->outputs = -1; -- 1.8.0

2 5

[alsa-devel] [PATCH v3 1/5] ASoC: Correct typo in SOC_VALUE_ENUM_SINGLE macro
by Charles Keepax 06 May '15

06 May '15

xnitmes is clearly intended to be xnitems, but all other macros just refer to this as xitems, so change it to that. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v2: - Use snd_soc_enum_item_to_val to get zero value from mux to account for non-value enums - Improve handling in snd_soc_dapm_put_enum_double of the situation where the register and widget views of the control values don't match Thanks, Charles include/sound/soc.h | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a73d855..45cfd69 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -190,8 +190,8 @@ #define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xitems, xtexts, xvalues) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .mask = xmask, .items = xitems, .texts = xtexts, .values = xvalues} -#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xnitmes, xtexts, xvalues) \ - SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xnitmes, xtexts, xvalues) +#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xitems, xtexts, xvalues) \ + SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xitems, xtexts, xvalues) #define SOC_ENUM_SINGLE_VIRT(xitems, xtexts) \ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, xitems, xtexts) #define SOC_ENUM(xname, xenum) \ -- 1.7.2.5

3 11

[alsa-devel] [PATCH V4 0/5] tas571x amplifier driver
by Kevin Cernekee 06 May '15

06 May '15

V3->V4: Eliminate unnecessary dev->of_node NULL check. Move the "should I sync this register?" test into a common helper function. Clear map->no_sync_defaults at the end of the regcache_sync functions. Kevin Cernekee (5): regmap: Add a helper function for regcache sync test regmap: Use regcache_mark_dirty() to indicate power loss or reset ASoC: tas571x: Add DT binding document ASoC: tas571x: New driver for TI TAS571x power amplifiers MAINTAINERS: Add entry for tas571x ASoC codec driver .../devicetree/bindings/sound/tas571x.txt | 41 ++ MAINTAINERS | 6 + drivers/base/regmap/internal.h | 3 + drivers/base/regmap/regcache.c | 45 +- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas571x.c | 515 +++++++++++++++++++++ sound/soc/codecs/tas571x.h | 33 ++ 8 files changed, 635 insertions(+), 15 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tas571x.txt create mode 100644 sound/soc/codecs/tas571x.c create mode 100644 sound/soc/codecs/tas571x.h -- 2.2.0.rc0.207.ga3a616c

2 10

[alsa-devel] [PATCH] ASoC: Intel: Enabled button jack for BSW platform with rt5650 codec
by yang.a.fang＠intel.com 06 May '15

06 May '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> rt5650 codec supports 4 buttons detections so enabled it Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 23 ++++++++++++++++++++++- 1 file changed, 22 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 8f96c21..1468c3d 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -43,6 +43,7 @@ struct cht_acpi_card { struct cht_mc_private { struct snd_soc_jack hp_jack; struct snd_soc_jack mic_jack; + struct snd_soc_jack btn_jack; struct cht_acpi_card *acpi_card; }; @@ -182,6 +183,7 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) struct snd_soc_codec *codec = runtime->codec; struct snd_soc_dai *codec_dai = runtime->codec_dai; struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); + struct snd_soc_jack *btn_jack; /* Select clk_i2s1_asrc as ASRC clock source */ rt5645_sel_asrc_clk_src(codec, @@ -214,7 +216,26 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) return ret; } - rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack, NULL); + if (ctx->acpi_card->codec_type == CODEC_TYPE_RT5650) { + + ret = snd_soc_card_jack_new(runtime->card, "Button Jack", + SND_JACK_BTN_0 | + SND_JACK_BTN_1 | + SND_JACK_BTN_2 | + SND_JACK_BTN_3, &ctx->btn_jack, + NULL, 0); + if (ret) { + dev_err(runtime->dev, + "Btn jack creation failed %d\n", ret); + return ret; + } + + btn_jack = &ctx->btn_jack; + } else { + btn_jack = NULL; + } + + rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack, btn_jack); return ret; } -- 1.7.9.5

5 12

[alsa-devel] [PATCH] ASoC: tas571x: Eliminate redundant dev->of_node NULL check
by Kevin Cernekee 06 May '15

06 May '15

of_match_device() checks if dev->of_node is NULL, so we don't need to do it again in the probe function. Signed-off-by: Kevin Cernekee <cernekee(a)chromium.org> --- This applies to the broonie/sound.git for-next branch. sound/soc/codecs/tas571x.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index b187ea53a7f9..85bcc374c8e8 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -377,6 +377,7 @@ static int tas571x_i2c_probe(struct i2c_client *client, { struct tas571x_private *priv; struct device *dev = &client->dev; + const struct of_device_id *of_id; int i, ret; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); @@ -384,18 +385,12 @@ static int tas571x_i2c_probe(struct i2c_client *client, return -ENOMEM; i2c_set_clientdata(client, priv); - if (dev->of_node) { - const struct of_device_id *of_id; - - of_id = of_match_device(tas571x_of_match, dev); - if (of_id) - priv->chip = of_id->data; - } - - if (!priv->chip) { + of_id = of_match_device(tas571x_of_match, dev); + if (!of_id) { dev_err(dev, "Unknown device type\n"); return -EINVAL; } + priv->chip = of_id->data; priv->mclk = devm_clk_get(dev, "mclk"); if (IS_ERR(priv->mclk) && PTR_ERR(priv->mclk) != -ENOENT) { -- 2.2.0.rc0.207.ga3a616c

2 1

[alsa-devel] [PATCH] ALSA: hda - Add headset mic quirk for Dell Inspiron 5548
by David Henningsson 06 May '15

06 May '15

This enables the headset microphone on Dell Inspiron 5548, or at least some variants of it. Cc: stable(a)kernel.org BugLink: https://bugs.launchpad.net/bugs/1452175 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 06199e4..c5413e9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5337,6 +5337,13 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x17, 0x40000000}, {0x1d, 0x40700001}, {0x21, 0x02211050}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell Inspiron 5548", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC255_STANDARD_PINS, + {0x12, 0x90a60180}, + {0x14, 0x90170130}, + {0x17, 0x40000000}, + {0x1d, 0x40700001}, + {0x21, 0x02211040}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC256_STANDARD_PINS, {0x13, 0x40000000}), -- 1.9.1

1 0

[alsa-devel] [PATCH v11 0/3] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 06 May '15

06 May '15

Based on 4.1-rc1 v11: - reduce the patch series to adding the tda998x codec only v10: - add the generic dt-card - define the audio ports from a DT graph of ports (Russell King) - reuse HDMI constants (Andrew Jackson - Jyri Sarha) - alloc rate_constraints in codec (Jyri Sarha) - fix bad number of channels (Jyri Sarha) - correct codec generation from config (Russell King - Jyri Sarha) - no module init/exit (Russell King) v9: - back to a TDA998x specific CODEC - more comments - change magic values to constants v8: - change some comments about the patches v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (3): drm/i2c: tda998x: Add support of a DT graph of ports drm/i2c: tda998x: Change drvdata for audio extension ASoC: tda998x: add a codec to the HDMI transmitter .../devicetree/bindings/drm/i2c/tda998x.txt | 51 +++++ drivers/gpu/drm/i2c/tda998x_drv.c | 220 +++++++++++++++++++-- include/sound/tda998x.h | 30 +++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 174 ++++++++++++++++ 6 files changed, 471 insertions(+), 11 deletions(-) create mode 100644 include/sound/tda998x.h create mode 100644 sound/soc/codecs/tda998x.c -- 2.1.4

2 4

Re: [alsa-devel] [PATCH 2/2] ASoC: zx: Add zx296702 SPDIF support
by Mark Brown 06 May '15

06 May '15

On Tue, May 05, 2015 at 09:43:56AM +0800, Jun Nie wrote: > 2015-05-04 21:54 GMT+08:00 Mark Brown <broonie(a)kernel.org>: > > So this special HDMI S/PDIF controller is directly integrated with the > > HDMI IP in the SoC and not usable as a generic S/PDIF controller? That > > seems to mirror system designs which have an external HDMI encoder which > > use S/PDIF or I2S to connect the audio portion to an external device. > The first SPDIF controller on ZTE SOC is dedicated for HDMI and the > 2nd one is for external output only via physical pad. The two > controller IPs are identical, also independent with each other and > HDMI in register address region. OK, so this does sound like the software should like one of the systems with an external HDMI encoder and the HDMI handled separately rather than in this driver anyway. > > I'd suggest for future versions reorganizing things so that you add the > > S/PDIF driver with only S/PDIF support and then have a separate patch > > that layers on the HDMI support. > Reasonable suggestion. Then I can have more time to think about HDMI > part. More comments is welcome if any. This sounds like the software should look like a TDA998x (which is having patches posted still) so just have the S/PDIF controllers as pure S/PDIF controllers and then a separate driver for the HDMI part that it's hooked up to which deals with the audio configuration for the HDMI side.

1 0

Re: [alsa-devel] Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 05 May '15

05 May '15

> > > seem internal subwoofer only use one channel and don't have any stereo to > > mono downmix according to your patch > > > > It is not easy to assign different name to internal 4.1 and external 5.1 > > speaker when alc888 only has five DAC > > > > Some playback volume control must be shared > > If you know what's missing, you could send a few lines of code and I will test > it, otherwise I'll have to experiment on my own. https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/patch/sound/pc… c3e837bbcc03e39991da5dd1f791c0 6cde9f7a45 As your patch indicate internal subwoofer use external center speaker channel and your codec does not support L/R swap You may need to reserve Audio mixer 0x26 for internal subwoofer playback volume by restrict the connection of the internal rear speaker excluding 0x26 so that last DAC can be assigned to the internal subwoofer similar to alc889_fixup_dac_route void alc888_fixup_dac_route(struct hda_codec *codec, const struct alc_fixup *fix, int action) { if (action == ALC_FIXUP_ACT_PRE_PROBE) { hda_nid_t conn1[4] = { 0x0c, 0x0d, 0x0e, 0x0f}; snd_hda_override_conn_list(codec, 0x17, 4, conn1); } else if (action == ALC_FIXUP_ACT_PROBE) { conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 }; snd_hda_override_conn_list(codec, 0x17, 5, conn); } } > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/patch/sound/pc… c3e837bbcc03e39991da5dd1f791c0 6cde9f7a45 > > > > Do you mean internal subwoofer has low pass filter > > I don't know. How to check it? > > > > what do you mean when you say 4.1 worked fine ? > > External 5.1 speakers working as 4.1 speakers because such profile was set. > > > > The channel map of internal 4.1 speaker are different from external 5.1 > > speaker > > Recently I also made such conclusion. > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=2d3… Seem 4.1 is not defined in snd_pcm_std_chmap and you need to define and add your 4.1 chmap similar to 2.1 https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=ee8…

1 0

[alsa-devel] [RFC PATCH] ASoC: wm8741: Add differential mono mode support
by Sergej Sawazki 05 May '15

05 May '15

The WM8741 DAC supports several differential output modes (stereo, stereo reversed, mono left, mono right). Add platform data and DT bindings to configure it. Signed-off-by: Sergej Sawazki <ce3a(a)gmx.de> --- Documentation/devicetree/bindings/sound/wm8741.txt | 11 ++ sound/soc/codecs/wm8741.c | 129 ++++++++++++++++++--- sound/soc/codecs/wm8741.h | 10 ++ 3 files changed, 137 insertions(+), 13 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/wm8741.txt b/Documentation/devicetree/bindings/sound/wm8741.txt index 74bda58..a133154 100644 --- a/Documentation/devicetree/bindings/sound/wm8741.txt +++ b/Documentation/devicetree/bindings/sound/wm8741.txt @@ -10,9 +10,20 @@ Required properties: - reg : the I2C address of the device for I2C, the chip select number for SPI. +Optional properties: + + - diff-mode: Differential output mode configuration. Default value for field + DIFF in register R8 (MODE_CONTROL_2). If absent, the default is 0, shall be: + 0 = stereo + 1 = mono left + 2 = stereo reversed + 3 = mono right + Example: codec: wm8741@1a { compatible = "wlf,wm8741"; reg = <0x1a>; + + diff-mode = <3>; }; diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 9e71c76..9b777a6 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -41,6 +41,7 @@ static const char *wm8741_supply_names[WM8741_NUM_SUPPLIES] = { /* codec private data */ struct wm8741_priv { + struct wm8741_platform_data pdata; struct regmap *regmap; struct regulator_bulk_data supplies[WM8741_NUM_SUPPLIES]; unsigned int sysclk; @@ -87,13 +88,27 @@ static int wm8741_reset(struct snd_soc_codec *codec) static const DECLARE_TLV_DB_SCALE(dac_tlv_fine, -12700, 13, 0); static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 400, 0); -static const struct snd_kcontrol_new wm8741_snd_controls[] = { +static const struct snd_kcontrol_new wm8741_snd_controls_stereo[] = { SOC_DOUBLE_R_TLV("Fine Playback Volume", WM8741_DACLLSB_ATTENUATION, WM8741_DACRLSB_ATTENUATION, 1, 255, 1, dac_tlv_fine), SOC_DOUBLE_R_TLV("Playback Volume", WM8741_DACLMSB_ATTENUATION, WM8741_DACRMSB_ATTENUATION, 0, 511, 1, dac_tlv), }; +static const struct snd_kcontrol_new wm8741_snd_controls_mono_left[] = { +SOC_SINGLE_TLV("Fine Playback Volume Left", WM8741_DACLLSB_ATTENUATION, + 1, 255, 1, dac_tlv_fine), +SOC_SINGLE_TLV("Playback Volume Left", WM8741_DACLMSB_ATTENUATION, + 0, 511, 1, dac_tlv), +}; + +static const struct snd_kcontrol_new wm8741_snd_controls_mono_right[] = { + SOC_SINGLE_TLV("Fine Playback Volume Right", WM8741_DACRLSB_ATTENUATION, + 1, 255, 1, dac_tlv_fine), + SOC_SINGLE_TLV("Playback Volume Right", WM8741_DACRMSB_ATTENUATION, + 0, 511, 1, dac_tlv), +}; + static const struct snd_soc_dapm_widget wm8741_dapm_widgets[] = { SND_SOC_DAPM_DAC("DACL", "Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DACR", "Playback", SND_SOC_NOPM, 0, 0), @@ -398,7 +413,7 @@ static struct snd_soc_dai_driver wm8741_dai = { .name = "wm8741", .playback = { .stream_name = "Playback", - .channels_min = 2, /* Mono modes not yet supported */ + .channels_min = 1, .channels_max = 2, .rates = WM8741_RATES, .formats = WM8741_FORMATS, @@ -416,6 +431,65 @@ static int wm8741_resume(struct snd_soc_codec *codec) #define wm8741_resume NULL #endif +static int wm8741_configure(struct snd_soc_codec *codec) +{ + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + + /* Configure differential mode */ + switch (wm8741->pdata.diff_mode) { + case WM8741_DIFF_MODE_STEREO: + case WM8741_DIFF_MODE_STEREO_REVERSED: + case WM8741_DIFF_MODE_MONO_LEFT: + case WM8741_DIFF_MODE_MONO_RIGHT: + snd_soc_update_bits(codec, WM8741_MODE_CONTROL_2, + WM8741_DIFF_MASK, + wm8741->pdata.diff_mode << WM8741_DIFF_SHIFT); + break; + default: + return -EINVAL; + } + + /* Change some default settings - latch VU */ + snd_soc_update_bits(codec, WM8741_DACLLSB_ATTENUATION, + WM8741_UPDATELL, WM8741_UPDATELL); + snd_soc_update_bits(codec, WM8741_DACLMSB_ATTENUATION, + WM8741_UPDATELM, WM8741_UPDATELM); + snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, + WM8741_UPDATERL, WM8741_UPDATERL); + snd_soc_update_bits(codec, WM8741_DACRMSB_ATTENUATION, + WM8741_UPDATERM, WM8741_UPDATERM); + + return 0; +} + +static int wm8741_add_controls(struct snd_soc_codec *codec) +{ + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + + switch (wm8741->pdata.diff_mode) { + case WM8741_DIFF_MODE_STEREO: + case WM8741_DIFF_MODE_STEREO_REVERSED: + snd_soc_add_codec_controls(codec, + wm8741_snd_controls_stereo, + ARRAY_SIZE(wm8741_snd_controls_stereo)); + break; + case WM8741_DIFF_MODE_MONO_LEFT: + snd_soc_add_codec_controls(codec, + wm8741_snd_controls_mono_left, + ARRAY_SIZE(wm8741_snd_controls_mono_left)); + break; + case WM8741_DIFF_MODE_MONO_RIGHT: + snd_soc_add_codec_controls(codec, + wm8741_snd_controls_mono_right, + ARRAY_SIZE(wm8741_snd_controls_mono_right)); + break; + default: + return -EINVAL; + } + + return 0; +} + static int wm8741_probe(struct snd_soc_codec *codec) { struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); @@ -434,15 +508,17 @@ static int wm8741_probe(struct snd_soc_codec *codec) goto err_enable; } - /* Change some default settings - latch VU */ - snd_soc_update_bits(codec, WM8741_DACLLSB_ATTENUATION, - WM8741_UPDATELL, WM8741_UPDATELL); - snd_soc_update_bits(codec, WM8741_DACLMSB_ATTENUATION, - WM8741_UPDATELM, WM8741_UPDATELM); - snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, - WM8741_UPDATERL, WM8741_UPDATERL); - snd_soc_update_bits(codec, WM8741_DACRMSB_ATTENUATION, - WM8741_UPDATERM, WM8741_UPDATERM); + ret = wm8741_configure(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to change default settings\n"); + goto err_enable; + } + + ret = wm8741_add_controls(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to add controls\n"); + goto err_enable; + } dev_dbg(codec->dev, "Successful registration\n"); return ret; @@ -467,8 +543,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8741 = { .remove = wm8741_remove, .resume = wm8741_resume, - .controls = wm8741_snd_controls, - .num_controls = ARRAY_SIZE(wm8741_snd_controls), .dapm_widgets = wm8741_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8741_dapm_widgets), .dapm_routes = wm8741_dapm_routes, @@ -493,6 +567,23 @@ static const struct regmap_config wm8741_regmap = { .readable_reg = wm8741_readable, }; +static int wm8741_set_pdata(struct device *dev, struct wm8741_priv *wm8741) +{ + const struct wm8741_platform_data *pdata = dev_get_platdata(dev); + u32 diff_mode; + + if (dev->of_node) { + if (of_property_read_u32(dev->of_node, "diff-mode", &diff_mode) + >= 0) + wm8741->pdata.diff_mode = diff_mode; + } else { + if (pdata != NULL) + memcpy(&wm8741->pdata, pdata, sizeof(wm8741->pdata)); + } + + return 0; +} + #if IS_ENABLED(CONFIG_I2C) static int wm8741_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -522,6 +613,12 @@ static int wm8741_i2c_probe(struct i2c_client *i2c, return ret; } + wm8741_set_pdata(&i2c->dev, wm8741); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to set pdata: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, wm8741); ret = snd_soc_register_codec(&i2c->dev, @@ -582,6 +679,12 @@ static int wm8741_spi_probe(struct spi_device *spi) return ret; } + wm8741_set_pdata(&spi->dev, wm8741); + if (ret != 0) { + dev_err(&spi->dev, "Failed to set pdata: %d\n", ret); + return ret; + } + spi_set_drvdata(spi, wm8741); ret = snd_soc_register_codec(&spi->dev, diff --git a/sound/soc/codecs/wm8741.h b/sound/soc/codecs/wm8741.h index 56c1b1d..c8835f6 100644 --- a/sound/soc/codecs/wm8741.h +++ b/sound/soc/codecs/wm8741.h @@ -194,6 +194,12 @@ #define WM8741_DITHER_SHIFT 0 /* DITHER - [1:0] */ #define WM8741_DITHER_WIDTH 2 /* DITHER - [1:0] */ +/* DIFF field values */ +#define WM8741_DIFF_MODE_STEREO 0 /* stereo normal */ +#define WM8741_DIFF_MODE_STEREO_REVERSED 2 /* stereo reversed */ +#define WM8741_DIFF_MODE_MONO_LEFT 1 /* mono left */ +#define WM8741_DIFF_MODE_MONO_RIGHT 3 /* mono right */ + /* * R32 (0x20) - ADDITONAL_CONTROL_1 */ @@ -208,4 +214,8 @@ #define WM8741_SYSCLK 0 +struct wm8741_platform_data { + u32 diff_mode; /* Differential Output Mode */ +}; + #endif -- 1.9.1

3 7

[alsa-devel] [PATCH] ALSA: hda/tegra - Fix oops
by Thierry Reding 05 May '15

05 May '15

From: Thierry Reding <treding(a)nvidia.com> Commit a41d122449be ("ALSA: hda - Embed bus into controller object") introduced a regression in the Tegra HDA driver that causes the following oops during boot: [ 2.333458] Unable to handle kernel NULL pointer dereference at virtual address 000004c4 [ 2.341537] pgd = c0004000 [ 2.344312] [000004c4] *pgd=00000000 [ 2.347898] Internal error: Oops: 5 [#1] PREEMPT SMP ARM [ 2.353200] Modules linked in: [ 2.356264] CPU: 0 PID: 1 Comm: swapper/0 Tainted: G W 4.1.0-rc2-next-20150505-00344-g8577890defbf #79 [ 2.366682] Hardware name: NVIDIA Tegra SoC (Flattened Device Tree) [ 2.372939] task: ee0d8b40 ti: ee0da000 task.ti: ee0da000 [ 2.378336] PC is at azx_bus_init+0x18/0xf4 [ 2.382516] LR is at hda_tegra_probe+0x6c/0x478 [ 2.387043] pc : [<c06156c4>] lr : [<c061cf00>] psr: 60000113 [ 2.387043] sp : ee0dbe38 ip : 00000000 fp : 00000000 [ 2.398501] r10: ed874c00 r9 : 000000fd r8 : 00000000 [ 2.403717] r7 : ed874c10 r6 : 00000000 r5 : 00000000 r4 : ed016810 [ 2.410232] r3 : c08a2ad4 r2 : c08a1ea0 r1 : 00000000 r0 : ed016810 [ 2.416750] Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel [ 2.424046] Control: 10c5387d Table: 8000406a DAC: 00000015 [ 2.429783] Process swapper/0 (pid: 1, stack limit = 0xee0da210) [ 2.435778] Stack: (0xee0dbe38 to 0xee0dc000) [ 2.440129] be20: 00000000 ed016810 [ 2.448297] be40: 00000000 c061cf00 00000000 ee0dbe5c ed8735d0 c0a7bc48 ed02fd50 ed016000 [ 2.456462] be60: c1250164 ed874c10 c0c66bf8 fffffdfb 00000000 000000fd c0b8dc98 c046664c [ 2.464628] be80: c0466608 c1250164 ed874c10 00000000 c0c66bf8 c0464eb4 ed874c10 c0c66bf8 [ 2.472793] bea0: ed874c44 c0c43458 00000000 c04650d0 00000000 c0c66bf8 c046503c c04633b4 [ 2.480959] bec0: ee11bea4 ed85f390 c0c66bf8 ed017ac0 00000000 c0464634 c0ab2b7c c0c66bf8 [ 2.489125] bee0: c0bfde20 c0c66bf8 c0bfde20 ed01ce40 c0b7b414 c04656e8 c04665b0 c0bfde20 [ 2.497291] bf00: c0bfde20 c0009770 ee0d8b40 c0c02488 60000113 00000000 00000000 00000003 [ 2.505458] bf20: 00000000 c0c02488 60000113 00000000 c0b54598 c0b16a90 ef7fcc57 c0041228 [ 2.513624] bf40: c0a9150c ef7fcc5f 00000006 00000006 00000000 c0bf1fa8 c0bf2354 00000006 [ 2.521790] bf60: c0b8dc90 c0c7c000 c0c7c000 c0b8dc98 00000000 c0b54dd8 00000006 00000006 [ 2.529956] bf80: c0b54598 00000000 00000000 c07ff08c 00000000 00000000 00000000 00000000 [ 2.538122] bfa0: 00000000 c07ff094 00000000 c000f5a0 00000000 00000000 00000000 00000000 [ 2.546286] bfc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 [ 2.554451] bfe0: 00000000 00000000 00000000 00000000 00000013 00000000 fffff7ff c013f264 [ 2.562624] [<c06156c4>] (azx_bus_init) from [<c061cf00>] (hda_tegra_probe+0x6c/0x478) [ 2.570535] [<c061cf00>] (hda_tegra_probe) from [<c046664c>] (platform_drv_probe+0x44/0xa4) [ 2.578879] [<c046664c>] (platform_drv_probe) from [<c0464eb4>] (driver_probe_device+0x174/0x2b8) [ 2.587739] [<c0464eb4>] (driver_probe_device) from [<c04650d0>] (__driver_attach+0x94/0x98) [ 2.596172] [<c04650d0>] (__driver_attach) from [<c04633b4>] (bus_for_each_dev+0x6c/0xa0) [ 2.604342] [<c04633b4>] (bus_for_each_dev) from [<c0464634>] (bus_add_driver+0x148/0x1f0) [ 2.612597] [<c0464634>] (bus_add_driver) from [<c04656e8>] (driver_register+0x78/0xf8) [ 2.620593] [<c04656e8>] (driver_register) from [<c0009770>] (do_one_initcall+0x8c/0x1d4) [ 2.628765] [<c0009770>] (do_one_initcall) from [<c0b54dd8>] (kernel_init_freeable+0x144/0x1e4) [ 2.637459] [<c0b54dd8>] (kernel_init_freeable) from [<c07ff094>] (kernel_init+0x8/0xe8) [ 2.645543] [<c07ff094>] (kernel_init) from [<c000f5a0>] (ret_from_fork+0x14/0x34) This is caused by azx_bus_init() trying to dereference chip->card, which for the Tegra driver doesn't get initialized until sometime later during the call to hda_tegra_create(). Fix this by mimicking the behaviour of the Intel driver and defer HDA bus initialization until right before the call to snd_device_new(). Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 801e9fb4a467..db0bb50fa5b9 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -439,6 +439,10 @@ static int hda_tegra_create(struct snd_card *card, chip->single_cmd = false; chip->snoop = true; + err = azx_bus_init(chip, NULL, &hda_tegra_io_ops); + if (err < 0) + return err; + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { dev_err(card->dev, "Error creating device\n"); @@ -475,10 +479,6 @@ static int hda_tegra_probe(struct platform_device *pdev) return err; } - err = azx_bus_init(chip, NULL, &hda_tegra_io_ops); - if (err < 0) - goto out_free; - err = hda_tegra_create(card, driver_flags, hda); if (err < 0) goto out_free; -- 2.3.5

2 1

Re: [alsa-devel] Add ALC298 alias name for Dell
by Takashi Iwai 05 May '15

05 May '15

At Tue, 5 May 2015 07:19:04 +0000, Kailang wrote: > > Hi Takashi, > > Dell has new platform use ALC298 codec. > ALC298 for Dell is call ALC3266. Thanks, applied. Takashi > > BR, > Kailang > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 0000-add-alc298-alias-name-for-Dell.patch <application/octet-stream (base64)>] >

1 0

[alsa-devel] [PATCH] ASoC: Intel: Fixed kbuild warnings
by yang.a.fang＠intel.com 05 May '15

05 May '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> fix following sparse warnings: (new ones prefixed by >>) >> sound/soc/intel/boards/cht_bsw_max98090_ti.c:168:37: sparse: >> incorrect type in argument 2 (different base types) sound/soc/intel/boards/cht_bsw_max98090_ti.c:168:37: expected unsigned int [unsigned] val sound/soc/intel/boards/cht_bsw_max98090_ti.c:168:37: got restricted snd_pcm_format_t [usertype] <noident> Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 4 +--- 1 file changed, 1 insertion(+), 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 3c518b1..1be0794 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -163,9 +163,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP2 to 24-bit */ - snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S24_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/6] ASoC: rt5645: remove RT5645_I2S_BCLK_MS1 control
by Bard Liao 05 May '15

05 May '15

RT5645_I2S_BCLK_MS1 (reg 0x73 [5]) is reserverd in rt5645 and rt5650. This function is move to TDM control. We can configure it by snd_soc_dai_set_tdm_slot's slot_width parameter. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 5 ++--- sound/soc/codecs/rt5645.h | 4 ---- 2 files changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 8bad9fe..33ebe88 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2104,9 +2104,8 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, switch (dai->id) { case RT5645_AIF1: - mask_clk = RT5645_I2S_BCLK_MS1_MASK | RT5645_I2S_PD1_MASK; - val_clk = bclk_ms << RT5645_I2S_BCLK_MS1_SFT | - pre_div << RT5645_I2S_PD1_SFT; + mask_clk = RT5645_I2S_PD1_MASK; + val_clk = pre_div << RT5645_I2S_PD1_SFT; snd_soc_update_bits(codec, RT5645_I2S1_SDP, (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 4473636..fa5c560 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -942,10 +942,6 @@ #define RT5645_I2S2_SDI_I2S2 (0x1 << 6) /* ADC/DAC Clock Control 1 (0x73) */ -#define RT5645_I2S_BCLK_MS1_MASK (0x1 << 15) -#define RT5645_I2S_BCLK_MS1_SFT 15 -#define RT5645_I2S_BCLK_MS1_32 (0x0 << 15) -#define RT5645_I2S_BCLK_MS1_64 (0x1 << 15) #define RT5645_I2S_PD1_MASK (0x7 << 12) #define RT5645_I2S_PD1_SFT 12 #define RT5645_I2S_PD1_1 (0x0 << 12) -- 1.8.1.1.439.g50a6b54

2 11

[alsa-devel] need information about dapm events, bias and mute management
by gpe 05 May '15

05 May '15

Hi, I'm working on an asoc codec but I'm facing some problems with the DAPM events the bias and the mute management. The kernel is a 3.4.91. Globally my codec is functionnal but I've some glitch. By using DAPM events (PRE_PMU, POST_PMU) and bias management I've suppressed the glitchs at the start but I've always some glitchs at the end of a playback. I'm trying to suppress them with PRE_PMD but it not works. When I test my code I see that some power off are done before the call of the SND_SOC_DAPM_PRE_PMD event. Is it normal ? Another question is regarding the mute management (snd_soc_dai_digital_mute) which is done after all the power off ... I expected it to be done just at the end of the playback. Last question about the bias management (set_bias_level in snd_soc_codec_driver). During the start I see the bias level incrementing but during the stop there is no action on the bias level. Why ? Regards. PS: I'm a newbie on codec development so don't hit to hard on my head if my questions are stupid. I've read many docs but I don't found a doc which describes the interactions between the APIs. PS2: sorry for my bad english. -- +-------+ | gpe92 | +-------+

1 0

[alsa-devel] Software based mmap support and dmix
by Yogesh Soni 05 May '15

05 May '15

Hi all, I am trying to have dmix plugin work ( I succeeded) with the audio card that does not have hardware mmap support and uses pcm_indirect method to map software buffer for mmap feature. Most of the plugins were working quite well with mmap/pcm_indirect combo. However, dmix plugin did not work (no sound). I debugged and realized that appl_ptr in pcm_direct.h(snd_pcm_indirect_playback_transfer()) was always 0 and hence nothing got copied to slave pcm buffer. I then discovered that dmix operates in freewheeling-mode.. I have made a small workaround patch (I believe it can be cleaned up) in case of free-wheeling mode, basically using sw_ready to hold bytes that need to be copied. I used rec->sw_ready because it looked safest to use for this purpose. As a result of this patch, normally period_size bytes are copied to pcm buffer every period and sound is playing smooth. I am a noob in alsa and linux in general. So, I would like your opinion on the patch. I would much prefer if I could write a patch that is generic and clean enough, so that it could be used by others. New pcm-indirect.h attached. Thanks a lot for your help. Regards Yogesh

1 0

[alsa-devel] [PATCH 1/7] ASoC: ad1836: Replace direct snd_soc_codec dapm field access
by Lars-Peter Clausen 04 May '15

04 May '15

The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm with snd_soc_codec_get_dapm(). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ad1836.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 685998d..95f0bec 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -251,7 +251,7 @@ static int ad1836_resume(struct snd_soc_codec *codec) static int ad1836_probe(struct snd_soc_codec *codec) { struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); int num_dacs, num_adcs; int ret = 0; int i; -- 1.8.0

2 7

[alsa-devel] [PATCH V3 0/4] tas571x amplifier driver
by Kevin Cernekee 04 May '15

04 May '15

V2->V3: Revert earlier changes to the regcache APIs. Change regcache_sync() so that regcache_mark_dirty() is used to indicate that the HW was reset. This allows tas571x (which does NOT call regcache_mark_dirty()) to use regcache_sync() to write back any register changes that occurred while in cache_only (tas571x pdn) mode. One assumption here is that regcache_sync() will be called once on resume, before any other writes. If it fails, the no_sync_defaults flag will be cleared and the next sync attempt, if any, will try to write out all contents of the cache. Kevin Cernekee (4): regmap: Use regcache_mark_dirty() to indicate power loss or reset ASoC: tas571x: Add DT binding document ASoC: tas571x: New driver for TI TAS571x power amplifiers MAINTAINERS: Add entry for tas571x ASoC codec driver .../devicetree/bindings/sound/tas571x.txt | 41 ++ MAINTAINERS | 6 + drivers/base/regmap/internal.h | 3 + drivers/base/regmap/regcache.c | 61 ++- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas571x.c | 520 +++++++++++++++++++++ sound/soc/codecs/tas571x.h | 33 ++ 8 files changed, 651 insertions(+), 20 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tas571x.txt create mode 100644 sound/soc/codecs/tas571x.c create mode 100644 sound/soc/codecs/tas571x.h -- 2.2.0.rc0.207.ga3a616c

3 10

[alsa-devel] [PATCH 1/4] ASoC: Intel: Add support max98090 in sst driver
by yang.a.fang＠intel.com 04 May '15

04 May '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> Added entry in sst driver to support max98090 codec for intel Braswell platform. Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/intel/atom/sst/sst_acpi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index fc02a48..bb19b58 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -356,6 +356,8 @@ static struct sst_machines sst_acpi_chv[] = { &chv_platform_data }, {"10EC5650", "cht-bsw", "cht-bsw-rt5645", NULL, "intel/fw_sst_22a8.bin", &chv_platform_data }, + {"193C9890", "cht-bsw", "cht-bsw-max98090", NULL, + "intel/fw_sst_22a8.bin", &chv_platform_data }, {}, }; -- 1.7.9.5

7 23

[alsa-devel] [PATCH] ASoC: uda1380: Avoid accessing i2c bus when codec is disabled
by Vasily Khoruzhick 04 May '15

04 May '15

set_dai_fmt_both() callback is called from snd_soc_runtime_set_dai_fmt() which is called from snd_soc_register_card(), but at this time codec is not powered on yet. Replace direct i2c write with regcache write. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- sound/soc/codecs/uda1380.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index dc7778b..c3c33bd 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -437,7 +437,7 @@ static int uda1380_set_dai_fmt_both(struct snd_soc_dai *codec_dai, if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) return -EINVAL; - uda1380_write(codec, UDA1380_IFACE, iface); + uda1380_write_reg_cache(codec, UDA1380_IFACE, iface); return 0; } -- 2.3.5

2 1

Re: [alsa-devel] [PATCH] ASoC: simple-card: support platform in dts parse
by Mark Brown 04 May '15

04 May '15

On Wed, Apr 29, 2015 at 06:11:07PM +0800, Jun Nie wrote: > Support platform in dts parse so that dma pcm component > can be added in dts. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH 2/2] ASoC: zx: Add zx296702 SPDIF support
by Mark Brown 04 May '15

04 May '15

On Mon, May 04, 2015 at 09:13:48PM +0800, Jun Nie wrote: > 2015-05-01 18:55 GMT+08:00 Mark Brown <broonie(a)kernel.org>: > > This is adding a new device with DT bindings but there is no bindings > > document. All new DT bindings must be documented. > Plan to add a separate patch to device tree. Or you prefer to add > device tree patch in this patch set and loop device tree maintainer? You *always* need to document new device tree bindings at the same time as adding new code. > There are more than one SPDIF controller on ZTE SOC. One of them > support HDMI output, while the other SPDIF controller support direct > output. So I want to separate SPDIF and HDMI and add SPDIF id 0 test So this special HDMI S/PDIF controller is directly integrated with the HDMI IP in the SoC and not usable as a generic S/PDIF controller? That seems to mirror system designs which have an external HDMI encoder which use S/PDIF or I2S to connect the audio portion to an external device. > Do you have any suggestion on what driver to look at for the interface > between audio and video? Half the problem right now is that the people working on HDMI don't appear to be talking to each other very much so everything is very inconsistent with no real code sharing. I don't really feel I've got a good enough picture of what the hardware looks like in general to be 100% clear on what the best way is forward beyond the fact that we should be sharing the EDID parsing code. I'd suggest for future versions reorganizing things so that you add the S/PDIF driver with only S/PDIF support and then have a separate patch that layers on the HDMI support.

1 0

Re: [alsa-devel] Fix typo for ALC286/ALC288
by Takashi Iwai 04 May '15

04 May '15

At Mon, 4 May 2015 07:59:07 +0000, Kailang wrote: > > Hi Takashi, > > Sorry. > It had typo for my prior patch. OK, applied now. Thanks. Takashi > > BR, > Kailang > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 0000-fix-typo-for-ALC286.patch <application/octet-stream (base64)>] >

1 0

[alsa-devel] [PATCH] ASoC: rsnd: Use generic names for device nodes
by Geert Uytterhoeven 04 May '15

04 May '15

rcar_sound -> sound Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index f316ce1f214a03e3..62ece4c59da759a4 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -47,7 +47,7 @@ DAI subnode properties: Example: -rcar_sound: rcar_sound@ec500000 { +rcar_sound: sound@ec500000 { #sound-dai-cells = <1>; compatible = "renesas,rcar_sound-r8a7791", "renesas,rcar_sound-gen2"; reg = <0 0xec500000 0 0x1000>, /* SCU */ -- 1.9.1

3 2

[alsa-devel] [PATCH 1/5] ASoC: lowland: Use card DAPM context to access widgets
by Lars-Peter Clausen 04 May '15

04 May '15

The dapm field of the snd_soc_codec struct will eventually be removed (replaced with the DAPM context from the component embedded inside the CODEC). Replace its usage with the card's DAPM context. The idea is that DAPM is hierarchical and with the card at the root it is possible to access widgets from other contexts through the card context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/samsung/lowland.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index 5f15609..0d0f582 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -72,7 +72,7 @@ static int lowland_wm9081_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; - snd_soc_dapm_nc_pin(&codec->dapm, "LINEOUT"); + snd_soc_dapm_nc_pin(&rtd->card->dapm, "LINEOUT"); /* At any time the WM9081 is active it will have this clock */ return snd_soc_codec_set_sysclk(codec, WM9081_SYSCLK_MCLK, 0, -- 1.8.0

3 7

[alsa-devel] [PATCH] ASoC: da7210: Fix dependency to allow build with SND_SOC_I2C_AND_SPI
by Axel Lin 04 May '15

04 May '15

Since commit aa0e25caafb7 ("ASoC: da7210: Add support for spi regmap"), the da7210 codec driver supports both I2C and SPI buses. Thus update the dependency accordingly. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..ac84ac4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -54,7 +54,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C select SND_SOC_CX20442 if TTY - select SND_SOC_DA7210 if I2C + select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: samsung: wolfson: Improve compile test coverage
by Lars-Peter Clausen 04 May '15

04 May '15

While the the Wolfson machine drivers have a runtime dependency on a specific machine there is no compile time dependency. Allow to lets those drivers to be selected when COMPILE_TEST is selected to improve the compile time test coverage. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/samsung/Kconfig | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 0632a36..3744c9e 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -174,7 +174,8 @@ config SND_SOC_SMDK_WM8994_PCM config SND_SOC_SPEYSIDE tristate "Audio support for Wolfson Speyside" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C && SPI_MASTER + depends on SND_SOC_SAMSUNG && I2C && SPI_MASTER + depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select SND_SOC_WM8996 select SND_SOC_WM9081 @@ -183,13 +184,15 @@ config SND_SOC_SPEYSIDE config SND_SOC_TOBERMORY tristate "Audio support for Wolfson Tobermory" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && INPUT && I2C + depends on SND_SOC_SAMSUNG && INPUT && I2C + depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select SND_SOC_WM8962 config SND_SOC_BELLS tristate "Audio support for Wolfson Bells" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA && I2C && SPI_MASTER + depends on SND_SOC_SAMSUNG && MFD_ARIZONA && I2C && SPI_MASTER + depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select SND_SOC_WM5102 select SND_SOC_WM5110 @@ -199,14 +202,16 @@ config SND_SOC_BELLS config SND_SOC_LOWLAND tristate "Audio support for Wolfson Lowland" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C + depends on SND_SOC_SAMSUNG && I2C + depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select SND_SOC_WM5100 select SND_SOC_WM9081 config SND_SOC_LITTLEMILL tristate "Audio support for Wolfson Littlemill" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C + depends on SND_SOC_SAMSUNG && I2C + depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select MFD_WM8994 select SND_SOC_WM8994 -- 1.8.0

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Re: [alsa-devel] [PATCH v3 3/8] mfd: arizona: Add support for WM8998 and WM1814
by Paul Bolle 04 May '15

04 May '15

Something I didn't notice when v2 came along. On Fri, 2015-05-01 at 16:15 +0100, Richard Fitzgerald wrote: > --- a/drivers/mfd/Kconfig > +++ b/drivers/mfd/Kconfig > +config MFD_WM8998 > + bool "Wolfson Microelectronics WM8998" > + depends on MFD_ARIZONA > + help > + Support for Wolfson Microelectronics WM8998 low power audio SoC > + > config MFD_WM8400 > bool "Wolfson Microelectronics WM8400" > select MFD_CORE > --- a/drivers/mfd/Makefile > +++ b/drivers/mfd/Makefile > ifneq ($(CONFIG_MFD_WM8997),n) > obj-$(CONFIG_MFD_ARIZONA) += wm8997-tables.o > endif > +ifneq ($(CONFIG_MFD_WM8998),n) > +obj-$(CONFIG_MFD_ARIZONA) += wm8998-tables.o > +endif Playing a bit with the current version of drivers/mfd/Makefile it seems these ifneq ($(CONFIG_MFD_WM[...]),n) tests will always be true. Because, as far as I know, the Kconfig macros used in those tests will either be "y" or the empty string, but never "n". (I don't speak Makefilese fluent enough to know how to add simple debugging prints for the values of the CONFIG_MFD_WM[...] macros to actually test this, so correct me if I'm wrong here.) So I think that, effectively, these wm[...]-tables.o objects will always be built if CONFIG_MFD_ARIZONA is set. Is that the intention? If not, perhaps these test should read ifeq ($(CONFIG_MFD_WM[...]),y) Thanks, Paul Bolle

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[alsa-devel] [PATCH v5 0/2] Add support for select accessory detect mode to HPDETL or HPDETR
by Inha Song 04 May '15

04 May '15

This set of patches adds support for select accessory detect mode to HPDETL or HPDETR. Changes in v5: - Rebased on extcon-next branch. Changes in v4: - Set the hpdet_channel default value to HPDETL in variable declaration. - Fix the indentation in binding documentation. Changes in v3: - Set the hpdet_channel default value to HPDETL If the value is unknown or invalid. Changes in v2: - Use the value in pdata instead of hpdet_channel in extcon_info. - Wrap arizona_extcon_of_get in IS_ENABLED(CONFIG_OF). - Change hpdet_channel type to unsigned from signed in pdata. - Move ARIZONA_ACCDET_MODE_* define to dt-binding header and directly set it to pdata. Inha Song (2): extcon: arizona: Add support for select accessory detect mode when headphone detection mfd: arizona: Update DT binding to support hpdet channel Documentation/devicetree/bindings/mfd/arizona.txt | 6 ++++ drivers/extcon/extcon-arizona.c | 38 ++++++++++++++++++----- include/dt-bindings/mfd/arizona.h | 4 +++ include/linux/mfd/arizona/pdata.h | 3 ++ 4 files changed, 43 insertions(+), 8 deletions(-) -- 2.0.0.390.gcb682f8

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[alsa-devel] [RESEND RFC PATCH 0/3] ASoC: Mediatek: Add support for MT8173 SOC
by Koro Chen 04 May '15

04 May '15

Refactor and resend this series because of the 60KB limitation. This adds basic support for the Mediatek AFE unit for MT8173 SoC. This patch is based on Linux 4.0-rc1, Sascha's clk patch [1], and his SCPSYS power patch [2]. The AFE unit comprises several memory interfaces that communicate with CPU, a multi input multi output digital audio interconnect, and several external interfaces (e.g. I2S). The AFE unit is a DMA master, so no external DMA engine is needed. Each external interface is presented as a DAI to ASoC. A memory interface must be connected via the interconnect to an external interface. The connection pathes are configured through the device tree. [1] http://lists.infradead.org/pipermail/linux-mediatek/2015-March/000079.html [2] http://lists.infradead.org/pipermail/linux-mediatek/2015-March/000132.html Koro Chen (3): ASoC: mediatek: Add binding support for AFE driver ASoC: mediatek: Add AFE connection control ASoC: mediatek: Add AFE platform driver .../devicetree/bindings/sound/mtk-afe-pcm.txt | 105 ++ include/dt-bindings/sound/mtk-afe.h | 36 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/mediatek/Kconfig | 8 + sound/soc/mediatek/Makefile | 2 + sound/soc/mediatek/mtk-afe-common.h | 105 ++ sound/soc/mediatek/mtk-afe-connection.c | 416 ++++++ sound/soc/mediatek/mtk-afe-connection.h | 30 + sound/soc/mediatek/mtk-afe-pcm.c | 1497 ++++++++++++++++++++ 10 files changed, 2201 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mtk-afe-pcm.txt create mode 100644 include/dt-bindings/sound/mtk-afe.h create mode 100644 sound/soc/mediatek/Kconfig create mode 100644 sound/soc/mediatek/Makefile create mode 100644 sound/soc/mediatek/mtk-afe-common.h create mode 100644 sound/soc/mediatek/mtk-afe-connection.c create mode 100644 sound/soc/mediatek/mtk-afe-connection.h create mode 100644 sound/soc/mediatek/mtk-afe-pcm.c --

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Re: [alsa-devel] Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 03 May '15

03 May '15

> > > My problem is that when I set channel output to "Analog output surround > > > 5.1" > > > (or 2.1 or 4.1), I hear no sound in front speakers, but in case of 5.1 or > > > 4.1 You need pulseaudio lfe-filter for 2.1 http://cgit.freedesktop.org/pulseaudio/pulseaudio/log/?qt=grep&q=lfe-filter do you mean surround21 need 6 channel instead of channels with 5.1 internal /external speaker ? http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=48f1b308cc66152e… > > > I can hear sound in rear speakers. I use kernel 3.19.5 and it didn't > > > happen in > > > earlier kernels that is 3.18 and in 3.17 for sure. > > > > > > My configuration is: > > > OS: Fedora 21 64 bit > > > SoundCard: ALC888 > > > Model name: acer-aspire-4930g > > > Laptop: Acer Aspire 5930g > > > Speakers: external 5.1 with 3 jacks > > > Sound channel configuration module is standard configuration module from > > > KDE. > > > > > > Additional info: > > > 1) When channel output is 5.1 and I unplug green jack and leave two others > > > jacks connected then I can hear sound in laptop internal speakers. auto mute only check headphone jack or line out jack but not those retasked mio jacks > > > 2) Earlier (kernel 3.18 and 3.17) I didn't have possibility to choose 2.1 > > > configuration and additionally to 4.1 configuration I could also choose > > > 4.0 > > > configuration. > > > > > > You need to switch "Channel mode" to "6 ch" if you need to use external > > > > 5.1 speakers since you have 5.1 internal speakers > > > > this change pin-ctl of headphone jack, mic jack and line in to OUT > > > > post the output of alsa-info.sh > > I always switched to "6 ch" through alsamixer. It always worked as expected, > but recently I've got problems with it so it regressed somewhere. > > > My alsa-info.sh with "6 ch" set through alsamixer and "Analog output surround > 5.1 + Analog input stereo" set through KDE configuration module. > http://www.alsa-project.org/db/?f=ff3f5a1f7f3e4662c3fa9bcf0158474a076b4cc4 > > With that I hear all channels except front left and front right. autoconfig for ALC888: line_outs=3 (0x14/0x17/0x16/0x0/0x0) type:speaker speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) hp_outs=1 (0x15/0x0/0x0/0x0/0x0) mono: mono_out=0x0 dig-out=0x1e/0x0 inputs: Mic=0x18 Internal Mic=0x12 Line=0x1a ==> Best config: lo_type=2, wired=1, mio=1 multi_outs = 15/0/0/0 : 2/3/4/0 (type HP) out path: depth=3 '02:0c:15' multi_ios(2) = 1a/18 : 3/4 mio path: depth=3 '03:0d:1a' mio path: depth=3 '04:0e:18' spk_outs = 14/17/16/0 : 5/25/4/0 spk path: depth=3 '05:0f:14' spk path: depth=3 '25:26:17' spk path: depth=3 '04:0e:16' Node 0x15 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Front Headphone Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001003e: IN OUT HP EAPD Detect Trigger EAPD 0x2: EAPD Pin Default 0x0221101f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 it is still headphone jack with pin-ctl : 0xc0 OUT HP while mic jack and line in jack 's pin ctl are changed to 0x40 OUT channel mode only change two mio jacks from IN to OUT May need to use add_out_jack_mode=1 instead of add_jack_modes=1 when there are Mic and line in jacks Use Headphone Jack Mode control to switch HP to Line Out for pin-ctl and EAPD https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=f81… > To exclude that my speakers are broken or wrong connected: I get sound from > front left and front right if i switch in KDE configuration module to "Analog > duplex stereo", but then I get no sound from rest of my speakers. http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… pulseaudio mute external center/lfe controls when headphone jack is plugged since pulseaudio not aware that the role of headphone has been changed [Element Center] switch = off volume = off [Element LFE] switch = off volume = off > > > KDE configuration module lets me choose "connector" configuration but only in > stereo mode. > In stereo mode I can choose between "headphones" and "speakers". > If it is set to "headphones" I hear sound in front left and front right > speakers. > If it is set to "speakers" I hear sound in rear left and rear right speakers. > you need to switch "channel mode" to "2 ch" if you want to use your 5.1 internal speaker https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=786… There are difference between static model and generic parser

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Re: [alsa-devel] [PATCH] ALSA: hda - Add dock support for ThinkPad X250 (17aa:2226)
by Takashi Iwai 02 May '15

02 May '15

At Sat, 11 Apr 2015 09:31:35 +0200, Yves-Alexis Perez wrote: > > This model uses the same dock port as the previous generation. > > Signed-off-by: Yves-Alexis Perez <corsac(a)debian.org> Thanks, applied with Cc to stable. But, at the next time, please put maintainers to Cc. Takashi > --- > 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 7438213..2087728 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -5030,6 +5030,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { > SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), > SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), > SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), > + SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), > SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), > SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), > SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), > -- > 2.1.4 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ >

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