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[alsa-devel] [ANNOUNCE] tinycompress: release 0.2.0
by Vinod Koul 07 Jun '13

07 Jun '13

Hello folks, Time to cut another release for the library i am working on. tinycompress library - the user mode library for using the ALSA compressed APIs has received following new features and now is v0.2.0 - support for non blocking I/O - support for recording compressed audio content - no poll when writing last file buffer - get_alsa_rate helper for rate conversion - bunch of other fixes - now all releases will be tagged accordingly For details & contributions, pls see the below changelog. Lakshmi N Vinnakota (1): fix error reporting in tinycompress Richard Fitzgerald (6): tinycompress: cplay: Correct a couple of small message typos compress: Add support for capture streams compress: fix hpointer error when no sample rate compress: copy final version of config struct compress: do not poll if enough space to write remaining data compress: Add non-blocking I/O Vinod Koul (4): compress: check for config first add utils file with rate conversion helper cplay: use get_alsa_rate to convert sample rate Tinycompress version 0.2.0 Android.mk | 2 +- compress.c | 171 +++++++++++++++++++++++++++-------- cplay.c | 52 ++--------- include/tinycompress/tinycompress.h | 27 +++++- include/tinycompress/version.h | 4 +- makefile.linux | 2 +- utils.c | 96 ++++++++++++++++++++ 7 files changed, 263 insertions(+), 91 deletions(-) create mode 100644 utils.c Thanks ~Vinod --

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[alsa-devel] [PATCH V4] ASoC: fsl: add imx-wm8962 machine driver
by Nicolin Chen 07 Jun '13

07 Jun '13

This is the initial imx-wm8962 device-tree-only machine driver working with fsl_ssi driver. More features can be added on top of it later. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- ChangeLog: v1->v2: * Fixed a typo in probe(): *** data->dai.ops = &imx_hifi_ops, ==> data->dai.ops = &imx_hifi_ops; v2->v3: * Moved Headphone/Microphone jack detection GPIO port to Optional properties. * Use set_bias_level() instead of hw_params/hw_free() to open/close FLL. * Use devm_clk_get() instead of clk_get(). * Dropped clock-frequency grabbing code. * Fixed a typo: *** data->card.late_probe = imx_wm8962_late_probe, *** ==> *** data->card.late_probe = imx_wm8962_late_probe; v3->v4: * Revised binding doc to describe two gpio pins more clearly. .../devicetree/bindings/sound/imx-audio-wm8962.txt | 55 ++++ sound/soc/fsl/Kconfig | 12 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8962.c | 336 ++++++++++++++++++++ 4 files changed, 405 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt create mode 100644 sound/soc/fsl/imx-wm8962.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt new file mode 100644 index 0000000..a34b94b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt @@ -0,0 +1,55 @@ +Freescale i.MX audio complex with WM8962 codec + +Required properties: +- compatible : "fsl,imx-audio-wm8962" +- model : The user-visible name of this sound complex +- ssi-controller : The phandle of the i.MX SSI controller +- audio-codec : The phandle of the WM8962 audio codec +- audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names could be power + supplies, WM8962 pins, and the jacks on the board: + + Power supplies: + * Mic Bias + + Board connectors: + * Mic Jack + * Headphone Jack + * Ext Spk + +- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) +- mux-ext-port : The external port of the i.MX audio muxer + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Optional properties: +- hp-det-gpios : The gpio pin to detect plug in/out event that happens to + Headphone jack. + +- mic-det-gpios: The gpio pin to detect plug in/out event that happens to + Micphone jack. + +Example: + +sound { + compatible = "fsl,imx6q-sabresd-wm8962", + "fsl,imx-audio-wm8962"; + model = "wm8962-audio"; + ssi-controller = <&ssi2>; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUTL", + "Headphone Jack", "HPOUTR", + "Ext Spk", "SPKOUTL", + "Ext Spk", "SPKOUTR", + "MICBIAS", "AMIC", + "IN3R", "MICBIAS", + "DMIC", "MICBIAS", + "DMICDAT", "DMIC"; + mux-int-port = <2>; + mux-ext-port = <3>; + hp-det-gpios = <&gpio7 8 1>; /* active low */ + mic-det-gpios = <&gpio1 9 1>; /* active low */ +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 7860cc2..aa43854 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -168,6 +168,18 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8962 + tristate "SoC Audio support for i.MX boards with wm8962" + depends on OF && I2C + select SND_SOC_WM8962 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + help + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8962 codec. + config SND_SOC_IMX_SGTL5000 tristate "SoC Audio support for i.MX boards with sgtl5000" depends on OF && I2C diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 91883f8..d4b4aa8 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -42,6 +42,7 @@ snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o @@ -49,4 +50,5 @@ obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c new file mode 100644 index 0000000..0b2eb65 --- /dev/null +++ b/sound/soc/fsl/imx-wm8962.c @@ -0,0 +1,336 @@ +/* + * Copyright 2013 Freescale Semiconductor, Inc. + * + * Based on imx-sgtl5000.c + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/of_i2c.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <linux/clk.h> +#include <linux/clk-provider.h> +#include <linux/irq.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/pinctrl/consumer.h> + +#include "../codecs/wm8962.h" +#include "imx-audmux.h" + +#define DAI_NAME_SIZE 32 + +struct imx_wm8962_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + struct clk *codec_clk; + unsigned int clk_frequency; +}; + +struct imx_priv { + int hp_gpio; + int hp_active_low; + int hp_status; + int amic_gpio; + int amic_active_low; + int amic_status; + struct platform_device *pdev; + struct snd_pcm_substream *first_stream; + struct snd_pcm_substream *second_stream; +}; +static struct imx_priv card_priv; + +static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static int sample_rate = 44100; +static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; + +static int imx_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + sample_rate = params_rate(params); + sample_format = params_format(params); + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_hifi_hw_params, +}; + +static int imx_wm8962_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + unsigned int pll_out; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + if (sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = sample_rate * 384; + else + pll_out = sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, data->clk_frequency, + pll_out); + if (ret < 0) { + pr_err("Failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_FLL, pll_out, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK: %d\n", ret); + return ret; + } + } + break; + + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_MCLK, data->clk_frequency, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to switch away from FLL: %d\n", + ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL: %d\n", ret); + return ret; + } + } + break; + + default: + break; + } + + dapm->bias_level = level; + + return 0; +} + +static int imx_wm8962_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + data->clk_frequency, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(card->dev, "Failed to set sysclk in %s\n", __func__); + + return ret; +} + +static int imx_wm8962_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct imx_priv *priv = &card_priv; + struct i2c_client *codec_dev; + struct imx_wm8962_data *data; + int int_port, ext_port; + int ret; + + priv->pdev = pdev; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(&pdev->dev, "audmux internal port setup failed\n"); + return ret; + } + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(&pdev->dev, "audmux external port setup failed\n"); + return ret; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + return -EINVAL; + } + + priv->hp_gpio = of_get_named_gpio_flags(np, "hp-det-gpios", 0, + (enum of_gpio_flags *)&priv->hp_active_low); + priv->amic_gpio = of_get_named_gpio_flags(np, "mic-det-gpios", 0, + (enum of_gpio_flags *)&priv->amic_active_low); + + priv->first_stream = NULL; + priv->second_stream = NULL; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); + if (IS_ERR(data->codec_clk)) { + ret = PTR_ERR(data->codec_clk); + dev_err(&codec_dev->dev, "could not get codec clk: %d\n", ret); + goto fail; + } + + data->clk_frequency = clk_get_rate(data->codec_clk); + clk_prepare_enable(data->codec_clk); + + data->dai.name = "HiFi"; + data->dai.stream_name = "HiFi"; + data->dai.codec_dai_name = "wm8962"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.platform_of_node = ssi_np; + data->dai.ops = &imx_hifi_ops; + data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto clk_fail; + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto clk_fail; + data->card.num_links = 1; + data->card.dai_link = &data->dai; + data->card.dapm_widgets = imx_wm8962_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); + + data->card.late_probe = imx_wm8962_late_probe; + data->card.set_bias_level = imx_wm8962_set_bias_level; + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto clk_fail; + } + + platform_set_drvdata(pdev, data); + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; + +clk_fail: + if (!IS_ERR(data->codec_clk)) + clk_disable_unprepare(data->codec_clk); +fail: + if (ssi_np) + of_node_put(ssi_np); + if (codec_np) + of_node_put(codec_np); + + return ret; +} + +static int imx_wm8962_remove(struct platform_device *pdev) +{ + struct imx_wm8962_data *data = platform_get_drvdata(pdev); + + if (!IS_ERR(data->codec_clk)) + clk_disable_unprepare(data->codec_clk); + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_wm8962_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8962", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids); + +static struct platform_driver imx_wm8962_driver = { + .driver = { + .name = "imx-wm8962", + .owner = THIS_MODULE, + .of_match_table = imx_wm8962_dt_ids, + }, + .probe = imx_wm8962_probe, + .remove = imx_wm8962_remove, +}; +module_platform_driver(imx_wm8962_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8962"); -- 1.7.1

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[alsa-devel] Question on wait_for_avail function
by gsantosh＠codeaurora.org 07 Jun '13

07 Jun '13

Hi Liam, Consider an use case 1) Write is blocked as there is no space to write the data, after the return from snd_pcm_playback_avail()function in wait_for_avail 2) HAL issues pause in parallel to the write I see that the following function snd_pcm_post_pause unblock the write wake_up(&runtime->sleep); wake_up(&runtime->tsleep); But i do not see any case handling for the pause state in the wait_for_avail function, it is not coming out of the for loop as there will not be any change in the buffer states. Please explain what is the intention behind the wake in pause callback. Want to understand why there is no handling of pause in wait_for_avail function. and also is there any restriction from ALSA framework that we should not issue pause while write is blocked or going on. Regards, Santosh M G.

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[alsa-devel] [PATCH V3] ASoC: fsl: add imx-wm8962 machine driver
by Nicolin Chen 07 Jun '13

07 Jun '13

This is the initial imx-wm8962 device-tree-only machine driver working with fsl_ssi driver. More features can be added on top of it later. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- ChangeLog: v1->v2: * Fixed a typo in probe(): *** data->dai.ops = &imx_hifi_ops, ==> data->dai.ops = &imx_hifi_ops; v2->v3: * Moved Headphone/Microphone jack detection GPIO port to Optional properties. * Use set_bias_level() instead of hw_params/hw_free() to open/close FLL. * Use devm_clk_get() instead of clk_get(). * Dropped clock-frequency grabbing code. * Fixed a typo: *** data->card.late_probe = imx_wm8962_late_probe, *** ==> *** data->card.late_probe = imx_wm8962_late_probe; .../devicetree/bindings/sound/imx-audio-wm8962.txt | 52 +++ sound/soc/fsl/Kconfig | 12 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8962.c | 336 ++++++++++++++++++++ 4 files changed, 402 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt create mode 100644 sound/soc/fsl/imx-wm8962.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt new file mode 100644 index 0000000..bb76097 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt @@ -0,0 +1,52 @@ +Freescale i.MX audio complex with WM8962 codec + +Required properties: +- compatible : "fsl,imx-audio-wm8962" +- model : The user-visible name of this sound complex +- ssi-controller : The phandle of the i.MX SSI controller +- audio-codec : The phandle of the WM8962 audio codec +- audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names could be power + supplies, WM8962 pins, and the jacks on the board: + + Power supplies: + * Mic Bias + + Board connectors: + * Mic Jack + * Headphone Jack + * Ext Spk + +- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) +- mux-ext-port : The external port of the i.MX audio muxer + +Optional properties: +- hp-det-gpios : The gpio port of Headphone jack detection. +- mic-det-gpios: The gpio port of Micphone jack detection. + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + +sound { + compatible = "fsl,imx6q-sabresd-wm8962", + "fsl,imx-audio-wm8962"; + model = "wm8962-audio"; + ssi-controller = <&ssi2>; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUTL", + "Headphone Jack", "HPOUTR", + "Ext Spk", "SPKOUTL", + "Ext Spk", "SPKOUTR", + "MICBIAS", "AMIC", + "IN3R", "MICBIAS", + "DMIC", "MICBIAS", + "DMICDAT", "DMIC"; + mux-int-port = <2>; + mux-ext-port = <3>; + hp-det-gpios = <&gpio7 8 1>; /* active low */ + mic-det-gpios = <&gpio1 9 1>; /* active low */ +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 7860cc2..aa43854 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -168,6 +168,18 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8962 + tristate "SoC Audio support for i.MX boards with wm8962" + depends on OF && I2C + select SND_SOC_WM8962 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + help + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8962 codec. + config SND_SOC_IMX_SGTL5000 tristate "SoC Audio support for i.MX boards with sgtl5000" depends on OF && I2C diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 91883f8..d4b4aa8 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -42,6 +42,7 @@ snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o @@ -49,4 +50,5 @@ obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c new file mode 100644 index 0000000..0b2eb65 --- /dev/null +++ b/sound/soc/fsl/imx-wm8962.c @@ -0,0 +1,336 @@ +/* + * Copyright 2013 Freescale Semiconductor, Inc. + * + * Based on imx-sgtl5000.c + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/of_i2c.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <linux/clk.h> +#include <linux/clk-provider.h> +#include <linux/irq.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/pinctrl/consumer.h> + +#include "../codecs/wm8962.h" +#include "imx-audmux.h" + +#define DAI_NAME_SIZE 32 + +struct imx_wm8962_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + struct clk *codec_clk; + unsigned int clk_frequency; +}; + +struct imx_priv { + int hp_gpio; + int hp_active_low; + int hp_status; + int amic_gpio; + int amic_active_low; + int amic_status; + struct platform_device *pdev; + struct snd_pcm_substream *first_stream; + struct snd_pcm_substream *second_stream; +}; +static struct imx_priv card_priv; + +static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static int sample_rate = 44100; +static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; + +static int imx_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + sample_rate = params_rate(params); + sample_format = params_format(params); + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_hifi_hw_params, +}; + +static int imx_wm8962_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + unsigned int pll_out; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + if (sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = sample_rate * 384; + else + pll_out = sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, data->clk_frequency, + pll_out); + if (ret < 0) { + pr_err("Failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_FLL, pll_out, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK: %d\n", ret); + return ret; + } + } + break; + + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_MCLK, data->clk_frequency, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to switch away from FLL: %d\n", + ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL: %d\n", ret); + return ret; + } + } + break; + + default: + break; + } + + dapm->bias_level = level; + + return 0; +} + +static int imx_wm8962_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + data->clk_frequency, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(card->dev, "Failed to set sysclk in %s\n", __func__); + + return ret; +} + +static int imx_wm8962_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct imx_priv *priv = &card_priv; + struct i2c_client *codec_dev; + struct imx_wm8962_data *data; + int int_port, ext_port; + int ret; + + priv->pdev = pdev; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(&pdev->dev, "audmux internal port setup failed\n"); + return ret; + } + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(&pdev->dev, "audmux external port setup failed\n"); + return ret; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + return -EINVAL; + } + + priv->hp_gpio = of_get_named_gpio_flags(np, "hp-det-gpios", 0, + (enum of_gpio_flags *)&priv->hp_active_low); + priv->amic_gpio = of_get_named_gpio_flags(np, "mic-det-gpios", 0, + (enum of_gpio_flags *)&priv->amic_active_low); + + priv->first_stream = NULL; + priv->second_stream = NULL; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); + if (IS_ERR(data->codec_clk)) { + ret = PTR_ERR(data->codec_clk); + dev_err(&codec_dev->dev, "could not get codec clk: %d\n", ret); + goto fail; + } + + data->clk_frequency = clk_get_rate(data->codec_clk); + clk_prepare_enable(data->codec_clk); + + data->dai.name = "HiFi"; + data->dai.stream_name = "HiFi"; + data->dai.codec_dai_name = "wm8962"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.platform_of_node = ssi_np; + data->dai.ops = &imx_hifi_ops; + data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto clk_fail; + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto clk_fail; + data->card.num_links = 1; + data->card.dai_link = &data->dai; + data->card.dapm_widgets = imx_wm8962_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); + + data->card.late_probe = imx_wm8962_late_probe; + data->card.set_bias_level = imx_wm8962_set_bias_level; + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto clk_fail; + } + + platform_set_drvdata(pdev, data); + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; + +clk_fail: + if (!IS_ERR(data->codec_clk)) + clk_disable_unprepare(data->codec_clk); +fail: + if (ssi_np) + of_node_put(ssi_np); + if (codec_np) + of_node_put(codec_np); + + return ret; +} + +static int imx_wm8962_remove(struct platform_device *pdev) +{ + struct imx_wm8962_data *data = platform_get_drvdata(pdev); + + if (!IS_ERR(data->codec_clk)) + clk_disable_unprepare(data->codec_clk); + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_wm8962_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8962", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids); + +static struct platform_driver imx_wm8962_driver = { + .driver = { + .name = "imx-wm8962", + .owner = THIS_MODULE, + .of_match_table = imx_wm8962_dt_ids, + }, + .probe = imx_wm8962_probe, + .remove = imx_wm8962_remove, +}; +module_platform_driver(imx_wm8962_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8962"); -- 1.7.1

2 1

[alsa-devel] [PATCH V3] ASoC: WM8962: Add device tree binding
by Nicolin Chen 07 Jun '13

07 Jun '13

Document the device tree binding for the WM8962 codec, and modify the driver to extract platform data from the device tree, if present. Based on work of WM8903 by Stephen Warren <swarren(a)nvidia.com> Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- ChangeLog: v1->v2: * Dropped out-of-band value error msg by regarding it as default value. v2->v3: * Revised binding doc to warn users if property absent then use default value. Documentation/devicetree/bindings/sound/wm8962.txt | 23 +++++++++++++ sound/soc/codecs/wm8962.c | 35 +++++++++++++++++++- 2 files changed, 57 insertions(+), 1 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/wm8962.txt b/Documentation/devicetree/bindings/sound/wm8962.txt index dceb3b1..7f82b59 100644 --- a/Documentation/devicetree/bindings/sound/wm8962.txt +++ b/Documentation/devicetree/bindings/sound/wm8962.txt @@ -8,9 +8,32 @@ Required properties: - reg : the I2C address of the device. +Optional properties: + - spk-mono: This is a boolean property. If present, the SPK_MONO bit + of R51 (Class D Control 2) gets set, indicating that the speaker is + in mono mode. + + - mic-cfg : Default register value for R48 (Additional Control 4). + If absent, the default should be the register default. + + - gpio-cfg : A list of GPIO configuration register values. The list must + be 6 entries long. If absent, no configuration of these registers is + performed. And note that only the value within [0x0, 0xffff] is valid. + Any other value is regarded as setting the GPIO register by its reset + value 0x0. + Example: codec: wm8962@1a { compatible = "wlf,wm8962"; reg = <0x1a>; + + gpio-cfg = < + 0x0000 /* 0:Default */ + 0x0000 /* 1:Default */ + 0x0013 /* 2:FN_DMICCLK */ + 0x0000 /* 3:Default */ + 0x8014 /* 4:FN_DMICCDAT */ + 0x0000 /* 5:Default */ + >; }; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d56dd86..26219ea 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3584,6 +3584,34 @@ static const struct regmap_config wm8962_regmap = { .cache_type = REGCACHE_RBTREE, }; +static int wm8962_set_pdata_from_of(struct i2c_client *i2c, + struct wm8962_pdata *pdata) +{ + const struct device_node *np = i2c->dev.of_node; + u32 val32; + int i; + + if (of_property_read_bool(np, "spk-mono")) + pdata->spk_mono = true; + + if (of_property_read_u32(np, "mic-cfg", &val32) >= 0) + pdata->mic_cfg = val32; + + if (of_property_read_u32_array(np, "gpio-cfg", pdata->gpio_init, + ARRAY_SIZE(pdata->gpio_init)) >= 0) + for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++) { + /* + * The range of GPIO register value is [0x0, 0xffff] + * While the default value of each register is 0x0 + * Any other value will be regarded as default value + */ + if (pdata->gpio_init[i] > 0xffff) + pdata->gpio_init[i] = 0x0; + } + + return 0; +} + static int wm8962_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -3604,8 +3632,13 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, wm8962->irq = i2c->irq; /* If platform data was supplied, update the default data in priv */ - if (pdata) + if (pdata) { memcpy(&wm8962->pdata, pdata, sizeof(struct wm8962_pdata)); + } else if (i2c->dev.of_node) { + ret = wm8962_set_pdata_from_of(i2c, &wm8962->pdata); + if (ret != 0) + return ret; + } for (i = 0; i < ARRAY_SIZE(wm8962->supplies); i++) wm8962->supplies[i].supply = wm8962_supply_names[i]; -- 1.7.1

2 1

[alsa-devel] [tinycompress]how should I call compress_open for aac ltp file playback?
by Keyon.Jey 07 Jun '13

07 Jun '13

I am trying to use tinycompress for hwdecoder. everything works fine, but after compress_open (with profile = SND_AUDIOPROFILE_AAC level=0 ch_mode=0 rate_control=0), the playback of my aac-ltp-16bit-48KHz-Stereo file is faster than it should be. Am I wrong? struct snd_codec { __u32 id; __u32 ch_in; __u32 ch_out; __u32 sample_rate; __u32 bit_rate; __u32 rate_control; __u32 profile; __u32 level; __u32 ch_mode; __u32 format; __u32 align; union snd_codec_options options; __u32 reserved[3]; }; Keyon

3 4

[alsa-devel] [PATCH V2] ASoC: fsl: add imx-wm8962 machine driver
by Nicolin Chen 07 Jun '13

07 Jun '13

This is the initial imx-wm8962 device-tree-only machine driver working with fsl_ssi driver. More features can be added on top of it later. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- ChangeLog: v1->v2: * fixed a typo in probe(): * data->dai.ops = &imx_hifi_ops, ==> data->dai.ops = &imx_hifi_ops; --- .../devicetree/bindings/sound/imx-audio-wm8962.txt | 61 +++ sound/soc/fsl/Kconfig | 12 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8962.c | 387 ++++++++++++++++++++ 4 files changed, 462 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt create mode 100644 sound/soc/fsl/imx-wm8962.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt new file mode 100644 index 0000000..cb188a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt @@ -0,0 +1,61 @@ +Freescale i.MX audio complex with WM8962 codec + +Required properties: +- compatible : "fsl,imx-audio-wm8962" +- model : The user-visible name of this sound complex +- ssi-controller : The phandle of the i.MX SSI controller +- audio-codec : The phandle of the WM8962 audio codec +- audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names could be power + supplies, WM8962 pins, and the jacks on the board: + + Power supplies: + * Mic Bias + + WM8962 pins: + * HPOUTL + * HPOUTR + * SPKOUTL + * SPKOUTR + * MICBIAS + * IN3R + * DMIC + * DMICDAT + + Board connectors: + * Mic Jack + * Headphone Jack + * Ext Spk + +- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) +- mux-ext-port : The external port of the i.MX audio muxer + +- hp-det-gpios : The gpio port of Headphone detection. +- mic-det-gpios: The gpio port of Micphone detection. + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + +sound { + compatible = "fsl,imx6q-sabresd-wm8962", + "fsl,imx-audio-wm8962"; + model = "wm8962-audio"; + ssi-controller = <&ssi2>; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUTL", + "Headphone Jack", "HPOUTR", + "Ext Spk", "SPKOUTL", + "Ext Spk", "SPKOUTR", + "MICBIAS", "AMIC", + "IN3R", "MICBIAS", + "DMIC", "MICBIAS", + "DMICDAT", "DMIC"; + mux-int-port = <2>; + mux-ext-port = <3>; + hp-det-gpios = <&gpio7 8 1>; /* active low */ + mic-det-gpios = <&gpio1 9 1>; /* active low */ +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 7860cc2..aa43854 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -168,6 +168,18 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8962 + tristate "SoC Audio support for i.MX boards with wm8962" + depends on OF && I2C + select SND_SOC_WM8962 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + help + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8962 codec. + config SND_SOC_IMX_SGTL5000 tristate "SoC Audio support for i.MX boards with sgtl5000" depends on OF && I2C diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 91883f8..d4b4aa8 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -42,6 +42,7 @@ snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o @@ -49,4 +50,5 @@ obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c new file mode 100644 index 0000000..cb54da4 --- /dev/null +++ b/sound/soc/fsl/imx-wm8962.c @@ -0,0 +1,387 @@ +/* + * Copyright 2013 Freescale Semiconductor, Inc. + * + * Based on imx-sgtl5000.c + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/of_i2c.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <linux/clk.h> +#include <linux/irq.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/pinctrl/consumer.h> + +#include "../codecs/wm8962.h" +#include "imx-audmux.h" + +#define DAI_NAME_SIZE 32 + +struct imx_wm8962_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + struct clk *codec_clk; + unsigned int clk_frequency; +}; + +struct imx_priv { + int hp_gpio; + int hp_active_low; + int hp_status; + int amic_gpio; + int amic_active_low; + int amic_status; + struct platform_device *pdev; + struct snd_pcm_substream *first_stream; + struct snd_pcm_substream *second_stream; +}; +static struct imx_priv card_priv; + +static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static int check_hw_params(struct snd_pcm_substream *substream, + snd_pcm_format_t sample_format, unsigned int sample_rate) +{ + struct imx_priv *priv = &card_priv; + struct device *dev = substream->pcm->card->dev; + + substream->runtime->sample_bits = + snd_pcm_format_physical_width(sample_format); + substream->runtime->rate = sample_rate; + substream->runtime->format = sample_format; + + if (!priv->first_stream) { + priv->first_stream = substream; + } else { + priv->second_stream = substream; + + if (priv->first_stream->runtime->rate != + priv->second_stream->runtime->rate) { + dev_err(dev, "KEEP THE SAME SAMPLE RATE: %d!\n", + priv->first_stream->runtime->rate); + return -EINVAL; + } + if (priv->first_stream->runtime->sample_bits != + priv->second_stream->runtime->sample_bits) { + snd_pcm_format_t first_format = + priv->first_stream->runtime->format; + + dev_err(dev, "KEEP THE SAME FORMAT: %s!\n", + snd_pcm_format_name(first_format)); + return -EINVAL; + } + } + + return 0; +} + +static int imx_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct device *dev = substream->pcm->card->dev; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + unsigned int sample_rate = params_rate(params); + unsigned int sample_format = params_format(params); + int ret = 0; + unsigned int pll_out; + + /* + * WM8962 doesn't support two substreams in different sample rates or + * sample formats. So check the two parameters of two substreams' when + * there're two substreams of playback and of capture in the same time. + */ + ret = check_hw_params(substream, sample_format, sample_rate); + if (ret < 0) { + dev_err(dev, "Failed to match hw params: %d\n", ret); + return ret; + } + + if (sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = sample_rate * 384; + else + pll_out = sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, + data->clk_frequency, pll_out); + if (ret < 0) { + dev_err(dev, "Failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, + pll_out, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(dev, "Failed to set SYSCLK: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct device *dev = substream->pcm->card->dev; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + int ret; + + /* We don't need to handle anything if there's no substream running */ + if (!priv->first_stream) + return 0; + + if (priv->first_stream == substream) + priv->first_stream = priv->second_stream; + priv->second_stream = NULL; + + if (!priv->first_stream) { + /* + * SYSCLK of wm8962's not disabled yet. If we wanna close FLL, + * we need to keep SYSCLK alive by changing its source to MCLK. + */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + data->clk_frequency, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(dev, "Failed to set SYSCLK: %d\n", ret); + return ret; + } + + /* Disable FLL and let codec do pm_runtime_put() */ + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, 0, 0, 0); + if (ret < 0) { + dev_err(dev, "Failed to disable FLL: %d\n", ret); + return ret; + } + } + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_hifi_hw_params, + .hw_free = imx_hifi_hw_free, +}; + +static int imx_wm8962_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + data->clk_frequency, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(card->dev, "Failed to set sysclk in %s\n", __func__); + + return ret; +} + +static int imx_wm8962_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct imx_priv *priv = &card_priv; + struct i2c_client *codec_dev; + struct imx_wm8962_data *data; + int int_port, ext_port; + int ret; + + priv->pdev = pdev; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(&pdev->dev, "audmux internal port setup failed\n"); + return ret; + } + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(&pdev->dev, "audmux external port setup failed\n"); + return ret; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + return -EINVAL; + } + + priv->hp_gpio = of_get_named_gpio_flags(np, "hp-det-gpios", 0, + (enum of_gpio_flags *)&priv->hp_active_low); + priv->amic_gpio = of_get_named_gpio_flags(np, "mic-det-gpios", 0, + (enum of_gpio_flags *)&priv->amic_active_low); + + priv->first_stream = NULL; + priv->second_stream = NULL; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + data->codec_clk = clk_get(&codec_dev->dev, NULL); + if (IS_ERR(data->codec_clk)) { + /* assuming clock enabled by default */ + data->codec_clk = NULL; + ret = of_property_read_u32(codec_np, "clock-frequency", + &data->clk_frequency); + if (ret) { + dev_err(&codec_dev->dev, + "clock-frequency missing or invalid\n"); + goto fail; + } + } else { + data->clk_frequency = clk_get_rate(data->codec_clk); + clk_prepare_enable(data->codec_clk); + } + + data->dai.name = "HiFi"; + data->dai.stream_name = "HiFi"; + data->dai.codec_dai_name = "wm8962"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.platform_of_node = ssi_np; + data->dai.ops = &imx_hifi_ops; + data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto clk_fail; + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto clk_fail; + data->card.num_links = 1; + data->card.dai_link = &data->dai; + data->card.dapm_widgets = imx_wm8962_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); + + data->card.late_probe = imx_wm8962_late_probe, + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto clk_fail; + } + + platform_set_drvdata(pdev, data); + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; + +clk_fail: + if (data->codec_clk) { + clk_disable_unprepare(data->codec_clk); + clk_put(data->codec_clk); + } +fail: + if (ssi_np) + of_node_put(ssi_np); + if (codec_np) + of_node_put(codec_np); + + return ret; +} + +static int imx_wm8962_remove(struct platform_device *pdev) +{ + struct imx_wm8962_data *data = platform_get_drvdata(pdev); + + if (data->codec_clk) { + clk_disable_unprepare(data->codec_clk); + clk_put(data->codec_clk); + } + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_wm8962_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8962", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids); + +static struct platform_driver imx_wm8962_driver = { + .driver = { + .name = "imx-wm8962", + .owner = THIS_MODULE, + .of_match_table = imx_wm8962_dt_ids, + }, + .probe = imx_wm8962_probe, + .remove = imx_wm8962_remove, +}; +module_platform_driver(imx_wm8962_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8962"); -- 1.7.1

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[alsa-devel] [tinycompress][PATCH 0/2] Add support for non-blocking I/O
by Richard Fitzgerald 07 Jun '13

07 Jun '13

(Applies on top of the two patches I just uploaded to fix the write logic) Adds support for non-blocking write and read. The API for this is based on the non-blocking API used in the standard PCM alsalib.

2 3

[alsa-devel] [tinycompress][PATCH 0/2] Fixes for write loop logic
by Richard Fitzgerald 07 Jun '13

07 Jun '13

Two small patches which correct the logic of compress_write() so that it will work correctly if asked to write a buffer shorter than the fragment length. The first patch is needed to ensure that if the client requests auto-configuration of fragment size the internal copy of the config contains the configured fragment size. The second patch contains the actual fix to the write loop logic. Richard Fitzgerald (2): compress: copy final version of config struct compress: do not poll if enough space to write remaining data compress.c | 13 ++++++------- 1 files changed, 6 insertions(+), 7 deletions(-) -- 1.7.2.5

2 3

[alsa-devel] Sound issues with ALC269VB-based hardware
by Anisse Astier 06 Jun '13

06 Jun '13

Hi, So I have this hardware that has a few issues (all tested with tiwai/sound for-next) Audio out/speakers: - Audio does not work by default. It needs plugging a jack in either headphone or mic. - disabling "Enable Automute" does make it work. But then it doesn't mute when jack is plugged (obviously). Mic: - needs model=laptop-dmic so that working "Internal Mic 1" appears - "Internal Mic" is useless. - no jack sense, need to select manually "Internal Mic 1" or "Mic" (external) as Input Source Please find alsa-info.sh with and without model=laptop-dmic in attachement. Regards, Anisse

3 17

[alsa-devel] Generic clock divider indices
by Daniel Mack 06 Jun '13

06 Jun '13

Hi Mark, Liam, we're using generic ASoC platform code for some audio devices, which is agnostic of the actual Codecs that it deals with, as the information is provided by DT nodes only. That works very well so far, but I'm now facing a case where I need to pass a special clock divider information down to the codec driver. The fact that the IDs of the dividers can be arbitrarily defined by the drivers leads to the unfortunate situation that I now have to make the platform code aware of the codec drivers again. Would it be an idea to define some commonly used dividers in the core? If they start at 0x80000000, they won't collide with existing ones, and we could simply add them as alternative case statements to existing drivers. Some commonly used ones I can think of are MCLK / BCLK MCLK / LRCLK BCLK / LRCLK That way, platform code could just pass the values dows to the codec drivers, whether or not they know what to do with it. What do you think? Daniel

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Re: [alsa-devel] "https://bugtrack.alsa-project.org/alsa-bug" Tascam US122L
by Takashi Iwai 06 Jun '13

06 Jun '13

Please don't drop Cc to ML. At Thu, 6 Jun 2013 15:03:56 +0100, James Miller wrote: > I'm using us122l. I've read this : > https://bugzilla.kernel.org/show_bug.cgi?id=53241 OK, then try to stace and see what gives error. Also, build your driver with CONFIG_SND_DEBUG_VERBOSE=y and see the kernel messages, too. The problem looks like the mmap. The strange thing is that the mmap code hasn't been changed so much and it's relatively simple in us122l.c. > and the bug is being addressed but there seems like no change so far. When > you say the drivers are pretty hackish - do you think that I might be able > to address them myself? I am about to begin some driver programming and > have a little previous development experience plus I'd be really stoked for > a decent us122l driver. If you know of any resources that might help with > the module driver code then I'd be grateful. Thanks. Nothing special I know of, the standard kernel coding book like "Linux Device Drivers" should be a good starting point to learn. Takashi > On Thu, Jun 6, 2013 at 2:33 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > At Thu, 6 Jun 2013 13:56:32 +0100, > > James Miller wrote: > > > > > > Hi, I can't visit this bugtracker page (above) and am also wondering if > > > there is any progress on the Tascam US122L. I understand someone from > > the > > > kxaudio team has tried a kernel bisect but that the problem lies in the > > > driver with a vm switch. > > > > > > http://www.remastersys.com/forums/index.php?topic=3013.0 > > > > > > somewhere I read that this was an alsa problem but I found this kernel > > > patch (referred to) and it has been suggested that it is this that is the > > > cause. I am not a prof. developer but might have a go at recompiling my > > > kernel after changing the module so that I can get my soundcard working > > > with kernel 3.8.13-rt(patch). Do you guys know anything about this that > > > might help me? > > > > > > https://patchwork.kernel.org/patch/1258951/ > > > > Which driver are you using? Is snd-usb-us122l or snd-usb-usx2y? > > > > If the patch above is really the culprit of breakage, the driver must > > have been already broken before that and it just had worked with > > luck. Both drivers are pretty hackish, and need real fixes... > > > > > > Takashi > > > > > > -- > James Stewart Miller Bsc(hons) Psych. > [2 <text/html; ISO-8859-1 (quoted-printable)>] >

1 0

[alsa-devel] "https://bugtrack.alsa-project.org/alsa-bug" Tascam US122L
by James Miller 06 Jun '13

06 Jun '13

Hi, I can't visit this bugtracker page (above) and am also wondering if there is any progress on the Tascam US122L. I understand someone from the kxaudio team has tried a kernel bisect but that the problem lies in the driver with a vm switch. http://www.remastersys.com/forums/index.php?topic=3013.0 somewhere I read that this was an alsa problem but I found this kernel patch (referred to) and it has been suggested that it is this that is the cause. I am not a prof. developer but might have a go at recompiling my kernel after changing the module so that I can get my soundcard working with kernel 3.8.13-rt(patch). Do you guys know anything about this that might help me? https://patchwork.kernel.org/patch/1258951/ thanks James ubuntu studio on mac pro 2006 3.8.13-rt9 #1 SMP PREEMPT RT Thu May 30 18:54:25 BST 2013 x86_64 x86_64 x86_64 GNU/Linux -- James Stewart Miller Bsc(hons) Psych.

2 1

[alsa-devel] [PATCH V2 0/2] Add device tree binding support to WM8962 codec driver
by Nicolin Chen 06 Jun '13

06 Jun '13

*** These two patch added devicetree binding support to WM8962 codec driver *** so that it can fetch pdata not only from old platform_data but also from *** newer dts binding. --- ChangeLog: v1->v2: * PATCH 1/2: *** Embedded a copy of struct wm8962_pdata instead of its pointer *** Removed all 'if (pdata)' checking code as it is always valid. * PATCH 2/2: *** Dropped out-of-band value error msg by regarding it as default value. Nicolin Chen (2): ASoC: WM8962: Create default platform data structure ASoC: WM8962: Add device tree binding Documentation/devicetree/bindings/sound/wm8962.txt | 20 ++++ sound/soc/codecs/wm8962.c | 99 +++++++++++++------ 2 files changed, 88 insertions(+), 31 deletions(-)

2 4

[alsa-devel] [PATCH 1/2] ALSA: hda - Introduce bit flags to snd_hda_codec_read/write()
by Takashi Iwai 06 Jun '13

06 Jun '13

snd_hda_codec_read(), snd_hda_codec_write() & co take the argument "direct" that indicates whether the given NID is a direct reference or an indirect reference. However, the indirect reference is practically unimplemented and never exists. And moreover, we don't need the indication of indirect reference at this high level, as NID can be represented in 16bit values at this point. Meanwhile, there are some cases where it'd be nice to give some operational options to these functions. So, we can reuse this argument as a new bit flag! Pretty frugal, eh? All callers so far pass zero to this argument, thus there is no behavior change by this replacement. The real usage of this new bit option will be added in the following patches. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 41 ++++++++++++++++++++--------------------- sound/pci/hda/hda_codec.h | 8 ++++---- 2 files changed, 24 insertions(+), 25 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 679fba4..503869a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -185,20 +185,19 @@ EXPORT_SYMBOL_HDA(snd_hda_get_jack_type); * Compose a 32bit command word to be sent to the HD-audio controller */ static inline unsigned int -make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, +make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int flags, unsigned int verb, unsigned int parm) { u32 val; - if ((codec->addr & ~0xf) || (direct & ~1) || (nid & ~0x7f) || + if ((codec->addr & ~0xf) || (nid & ~0x7f) || (verb & ~0xfff) || (parm & ~0xffff)) { - printk(KERN_ERR "hda-codec: out of range cmd %x:%x:%x:%x:%x\n", - codec->addr, direct, nid, verb, parm); + printk(KERN_ERR "hda-codec: out of range cmd %x:%x:%x:%x\n", + codec->addr, nid, verb, parm); return ~0; } val = (u32)codec->addr << 28; - val |= (u32)direct << 27; val |= (u32)nid << 20; val |= verb << 8; val |= parm; @@ -209,7 +208,7 @@ make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, * Send and receive a verb */ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, - unsigned int *res) + int flags, unsigned int *res) { struct hda_bus *bus = codec->bus; int err; @@ -255,7 +254,7 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, * snd_hda_codec_read - send a command and get the response * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -264,12 +263,12 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, * Returns the obtained response value, or -1 for an error. */ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, - int direct, + int flags, unsigned int verb, unsigned int parm) { - unsigned cmd = make_codec_cmd(codec, nid, direct, verb, parm); + unsigned cmd = make_codec_cmd(codec, nid, flags, verb, parm); unsigned int res; - if (codec_exec_verb(codec, cmd, &res)) + if (codec_exec_verb(codec, cmd, flags, &res)) return -1; return res; } @@ -279,7 +278,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_read); * snd_hda_codec_write - send a single command without waiting for response * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -287,12 +286,12 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_read); * * Returns 0 if successful, or a negative error code. */ -int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, - unsigned int verb, unsigned int parm) +int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, + unsigned int verb, unsigned int parm) { - unsigned int cmd = make_codec_cmd(codec, nid, direct, verb, parm); + unsigned int cmd = make_codec_cmd(codec, nid, flags, verb, parm); unsigned int res; - return codec_exec_verb(codec, cmd, + return codec_exec_verb(codec, cmd, flags, codec->bus->sync_write ? &res : NULL); } EXPORT_SYMBOL_HDA(snd_hda_codec_write); @@ -3582,7 +3581,7 @@ EXPORT_SYMBOL_HDA(snd_hda_create_spdif_in_ctls); * snd_hda_codec_write_cache - send a single command with caching * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -3591,7 +3590,7 @@ EXPORT_SYMBOL_HDA(snd_hda_create_spdif_in_ctls); * Returns 0 if successful, or a negative error code. */ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm) + int flags, unsigned int verb, unsigned int parm) { int err; struct hda_cache_head *c; @@ -3600,7 +3599,7 @@ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, cache_only = codec->cached_write; if (!cache_only) { - err = snd_hda_codec_write(codec, nid, direct, verb, parm); + err = snd_hda_codec_write(codec, nid, flags, verb, parm); if (err < 0) return err; } @@ -3624,7 +3623,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_write_cache); * snd_hda_codec_update_cache - check cache and write the cmd only when needed * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -3635,7 +3634,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_write_cache); * Returns 0 if successful, or a negative error code. */ int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm) + int flags, unsigned int verb, unsigned int parm) { struct hda_cache_head *c; u32 key; @@ -3651,7 +3650,7 @@ int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid, return 0; } mutex_unlock(&codec->bus->cmd_mutex); - return snd_hda_codec_write_cache(codec, nid, direct, verb, parm); + return snd_hda_codec_write_cache(codec, nid, flags, verb, parm); } EXPORT_SYMBOL_HDA(snd_hda_codec_update_cache); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c93f902..39a658e 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -945,9 +945,9 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec); * low level functions */ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, - int direct, + int flags, unsigned int verb, unsigned int parm); -int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, +int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, unsigned int verb, unsigned int parm); #define snd_hda_param_read(codec, nid, param) \ snd_hda_codec_read(codec, nid, 0, AC_VERB_PARAMETERS, param) @@ -986,11 +986,11 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex); /* cached write */ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm); + int flags, unsigned int verb, unsigned int parm); void snd_hda_sequence_write_cache(struct hda_codec *codec, const struct hda_verb *seq); int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm); + int flags, unsigned int verb, unsigned int parm); void snd_hda_codec_resume_cache(struct hda_codec *codec); /* both for cmd & amp caches */ void snd_hda_codec_flush_cache(struct hda_codec *codec); -- 1.8.3

1 1

[alsa-devel] [PATCH] ALSA: hda - Drop hard dependency on CONFIG_SND_DYNAMIC_MINORS
by Takashi Iwai 06 Jun '13

06 Jun '13

Currently HDMI codec driver sets the hard dependency (reverse selection) on CONFIG_SND_DYNAMIC_MINORS because the recent codecs may support more than two PCMs. But, this doesn't mean that we need always this option, since there can be a single PCM stream even with the modern codecs. This patch drops the hard dependency again but give more sensible error message when no enough PCMs are available due to the lack of this option. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/Kconfig | 1 - sound/pci/hda/hda_codec.c | 19 +++++++++++++++---- 2 files changed, 15 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 80a7d44..0c5371a 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -140,7 +140,6 @@ config SND_HDA_CODEC_VIA config SND_HDA_CODEC_HDMI bool "Build HDMI/DisplayPort HD-audio codec support" - select SND_DYNAMIC_MINORS default y help Say Y here to include HDMI and DisplayPort HD-audio codec diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 55108b5..679fba4 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4461,12 +4461,13 @@ const char *snd_hda_pcm_type_name[HDA_PCM_NTYPES] = { /* * get the empty PCM device number to assign - * - * note the max device number is limited by HDA_MAX_PCMS, currently 10 */ -static int get_empty_pcm_device(struct hda_bus *bus, int type) +static int get_empty_pcm_device(struct hda_bus *bus, unsigned int type) { /* audio device indices; not linear to keep compatibility */ + /* assigned to static slots up to dev#10; if more needed, assign + * the later slot dynamically (when CONFIG_SND_DYNAMIC_MINORS=y) + */ static int audio_idx[HDA_PCM_NTYPES][5] = { [HDA_PCM_TYPE_AUDIO] = { 0, 2, 4, 5, -1 }, [HDA_PCM_TYPE_SPDIF] = { 1, -1 }, @@ -4480,18 +4481,28 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) return -EINVAL; } - for (i = 0; audio_idx[type][i] >= 0 ; i++) + for (i = 0; audio_idx[type][i] >= 0; i++) { +#ifndef CONFIG_SND_DYNAMIC_MINORS + if (audio_idx[type][i] >= 8) + break; +#endif if (!test_and_set_bit(audio_idx[type][i], bus->pcm_dev_bits)) return audio_idx[type][i]; + } +#ifdef CONFIG_SND_DYNAMIC_MINORS /* non-fixed slots starting from 10 */ for (i = 10; i < 32; i++) { if (!test_and_set_bit(i, bus->pcm_dev_bits)) return i; } +#endif snd_printk(KERN_WARNING "Too many %s devices\n", snd_hda_pcm_type_name[type]); +#ifndef CONFIG_SND_DYNAMIC_MINORS + snd_printk(KERN_WARNING "Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y\n"); +#endif return -EAGAIN; } -- 1.8.3

1 0

[alsa-devel] hda question: 'Headphone Playback Switch' cannot unmute the pin's output amplifier?
by Lin, Mengdong 06 Jun '13

06 Jun '13

Hi, I'm using a Realtek HD-A codec ALC282. But the headphone output amplifier is always muted and I cannot unmute it by amixer. I tried to unmute the pin via control 'Headphone Playback Switch', but snd_hda_mixer_amp_switch_put() gets a ZERO value from ucontrol->value.integer.value, thus the pin will be muted. Could somebody give me some tips to further check this issue? Many thanks! Here are the headphone pin info: Node 0x21 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000001c: OUT HP Detect Pin Default 0x0321401f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Green 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: 2 0x0c 0x0d* Here is the pin's controls listed by amixer: numid=2,iface=MIXER,name='Headphone Playback Switch' Then I use this command to unmute the pin's amplifier: amixer -c 1 cset numid=2,iface=MIXER,name='Headphone Playback Switch' on My debug message show that snd_hda_mixer_amp_switch_put() gets zero value , indicating to 'mute': [ 346.263519] snd_hda_mixer_amp_switch_put: nid 0x0021, chs 3, control value [0x00000000, 0x00000000] [ 346.263527] snd_hda_codec_amp_update: nid 0x0021 [ 346.263532] codec_amp_update: nid 0x0021, ch 0, direction 1 (OUT 1), index 0, val 0x00000080 [ 346.263536] codec_amp_update: amp val not change [ 346.263538] snd_hda_codec_amp_update: nid 0x0021 [ 346.263542] codec_amp_update: nid 0x0021, ch 1, direction 1 (OUT 1), index 0, val 0x00000080 [ 346.263544] codec_amp_update: amp val not change Thanks Mengdong

3 5

[alsa-devel] [PATCH 0/2] Add device tree binding support to WM8962 codec driver
by Nicolin Chen 06 Jun '13

06 Jun '13

*** These two patch added devicetree binding support to WM8962 codec driver *** so that it can fetch pdata not only from old platform_data but also from *** newer dts binding. Nicolin Chen (2): ASoC: WM8962: Create default platform data structure ASoC: WM8962: Add device tree binding Documentation/devicetree/bindings/sound/wm8962.txt | 20 +++++++ sound/soc/codecs/wm8962.c | 56 ++++++++++++++++++-- 2 files changed, 72 insertions(+), 4 deletions(-)

2 6

Re: [alsa-devel] [PATCH V5 4/5] ARM: dts: add clock provider information for i2s controllers in Exynos5250
by Padma Venkat 06 Jun '13

06 Jun '13

Hi Doug, On Tue, Jun 4, 2013 at 10:49 PM, Doug Anderson <dianders(a)chromium.org> wrote: > Padmavathi, > > On Tue, Jun 4, 2013 at 5:28 AM, Padmavathi Venna <padma.v(a)samsung.com> wrote: >> @@ -471,6 +477,8 @@ >> dmas = <&pdma1 12 >> &pdma1 11>; >> dma-names = "tx", "rx"; >> + clocks = <&clock 307>; >> + clock-names = "iis"; > > ...actually, glancing at the driver I'm a little surprised that you > don't need to list "i2s_opclk0". Did you test i2s1? I didn't test i2s1 but i2s_opclk0 is also required. I will resubmit the patch. > >> pinctrl-names = "default"; >> pinctrl-0 = <&i2s1_bus>; >> }; >> @@ -481,6 +489,8 @@ >> dmas = <&pdma0 12 >> &pdma0 11>; >> dma-names = "tx", "rx"; >> + clocks = <&clock 308>; >> + clock-names = "iis"; > > ...and here. > > > -Doug Thanks Padma

1 0

[alsa-devel] [PATCH] ASoC: dapm: Treat DAI widgets like AIF widgets for power
by Mark Brown 06 Jun '13

06 Jun '13

Even though they are virtual widgets DAI widgets still get counted for the DAPM context power management so we can't just use the active state to check if they should be powered as they may not be part of a complete path. Instead split them into input and output widgets and do the same power checks as we perform on AIFs. Reported-by: Stephen Warren <swarren(a)nvidia.com> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- Compile tested only thus far. include/sound/soc-dapm.h | 3 ++- sound/soc/soc-dapm.c | 49 ++++++++++++++++++++++++---------------------- sound/soc/soc-pcm.c | 7 ++++++- 3 files changed, 34 insertions(+), 25 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index f4f85dd..69e8e07 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -452,7 +452,8 @@ enum snd_soc_dapm_type { snd_soc_dapm_aif_in, /* audio interface input */ snd_soc_dapm_aif_out, /* audio interface output */ snd_soc_dapm_siggen, /* signal generator */ - snd_soc_dapm_dai, /* link to DAI structure */ + snd_soc_dapm_dai_in, /* link to DAI structure */ + snd_soc_dapm_dai_out, snd_soc_dapm_dai_link, /* link between two DAI structures */ }; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3507346..fdb14e4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -55,7 +55,8 @@ static int dapm_up_seq[] = { [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_micbias] = 2, [snd_soc_dapm_dai_link] = 2, - [snd_soc_dapm_dai] = 3, + [snd_soc_dapm_dai_in] = 3, + [snd_soc_dapm_dai_out] = 3, [snd_soc_dapm_aif_in] = 3, [snd_soc_dapm_aif_out] = 3, [snd_soc_dapm_mic] = 4, @@ -92,7 +93,8 @@ static int dapm_down_seq[] = { [snd_soc_dapm_value_mux] = 9, [snd_soc_dapm_aif_in] = 10, [snd_soc_dapm_aif_out] = 10, - [snd_soc_dapm_dai] = 10, + [snd_soc_dapm_dai_in] = 10, + [snd_soc_dapm_dai_out] = 10, [snd_soc_dapm_dai_link] = 11, [snd_soc_dapm_clock_supply] = 12, [snd_soc_dapm_regulator_supply] = 12, @@ -419,7 +421,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: case snd_soc_dapm_hp: case snd_soc_dapm_mic: case snd_soc_dapm_spk: @@ -813,7 +816,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_out: if (widget->active) { widget->outputs = snd_soc_dapm_suspend_check(widget); return widget->outputs; @@ -909,7 +912,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_dac: case snd_soc_dapm_aif_in: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_in: if (widget->active) { widget->inputs = snd_soc_dapm_suspend_check(widget); return widget->inputs; @@ -1128,16 +1131,6 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) return out != 0 && in != 0; } -static int dapm_dai_check_power(struct snd_soc_dapm_widget *w) -{ - DAPM_UPDATE_STAT(w, power_checks); - - if (w->active) - return w->active; - - return dapm_generic_check_power(w); -} - /* Check to see if an ADC has power */ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) { @@ -2327,7 +2320,8 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: case snd_soc_dapm_dai_link: list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &wsink->sources); @@ -3134,10 +3128,12 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, break; case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: + case snd_soc_dapm_dai_out: w->power_check = dapm_adc_check_power; break; case snd_soc_dapm_dac: case snd_soc_dapm_aif_in: + case snd_soc_dapm_dai_in: w->power_check = dapm_dac_check_power; break; case snd_soc_dapm_pga: @@ -3157,9 +3153,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_clock_supply: w->power_check = dapm_supply_check_power; break; - case snd_soc_dapm_dai: - w->power_check = dapm_dai_check_power; - break; default: w->power_check = dapm_always_on_check_power; break; @@ -3380,7 +3373,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.reg = SND_SOC_NOPM; if (dai->driver->playback.stream_name) { - template.id = snd_soc_dapm_dai; + template.id = snd_soc_dapm_dai_in; template.name = dai->driver->playback.stream_name; template.sname = dai->driver->playback.stream_name; @@ -3398,7 +3391,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, } if (dai->driver->capture.stream_name) { - template.id = snd_soc_dapm_dai; + template.id = snd_soc_dapm_dai_out; template.name = dai->driver->capture.stream_name; template.sname = dai->driver->capture.stream_name; @@ -3428,8 +3421,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) /* For each DAI widget... */ list_for_each_entry(dai_w, &card->widgets, list) { - if (dai_w->id != snd_soc_dapm_dai) + switch (w->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + break; + default: continue; + } dai = dai_w->priv; @@ -3438,8 +3436,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) if (w->dapm != dai_w->dapm) continue; - if (w->id == snd_soc_dapm_dai) + switch (w->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: continue; + default: + break; + } if (!w->sname) continue; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a14335e..b6c6403 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -931,8 +931,13 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, /* Create any new FE <--> BE connections */ for (i = 0; i < list->num_widgets; i++) { - if (list->widgets[i]->id != snd_soc_dapm_dai) + switch (list->widgets[i]->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + break; + default: continue; + } /* is there a valid BE rtd for this widget */ be = dpcm_get_be(card, list->widgets[i], stream); -- 1.7.10.4

3 2

[alsa-devel] [PATCH] ASoC: max98090: Guard runtime PM callbacks
by Mark Brown 05 Jun '13

05 Jun '13

Otherwise the functions will be defined but unreferenced when runtime PM is disabled, generating warnings. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/max98090.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 1bf96bd..ad5313f 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2350,6 +2350,7 @@ static int max98090_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME static int max98090_runtime_resume(struct device *dev) { struct max98090_priv *max98090 = dev_get_drvdata(dev); @@ -2369,6 +2370,7 @@ static int max98090_runtime_suspend(struct device *dev) return 0; } +#endif static const struct dev_pm_ops max98090_pm = { SET_RUNTIME_PM_OPS(max98090_runtime_suspend, -- 1.7.10.4

1 0

[alsa-devel] [PATCH (alsa-lib)] chmap: Always succeed setting the map to what it already is
by David Henningsson 05 Jun '13

05 Jun '13

If we try to set the channel map to what it already is, this should always succeed. E g, speaker-test can do this sometimes. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/pcm/pcm.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 7ec1f0e..ca4d416 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -7361,6 +7361,13 @@ OBSOLETE1(snd_pcm_sw_params_get_silence_size, ALSA_0.9, ALSA_0.9.0rc4); #endif /* DOC_HIDDEN */ +static int chmap_equal(const snd_pcm_chmap_t *a, const snd_pcm_chmap_t *b) +{ + if (a->channels != b->channels) + return 0; + return !memcmp(a->pos, b->pos, a->channels * sizeof(a->pos[0])); +} + /** * \!brief Query the available channel maps * \param pcm PCM handle to query @@ -7415,6 +7422,10 @@ snd_pcm_chmap_t *snd_pcm_get_chmap(snd_pcm_t *pcm) */ int snd_pcm_set_chmap(snd_pcm_t *pcm, const snd_pcm_chmap_t *map) { + const snd_pcm_chmap_t *oldmap = snd_pcm_get_chmap(pcm); + if (oldmap && chmap_equal(oldmap, map)) + return 0; + if (!pcm->ops->set_chmap) return -ENXIO; return pcm->ops->set_chmap(pcm, map); -- 1.7.9.5

2 1

Re: [alsa-devel] CONFIG_SND_DEBUG
by David Henningsson 05 Jun '13

05 Jun '13

On 06/05/2013 12:53 PM, Andy Whitcroft wrote: > On Wed, Jun 05, 2013 at 09:47:30AM +0200, David Henningsson wrote: >>>>> On Wed, 2013-06-05 at 08:54 +0200, Takashi Iwai wrote: >>>>>> At Tue, 04 Jun 2013 23:52:01 -0700, Joe Perches wrote: >>>>>>> On Wed, 2013-06-05 at 08:32 +0200, Takashi Iwai wrote: >>>>>>>> Yes. These are snd_printd() just to be conditionally built in. >>>>>>>> But in most cases it's rather useful to print them (as most distros >>>>>>>> set CONFIG_SND_DEBUG=y). >>>>>>> Ubuntu doesn't, I believe Fedora doesn't. >>>>>> Then they should have done so :) >>>>> But they don't, so what distros do? >>>> >>>> RedHat (including Fedora) and SUSE do at least. >>> >>> Mandriva does too. (still looking around for others) >>> >>> We can ask Ubuntu to enable CONFIG_SND_DEBUG. >>> (cc'd Andy Whitcroft, Leann Ogasawara and David Henningsson) >>> >>> Maybe there are others Canonical folk that >>> should be cc'd? >> >> Adding kernel team mailing list to CC. >> >> Andy/Leann - apparently CONFIG_SND_DEBUG is on by default from >> upstream, and we explicitly disable it. Is there any reason why we >> do that? > > config SND_DEBUG > bool "Debug" > help > Say Y here to enable ALSA debug code. > > It is off by default in upstream, and the really helpful description > would cirtainly tend to lead to it being disabled. But if it is helpful > to your debugging efforts David then I suspect we can enable it in Saucy > and see what happens. Okay, so then the ball is back in Takashi's area - if we're recommended to turn CONFIG_SND_DEBUG on, why is it off by default in the upstream Linux kernel? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

3 4

[alsa-devel] [PATCH 1/2] ASoC: arizona: Correct AEC loopback enable
by Charles Keepax 05 Jun '13

05 Jun '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 3 ++- sound/soc/codecs/wm5110.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index e895d39..100fdad 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1120,7 +1120,8 @@ SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, ARIZONA_DSP_WIDGETS(DSP1, "DSP1"), SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm5102_aec_loopback_mux), SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index ba38f06..88ad7db 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -503,7 +503,8 @@ SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5110_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm5110_aec_loopback_mux), SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), -- 1.7.2.5

2 3

[alsa-devel] [PATCH 0/2] snd-firewire-lib: add MIDI stream support
by o-takashi＠sakamocchi.jp 05 Jun '13

05 Jun '13

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> This series of patch add MIDI stream support to snd-firewire-lib according to IEC 61883-6 and MMA/AMEI RP-027. This series of patch is based on my previous patches for snd-firewire-lib. [PATCH v2 0/3] snd-firewire-lib: add handling CMP output connection http://mailman.alsa-project.org/pipermail/alsa-devel/2013-April/061607.html [PATCH v2 0/4] snd-firewire-lib: add handling AMDTP receive stream http://mailman.alsa-project.org/pipermail/alsa-devel/2013-April/061611.html Takashi Sakamoto (2): Add amdtp_stream_running() and amdtp_stream_pcm_running() Add MIDI stream support with data rate restriction sound/firewire/amdtp.c | 171 +++++++++++++++++++++++++++++++++++++++++++----- sound/firewire/amdtp.h | 37 +++++++++++ 2 files changed, 193 insertions(+), 15 deletions(-) -- 1.7.10.4

3 5

Re: [alsa-devel] [PATCH] ASoC: add RT5640 CODEC driver
by Stephen Warren 05 Jun '13

05 Jun '13

On 05/31/2013 01:04 AM, bardliao(a)realtek.com wrote: > From: Bard Liao <bardliao(a)realtek.com> > > * Use regmap_range_cfg to replace index read/write function. > * Remove I2S3 related code since there is no I2S3 in ALC5640. > * Remove Voice DSP related code since there is no Voice DSP in ALC5640. > * Remove MICBIAS2 since there is no MICBIAS2 in ALC5640. > * Change DMIC1/2 CLK to DMIC1/2 Power since it is for enable/disable DMIC1/2 > * Modify some texts for consistent coding style. > * Merge sto adc l/r mux since it shares the same control bits. > * Other minor changes. > +static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { > + /* Input Lines */ > + SND_SOC_DAPM_INPUT("DMIC1"), > + SND_SOC_DAPM_INPUT("DMIC2"), Can you explain that more? In the spec for this part, I don't see any DMIC1/2 input pins. Instead, I think they're alternative uses for the IN1P/N pins, right? > + SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, > + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), > + SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, > + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), In the version of the driver I posted, I had replaced this with a single "HPO MIX" DAPM_MIXER, because there is a single bit in HW that controls both the L and R channels. I think you should incorporate that same change. That relies on commit 85762e7 "ASoC: dapm: Implement mixer control sharing". In an earlier review, IIRC, Mark had asked you to implement platform data to configure whether the microphones were single-ended or not. I don't see that in this patch. Finally, I really would prefer if you could implement the support for the LDO1_EN GPIO via platform data. Getting that right requires a bit of knowledge of how set_bias_level() is supposed to work on this CODEC, and I think Realtek have the best information to get that right. I'll certainly send a patch to implement this all by device tree after that. Anyway, I'll go test this patch and see if it works for me on my HW.

2 2

[alsa-devel] [PATCH] ASoC: fsl: add imx-wm8962 machine driver
by Nicolin Chen 05 Jun '13

05 Jun '13

This is the initial imx-wm8962 device-tree-only machine driver working with fsl_ssi driver. More features can be added on top of it later. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- .../devicetree/bindings/sound/imx-audio-wm8962.txt | 61 +++ sound/soc/fsl/Kconfig | 12 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8962.c | 387 ++++++++++++++++++++ 4 files changed, 462 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt create mode 100644 sound/soc/fsl/imx-wm8962.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt new file mode 100644 index 0000000..cb188a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt @@ -0,0 +1,61 @@ +Freescale i.MX audio complex with WM8962 codec + +Required properties: +- compatible : "fsl,imx-audio-wm8962" +- model : The user-visible name of this sound complex +- ssi-controller : The phandle of the i.MX SSI controller +- audio-codec : The phandle of the WM8962 audio codec +- audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names could be power + supplies, WM8962 pins, and the jacks on the board: + + Power supplies: + * Mic Bias + + WM8962 pins: + * HPOUTL + * HPOUTR + * SPKOUTL + * SPKOUTR + * MICBIAS + * IN3R + * DMIC + * DMICDAT + + Board connectors: + * Mic Jack + * Headphone Jack + * Ext Spk + +- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) +- mux-ext-port : The external port of the i.MX audio muxer + +- hp-det-gpios : The gpio port of Headphone detection. +- mic-det-gpios: The gpio port of Micphone detection. + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + +sound { + compatible = "fsl,imx6q-sabresd-wm8962", + "fsl,imx-audio-wm8962"; + model = "wm8962-audio"; + ssi-controller = <&ssi2>; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUTL", + "Headphone Jack", "HPOUTR", + "Ext Spk", "SPKOUTL", + "Ext Spk", "SPKOUTR", + "MICBIAS", "AMIC", + "IN3R", "MICBIAS", + "DMIC", "MICBIAS", + "DMICDAT", "DMIC"; + mux-int-port = <2>; + mux-ext-port = <3>; + hp-det-gpios = <&gpio7 8 1>; /* active low */ + mic-det-gpios = <&gpio1 9 1>; /* active low */ +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 7860cc2..aa43854 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -168,6 +168,18 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8962 + tristate "SoC Audio support for i.MX boards with wm8962" + depends on OF && I2C + select SND_SOC_WM8962 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + help + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8962 codec. + config SND_SOC_IMX_SGTL5000 tristate "SoC Audio support for i.MX boards with sgtl5000" depends on OF && I2C diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 91883f8..d4b4aa8 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -42,6 +42,7 @@ snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o @@ -49,4 +50,5 @@ obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c new file mode 100644 index 0000000..7aa42d8 --- /dev/null +++ b/sound/soc/fsl/imx-wm8962.c @@ -0,0 +1,387 @@ +/* + * Copyright 2013 Freescale Semiconductor, Inc. + * + * Based on imx-sgtl5000.c + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/of_i2c.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <linux/clk.h> +#include <linux/irq.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/pinctrl/consumer.h> + +#include "../codecs/wm8962.h" +#include "imx-audmux.h" + +#define DAI_NAME_SIZE 32 + +struct imx_wm8962_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + struct clk *codec_clk; + unsigned int clk_frequency; +}; + +struct imx_priv { + int hp_gpio; + int hp_active_low; + int hp_status; + int amic_gpio; + int amic_active_low; + int amic_status; + struct platform_device *pdev; + struct snd_pcm_substream *first_stream; + struct snd_pcm_substream *second_stream; +}; +static struct imx_priv card_priv; + +static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static int check_hw_params(struct snd_pcm_substream *substream, + snd_pcm_format_t sample_format, unsigned int sample_rate) +{ + struct imx_priv *priv = &card_priv; + struct device *dev = substream->pcm->card->dev; + + substream->runtime->sample_bits = + snd_pcm_format_physical_width(sample_format); + substream->runtime->rate = sample_rate; + substream->runtime->format = sample_format; + + if (!priv->first_stream) { + priv->first_stream = substream; + } else { + priv->second_stream = substream; + + if (priv->first_stream->runtime->rate != + priv->second_stream->runtime->rate) { + dev_err(dev, "KEEP THE SAME SAMPLE RATE: %d!\n", + priv->first_stream->runtime->rate); + return -EINVAL; + } + if (priv->first_stream->runtime->sample_bits != + priv->second_stream->runtime->sample_bits) { + snd_pcm_format_t first_format = + priv->first_stream->runtime->format; + + dev_err(dev, "KEEP THE SAME FORMAT: %s!\n", + snd_pcm_format_name(first_format)); + return -EINVAL; + } + } + + return 0; +} + +static int imx_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct device *dev = substream->pcm->card->dev; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + unsigned int sample_rate = params_rate(params); + unsigned int sample_format = params_format(params); + int ret = 0; + unsigned int pll_out; + + /* + * WM8962 doesn't support two substreams in different sample rates or + * sample formats. So check the two parameters of two substreams' when + * there're two substreams of playback and of capture in the same time. + */ + ret = check_hw_params(substream, sample_format, sample_rate); + if (ret < 0) { + dev_err(dev, "Failed to match hw params: %d\n", ret); + return ret; + } + + if (sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = sample_rate * 384; + else + pll_out = sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, + data->clk_frequency, pll_out); + if (ret < 0) { + dev_err(dev, "Failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, + pll_out, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(dev, "Failed to set SYSCLK: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct device *dev = substream->pcm->card->dev; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + int ret; + + /* We don't need to handle anything if there's no substream running */ + if (!priv->first_stream) + return 0; + + if (priv->first_stream == substream) + priv->first_stream = priv->second_stream; + priv->second_stream = NULL; + + if (!priv->first_stream) { + /* + * SYSCLK of wm8962's not disabled yet. If we wanna close FLL, + * we need to keep SYSCLK alive by changing its source to MCLK. + */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + data->clk_frequency, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(dev, "Failed to set SYSCLK: %d\n", ret); + return ret; + } + + /* Disable FLL and let codec do pm_runtime_put() */ + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, 0, 0, 0); + if (ret < 0) { + dev_err(dev, "Failed to disable FLL: %d\n", ret); + return ret; + } + } + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_hifi_hw_params, + .hw_free = imx_hifi_hw_free, +}; + +static int imx_wm8962_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + data->clk_frequency, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(card->dev, "Failed to set sysclk in %s\n", __func__); + + return ret; +} + +static int imx_wm8962_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct imx_priv *priv = &card_priv; + struct i2c_client *codec_dev; + struct imx_wm8962_data *data; + int int_port, ext_port; + int ret; + + priv->pdev = pdev; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(&pdev->dev, "audmux internal port setup failed\n"); + return ret; + } + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(&pdev->dev, "audmux external port setup failed\n"); + return ret; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + return -EINVAL; + } + + priv->hp_gpio = of_get_named_gpio_flags(np, "hp-det-gpios", 0, + (enum of_gpio_flags *)&priv->hp_active_low); + priv->amic_gpio = of_get_named_gpio_flags(np, "mic-det-gpios", 0, + (enum of_gpio_flags *)&priv->amic_active_low); + + priv->first_stream = NULL; + priv->second_stream = NULL; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + data->codec_clk = clk_get(&codec_dev->dev, NULL); + if (IS_ERR(data->codec_clk)) { + /* assuming clock enabled by default */ + data->codec_clk = NULL; + ret = of_property_read_u32(codec_np, "clock-frequency", + &data->clk_frequency); + if (ret) { + dev_err(&codec_dev->dev, + "clock-frequency missing or invalid\n"); + goto fail; + } + } else { + data->clk_frequency = clk_get_rate(data->codec_clk); + clk_prepare_enable(data->codec_clk); + } + + data->dai.name = "HiFi"; + data->dai.stream_name = "HiFi"; + data->dai.codec_dai_name = "wm8962"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.platform_of_node = ssi_np; + data->dai.ops = &imx_hifi_ops, + data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto clk_fail; + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto clk_fail; + data->card.num_links = 1; + data->card.dai_link = &data->dai; + data->card.dapm_widgets = imx_wm8962_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); + + data->card.late_probe = imx_wm8962_late_probe, + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto clk_fail; + } + + platform_set_drvdata(pdev, data); + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; + +clk_fail: + if (data->codec_clk) { + clk_disable_unprepare(data->codec_clk); + clk_put(data->codec_clk); + } +fail: + if (ssi_np) + of_node_put(ssi_np); + if (codec_np) + of_node_put(codec_np); + + return ret; +} + +static int imx_wm8962_remove(struct platform_device *pdev) +{ + struct imx_wm8962_data *data = platform_get_drvdata(pdev); + + if (data->codec_clk) { + clk_disable_unprepare(data->codec_clk); + clk_put(data->codec_clk); + } + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_wm8962_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8962", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids); + +static struct platform_driver imx_wm8962_driver = { + .driver = { + .name = "imx-wm8962", + .owner = THIS_MODULE, + .of_match_table = imx_wm8962_dt_ids, + }, + .probe = imx_wm8962_probe, + .remove = imx_wm8962_remove, +}; +module_platform_driver(imx_wm8962_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8962"); -- 1.7.1

1 0

Re: [alsa-devel] USB audio Gadget - code 10 error
by anish singh 05 Jun '13

05 Jun '13

adding alsa-devel On Tue, Jun 4, 2013 at 10:35 PM, Eamon Hayden <eamon.hayden(a)cloudiumsystems.com> wrote: > I’m trying to use the linux usb audio gadget but I get the error : “This > device cannot start (code 10) when I connect to a windows 7/8 box. I don’t > have this problem when I connect to ubunto 12.0.4 > > > > I’m current using kernel 3.6.9, but have also had the issue with an older > kernel (2.6.39.4). > > > > I’m not sure if this is an appropriate issue for this mail list, but I need > to get a resolution to this... > > > > Thanks for your help. > > Eamon. > > > _______________________________________________ > Kernelnewbies mailing list > Kernelnewbies(a)kernelnewbies.org > http://lists.kernelnewbies.org/mailman/listinfo/kernelnewbies >

2 1

[alsa-devel] [PATCH] ASoC: tegra: add runtime PM to resume functions
by Stephen Warren 04 Jun '13

04 Jun '13

From: Stephen Warren <swarren(a)nvidia.com> Tegra HW needs clocks etc. active when touching registers. Make sure they are active during resume, by calling pm_runtime_get_sync() before touching HW. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra30_ahub.c | 4 ++++ sound/soc/tegra/tegra30_i2s.c | 9 ++++++++- 2 files changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 0f4787c..d554d46 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -640,8 +640,12 @@ static int tegra30_ahub_resume(struct device *dev) { int ret; + ret = pm_runtime_get_sync(dev); + if (ret < 0) + return ret; ret = regcache_sync(ahub->regmap_ahub); ret |= regcache_sync(ahub->regmap_apbif); + pm_runtime_put(dev); return ret; } diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index bd0ebc0..d04146c 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -527,8 +527,15 @@ static int tegra30_i2s_suspend(struct device *dev) static int tegra30_i2s_resume(struct device *dev) { struct tegra30_i2s *i2s = dev_get_drvdata(dev); + int ret; - return regcache_sync(i2s->regmap); + ret = pm_runtime_get_sync(dev); + if (ret < 0) + return ret; + ret = regcache_sync(i2s->regmap); + pm_runtime_put(dev); + + return ret; } #endif -- 1.8.1.5

2 1

[alsa-devel] [PATCH V2] ASoC: sgtl5000: Fix VAG_POWER enabling/disabling order
by Marek Vasut 04 Jun '13

04 Jun '13

The VAG_POWER must be enabled after all other bits in CHIP_ANA_POWER and disabled before any other bit in CHIP_ANA_POWER. See the SGTL5000 datasheet (Table 31, BIT 7, page 42-43). Failing to follow this order will result in ugly loud "POP" noise at the end of playback. To achieve such order, use the _PRE and _POST DAPM widgets to trigger the power_vag_event, where the event type check has to be fixed accordingly as well. Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Dong Aisheng <dong.aisheng(a)linaro.org> Cc: Eric Nelson <eric.nelson(a)boundarydevices.com> Cc: Fabio Estevam <fabio.estevam(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/sgtl5000.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) V2: Remove that mdelay/msleep regression diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 92bbfec..1994f01 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -151,12 +151,12 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { switch (event) { - case SND_SOC_DAPM_PRE_PMU: + case SND_SOC_DAPM_POST_PMU: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); break; - case SND_SOC_DAPM_POST_PMD: + case SND_SOC_DAPM_PRE_PMD: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, 0); msleep(400); @@ -217,12 +217,11 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { 0, SGTL5000_CHIP_DIG_POWER, 1, 0), - SND_SOC_DAPM_SUPPLY("VAG_POWER", SGTL5000_CHIP_ANA_POWER, 7, 0, - power_vag_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), + + SND_SOC_DAPM_PRE("VAG_POWER_PRE", power_vag_event), + SND_SOC_DAPM_POST("VAG_POWER_POST", power_vag_event), }; /* routes for sgtl5000 */ @@ -230,16 +229,13 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ - {"ADC", NULL, "VAG_POWER"}, {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ - {"DAC", NULL, "VAG_POWER"}, {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ - {"LINE_IN", NULL, "VAG_POWER"}, {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */ {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */ -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: omap-hdmi-codec: make the driver common for other SoCs
by Barry Song 04 Jun '13

04 Jun '13

to support HDMI on CSR SiRFprimaII and atlasVI, we need one more HDMI pseudo codec, rather than add a new driver, we can make omap HDMI codec common for other SoCs as well. then the omap-hdmi codec becomes a generic HDMI pseudo- codec as HDMI audio features depend on HDMI specification not on SoCs. Cc: Ricardo Neri <ricardo.neri(a)ti.com> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- sound/soc/codecs/Kconfig | 4 ++-- sound/soc/codecs/Makefile | 4 ++-- sound/soc/codecs/{omap-hdmi.c => hdmi.c} | 26 +++++++++++++------------- sound/soc/omap/omap-hdmi-card.c | 2 +- 4 files changed, 18 insertions(+), 18 deletions(-) rename sound/soc/codecs/{omap-hdmi.c => hdmi.c} (69%) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00..d8c4f3d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -53,7 +53,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C - select SND_SOC_OMAP_HDMI_CODEC if OMAP4_DSS_HDMI + select SND_SOC_HDMI_CODEC select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_SGTL5000 if I2C @@ -287,7 +287,7 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate -config SND_SOC_OMAP_HDMI_CODEC +config SND_SOC_HDMI_CODEC tristate config SND_SOC_PCM3008 diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b9e41c9..49ff127 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -41,7 +41,7 @@ snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o -snd-soc-omap-hdmi-codec-objs := omap-hdmi.o +snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -168,7 +168,7 @@ obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o -obj-$(CONFIG_SND_SOC_OMAP_HDMI_CODEC) += snd-soc-omap-hdmi-codec.o +obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/omap-hdmi.c b/sound/soc/codecs/hdmi.c similarity index 69% rename from sound/soc/codecs/omap-hdmi.c rename to sound/soc/codecs/hdmi.c index 529d064..2bcae2b 100644 --- a/sound/soc/codecs/omap-hdmi.c +++ b/sound/soc/codecs/hdmi.c @@ -1,5 +1,5 @@ /* - * ALSA SoC codec driver for HDMI audio on OMAP processors. + * ALSA SoC codec driver for HDMI audio codecs. * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/ * Author: Ricardo Neri <ricardo.neri(a)ti.com> * @@ -23,10 +23,10 @@ #define DRV_NAME "hdmi-audio-codec" -static struct snd_soc_codec_driver omap_hdmi_codec; +static struct snd_soc_codec_driver hdmi_codec; -static struct snd_soc_dai_driver omap_hdmi_codec_dai = { - .name = "omap-hdmi-hifi", +static struct snd_soc_dai_driver hdmi_codec_dai = { + .name = "hdmi-hifi", .playback = { .channels_min = 2, .channels_max = 8, @@ -39,31 +39,31 @@ static struct snd_soc_dai_driver omap_hdmi_codec_dai = { }, }; -static int omap_hdmi_codec_probe(struct platform_device *pdev) +static int hdmi_codec_probe(struct platform_device *pdev) { - return snd_soc_register_codec(&pdev->dev, &omap_hdmi_codec, - &omap_hdmi_codec_dai, 1); + return snd_soc_register_codec(&pdev->dev, &hdmi_codec, + &hdmi_codec_dai, 1); } -static int omap_hdmi_codec_remove(struct platform_device *pdev) +static int hdmi_codec_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; } -static struct platform_driver omap_hdmi_codec_driver = { +static struct platform_driver hdmi_codec_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, }, - .probe = omap_hdmi_codec_probe, - .remove = omap_hdmi_codec_remove, + .probe = hdmi_codec_probe, + .remove = hdmi_codec_remove, }; -module_platform_driver(omap_hdmi_codec_driver); +module_platform_driver(hdmi_codec_driver); MODULE_AUTHOR("Ricardo Neri <ricardo.neri(a)ti.com>"); -MODULE_DESCRIPTION("ASoC OMAP HDMI codec driver"); +MODULE_DESCRIPTION("ASoC generic HDMI codec driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/omap/omap-hdmi-card.c b/sound/soc/omap/omap-hdmi-card.c index d4eaa92..7e66e9c 100644 --- a/sound/soc/omap/omap-hdmi-card.c +++ b/sound/soc/omap/omap-hdmi-card.c @@ -35,7 +35,7 @@ static struct snd_soc_dai_link omap_hdmi_dai = { .cpu_dai_name = "omap-hdmi-audio-dai", .platform_name = "omap-pcm-audio", .codec_name = "hdmi-audio-codec", - .codec_dai_name = "omap-hdmi-hifi", + .codec_dai_name = "hdmi-hifi", }; static struct snd_soc_card snd_soc_omap_hdmi = { -- 1.8.2.3

2 1

[alsa-devel] [RFC PATCH] ALSA: hda - Ignore default pin config on Haswell HDMI codecs
by David Henningsson 04 Jun '13

04 Jun '13

According to Daniel Vetter, these pin configs should be ignored, because the graphics driver/userspace could route audio to these pins even if they are disabled by BIOS. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) It would be good to have this acked by someone at Intel before committing. diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e12f7a0..8c1bede 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1307,12 +1307,14 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) if (!(caps & (AC_PINCAP_HDMI | AC_PINCAP_DP))) return 0; - config = snd_hda_codec_get_pincfg(codec, pin_nid); - if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) - return 0; - if (codec->vendor_id == 0x80862807) intel_haswell_fixup_connect_list(codec, pin_nid); + else { + /* On Haswell, default pin config should be ignored */ + config = snd_hda_codec_get_pincfg(codec, pin_nid); + if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) + return 0; + } pin_idx = spec->num_pins; per_pin = snd_array_new(&spec->pins); -- 1.7.9.5

3 4

[alsa-devel] dsnoop memory leak
by Иван Больсунов 04 Jun '13

04 Jun '13

Since alsa bug tracker does not work for me for weeks, I have to report bug here. Sorry if it a wrong place. So, to reproduce memory leak (Alsa lib used v.1.0.27), you have to use next asound.conf: pcm_slave.myslave { pcm "hw:0,0" } pcm.mysnoop { type dsnoop ipc_key 1024 slave myslave } Then you simply do: snd_pcm_open(&pcm_handle, "mysnoop", SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK); snd_pcm_close(pcm_handle); snd_config_update_free_global(); Valgrind will show something like this: ==7889== ==7889== HEAP SUMMARY: ==7889== in use at exit: 15,854 bytes in 871 blocks ==7889== total heap usage: 333,212 allocs, 332,341 frees, 11,695,490 bytes allocated ==7889== ==7889== 15,834 (6,264 direct, 9,570 indirect) bytes in 174 blocks are definitely lost in loss record 6 of 6 ==7889== at 0x4027CBF: calloc (vg_replace_malloc.c:566) ==7889== by 0x406B416: _snd_config_make (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406C69C: snd_config_make (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406C861: snd_config_make_compound (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406D00C: _snd_config_copy (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406F112: snd_config_walk (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406FE04: snd_config_copy (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x4070370: snd_config_expand (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x407107E: snd_config_search_definition (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x40D2C22: _snd_pcm_direct_get_slave_ipc_offset (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x40D68A2: snd1_pcm_direct_parse_open_conf (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x40D28CA: _snd_pcm_dsnoop_open (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x40909C4: snd_pcm_open_conf (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x4091036: snd_pcm_open_noupdate (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x804873D: main (in /home/blah/al/a.out) ==7889== ==7889== LEAK SUMMARY: ==7889== definitely lost: 6,264 bytes in 174 blocks ==7889== indirectly lost: 9,570 bytes in 696 blocks ==7889== possibly lost: 0 bytes in 0 blocks ==7889== still reachable: 20 bytes in 1 blocks ==7889== suppressed: 0 bytes in 0 blocks ==7889== Reachable blocks (those to which a pointer was found) are not shown. ==7889== To see them, rerun with: --leak-check=full --show-reachable=yes ==7889== ==7889== For counts of detected and suppressed errors, rerun with: -v ==7889== ERROR SUMMARY: 1 errors from 1 contexts (suppressed: 0 from 0)

2 1

[alsa-devel] dsnoop memory leak
by Ivan Bolsunov 04 Jun '13

04 Jun '13

Hi guys, on may 31 I sent message, can you at least confirm what bug exist? Thanks.

1 0

[alsa-devel] [PATCH V5 0/5] clk: Samsung: audss: Register audio subsytem clocks using common clk framework
by Padmavathi Venna 04 Jun '13

04 Jun '13

Samsung S5PV210 and Exynos SoC has a separate subsystem for audio. This subsystem has a internal clock controller which controls i2s0 and pcm0 clocks. This patch series adds the Samsung audio subsytem clock to the common clock framework and provides the I2S controllers clock information in the dtsi file. This patch series is made based on Kukjin Kim for-next branch Changes since V4: - Reworked on the nits given by Doug. - Removed mout_audss and mout_i2s from i2s nodes as we are not getting these clocks in the i2s driver. - Modified the I2S binding documentation for clocks and pinmux. Changes since V3: - Replaced samsung with exynos in the macro prefixes and function names as this driver supports mainly exynos and s5p family. - Added Reviewed-by:Sylwester Nawrocki <s.nawrocki(a)samsung.com> Changes since V2: - Removed s5pv210 compatible name from driver as it is not yet supported which is different from Exynos series audio subsystem clock conroller. - Removed clkdev lookup support and added alias names in the i2s0 controller node. Changes since V1: - Reworked on all review comments by Sylwester Nawrocki - Added a header file for all clock indexes as requested by Sylwester - Added different compatible names for s5pv210, exynos4 and exynos5 - Registered the pcm clocks with common clock framework Padmavathi Venna (5): ARM: samsung: use #include for all device trees clk: samsung: register audio subsystem clocks using common clock framework ARM: dts: add Exynos audio subsystem clock controller node ARM: dts: add clock provider information for i2s controllers in Exynos5250 ARM: dts: Update Samsung I2S documentation .../devicetree/bindings/clock/clk-exynos-audss.txt | 64 ++++++++++ .../devicetree/bindings/sound/samsung-i2s.txt | 40 ++---- arch/arm/boot/dts/exynos4.dtsi | 2 +- arch/arm/boot/dts/exynos4210-origen.dts | 2 +- arch/arm/boot/dts/exynos4210-smdkv310.dts | 2 +- arch/arm/boot/dts/exynos4210-trats.dts | 2 +- arch/arm/boot/dts/exynos4210-universal_c210.dts | 2 +- arch/arm/boot/dts/exynos4210.dtsi | 4 +- arch/arm/boot/dts/exynos4212.dtsi | 2 +- arch/arm/boot/dts/exynos4412-odroidx.dts | 2 +- arch/arm/boot/dts/exynos4412-origen.dts | 2 +- arch/arm/boot/dts/exynos4412-smdk4412.dts | 2 +- arch/arm/boot/dts/exynos4412.dtsi | 2 +- arch/arm/boot/dts/exynos4x12.dtsi | 4 +- arch/arm/boot/dts/exynos5250-arndale.dts | 2 +- arch/arm/boot/dts/exynos5250-smdk5250.dts | 2 +- arch/arm/boot/dts/exynos5250-snow.dts | 4 +- arch/arm/boot/dts/exynos5250.dtsi | 20 +++- arch/arm/boot/dts/exynos5440-sd5v1.dts | 2 +- arch/arm/boot/dts/exynos5440-ssdk5440.dts | 2 +- arch/arm/boot/dts/exynos5440.dtsi | 2 +- arch/arm/boot/dts/s3c2416-smdk2416.dts | 2 +- arch/arm/boot/dts/s3c2416.dtsi | 4 +- arch/arm/boot/dts/s3c24xx.dtsi | 2 +- drivers/clk/samsung/Makefile | 1 + drivers/clk/samsung/clk-exynos-audss.c | 133 ++++++++++++++++++++ include/dt-bindings/clk/exynos-audss-clk.h | 25 ++++ 27 files changed, 279 insertions(+), 54 deletions(-) create mode 100644 Documentation/devicetree/bindings/clock/clk-exynos-audss.txt create mode 100644 drivers/clk/samsung/clk-exynos-audss.c create mode 100644 include/dt-bindings/clk/exynos-audss-clk.h -- 1.7.4.4

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Re: [alsa-devel] [PATCH] ASoC: add RT5640 CODEC driver
by Mark Brown 04 Jun '13

04 Jun '13

On Fri, May 31, 2013 at 03:04:59PM +0800, bardliao(a)realtek.com wrote: > From: Bard Liao <bardliao(a)realtek.com> This looks pretty good, a few small things below and probably one actual bug. > > * Use regmap_range_cfg to replace index read/write function. > * Remove I2S3 related code since there is no I2S3 in ALC5640. > * Remove Voice DSP related code since there is no Voice DSP in ALC5640. > * Remove MICBIAS2 since there is no MICBIAS2 in ALC5640. > * Change DMIC1/2 CLK to DMIC1/2 Power since it is for enable/disable DMIC1/2 > * Modify some texts for consistent coding style. > * Merge sto adc l/r mux since it shares the same control bits. > * Other minor changes. Put stuff like this after the --- so it doesn't end up in the commit log. > + switch (event) { > + case SND_SOC_DAPM_POST_PMU: > + snd_soc_update_bits(codec, RT5640_PWR_DIG1, 0x0001, 0x0001); > + regmap_update_bits(rt5640->regmap, 0x1c, 0xf000, 0xf000); > + break; It's a bit weird to be using snd_soc_update_bits() and regmap_update_bits()... > + SND_SOC_DAPM_SUPPLY("DMIC1 Power", SND_SOC_NOPM, 0, 0, > + rt5640_set_dmic1_event, > + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), > + SND_SOC_DAPM_SUPPLY("DMIC2 Power", SND_SOC_NOPM, 0, 0, > + rt5640_set_dmic2_event, > + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), The enable bits could be listed here - this would be a bit more idiomatic, would save the PMD event and would enable the writes to be combined. > + case SND_SOC_BIAS_STANDBY: > + if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { > + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, > + RT5640_PWR_VREF1 | RT5640_PWR_MB | > + RT5640_PWR_BG | RT5640_PWR_VREF2, > + RT5640_PWR_VREF1 | RT5640_PWR_MB | > + RT5640_PWR_BG | RT5640_PWR_VREF2); > + mdelay(10); > + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, > + RT5640_PWR_FV1 | RT5640_PWR_FV2, > + RT5640_PWR_FV1 | RT5640_PWR_FV2); > + regcache_cache_only(rt5640->regmap, false); > + regcache_sync(rt5640->regmap); This looks wrong - you're writing to the device before you take it out of cache only mode so your delay in the above sequence will have no effect. > + val = snd_soc_read(codec, RT5640_VENDOR_ID2); > + if ((val != RT5640_DEVICE_ID)) { > + dev_err(codec->dev, > + "Device with ID register %x is not rt5640/39\n", val); > + return -ENODEV; > + } > + > + rt5640_reset(codec); Do these in the I2C probe function before you register the CODEC, that way we don't try to start the sound card if we can't talk to the device. > + ret = regmap_register_patch(rt5640->regmap, init_list, > + ARRAY_SIZE(init_list)); > + if (ret != 0) > + dev_warn(codec->dev, "Failed to apply regmap patch: %d\n", > + ret); This should also be in the I2C probe.

2 2

[alsa-devel] regression: from 3.8 to 3.9: headphones output no sound on Intel HDA, codec VIA VT1802
by Alex Riesen 04 Jun '13

04 Jun '13

Hi, I just noticed (use the headphones rarely) that the headphones on this System76 Lemur Ultra (lemu4) stopped working. There is absolutely no output. I tried reverting all changes to patch_via.c since v3.8, and have the sound back, but there wasn't much to revert: Revert "ALSA: hda - Rearrange INPUT_PIN_ATTR_*" Revert "ALSA: hda - Use generic parser for VIA codec driver" Revert "ALSA: hda - Enable parsing the independent HP mode as default for VIA codecs" Revert "ALSA: hda/via - Fix wrong checks of power state bits" Revert "ALSA: hda - Fix phantom jacks on VT1708" Revert "ALSA: hda - detect jacks on VT1708 even when no streams are active" Revert "ALSA: hda - Use generic array for loopback list management" and one change (which I indeed suspect) is large: the use of generic parser. The problem is also present in 3.10-rc1. The output of alsa-info.sh attached, for 3.8.12 and 3.9.2 respectively. Regards, Alex Riesen

2 30

Re: [alsa-devel] [PATCH] ASoC: add RT5640 CODEC driver
by Stephen Warren 04 Jun '13

04 Jun '13

On 05/31/2013 01:15 AM, Bard Liao wrote: > Stephen, > > I remove the DT support from this patch. > Can you add the DT support as a separate patch? Yes, I'll do that. I assume you also want me to add the code to control the LDO1_EN GPIO, since that seems missing from this patch. I thought you said you were going to add it. I assume you tested this on Dalmore by manually setting the GPIO from U-Boot before booting the kernel? I'll try to get to this ASAP; perhaps not today though - we'll see how long the other things I have stacked up take.

2 1

Re: [alsa-devel] [PATCH V4 4/4] ARM: dts: add clock provider information for i2s controllers in Exynos5250
by Padma Venkat 04 Jun '13

04 Jun '13

Hi Doug, On Tue, Jun 4, 2013 at 1:48 AM, Doug Anderson <dianders(a)chromium.org> wrote: > Padmavathi, > > On Sun, Jun 2, 2013 at 10:19 PM, Padmavathi Venna <padma.v(a)samsung.com> wrote: >> + clocks = <&clock_audss EXYNOS_I2S_BUS>, >> + <&clock_audss EXYNOS_I2S_BUS>, >> + <&clock_audss EXYNOS_SCLK_I2S>, >> + <&clock_audss EXYNOS_MOUT_AUDSS>, >> + <&clock_audss EXYNOS_MOUT_I2S>; >> + clock-names = "iis", "i2s_opclk0", "i2s_opclk1", >> + "mout_audss", "mout_i2s"; > > Are there bindings for these clocks? Would be nice to see a > description for what they are supposed to be. Yes. They have bindings. I will add the description. > > I looked up i2s_opclk0 / i2s_opclk1 and they look reasonable at a > quick glance. ...and it does seem right that iis and i2s_opclk0 are > the same. ...but I don't see any place that uses the last two. Is > there an in-flight patch? I don't have any patch for these right now. These are used internally. But as these are i2s0 base clocks I added here. > > -Doug Thanks Padma

1 0

Re: [alsa-devel] [PATCH V4 3/4] ARM: dts: add Exynos audio subsystem clock controller node
by Padma Venkat 04 Jun '13

04 Jun '13

Hi Doug, On Tue, Jun 4, 2013 at 1:43 AM, Doug Anderson <dianders(a)chromium.org> wrote: > Padmavathi, > > On Sun, Jun 2, 2013 at 10:19 PM, Padmavathi Venna <padma.v(a)samsung.com> wrote: >> Audio subsystem introduced in s5pv210 and exynos platforms >> which has a internal clock controller. This patch adds a node >> for the same on exynos5250. >> >> Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> >> Reviewed-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> >> --- >> arch/arm/boot/dts/exynos5250.dtsi | 6 ++++++ >> 1 files changed, 6 insertions(+), 0 deletions(-) >> >> diff --git a/arch/arm/boot/dts/exynos5250.dtsi b/arch/arm/boot/dts/exynos5250.dtsi >> index bccda67..388983e 100644 >> --- a/arch/arm/boot/dts/exynos5250.dtsi >> +++ b/arch/arm/boot/dts/exynos5250.dtsi >> @@ -72,6 +72,12 @@ >> #clock-cells = <1>; >> }; >> >> + clock_audss: audss-clock-controller@3810000 { I removed this leading 0 as Tomasz Figa suggested. > > Nit: other places in the same file have the leading 0, like > i2s0: i2s@03830000 { This was the patch which got merged earlier. So I didn't modify this. Is it okey if I remove leading 0 for both of the nodes now? > > So you could follow suit and do: > > clock_audss: audss-clock-controller@03810000 { > > -Doug Thanks Padma

1 0

[alsa-devel] cplay error "Can't find sync world"
by Keyon.Jey 04 Jun '13

04 Jun '13

hi, all, when I try use clpay -c 1 -d 2 test.mp3 to use tinycompress hwdec on my phone, "Can't find sync world" always occurred, no matter what mp3 music I try to play. vinod/guys, can you help? regards, Keyon

3 3

[alsa-devel] [PATCH] ASoC: dapm: check suspend state in dapm_dai_check_power()
by Stephen Warren 03 Jun '13

03 Jun '13

From: Stephen Warren <swarren(a)nvidia.com> Currently, when the system enters suspend with audio playing, since the DAI widgets are active, the associated CODEC is considered active, and hence is not transition into a low-power bias state as the system suspends. This likely causes audio failure after resume, since the CODEC power has quite possibly been removed during suspend, and the bias level transitions in the CODEC driver are needed to recover from this. At the very least, this issue likely causes warnings in the kernel log due to the CODEC being left on over the suspend transition. To solve this, modify dapm_dai_check_power() not to consider DAI widgets active if the system is entering suspend. Suggested-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3507346..bad545a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1133,7 +1133,7 @@ static int dapm_dai_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); if (w->active) - return w->active; + return snd_soc_dapm_suspend_check(w); return dapm_generic_check_power(w); } -- 1.7.10.4

3 4

[alsa-devel] [PATCH 1/2] ASoC: tegra: implement suspend/resume for Tegra30 AHUB
by Stephen Warren 03 Jun '13

03 Jun '13

From: Stephen Warren <swarren(a)nvidia.com> Add tegra30_ahub_{suspend,resume}. These use regcache functions to restore all HW registers after power loss during a suspend/resume cycle. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra30_ahub.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 23e592f..0f4787c 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -627,9 +627,30 @@ static int tegra30_ahub_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int tegra30_ahub_suspend(struct device *dev) +{ + regcache_mark_dirty(ahub->regmap_ahub); + regcache_mark_dirty(ahub->regmap_apbif); + + return 0; +} + +static int tegra30_ahub_resume(struct device *dev) +{ + int ret; + + ret = regcache_sync(ahub->regmap_ahub); + ret |= regcache_sync(ahub->regmap_apbif); + + return ret; +} +#endif + static const struct dev_pm_ops tegra30_ahub_pm_ops = { SET_RUNTIME_PM_OPS(tegra30_ahub_runtime_suspend, tegra30_ahub_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(tegra30_ahub_suspend, tegra30_ahub_resume) }; static struct platform_driver tegra30_ahub_driver = { -- 1.7.10.4

2 2

[alsa-devel] [PATCH 1/4] ASoC: wm0010: Clear IRQ as wake source and include missing header
by Charles Keepax 03 Jun '13

03 Jun '13

Both clear the IRQ as being a wake source when we are finished with it and include a missing header file that is required. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index b6df319..f5e8356 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -14,6 +14,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> +#include <linux/interrupt.h> #include <linux/irqreturn.h> #include <linux/init.h> #include <linux/spi/spi.h> @@ -1002,6 +1003,8 @@ static int wm0010_spi_remove(struct spi_device *spi) gpio_set_value_cansleep(wm0010->gpio_reset, wm0010->gpio_reset_value); + irq_set_irq_wake(wm0010->irq, 0); + if (wm0010->irq) free_irq(wm0010->irq, wm0010); -- 1.7.2.5

2 2

[alsa-devel] [PATCH 1/1] ASoC: davinci-mcasp: Remove redundant use of of_match_ptr macro
by Sachin Kamat 03 Jun '13

03 Jun '13

'mcasp_dt_ids' is always compiled in. Hence of_match_ptr is not necessary. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/davinci/davinci-mcasp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 56ecfc7..94f1e46 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1023,7 +1023,7 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( struct device_node *np = pdev->dev.of_node; struct snd_platform_data *pdata = NULL; const struct of_device_id *match = - of_match_device(of_match_ptr(mcasp_dt_ids), &pdev->dev); + of_match_device(mcasp_dt_ids, &pdev->dev); const u32 *of_serial_dir32; u8 *of_serial_dir; @@ -1256,7 +1256,7 @@ static struct platform_driver davinci_mcasp_driver = { .driver = { .name = "davinci-mcasp", .owner = THIS_MODULE, - .of_match_table = of_match_ptr(mcasp_dt_ids), + .of_match_table = mcasp_dt_ids, }, }; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: mxs: mxs-saif: Fix the playback/record capabilities
by Fabio Estevam 03 Jun '13

03 Jun '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Running "aplay -l" and "arecord -l" results in the following outputs: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 On mx28 there are two serial audio interface ports (SAIF0 and SAIF1) and each one of them are unidirectional. Create a mxs_saif_dai array containing a playback element and a record element only in order to properly describe the dai links. After this change we can correctly report the capabilities as follows: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: HiFi Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 1: HiFi Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- This patch also needs: "[PATCH] ASoC: pcm: Require both CODEC and CPU support when declaring stream caps" from Mark Brown. sound/soc/mxs/mxs-saif.c | 38 ++++++++++++++++++++++---------------- 1 file changed, 22 insertions(+), 16 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 49d8700..7f3e67a 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -607,22 +607,28 @@ static int mxs_saif_dai_probe(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_driver mxs_saif_dai = { - .name = "mxs-saif", - .probe = mxs_saif_dai_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = MXS_SAIF_RATES, - .formats = MXS_SAIF_FORMATS, +static struct snd_soc_dai_driver mxs_saif_dai[2] = { + { + .name = "saif0", + .probe = mxs_saif_dai_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = MXS_SAIF_RATES, + .formats = MXS_SAIF_FORMATS, + }, + .ops = &mxs_saif_dai_ops, + }, { + .name = "saif1", + .probe = mxs_saif_dai_probe, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = MXS_SAIF_RATES, + .formats = MXS_SAIF_FORMATS, + }, + .ops = &mxs_saif_dai_ops, }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = MXS_SAIF_RATES, - .formats = MXS_SAIF_FORMATS, - }, - .ops = &mxs_saif_dai_ops, }; static const struct snd_soc_component_driver mxs_saif_component = { @@ -735,7 +741,7 @@ static int mxs_saif_probe(struct platform_device *pdev) platform_set_drvdata(pdev, saif); ret = snd_soc_register_component(&pdev->dev, &mxs_saif_component, - &mxs_saif_dai, 1); + &mxs_saif_dai[saif->id], 1); if (ret) { dev_err(&pdev->dev, "register DAI failed\n"); return ret; -- 1.8.1.2

3 5

[alsa-devel] [PATCH] ASoC: pcm: Require both CODEC and CPU support when declaring stream caps
by Mark Brown 03 Jun '13

03 Jun '13

When declaring playback and capture capabilities check for both CODEC side and CPU side support rather than only checking for CODEC side support. While it is unusual some CPUs do have unidirectional DAIs. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 73bb8ee..a9fddf0 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2011,9 +2011,11 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (cpu_dai->driver->capture.channels_min) capture = 1; } else { - if (codec_dai->driver->playback.channels_min) + if (codec_dai->driver->playback.channels_min && + cpu_dai->driver->playback.channels_min) playback = 1; - if (codec_dai->driver->capture.channels_min) + if (codec_dai->driver->capture.channels_min && + cpu_dai->driver->capture.channels_min) capture = 1; } -- 1.7.10.4

4 4

[alsa-devel] No headphone control for ALC892 on Zotac Motherboard D2550ITXS-B-E
by Halim Sahin 03 Jun '13

03 Jun '13

Hi, Attached is output of alsa-info. Updated to Alsa-driver 1.0.25 running with 2.6.34 kernel. Frontpanel of my case has mic and headphone output connectors. front mic works but no headphones. Card works except headphones Can you please help? -- Halim Sahin E-Mail: halim.sahin (at) t-online.de

3 5

[alsa-devel] [PATCH] ALSA: usb-audio: fix Roland/Cakewalk UM-3G support
by Clemens Ladisch 03 Jun '13

03 Jun '13

Commit 927c9423dd5f2d1c0b93d5e694ab84b4a5559713 (ALSA: usb-audio: add Edirol UM-3G support) used a wrong quirk type, which would make the driver refuse to attach with the error message "MIDIStreaming interface descriptor not found". Cc: <stable(a)vger.kernel.org> # 3.3 and later Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/quirks-table.h | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1792,7 +1792,11 @@ USB_DEVICE_VENDOR_SPEC(0x0582, 0x0108), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .ifnum = 0, - .type = QUIRK_MIDI_STANDARD_INTERFACE + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0007, + .in_cables = 0x0007 + } } }, {

2 1

[alsa-devel] [PATCH] ASoC: dwc: debug message correction.
by Rajeev Kumar 03 Jun '13

03 Jun '13

From: Rajeev Kumar <rajeevk-dlh.kumar(a)st.com> Debug message correction. Signed-off-by: Rajeev Kumar <rajeevk-dlh.kumar(a)st.com> --- sound/soc/dwc/designware_i2s.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index deb30d5..deb975e 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -392,7 +392,7 @@ static int dw_i2s_probe(struct platform_device *pdev) } if (cap & DWC_I2S_PLAY) { - dev_dbg(&pdev->dev, " SPEAr: play supported\n"); + dev_dbg(&pdev->dev, " designware: play supported\n"); dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; dw_i2s_dai->playback.channels_max = pdata->channel; dw_i2s_dai->playback.formats = pdata->snd_fmts; @@ -400,7 +400,7 @@ static int dw_i2s_probe(struct platform_device *pdev) } if (cap & DWC_I2S_RECORD) { - dev_dbg(&pdev->dev, "SPEAr: record supported\n"); + dev_dbg(&pdev->dev, "designware: record supported\n"); dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; dw_i2s_dai->capture.channels_max = pdata->channel; dw_i2s_dai->capture.formats = pdata->snd_fmts; -- 1.7.6.5

1 0

[alsa-devel] [PATCH V2] ARM: dts: wm8994: Add wm8994 regulator support on smdk5250.
by Padmavathi Venna 03 Jun '13

03 Jun '13

This patch adds the required regulator supplies and properties for wm8994 codec on smdk5250 board. Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> --- Changes since V1: - Clubbed the same supply regulators as suggested by Mark. arch/arm/boot/dts/exynos5250-smdk5250.dts | 37 +++++++++++++++++++++++++++- 1 files changed, 35 insertions(+), 2 deletions(-) diff --git a/arch/arm/boot/dts/exynos5250-smdk5250.dts b/arch/arm/boot/dts/exynos5250-smdk5250.dts index f2a025e..dc2ec25 100644 --- a/arch/arm/boot/dts/exynos5250-smdk5250.dts +++ b/arch/arm/boot/dts/exynos5250-smdk5250.dts @@ -37,6 +37,30 @@ }; }; + vdd:fixed-regulator@0 { + compatible = "regulator-fixed"; + regulator-name = "vdd-supply"; + regulator-min-microvolt = <1800000>; + regulator-max-microvolt = <1800000>; + regulator-always-on; + }; + + dbvdd:fixed-regulator@1 { + compatible = "regulator-fixed"; + regulator-name = "dbvdd-supply"; + regulator-min-microvolt = <3300000>; + regulator-max-microvolt = <3300000>; + regulator-always-on; + }; + + spkvdd:fixed-regulator@2 { + compatible = "regulator-fixed"; + regulator-name = "spkvdd-supply"; + regulator-min-microvolt = <5000000>; + regulator-max-microvolt = <5000000>; + regulator-always-on; + }; + i2c@12C70000 { samsung,i2c-sda-delay = <100>; samsung,i2c-max-bus-freq = <20000>; @@ -47,8 +71,17 @@ }; wm8994: wm8994@1a { - compatible = "wlf,wm8994"; - reg = <0x1a>; + compatible = "wlf,wm8994"; + reg = <0x1a>; + + gpio-controller; + #gpio-cells = <2>; + + AVDD2-supply = <&vdd>; + CPVDD-supply = <&vdd>; + DBVDD-supply = <&dbvdd>; + SPKVDD1-supply = <&spkvdd>; + SPKVDD2-supply = <&spkvdd>; }; }; -- 1.7.4.4

1 0

[alsa-devel] [PATCH V4 0/4] clk: Samsung: audss: Register audio subsytem clocks using common clk framework
by Padmavathi Venna 03 Jun '13

03 Jun '13

Samsung S5PV210 and Exynos SoC has a separate subsystem for audio. This subsystem has a internal clock controller which controls i2s0 and pcm0 clocks. This patch series adds the Samsung audio subsytem clock to the common clock framework and provides the I2S controllers clock information in the dtsi file. This patch series is made based on Kukjin Kim for-next branch Changes since V3: - Replaced samsung with exynos in the macro prefixes and function names as this driver supports mainly exynos and s5p family. - Added Reviewed-by:Sylwester Nawrocki <s.nawrocki(a)samsung.com> Changes since V2: - Removed s5pv210 compatible name from driver as it is not yet supported which is different from Exynos series audio subsystem clock conroller. - Removed clkdev lookup support and added alias names in the i2s0 controller node. Changes since V1: - Reworked on all review comments by Sylwester Nawrocki - Added a header file for all clock indexes as requested by Sylwester - Added different compatible names for s5pv210, exynos4 and exynos5 - Registered the pcm clocks with common clock framework Padmavathi Venna (4): ARM: samsung: use #include for all device trees clk: samsung: register audio subsystem clocks using common clock framework ARM: dts: add Exynos audio subsystem clock controller node ARM: dts: add clock provider information for i2s controllers in Exynos5250 .../devicetree/bindings/clock/clk-exynos-audss.txt | 64 ++++++++++ arch/arm/boot/dts/exynos4.dtsi | 2 +- arch/arm/boot/dts/exynos4210-origen.dts | 2 +- arch/arm/boot/dts/exynos4210-smdkv310.dts | 2 +- arch/arm/boot/dts/exynos4210-trats.dts | 2 +- arch/arm/boot/dts/exynos4210-universal_c210.dts | 2 +- arch/arm/boot/dts/exynos4210.dtsi | 4 +- arch/arm/boot/dts/exynos4212.dtsi | 2 +- arch/arm/boot/dts/exynos4412-odroidx.dts | 2 +- arch/arm/boot/dts/exynos4412-origen.dts | 2 +- arch/arm/boot/dts/exynos4412-smdk4412.dts | 2 +- arch/arm/boot/dts/exynos4412.dtsi | 2 +- arch/arm/boot/dts/exynos4x12.dtsi | 4 +- arch/arm/boot/dts/exynos5250-arndale.dts | 2 +- arch/arm/boot/dts/exynos5250-smdk5250.dts | 2 +- arch/arm/boot/dts/exynos5250-snow.dts | 4 +- arch/arm/boot/dts/exynos5250.dtsi | 23 +++- arch/arm/boot/dts/exynos5440-sd5v1.dts | 2 +- arch/arm/boot/dts/exynos5440-ssdk5440.dts | 2 +- arch/arm/boot/dts/exynos5440.dtsi | 2 +- arch/arm/boot/dts/s3c2416-smdk2416.dts | 2 +- arch/arm/boot/dts/s3c2416.dtsi | 4 +- arch/arm/boot/dts/s3c24xx.dtsi | 2 +- drivers/clk/samsung/Makefile | 1 + drivers/clk/samsung/clk-exynos-audss.c | 133 ++++++++++++++++++++ include/dt-bindings/clk/exynos-audss-clk.h | 25 ++++ 26 files changed, 269 insertions(+), 27 deletions(-) create mode 100644 Documentation/devicetree/bindings/clock/clk-exynos-audss.txt create mode 100644 drivers/clk/samsung/clk-exynos-audss.c create mode 100644 include/dt-bindings/clk/exynos-audss-clk.h -- 1.7.4.4

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[alsa-devel] [PATCH] ASoC: dwc: Folder path correction in file header.
by Rajeev Kumar 03 Jun '13

03 Jun '13

From: Rajeev Kumar <rajeevk-dlh.kumar(a)st.com> Folder path correction in file header. Signed-off-by: Rajeev Kumar <rajeevk-dlh.kumar(a)st.com> --- sound/soc/dwc/designware_i2s.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index deb30d5..a11050c 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -1,7 +1,7 @@ /* * ALSA SoC Synopsys I2S Audio Layer * - * sound/soc/spear/designware_i2s.c + * sound/soc/dwc/designware_i2s.c * * Copyright (C) 2010 ST Microelectronics * Rajeev Kumar <rajeev-dlh.kumar(a)st.com> -- 1.7.6.5

1 0

[alsa-devel] how to use the invert of the field of struct snd_kcontrl_new ?
by Zhang wei 03 Jun '13

03 Jun '13

Is there some details documents about it? When need I set it as 1 and 0? Thanks BR zw

1 0

[alsa-devel] aplay lists both Playback and Capture devices
by Fabio Estevam 02 Jun '13

02 Jun '13

Hi, When running 'aplay -l' and 'arecord -l' I see the following output on a mx28evk running a 3.10-rc kernel: /$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 I would expect to see only the Playback entry in aplay and only the Record entry in arecord. Is there anything I should change in sound/soc/mxs/mxs-sgtl5000.c in order to get the correct outputs? Regards, Fabio Estevam

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[alsa-devel] [PATCH 1/2] dma: Add a PAUSE_RESUME capability
by Lars-Peter Clausen 01 Jun '13

01 Jun '13

Add a capability which allows dmaengine consumers to query whether a dmaengine controller supports the DMA_PAUSE and DMA_RESUME commands. Sometimes a user of the dmaengine API needs to know in advance if a channel will support pause/resume. E.g. in ALSA the information whether it is possible to pause/resume a audio stream or whether it is necessary to abort and restart the stream is communicated to userspace so it can act accordingly. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/linux/dmaengine.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h index 96d3e4a..ad98dfb 100644 --- a/include/linux/dmaengine.h +++ b/include/linux/dmaengine.h @@ -74,6 +74,7 @@ enum dma_transaction_type { DMA_SLAVE, DMA_CYCLIC, DMA_INTERLEAVE, + DMA_PAUSE_RESUME, /* last transaction type for creation of the capabilities mask */ DMA_TX_TYPE_END, }; -- 1.8.0

3 3

[alsa-devel] [PATCH 1/2] ASoC: wm8994: fix the codec state machine
by Vinod Koul 01 Jun '13

01 Jun '13

few places the mic_detecting is not updated correctly when state changes, causing jack detection to go wrong Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/wm8994.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 49b2b00..43f82d2 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3547,6 +3547,7 @@ static void wm8958_mic_id(void *data, u16 status) snd_soc_jack_report(wm8994->micdet[0].jack, 0, wm8994->btn_mask | SND_JACK_HEADSET); + wm8994->mic_detecting = true; return; } @@ -3672,7 +3673,7 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) snd_soc_update_bits(codec, WM1811_JACKDET_CTRL, WM1811_JACKDET_DB, WM1811_JACKDET_DB); - wm8994->mic_detecting = false; + wm8994->mic_detecting = true; wm8994->jack_mic = false; snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); @@ -3696,6 +3697,7 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) /* Since we only report deltas force an update, ensures we * avoid bootstrapping issues with the core. */ snd_soc_jack_report(wm8994->micdet[0].jack, 0, 0); + wm8994->mic_detecting = true; pm_runtime_put(codec->dev); return IRQ_HANDLED; -- 1.7.0.4

2 3

Re: [alsa-devel] [PATCH V3 0/4] clk: Samsung: audss: Register audio subsytem clocks using common clk framework
by Padma Venkat 01 Jun '13

01 Jun '13

Hi Sylwester, On Sat, Jun 1, 2013 at 1:59 PM, Sylwester Nawrocki <sylvester.nawrocki(a)gmail.com> wrote: > Padmavathi, > > > On 05/28/2013 12:02 PM, Padmavathi Venna wrote: >> >> Samsung S5PV210 and Exynos SoC has a separate subsystem for audio. This >> subsystem >> has a internal clock controller which controls i2s0 and pcm0 clocks. This >> patch >> series adds the Samsung audio subsytem clock to the common clock framework >> and >> provides the I2S controllers clock information in the dtsi file. >> >> This patch series is made based on Kukjin Kim for-next branch >> >> Changes since V2: >> - Removed s5pv210 compatible name from driver as it is >> not yet supported which is different from Exynos series >> audio subsystem clock conroller. >> - Removed clkdev lookup support and added alias names in >> the i2s0 controller node. >> Changes since V1: >> - Reworked on all review comments by Sylwester Nawrocki >> - Added a header file for all clock indexes as requested by >> Sylwester >> - Added different compatible names for s5pv210, exynos4 and >> exynos5 >> - Registered the pcm clocks with common clock framework > > > Overall it looks good to me. I'm only a bit uncomfortable with using > SAMSUNG_ > prefix for the clock index definitions. Ideally it should be something more > specific to the SoC family. But I can't think of anything better at the > moment. > This covers the Exynos and S5P SoCs, hence EXYNOS_ would not be appropriate. Actuvally I added samsung prefix when I added s5pv210 compatible name. But I also feel Exynos would be better now. I will resend the patch with Exynos prefix and with your reviewed-by. Thanks Padma > > The patch series: > Reviewed-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> > > Thanks, > Sylwester > > >> Padmavathi Venna (4): >> ARM: samsung: use #include for all device trees >> clk: samsung: register audio subsystem clocks using common clock >> framework >> ARM: dts: add Exynos audio subsystem clock controller node >> ARM: dts: add clock provider information for i2s controllers in >> Exynos5250 >> >> .../bindings/clock/clk-samsung-audss.txt | 64 ++++++++++ >> arch/arm/boot/dts/exynos4.dtsi | 2 +- >> arch/arm/boot/dts/exynos4210-origen.dts | 2 +- >> arch/arm/boot/dts/exynos4210-smdkv310.dts | 2 +- >> arch/arm/boot/dts/exynos4210-trats.dts | 2 +- >> arch/arm/boot/dts/exynos4210-universal_c210.dts | 2 +- >> arch/arm/boot/dts/exynos4210.dtsi | 4 +- >> arch/arm/boot/dts/exynos4212.dtsi | 2 +- >> arch/arm/boot/dts/exynos4412-odroidx.dts | 2 +- >> arch/arm/boot/dts/exynos4412-origen.dts | 2 +- >> arch/arm/boot/dts/exynos4412-smdk4412.dts | 2 +- >> arch/arm/boot/dts/exynos4412.dtsi | 2 +- >> arch/arm/boot/dts/exynos4x12.dtsi | 4 +- >> arch/arm/boot/dts/exynos5250-arndale.dts | 2 +- >> arch/arm/boot/dts/exynos5250-smdk5250.dts | 2 +- >> arch/arm/boot/dts/exynos5250-snow.dts | 4 +- >> arch/arm/boot/dts/exynos5250.dtsi | 23 +++- >> arch/arm/boot/dts/exynos5440-sd5v1.dts | 2 +- >> arch/arm/boot/dts/exynos5440-ssdk5440.dts | 2 +- >> arch/arm/boot/dts/exynos5440.dtsi | 2 +- >> arch/arm/boot/dts/s3c2416-smdk2416.dts | 2 +- >> arch/arm/boot/dts/s3c2416.dtsi | 4 +- >> arch/arm/boot/dts/s3c24xx.dtsi | 2 +- >> drivers/clk/samsung/Makefile | 1 + >> drivers/clk/samsung/clk-samsung-audss.c | 133 >> ++++++++++++++++++++ >> include/dt-bindings/clk/samsung-audss-clk.h | 25 ++++ >> 26 files changed, 269 insertions(+), 27 deletions(-) >> create mode 100644 >> Documentation/devicetree/bindings/clock/clk-samsung-audss.txt >> create mode 100644 drivers/clk/samsung/clk-samsung-audss.c >> create mode 100644 include/dt-bindings/clk/samsung-audss-clk.h

1 0

[alsa-devel] [PATCH] ASoC: sgtl5000: Do not read registers prior to turning on the supplies
by Fabio Estevam 31 May '13

31 May '13

Commit b871f1ad (ASoC: sgtl5000: Read SGTL5000_CHIP_ID in i2c_probe()) placed the code for reading the codec revision prior to turning on the power supplies. Even though this works on some systems that always have the codec power supplies enabled, this is not correct, so revert this commit. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 39 +++++++++++++++------------------------ 1 file changed, 15 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1c3b20f..788af6c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1275,7 +1275,7 @@ static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) { - int reg; + u16 reg; int ret; int rev; int i; @@ -1303,17 +1303,23 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) /* wait for all power rails bring up */ udelay(10); - /* - * workaround for revision 0x11 and later, - * roll back to use internal LDO - */ - - ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); - if (ret) + /* read chip information */ + reg = snd_soc_read(codec, SGTL5000_CHIP_ID); + if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != + SGTL5000_PARTID_PART_ID) { + dev_err(codec->dev, + "Device with ID register %x is not a sgtl5000\n", reg); + ret = -ENODEV; goto err_regulator_disable; + } rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; + dev_info(codec->dev, "sgtl5000 revision 0x%x\n", rev); + /* + * workaround for revision 0x11 and later, + * roll back to use internal LDO + */ if (external_vddd && rev >= 0x11) { /* disable all regulator first */ regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), @@ -1497,7 +1503,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct sgtl5000_priv *sgtl5000; - int ret, reg, rev; + int ret; sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), GFP_KERNEL); @@ -1511,21 +1517,6 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, return ret; } - /* read chip information */ - ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); - if (ret) - return ret; - - if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != - SGTL5000_PARTID_PART_ID) { - dev_err(&client->dev, - "Device with ID register %x is not a sgtl5000\n", reg); - return -ENODEV; - } - - rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; - dev_info(&client->dev, "sgtl5000 revision 0x%x\n", rev); - i2c_set_clientdata(client, sgtl5000); /* Ensure sgtl5000 will start with sane register values */ -- 1.8.1.2

5 8

[alsa-devel] Improving or replacing snd_printk()
by Alan Stern 31 May '13

31 May '13

Takashi: Messages produced by snd_printk() aren't as informative as they could be -- especially when CONFIG_SND_VERBOSE_PRINTK isn't enabled, which is the case on most systems. The routine uses printk() rather then dev_dbg(), dev_info(), or dev_err(). As a result, the messages don't say what driver or device is involved. We just get something like: [15919.883604] cannot submit urb (err = -18) That's not much help to anybody. If you didn't know what an urb was, you wouldn't even realize that this referred to a USB device. And if you had multiple USB audio devices, you wouldn't know which one it referred to. Has there been any thought of improving snd_printk(), or even better, changing the call sites to use the dev_*() routines instead? Alan Stern

4 17

[alsa-devel] Fwd: regression: from 3.8 to 3.9: headphones output no sound on Intel HDA, codec VIA VT1802
by Evan 31 May '13

31 May '13

I am having this same issue (with the same hardware). I found a temporary work around for the problem by disabling "Auto-Mute Mode" with alsamixer and then muting my "Speaker" output. If I don't mute the "Speaker" I get audio through both my headphones and my speakers, but when I mute the speakers I get the audio just through the headphones. The output of alsa-info for 3.9.4 is here[1]. [1] http://www.alsa-project.org/db/?f=6120303f74b8e46930e15a544ba11b859de25c04 -E

2 1

[alsa-devel] [PATCH 1/3] ALSA: hdspm - Add support for 128-192kHz WordClock input
by Adrian Knoth 31 May '13

31 May '13

Allow WordClock input rates of 128, 176.4 and 192kHz. Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index ef3cbc0..e070ea8 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -400,8 +400,8 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_wc_freq0 (1<<5) /* input freq detected via autosync */ #define HDSPM_wc_freq1 (1<<6) /* 001=32, 010==44.1, 011=48, */ -#define HDSPM_wc_freq2 (1<<7) /* 100=64, 101=88.2, 110=96, */ -/* missing Bit for 111=128, 1000=176.4, 1001=192 */ +#define HDSPM_wc_freq2 (1<<7) /* 100=64, 101=88.2, 110=96, 111=128 */ +#define HDSPM_wc_freq3 0x800 /* 1000=176.4, 1001=192 */ #define HDSPM_SyncRef0 0x10000 /* Sync Reference */ #define HDSPM_SyncRef1 0x20000 @@ -412,13 +412,17 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_wc_valid (HDSPM_wcLock|HDSPM_wcSync) -#define HDSPM_wcFreqMask (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2) +#define HDSPM_wcFreqMask (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2|\ + HDSPM_wc_freq3) #define HDSPM_wcFreq32 (HDSPM_wc_freq0) #define HDSPM_wcFreq44_1 (HDSPM_wc_freq1) #define HDSPM_wcFreq48 (HDSPM_wc_freq0|HDSPM_wc_freq1) #define HDSPM_wcFreq64 (HDSPM_wc_freq2) #define HDSPM_wcFreq88_2 (HDSPM_wc_freq0|HDSPM_wc_freq2) #define HDSPM_wcFreq96 (HDSPM_wc_freq1|HDSPM_wc_freq2) +#define HDSPM_wcFreq128 (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2) +#define HDSPM_wcFreq176_4 (HDSPM_wc_freq3) +#define HDSPM_wcFreq192 (HDSPM_wc_freq0|HDSPM_wc_freq3) #define HDSPM_status1_F_0 0x0400000 #define HDSPM_status1_F_1 0x0800000 @@ -1181,6 +1185,15 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm) case HDSPM_wcFreq96: rate = 96000; break; + case HDSPM_wcFreq128: + rate = 128000; + break; + case HDSPM_wcFreq176_4: + rate = 176400; + break; + case HDSPM_wcFreq192: + rate = 192000; + break; default: rate = 0; break; -- 1.7.10.4

2 3

[alsa-devel] [PATCH] ALSA: hda - Add headset quirk for two Dell machines
by David Henningsson 31 May '13

31 May '13

This quirk is required for the headset mic to work on these two machines. BugLink: https://bugs.launchpad.net/bugs/1186170 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 59d2e91..aa4bf82 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3493,6 +3493,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), -- 1.7.9.5

2 1

[alsa-devel] [PATCH] [resend] snd-es1968: Add radio support for MediaForte M56VAP
by Ondrej Zary 31 May '13

31 May '13

Add support for TEA5757 tuner on MediaForte M56VAP sound+modem+radio card. The GPIO connection type is automatically detected (like snd-fm801 driver). Also add a safety subsystem vendor check to skip radio detection if vendor differs from ESS (so we don't touch GPIOs on laptops). Tested with SF64-PCE2 and M56VAP cards. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/es1968.c | 73 +++++++++++++++++++++++++++++++++++---------------- 1 files changed, 50 insertions(+), 23 deletions(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index a1f32b5..db08641 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -564,6 +564,7 @@ struct es1968 { #ifdef CONFIG_SND_ES1968_RADIO struct v4l2_device v4l2_dev; struct snd_tea575x tea; + unsigned int tea575x_tuner; #endif }; @@ -2557,37 +2558,47 @@ static int snd_es1968_input_register(struct es1968 *chip) bits 1=unmask write to given bit */ #define IO_DIR 8 /* direction register offset from GPIO_DATA bits 0/1=read/write direction */ -/* mask bits for GPIO lines */ -#define STR_DATA 0x0040 /* GPIO6 */ -#define STR_CLK 0x0080 /* GPIO7 */ -#define STR_WREN 0x0100 /* GPIO8 */ -#define STR_MOST 0x0200 /* GPIO9 */ + +/* GPIO to TEA575x maps */ +struct snd_es1968_tea575x_gpio { + u8 data, clk, wren, most; + char *name; +}; + +static struct snd_es1968_tea575x_gpio snd_es1968_tea575x_gpios[] = { + { .data = 6, .clk = 7, .wren = 8, .most = 9, .name = "SF64-PCE2" }, + { .data = 7, .clk = 8, .wren = 6, .most = 10, .name = "M56VAP" }, +}; + +#define get_tea575x_gpio(chip) \ + (&snd_es1968_tea575x_gpios[(chip)->tea575x_tuner]) + static void snd_es1968_tea575x_set_pins(struct snd_tea575x *tea, u8 pins) { struct es1968 *chip = tea->private_data; - unsigned long io = chip->io_port + GPIO_DATA; + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); u16 val = 0; - val |= (pins & TEA575X_DATA) ? STR_DATA : 0; - val |= (pins & TEA575X_CLK) ? STR_CLK : 0; - val |= (pins & TEA575X_WREN) ? STR_WREN : 0; + val |= (pins & TEA575X_DATA) ? (1 << gpio.data) : 0; + val |= (pins & TEA575X_CLK) ? (1 << gpio.clk) : 0; + val |= (pins & TEA575X_WREN) ? (1 << gpio.wren) : 0; - outw(val, io); + outw(val, chip->io_port + GPIO_DATA); } static u8 snd_es1968_tea575x_get_pins(struct snd_tea575x *tea) { struct es1968 *chip = tea->private_data; - unsigned long io = chip->io_port + GPIO_DATA; - u16 val = inw(io); - u8 ret; + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); + u16 val = inw(chip->io_port + GPIO_DATA); + u8 ret = 0; - ret = 0; - if (val & STR_DATA) + if (val & (1 << gpio.data)) ret |= TEA575X_DATA; - if (val & STR_MOST) + if (val & (1 << gpio.most)) ret |= TEA575X_MOST; + return ret; } @@ -2596,13 +2607,18 @@ static void snd_es1968_tea575x_set_direction(struct snd_tea575x *tea, bool outpu struct es1968 *chip = tea->private_data; unsigned long io = chip->io_port + GPIO_DATA; u16 odir = inw(io + IO_DIR); + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); if (output) { - outw(~(STR_DATA | STR_CLK | STR_WREN), io + IO_MASK); - outw(odir | STR_DATA | STR_CLK | STR_WREN, io + IO_DIR); + outw(~((1 << gpio.data) | (1 << gpio.clk) | (1 << gpio.wren)), + io + IO_MASK); + outw(odir | (1 << gpio.data) | (1 << gpio.clk) | (1 << gpio.wren), + io + IO_DIR); } else { - outw(~(STR_CLK | STR_WREN | STR_DATA | STR_MOST), io + IO_MASK); - outw((odir & ~(STR_DATA | STR_MOST)) | STR_CLK | STR_WREN, io + IO_DIR); + outw(~((1 << gpio.clk) | (1 << gpio.wren) | (1 << gpio.data) | (1 << gpio.most)), + io + IO_MASK); + outw((odir & ~((1 << gpio.data) | (1 << gpio.most))) + | (1 << gpio.clk) | (1 << gpio.wren), io + IO_DIR); } } @@ -2772,6 +2788,9 @@ static int snd_es1968_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); #ifdef CONFIG_SND_ES1968_RADIO + /* don't play with GPIOs on laptops */ + if (chip->pci->subsystem_vendor != 0x125d) + goto no_radio; err = v4l2_device_register(&pci->dev, &chip->v4l2_dev); if (err < 0) { snd_es1968_free(chip); @@ -2781,10 +2800,18 @@ static int snd_es1968_create(struct snd_card *card, chip->tea.private_data = chip; chip->tea.radio_nr = radio_nr; chip->tea.ops = &snd_es1968_tea_ops; - strlcpy(chip->tea.card, "SF64-PCE2", sizeof(chip->tea.card)); sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); - if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) - printk(KERN_INFO "es1968: detected TEA575x radio\n"); + for (i = 0; i < ARRAY_SIZE(snd_es1968_tea575x_gpios); i++) { + chip->tea575x_tuner = i; + if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) { + snd_printk(KERN_INFO "es1968: detected TEA575x radio type %s\n", + get_tea575x_gpio(chip)->name); + strlcpy(chip->tea.card, get_tea575x_gpio(chip)->name, + sizeof(chip->tea.card)); + break; + } + } +no_radio: #endif *chip_ret = chip; -- Ondrej Zary

2 1

[alsa-devel] [PATCH 1/1] ALSA: hda - add dock support for Thinkpad T431s
by Ebben Aries 31 May '13

31 May '13

Add a model/fixup string "lenovo-dock", for Thinkpad T431s, to allow sound in docking station. Tested on Lenovo T431s with ThinkPad Mini Dock Plus Series 3 Signed-off-by: Ebben Aries <earies(a)dscp.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 59d2e91..9658faf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3530,6 +3530,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), -- 1.7.12.1

2 1

[alsa-devel] [PATCH] ALSA: sis7019: fix error return code in sis_chip_create()
by Wei Yongjun 31 May '13

31 May '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Fix to return a negative error code in the pci_set_dma_mask() error handling case instead of 0, as done elsewhere in this function. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/pci/sis7019.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index f2639a0..e413b4e 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1341,7 +1341,8 @@ static int sis_chip_create(struct snd_card *card, if (rc) goto error_out; - if (pci_set_dma_mask(pci, DMA_BIT_MASK(30)) < 0) { + rc = pci_set_dma_mask(pci, DMA_BIT_MASK(30)); + if (rc < 0) { dev_err(&pci->dev, "architecture does not support 30-bit PCI busmaster DMA"); goto error_out_enabled; }

2 1

Re: [alsa-devel] [PATCH] ALSA: hda - Enable runtime pm for Haswell
by Wang, Xingchao 31 May '13

31 May '13

Hi Rafael, > -----Original Message----- > From: Wysocki, Rafael J > Sent: Wednesday, May 22, 2013 7:09 PM > To: Takashi Iwai > Cc: Wang, Xingchao; Girdwood, Liam R; Lin, Mengdong; Li, Jocelyn; > alsa-devel(a)alsa-project.org; Wang Xingchao > Subject: Re: [PATCH] ALSA: hda - Enable runtime pm for Haswell > > On 5/22/2013 7:51 AM, Takashi Iwai wrote: > > At Wed, 22 May 2013 03:56:00 +0000, > > Wang, Xingchao wrote: > >> Hi, > >> > >> Add Rafael in loop. > >> > >>> -----Original Message----- > >>> From: Takashi Iwai [mailto:tiwai@suse.de] > >>> Sent: Thursday, May 16, 2013 4:49 PM > >>> To: Wang Xingchao > >>> Cc: Girdwood, Liam R; Lin, Mengdong; Li, Jocelyn; > >>> alsa-devel(a)alsa-project.org; Wang, Xingchao > >>> Subject: Re: [PATCH] ALSA: hda - Enable runtime pm for Haswell > >>> > >>> At Thu, 16 May 2013 16:29:05 +0800, > >>> Wang Xingchao wrote: > >>>> Haswell doesnot support runtime pm by default. > >>>> This patch let haswell Display HD-A controller enter runtime > >>>> suspend, and bring more power saving whith power-well. > >>>> > >>>> Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> > >>> I don't think it's good to fiddle such a thing in the driver side. > >>> If Haswell can support runtime PM really, it should be set commonly. > >> I wonder whether it's HD-A driver's policy to only support runtime PM if the > device can support signal wakup? > >> According to Rafael, the device can support runtime PM regardless, no > matter it supports PME or not. > >> If so, we should remove the "if" condition check here. > > Well, if the decision is purely a driver issue, then we can get rid of > > PME check. > > Yes, it is. > > > But in that case, it should be simply like: > > > > azx_probe() { > > ... > > if (chip->driver_caps & AZX_DCAPS_PM_RUNTIME) > > pm_runtime_put_noidle(&pci->dev); > > ... > > } > > > > azx_remove() { > > ... > > if (chip->driver_caps & AZX_DCAPS_PM_RUNTIME) > > pm_runtime_get_noresume(&pci->dev); > > ... > > } > > > > > > AFAIU, calling pm_runtime_allow() enables the runtime PM *forcibly*. > > Usually this isn't a good thing. > > > > That's correct. Moreover, calling both pm_runtime_allow() *and* > pm_runtime_put_noidle() together would be a bug IMO. It's interesting something maybe wrong in my test: 1. withtout calling pm_runtime_allow(), azx_runteim_idle/suspend() will not be called. 2. another trick is on my Haswell ULT C stepping board, the runtime PM only work after exit from resume: echo mem > /sys/power/state if you donot let system enter suspend manually, the runtime pm will not be triggered. Is there any dependency between runtime pm suspend and normal suspend? Thanks --xingchao > > Thanks, > Rafael > > > > > >>>> --- > >>>> sound/pci/hda/hda_intel.c | 6 ++++++ > >>>> 1 file changed, 6 insertions(+) > >>>> > >>>> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > >>>> index bf27693..eb25888 100644 > >>>> --- a/sound/pci/hda/hda_intel.c > >>>> +++ b/sound/pci/hda/hda_intel.c > >>>> @@ -3755,6 +3755,12 @@ static int azx_probe(struct pci_dev *pci, > >>>> > >>>> if (pci_dev_run_wake(pci)) > >>>> pm_runtime_put_noidle(&pci->dev); > >>>> + else if (chip->driver_caps > >>>> + & AZX_DCAPS_I915_POWERWELL) { > >>>> + /* Haswell doesnot support runtime pm by default */ > >>>> + pm_runtime_put_noidle(&pci->dev); > >>>> + pm_runtime_allow(&pci->dev); > >>>> + } > >>>> > >>>> dev++; > >>>> complete_all(&chip->probe_wait); > >>>> -- > >>>> 1.7.9.5 > >>>>

2 2

[alsa-devel] [PATCH v7 00/11] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 31 May '13

31 May '13

Hi, This series adds DT support for phycore-ac97. Beside ac97 support, the series adds imx-pcm-fiq and generic DMA bindings to fsl-ssi. Regards, Markus Changes in v7: - Only one soc_ac97_ops struct for both ssi drivers (fsl-ssi and imx-ssi) to make it possible to compile a kernel with both drivers. - Other small bug/style fixes described in the changelogs of the patches. Changes in v6: - Rebased onto 3.10-rc3 with Shawn's pcm cleanup patches - Setup of reset functions moved to phycore-ac97.c Changes in v5: - Pass of_node via parent device to imx-pcm-dma. Changes in v4: - phycore-ac97 uses the iomux functions based on audmux phandle now. - fsl-ssi imx-pcm-fiq got some new comments about problems with imx-ssi and incompatible codecs Changes in v3: - Rebased onto pcm dma cleanup patches - A lot of cleanups for the fsl-ssi ac97 integration - Some small changes which are listed in the notes of the patches. Markus Pargmann (11): ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl: Move soc_ac97_ops from imx-ssi to fsl_ssi ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ASoC: fsl-ssi: imx ac97 support ARM: imx: Export ac97 reset functions ASoC: phycore-ac97: Add DT support ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97 ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc .../{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 10 + .../bindings/sound/phytec,phycore-ac97.txt | 14 + arch/arm/mach-imx/mach-pca100.c | 7 +- arch/arm/mach-imx/mach-pcm043.c | 7 +- sound/soc/fsl/Kconfig | 4 +- sound/soc/fsl/fsl_ssi.c | 450 +++++++++++++++++---- sound/soc/fsl/fsl_ssi.h | 7 + sound/soc/fsl/imx-pcm-dma.c | 1 - sound/soc/fsl/imx-pcm-fiq.c | 18 +- sound/soc/fsl/imx-pcm.h | 15 +- sound/soc/fsl/imx-ssi.c | 14 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/phycore-ac97.c | 236 +++++++++-- 13 files changed, 656 insertions(+), 128 deletions(-) rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (88%) create mode 100644 Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt

2 12

[alsa-devel] [PATCH] alsa-utils 1.0.27.1 still ships 90-alsa-restore.rules
by Elimar Riesebieter 30 May '13

30 May '13

Hi all, [0] tells: alsactl: remove 90-alsa-restore.rules from the tar ball but it still ships 90-alsa-restore.rules.in and installs the file as 90-alsa-restore.rules [0] http://www.alsa-project.org/main/index.php/Changes_v1.0.27_v1.0.27.1 Thanks Signed-off-by: Elimar Riesebieter <riesebie(a)lxtec.de> --- alsactl/90-alsa-restore.rules.in | 8 -------- alsactl/Makefile.am | 5 ----- 2 files changed, 13 deletions(-) diff --git a/alsactl/90-alsa-restore.rules.in b/alsactl/90-alsa-restore.rules.in deleted file mode 100644 index 88e12e0..0000000 --- a/alsactl/90-alsa-restore.rules.in +++ /dev/null @@ -1,8 +0,0 @@ -ACTION=="add", SUBSYSTEM=="sound", KERNEL=="controlC*", KERNELS!="card*", GOTO="alsa_restore_go" -GOTO="alsa_restore_end" - -LABEL="alsa_restore_go" -TEST!="@daemonswitch@", RUN+="@sbindir@/alsactl restore $attr{number}" -TEST=="@daemonswitch@", RUN+="@sbindir@/alsactl nrestore $attr{number}" - -LABEL="alsa_restore_end" diff --git a/alsactl/Makefile.am b/alsactl/Makefile.am index bac84eb..3bf8eca 100644 --- a/alsactl/Makefile.am +++ b/alsactl/Makefile.am @@ -11,8 +11,6 @@ alsactl_SOURCES=alsactl.c state.c lock.c utils.c init_parse.c daemon.c alsactl_CFLAGS=$(AM_CFLAGS) -D__USE_GNU -DSYS_ASOUNDRC=\"$(ASOUND_STATE_DIR)/asound.state\" -DSYS_PIDFILE=\"$(ALSACTL_PIDFILE_DIR)/alsactl.pid\" noinst_HEADERS=alsactl.h list.h init_sysdeps.c init_utils_string.c init_utils_run.c init_sysfs.c -udevrules_DATA = \ - 90-alsa-restore.rules if HAVE_SYSTEMD @@ -49,20 +47,17 @@ alsa-restore.service: alsa-restore.service.in alsa-store.service: alsa-store.service.in $(edit) -90-alsa-restore.rules: 90-alsa-restore.rules.in $(edit) EXTRA_DIST += \ alsa-state.service.in \ alsa-restore.service.in \ alsa-store.service.in \ - 90-alsa-restore.rules.in CLEANFILES = \ alsa-state.service \ alsa-restore.service \ alsa-store.service \ - 90-alsa-restore.rules %.7: %.xml xmlto man $?

2 5

[alsa-devel] [PATCH 1/5] ASoC: wm8994: Add digital loopback paths
by Mark Brown 30 May '13

30 May '13

There is loopback control within the audio interfaces, provide control of this as there are some obscure scenarios where this could be used in production. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 31 +++++++++++++++++++++++++++++-- 1 file changed, 29 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 561c2f9..2081fa1 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1499,6 +1499,24 @@ static const char *aif1dac_text[] = { "AIF1DACDAT", "AIF3DACDAT", }; +static const char *loopback_text[] = { + "None", "ADCDAT", +}; + +static const struct soc_enum aif1_loopback_enum = + SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, WM8994_AIF1_LOOPBACK_SHIFT, 2, + loopback_text); + +static const struct snd_kcontrol_new aif1_loopback = + SOC_DAPM_ENUM("AIF1 Loopback", aif1_loopback_enum); + +static const struct soc_enum aif2_loopback_enum = + SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, WM8994_AIF2_LOOPBACK_SHIFT, 2, + loopback_text); + +static const struct snd_kcontrol_new aif2_loopback = + SOC_DAPM_ENUM("AIF2 Loopback", aif2_loopback_enum); + static const struct soc_enum aif1dac_enum = SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 0, 2, aif1dac_text); @@ -1745,6 +1763,9 @@ SND_SOC_DAPM_ADC("DMIC1R", NULL, WM8994_POWER_MANAGEMENT_4, 2, 0), SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_MUX("AIF1 Loopback", SND_SOC_NOPM, 0, 0, &aif1_loopback), +SND_SOC_DAPM_MUX("AIF2 Loopback", SND_SOC_NOPM, 0, 0, &aif2_loopback), + SND_SOC_DAPM_POST("Debug log", post_ev), }; @@ -1876,9 +1897,9 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF1DAC2L", NULL, "AIF1DAC Mux" }, { "AIF1DAC2R", NULL, "AIF1DAC Mux" }, - { "AIF1DAC Mux", "AIF1DACDAT", "AIF1DACDAT" }, + { "AIF1DAC Mux", "AIF1DACDAT", "AIF1 Loopback" }, { "AIF1DAC Mux", "AIF3DACDAT", "AIF3DACDAT" }, - { "AIF2DAC Mux", "AIF2DACDAT", "AIF2DACDAT" }, + { "AIF2DAC Mux", "AIF2DACDAT", "AIF2 Loopback" }, { "AIF2DAC Mux", "AIF3DACDAT", "AIF3DACDAT" }, { "AIF2ADC Mux", "AIF2ADCDAT", "AIF2ADCL" }, { "AIF2ADC Mux", "AIF2ADCDAT", "AIF2ADCR" }, @@ -1929,6 +1950,12 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACL" }, { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACR" }, + /* Loopback */ + { "AIF1 Loopback", "ADCDAT", "AIF1ADCDAT" }, + { "AIF1 Loopback", "None", "AIF1DACDAT" }, + { "AIF2 Loopback", "ADCDAT", "AIF2ADCDAT" }, + { "AIF2 Loopback", "None", "AIF2DACDAT" }, + /* Sidetone */ { "Left Sidetone", "ADC/DMIC1", "ADCL Mux" }, { "Left Sidetone", "DMIC2", "DMIC2L" }, -- 1.7.10.4

2 5

[alsa-devel] [PATCH] hda_analyzer: don't fail to display audio selector if caps variables aren't available
by Anisse Astier 30 May '13

30 May '13

In particular, when an audio selector node has Amp-In caps: N/A, it allows to still interact with the selector, and not have TypeError traceback throwns because of NoneType variables. Signed-off-by: Anisse Astier <anisse(a)astier.eu> --- Hi Jared, I'm not sure if this is the proper way to fix this, so please consider this a formal report for the error thrown while parsing this node: Node 0x22 [Audio Selector] wcaps 0x30010b: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Connection: 7 0x18* 0x19 0x1a 0x1b 0x1d 0x0b 0x12 Regards, Anisse -- hda-analyzer/hda_guilib.py | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/hda-analyzer/hda_guilib.py b/hda-analyzer/hda_guilib.py index 93c85e1..62eddab 100644 --- a/hda-analyzer/hda_guilib.py +++ b/hda-analyzer/hda_guilib.py @@ -285,9 +285,9 @@ class NodeGui(gtk.ScrolledWindow): frame.set_border_width(4) vbox = gtk.VBox(False, 0) if caps: - str = 'Offset: %d\n' % caps.ofs - str += 'Number of steps: %d\n' % caps.nsteps - str += 'Step size: %d\n' % caps.stepsize + str = 'Offset: %s\n' % (caps.ofs if caps.ofs else "N/A") + str += 'Number of steps: %s\n' % (caps.nsteps if caps.nsteps else "N/A") + str += 'Step size: %s\n' % (caps.stepsize if caps.stepsize else "N/A") str += 'Mute: %s\n' % (caps.mute and "True" or "False") vbox.pack_start(self.__new_text_view(text=str), True, True, 0) idx = 0 -- 1.8.3.rc1

2 2

[alsa-devel] [PATCH 01/10] ASoC: blackfin: Remove unused bf5xx-{i2s, tdm, ac97}-pcm.h
by Lars-Peter Clausen 30 May '13

30 May '13

The structs defined in these files are completely unused, so remove both the structs and the files. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/blackfin/bf5xx-ac97-pcm.c | 1 - sound/soc/blackfin/bf5xx-ac97-pcm.h | 26 -------------------------- sound/soc/blackfin/bf5xx-ad1836.c | 1 - sound/soc/blackfin/bf5xx-ad193x.c | 1 - sound/soc/blackfin/bf5xx-ad1980.c | 1 - sound/soc/blackfin/bf5xx-ad73311.c | 1 - sound/soc/blackfin/bf5xx-i2s-pcm.c | 1 - sound/soc/blackfin/bf5xx-i2s-pcm.h | 26 -------------------------- sound/soc/blackfin/bf5xx-ssm2602.c | 1 - sound/soc/blackfin/bf5xx-tdm-pcm.c | 1 - sound/soc/blackfin/bf5xx-tdm-pcm.h | 18 ------------------ 11 files changed, 78 deletions(-) delete mode 100644 sound/soc/blackfin/bf5xx-ac97-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-i2s-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.h diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 7e2f360..53f8408 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -39,7 +39,6 @@ #include <asm/dma.h> -#include "bf5xx-ac97-pcm.h" #include "bf5xx-ac97.h" #include "bf5xx-sport.h" diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.h b/sound/soc/blackfin/bf5xx-ac97-pcm.h deleted file mode 100644 index d324d58..0000000 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.h +++ /dev/null @@ -1,26 +0,0 @@ -/* - * linux/sound/arm/bf5xx-ac97-pcm.h -- ALSA PCM interface for the Blackfin - * - * Copyright 2007 Analog Device 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. - */ - -#ifndef _BF5XX_AC97_PCM_H -#define _BF5XX_AC97_PCM_H - -struct bf5xx_pcm_dma_params { - char *name; /* stream identifier */ -}; - -struct bf5xx_gpio { - u32 sys; - u32 rx; - u32 tx; - u32 clk; - u32 frm; -}; - -#endif diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index d23f4b0..1528176 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -30,7 +30,6 @@ #include "../codecs/ad1836.h" -#include "bf5xx-tdm-pcm.h" #include "bf5xx-tdm.h" static struct snd_soc_card bf5xx_ad1836; diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index 0e55e9f..ce773b3 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -39,7 +39,6 @@ #include "../codecs/ad193x.h" -#include "bf5xx-tdm-pcm.h" #include "bf5xx-tdm.h" static struct snd_soc_card bf5xx_ad193x; diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c index b30f88b..3450e8f 100644 --- a/sound/soc/blackfin/bf5xx-ad1980.c +++ b/sound/soc/blackfin/bf5xx-ad1980.c @@ -48,7 +48,6 @@ #include "../codecs/ad1980.h" -#include "bf5xx-ac97-pcm.h" #include "bf5xx-ac97.h" static struct snd_soc_card bf5xx_board; diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 61cc91d..786bbdd 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -45,7 +45,6 @@ #include "../codecs/ad73311.h" #include "bf5xx-sport.h" -#include "bf5xx-i2s-pcm.h" #if CONFIG_SND_BF5XX_SPORT_NUM == 0 #define bfin_write_SPORT_TCR1 bfin_write_SPORT0_TCR1 diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 262c1de..8726c3a 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -39,7 +39,6 @@ #include <asm/dma.h> -#include "bf5xx-i2s-pcm.h" #include "bf5xx-sport.h" static void bf5xx_dma_irq(void *data) diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.h b/sound/soc/blackfin/bf5xx-i2s-pcm.h deleted file mode 100644 index 0c2c5a6..0000000 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.h +++ /dev/null @@ -1,26 +0,0 @@ -/* - * linux/sound/arm/bf5xx-i2s-pcm.h -- ALSA PCM interface for the Blackfin - * - * Copyright 2007 Analog Device 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. - */ - -#ifndef _BF5XX_I2S_PCM_H -#define _BF5XX_I2S_PCM_H - -struct bf5xx_pcm_dma_params { - char *name; /* stream identifier */ -}; - -struct bf5xx_gpio { - u32 sys; - u32 rx; - u32 tx; - u32 clk; - u32 frm; -}; - -#endif diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index 7dbeef1..9c19ccc 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -40,7 +40,6 @@ #include <linux/gpio.h> #include "../codecs/ssm2602.h" #include "bf5xx-sport.h" -#include "bf5xx-i2s-pcm.h" static struct snd_soc_card bf5xx_ssm2602; diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index 19e855d..a6b5457 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -39,7 +39,6 @@ #include <asm/dma.h> -#include "bf5xx-tdm-pcm.h" #include "bf5xx-tdm.h" #include "bf5xx-sport.h" diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.h b/sound/soc/blackfin/bf5xx-tdm-pcm.h deleted file mode 100644 index 7f8cc01..0000000 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.h +++ /dev/null @@ -1,18 +0,0 @@ -/* - * sound/soc/blackfin/bf5xx-tdm-pcm.h -- ALSA PCM interface for the Blackfin - * - * Copyright 2009 Analog Device 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. - */ - -#ifndef _BF5XX_TDM_PCM_H -#define _BF5XX_TDM_PCM_H - -struct bf5xx_pcm_dma_params { - char *name; /* stream identifier */ -}; - -#endif -- 1.8.0

2 14

[alsa-devel] [PATCH 0/4 V6] Power-well API implementation for Haswell
by Wang Xingchao 30 May '13

30 May '13

Hi all, This is V6 and here're some changes notes: change from V5-->V6: - Remove duplication code in new introduced probe work - move duplication code in azx_probe_continue - remove unused #ifdef - replace request_module with symbol_request - replace spin_lock_irq with spin_lock_irqsave in gfx side - other typo fixes (review by Takashi) change from V4-->V5: - fix reference count bug - new patch on general runtime pm support for audio pci device - new patch to avoid request_module() deadlock change between V3-->V4: - add new structure i915_power_well - initialize drm_device pointer at module init time - change function name change between V2-->V3: - make SND_HDA_I915 selectable - use snd_printdd to output message - add return error code check - use symbol_request to replace symbol_get - release power_well at azx_free - some typo fixes changes between V1-->V2: - use reference count to track power-well usage - remove external module, compiled into snd-hda-intel instead - manage symbols and module loading properly - remove IS_HSW macro, use flag instead - remove audio callback for gfx driver to avoid dependency - split whole patch into two pieces for easy review - more typo fixes Takashi Iwai (1): ALSA: hda - Move azx_first_init() into azx_probe_continue() Wang Xingchao (3): ALSA: hda - Fix runtime PM check ALSA: hda - Add power-welll support for haswell HDA i915/drm: Add private api for power well usage drivers/gpu/drm/i915/i915_dma.c | 6 +++ drivers/gpu/drm/i915/i915_drv.h | 12 +++++ drivers/gpu/drm/i915/intel_drv.h | 4 ++ drivers/gpu/drm/i915/intel_pm.c | 92 +++++++++++++++++++++++++++++++++++--- include/drm/i915_powerwell.h | 36 +++++++++++++++ sound/pci/hda/Kconfig | 10 +++++ sound/pci/hda/Makefile | 2 + sound/pci/hda/hda_i915.c | 75 +++++++++++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 35 +++++++++++++++ sound/pci/hda/hda_intel.c | 87 ++++++++++++++++++++++++++++------- 10 files changed, 335 insertions(+), 24 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.7.9.5

3 6

[alsa-devel] [PATCH] ASoC: OMAP: Remove obsolete Makefile line
by Paul Bolle 30 May '13

30 May '13

Support for omap2evm was removed in v3.0. But only one of its two lines in this Makefile was removed. Remove the second line too. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Eyeball tested only. sound/soc/omap/Makefile | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 2b22594..a725905 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -26,7 +26,6 @@ obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_TWL4030) += snd-soc-omap-twl4030.o -- 1.7.11.7

3 2

[alsa-devel] [PATCH] ASoC: wm0010: Set IRQ as a wake source
by Charles Keepax 30 May '13

30 May '13

The DSPs IRQ should be a wake source as several of the possible algorithms may run whilst the AP is asleepi and require to wake the AP to push or pull more data, such as compressed playback. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 370af0c..b6df319 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -972,6 +972,13 @@ static int wm0010_spi_probe(struct spi_device *spi) } wm0010->irq = irq; + ret = irq_set_irq_wake(irq, 1); + if (ret) { + dev_err(wm0010->dev, "Failed to set IRQ %d as wake source: %d\n", + irq, ret); + return ret; + } + if (spi->max_speed_hz) wm0010->board_max_spi_speed = spi->max_speed_hz; else -- 1.7.2.5

2 1

[alsa-devel] [PATCH v6 00/10] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 30 May '13

30 May '13

Hi, This series adds DT support for phycore-ac97. Beside ac97 support, the series adds imx-pcm-fiq and generic DMA bindings to fsl-ssi. Regards, Markus Changes in v6: - Rebased onto 3.10-rc3 with Shawn's pcm cleanup patches - Setup of reset functions moved to phycore-ac97.c Changes in v5: - Pass of_node via parent device to imx-pcm-dma. Changes in v4: - phycore-ac97 uses the iomux functions based on audmux phandle now. - fsl-ssi imx-pcm-fiq got some new comments about problems with imx-ssi and incompatible codecs Changes in v3: - Rebased onto pcm dma cleanup patches - A lot of cleanups for the fsl-ssi ac97 integration - Some small changes which are listed in the notes of the patches. Markus Pargmann (10): ASoC: phycore-ac97: Add DT support ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ARM: imx: Export ac97 reset functions ASoC: fsl-ssi: imx ac97 support ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97 ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc .../{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 10 + .../bindings/sound/phytec,phycore-ac97.txt | 14 + arch/arm/mach-imx/mach-pca100.c | 7 +- arch/arm/mach-imx/mach-pcm043.c | 7 +- sound/soc/fsl/Kconfig | 2 +- sound/soc/fsl/fsl_ssi.c | 441 +++++++++++++++++---- sound/soc/fsl/fsl_ssi.h | 7 + sound/soc/fsl/imx-pcm-dma.c | 1 - sound/soc/fsl/imx-pcm-fiq.c | 18 +- sound/soc/fsl/imx-pcm.h | 12 +- sound/soc/fsl/imx-ssi.c | 7 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/phycore-ac97.c | 241 +++++++++-- 13 files changed, 643 insertions(+), 125 deletions(-) rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (88%) create mode 100644 Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt

4 20

[alsa-devel] [PATCH 0/2] ASoC: cs42l52: HP gain enum and PCM Mixer dB scale fixes.
by Nicolas Schichan 29 May '13

29 May '13

Hi, The two patches in the serie contains the following fixes: - Use the correct shift value in hp_gain_enum, effectively allowing the higher gain values to be effective and the correct gain value to be reported to userland wrt what is written to the register. - Fix the PCM Mixer dB scale so that userland can report the correct dB gain values. Note that in its current form, the PCM Mixer control will only allow the range [-0.5 dB (0x7f), +12 dB (0x18)] to be programmed to the hardware. I don't know if it is intentional or not. Regards, Nicolas Schichan (2): ASoC: cs42l52: use correct PCM mixer TLV dB scale to match datasheet. ASoC: cs42l52: fix hp_gain_enum shift value. sound/soc/codecs/cs42l52.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) -- 1.7.10.4

3 6

[alsa-devel] [PATCH] ASoC: wm8994: Avoid leaking pm_runtime reference on removed jack race
by Mark Brown 29 May '13

29 May '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1ddb067..4aaa8a0 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3886,7 +3886,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) ret); } else if (!(ret & WM1811_JACKDET_LVL)) { dev_dbg(codec->dev, "Ignoring removed jack\n"); - return IRQ_HANDLED; + goto out; } } else if (!(reg & WM8958_MICD_STS)) { snd_soc_jack_report(wm8994->micdet[0].jack, 0, -- 1.7.10.4

1 0

[alsa-devel] UCM {Supported,Conflicting}Device confusion
by Juho Hämäläinen 29 May '13

29 May '13

Hello, Firstly as a sidenote, Supported/Conflicting is checked only one-way, if say devices A and B are enabled and say B has ConflictingDevice "C", and device C doesn't list anything, enabling of C is possible (and after that disabling C is not possible before B is disabled first). Now, looking at the way SupportedDevice and ConflictingDevice are implemented in ucm and I came to following conclusion: If no SupportedDevice or ConflictingDevice: All devices and combinations are compatible, so implicit "SupportedDevice ALL". If SupportedDevice is listed: Device can be enabled as long as at least one enabled device is in SupportedDevice list. If ConflictingDevice is listed: Device can be enabled if none of active devices are in ConflictingDevice list (noting aforementioned limit in checking this) This is at least how I figured all combinations of multiple devices can be defined with only zero or one either SupportedDevice or ConflictingDevice entry per device. Did I understand this right? -- juho

2 1

[alsa-devel] [PATCH 1/3] lx6464es: remove useles pci_set_drvdata(pdev, NULL)
by Andy Shevchenko 29 May '13

29 May '13

The driver core handles this. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/pci/lx6464es/lx6464es.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 298bc9b..3230e57 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -1139,7 +1139,6 @@ out_free: static void snd_lx6464es_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } -- 1.8.2.rc0.22.gb3600c3

2 10

[alsa-devel] ASoC updates for v3.10
by Mark Brown 29 May '13

29 May '13

The following changes since commit e6135fe960bba1d1a3a6b6818839e6d8d3f21293: ALSA: usb-audio: proc: use found syncmaxsize to determine feedback format (2013-05-17 08:05:34 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.10-rc3 for you to fetch changes up to 1ab9ecc24819a8cf8ee982aaf6fb83298f094b0d: Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linus (2013-05-28 15:41:37 +0100) The diffstat and so on being rather obscured by the fact that I've got a newer -rc merged in here than you do in your branch, sorry. ---------------------------------------------------------------- ASoC: Updates for v3.10 A series of driver specific updates, none particularly critical, plus one fix to the compressed API code to handle capture streams properly which is very safe for mainline as there's no current users. ---------------------------------------------------------------- Alan Stern (6): USB: xHCI: override bogus bulk wMaxPacketSize values USB: fix Kconfig logic for USB_UHCI_HCD USB: UHCI: fix for suspend of virtual HP controller USB: fix latency in uhci-hcd and ohci-hcd USB: OHCI: fix logic for scheduling isochronous URBs USB: remove remaining instances of USB_SUSPEND Alasdair G Kergon (1): dm thin: fix metadata dev resize detection Albert Pool (1): rtlwifi: rtl8192cu: Add new USB ID Alex Deucher (10): drm/radeon/dce2: use 10khz units for audio dto calculation drm/radeon: add chip family for Hainan drm/radeon: fill in GPU init for Hainan (v2) drm/radeon: don't touch DCE or VGA regs on Hainan (v3) drm/radeon: fill in ucode loading support for Hainan drm/radeon: radeon-asic updates for Hainan drm/radeon: track which asics have UVD drm/radeon: sun/hainan chips do not have UVD (v2) drm/radeon: add golden register settings for Hainan (v2) drm/radeon: add Hainan pci ids Alex Elder (33): rbd: fix leak of format 2 snapshot context rbd: revalidate only for mapping size changes rbd: update capacity in rbd_dev_refresh() rbd: kill rbd_update_mapping_size() rbd: don't print warning if not mapping a parent rbd: don't look up snapshot id in rbd_dev_mapping_set() rbd: kill rbd_dev_clear_mapping() rbd: always set read-only flag in rbd_add() rbd: set the mapping size and features later rbd: zero format 1 header structure earlier rbd: refactor rbd_header_from_disk() rbd: update in-core header directly rbd: simplify rbd_dev_v1_probe() rbd: get rid of trivial v1 header wrappers rbd: define rbd_dev_v2_header_info() rbd: fix an incorrect assertion condition rbd: support reading parent page data rbd: set mapping read-only flag in rbd_add() rbd: only set up watch for mapped images rbd: kill rbd_img_request_get() libceph: init sent and completed when starting rbd: fix parent request size assumption rbd: support reading parent page data for writes rbd: ignore zero-overlap parent rbd: get parent info on refresh rbd: don't release write request until necessary rbd: define rbd_dev_unparent() rbd: define parent image request routines rbd: reference count parent requests rbd: detect when clone image is flattened rbd: re-submit read request for flattened clone rbd: re-submit write request for flattened clone rbd: re-submit flattened write request (part 2) Alex Hung (1): ACPI video: ignore BIOS initial backlight value for HP 1000 Alexander Bondar (1): iwlwifi: mvm: Prevent setting assoc flag in MAC_CONTEXT_CMD Alexander Gordeev (1): powerpc: Fix irq_set_affinity() return values Alexander Shiyan (2): s390: remove non existent reference to GENERIC_KERNEL_THREAD cpufreq: Fix incorrect dependecies for ARM SA11xx drivers Alexander Sverdlin (1): i2c: suppress lockdep warning on delete_device Alexandre Bounine (4): rapidio: make enumeration/discovery configurable rapidio: add enumeration/discovery start from user space rapidio: documentation update for enumeration changes rapidio/tsi721: fix bug in MSI interrupt handling Alexandre Courbot (1): ARM: tegra: defconfig fixes Alexandre Oliva (1): btrfs: do away with non-whole_page extent I/O Alexandru Gheorghiu (1): drivers: char: ipmi: Replaced kmalloc and strcpy with kstrdup Alistair Popple (1): powerpc: Add an in memory udbg console Amerigo Wang (2): virtio_net: use default napi weight by default ipv6,gre: do not leak info to user-space Amitkumar Karwar (1): mwifiex: fix memory leak issue when driver unload Andreas Larsson (2): spi: Initialize cs_gpio and cs_gpios with -ENOENT spi: Return error from of_spi_register_master on bad "cs-gpios" property Andreas Philipp (1): Correct allowed raid levels on balance. Andrew Bresticker (1): ASoC: max98090: request IRQF_ONESHOT interrupt Andrew Lunn (2): ARM: Orion: Remove redundant init_dma_coherent_pool_size() mv643xx_eth: fix NAPI weight being > 64 Andrew Morton (1): revert "selftest: add simple test for soft-dirty bit" Andy Adamson (1): NFSv4.1 Fix a pNFS session draining deadlock Andy Grover (3): target: Remove unused struct members in se_dev_entry target: Use FD_MAX_SECTORS/FD_BLOCKSIZE for blockdevs using fileio target: Have dev/enable show if TCM device is configured Andy Shevchenko (1): dma: acpi-dma: parse CSRT to extract additional resources Aneesh Kumar K.V (3): powerpc/mm: Use the correct mask value when looking at pgtable address powerpc: Fix build errors STRICT_MM_TYPECHECKS mm/THP: use pmd_populate() to update the pmd with pgtable_t pointer Anton Blanchard (1): powerpc/kexec: Fix kexec when using VMX optimised memcpy Antonio Quartulli (2): batman-adv: make DAT drop ARP requests targeting local clients batman-adv: reorder clean up routine in order to avoid race conditions Ard Biesheuvel (1): ARM: 7723/1: crypto: sha1-armv4-large.S: fix SP handling Arend van Spriel (1): brcmfmac: announce P2P_DEVICE support in wiphy structure Arnd Bergmann (4): staging/drm: imx: add missing dependencies staging/solo6x10: depend on CONFIG_FONTS mfd: ab8500: Debugfs code depends on gpadc USB: EHCI: remove bogus #error Aron Xu (1): MIPS: N64: Wire getdents64(2) Axel Lin (7): wm831x_backup: Fix wrong kfree call for devdata->backup.name pm2301_charger: Fix module alias prefix hwmon: (iio_hwmon) Fix missing iio_channel_release_all call if devm_kzalloc fail hwmon: (iio_hwmon) Fix null pointer dereference gpio: mcp23s08: Fix build error when CONFIG_SPI_MASTER=y && CONFIG_I2C=m gpio: Don't override the error code in probe error handling iio: dac: Fix build error when CONFIG_SPI_MASTER=y && CONFIG_I2C=m Bastian Triller (1): ACPI / video: Add "Asus UL30A" to ACPI video detect blacklist Ben Hutchings (7): sfc: Delete EFX_PAGE_IP_ALIGN, equivalent to NET_IP_ALIGN sfc: Reduce RX scatter buffer size, and reduce alignment if appropriate perf: net_dropmonitor: Fix trace parameter order perf: net_dropmonitor: Fix symbol-relative addresses perf: net_dropmonitor: Do not assume ordering of dictionaries perf: net_dropmonitor: Use bisection in symbol lookup perf: net_dropmonitor: Remove progress indicator Ben Skeggs (9): drm/nouveau/bios: fix thinko in ZM_MASK_ADD opcode drm/nvc0/ce: disable ce1 on a number of chipsets drm/nvc0/ltcg: fix handling of disabled partitions drm/nve0/ltcg: poke the partition count into yet another register drm/nve0/fifo: prevent races between clients updating playlists drm/nvc0/fifo: prevent races between clients updating playlists drm/nvc0/fifo: prevent CHAN_TABLE_ERROR:CHANNEL_PENDING on fifo fini drm/nv50/fifo: prevent races between clients updating playlists drm/nouveau: ensure channels are stopped before saving fences for suspend Ben Widawsky (1): Revert "drm/i915: Calculate correct stolen size for GEN7+" Benjamin Herrenschmidt (10): powerpc/powernv: Properly drop characters if console is closed powerpc/powernv: Improve kexec reliability powerpc/powernv: Detect OPAL v3 API version powerpc/powernv: Fix starting of secondary CPUs on OPALv2 and v3 powerpc: Set show_unhandled_signals to 1 by default powerpc: Fix TLB cleanup at boot on POWER8 powerpc/pci: Fix bogus message at boot about empty memory resources powerpc/powernv: Fix condition for when to invalidate the TCE cache powerpc: Make radeon 32-bit MSI quirk work on powernv powerpc/powernv: Build a zImage.epapr Benjamin LaHaise (2): ipmi: ipmi_devintf: compat_ioctl method fails to take ipmi_mutex aio: fix kioctx not being freed after cancellation at exit time Bing Zhao (1): mwifiex: clear is_suspended flag when interrupt is received early Bjorn Helgaas (1): Revert "x86/pci/mrst: Use configuration mechanism 1 for 00:00.0, 00:02.0, 00:03.0" Bob Peterson (2): GFS2: two minor quota fixups GFS2: Use single-block reservations for directories Boris BREZILLON (1): ARM: at91/dt: fix macb pinctrl_macb_rmii_mii_alt definition Borislav Petkov (3): cpufreq, ondemand: Remove leftover debug line PM / hibernate: Correct documentation amd64_edac: Fix bogus sysfs file permissions Brian Behlendorf (1): drivers/block/brd.c: fix brd_lookup_page() race Brian King (1): powerpc/pseries: Make 32-bit MSI quirk work on systems lacking firmware support Catalin Marinas (3): arm64: Invoke the of_platform_populate() at arch_initcall() level arm64: Do not source kernel/time/Kconfig explicitly ACPI / scan: Fix memory leak on acpi_scan_init_hotplug() error path Charles Keepax (2): ASoC: soc-compress: Send correct stream event for capture start ASoC: wm5110: Correct DSP4R Mixer control name Chen Gang (3): arm64: Fix duplicate definition of early_console drivers/char/ipmi: memcpy, need additional 2 bytes to avoid memory overflow net: irda: using kzalloc() instead of kmalloc() to avoid strncpy() issue. Chris Cummins (1): drm: Use names of ioctls in debug traces Chris Mason (2): Btrfs: use a btrfs bioset instead of abusing bio internals Merge branch 'for-chris' of git://git.kernel.org/.../josef/btrfs-next Chris Wilson (1): drm/i915: Always normalize return timeout for wait_timeout_ioctl Christian Borntraeger (4): s390: disable pfmf for clear page instruction s390: fix gmap_ipte_notifier vs. software dirty pages s390/xpram: mark xpram as non-rotational s390/pgtable: fix ipte notify bit Christian Gmeiner (1): drivers/leds/leds-ot200.c: fix error caused by shifted mask Christian Ruppert (1): ARC: [TB10x] Remove redundant abilis,simple-pinctrl mechanism Christopher Harvey (4): drm/mgag200: Don't change unrelated registers during modeset drm/mgag200: Fix writes into MGA1064_PIX_CLK_CTL register drm/mgag200: Convert counter delays to jiffies drm/mgag200: Fix framebuffer base address programming Chuck Lever (1): NFS: Fix SETCLIENTID fallback if GSS is not available Cliff Wickman (1): mm/pagewalk.c: walk_page_range should avoid VM_PFNMAP areas Cong Wang (1): xfrm6: release dev before returning error Corey Minyard (1): ipmi: Improve error messages on failed irq enable Damien Lespiau (2): drm: Add missing break in the command line mode parsing code drm: Don't prune modes loudly when a connector is disconnected Dan Carpenter (6): xen/privcmd: fix condition in privcmd_close() ntb: off by one sanity checks NTB: fix pointer math issues ipg: fix an unsigned widening cast of '~' truncation issue USB: cxacru: potential underflow in cxacru_cm_get_array() isdn/kcapi: fix a small underflow Dan Magenheimer (1): staging: ramster: add how-to document Dan Williams (1): USB: option: add device IDs for Dell 5804 (Novatel E371) WWAN card Daniel Drake (1): mwifiex: fix setting of multicast filter Daniel Mack (1): ASoC: davinci: fix sample rotation Daniel Tang (1): ARM: 7712/1: Remove trailing whitespace in arch/arm/Makefile Daniel Vetter (3): drm/mm: fix dump table BUG drm: don't check modeset locks in panic handler Revert "drm/i915: revert eDP bpp clamping code changes" Darrick J. Wong (1): ttyprintk: Fix NULL pointer deref by setting tty_port ops after initializing port Dave Airlie (11): Merge branch 'for-linux-next' of git://people.freedesktop.org/~danvet/drm-intel into drm-next drm: remove unused wrapper macros drm/qxl: fix ioport interactions for kernel submitted commands. qxl: drop unused variable. drm/qxl: drop active_user_framebuffer as its unneeded qxl: fix bug with object eviction and update area drm/radeon: restore nomodeset operation (v2) drm/nouveau: fix build with nv50->nvc0 Merge branch 'drm-nouveau-fixes-3.10' of git://anongit.freedesktop.org/git/nouveau/linux-2.6 into drm-next Merge branch 'drm-fixes-3.10' of git://people.freedesktop.org/~agd5f/linux into drm-next Merge branch 'drm-fixes-3.10-sun' of git://people.freedesktop.org/~agd5f/linux into drm-next Dave Chinner (7): xfs: fix sub-page blocksize data integrity writes xfs: fix rounding in xfs_free_file_space xfs: Don't reference the EFI after it is freed xfs: fix missing KM_NOFS tags to keep lockdep happy xfs: xfs_da3_node_read_verify() doesn't handle XFS_ATTR3_LEAF_MAGIC xfs: xfs_attr_shortform_allfit() does not handle attr3 format. xfs: remote attribute lookups require the value length Dave Jones (1): Add missing module license tag to vring helpers. David Cohen (1): gpio-langwell: fix irq conflicts when DT is not used David Daney (3): Revert "MIPS: microMIPS: Support dynamic ASID sizing." Revert "MIPS: Allow ASID size to be determined at boot time." MIPS: Make virt_to_phys() work for all unmapped addresses. David Rientjes (2): mfd: tps65912: Select MFD_CORE usb, chipidea: fix link error when USB_EHCI_HCD is a module David S. Miller (9): Merge branch 'wireless' Merge branch 'qlcnic' Merge tag 'batman-adv-fix-for-davem' of git://git.open-mesh.org/linux-merge Merge branch 'master' of git://git.kernel.org/.../pablo/nf Merge tag 'batman-adv-fix-for-davem' of git://git.open-mesh.org/linux-merge Merge branch 'for-davem' of git://git.kernel.org/.../linville/wireless Merge branch 'perf_net_dropmonitor' Merge tag 'batman-adv-fix-for-davem' of git://git.open-mesh.org/linux-merge net: Revert unused variable changes. David Sterba (1): btrfs: annotate quota tree for lockdep David Woodhouse (1): powerpc: Provide __bswapdi2 Denis CIOCCA (1): iio:common:st: added disable function after read info raw data Denis Efremov (4): ipv4: ip_output: remove inline marking of EXPORT_SYMBOL functions staging: nvec: remove inline marking of EXPORT_SYMBOL functions iwlegacy: remove inline marking of EXPORT_SYMBOL functions MIPS: MSP71xx: remove inline marking of EXPORT_SYMBOL functions Dirk Behme (1): ARM i.MX6: correct MLB clock configuration Dirk Brandewie (6): cpufreq / intel_pstate: remove idle time and duration from sample and calculations cpufreq / intel_pstate: use lowest requested max performance cpufreq / intel_pstate: fix ffmpeg regression cpufreq / intel_pstate: Remove idle mode PID cpufreq / intel_pstate: remove #ifdef MODULE compile fence cpufreq / intel_pstate: Add additional supported CPU ID Emilio López (4): ARM: ux500: select REGULATOR net: ethernet: apple: drop unused variable net: ethernet: korina: drop unused variable net: ethernet: sun: drop unused variable Emmanuel Grumbach (2): iwlwifi: mvm: tell firmware to let multicast frames in iwlwifi: mvm: fix aggregation drain flow Eric Dumazet (8): ipv6: do not clear pinet6 field tcp: fix tcp_md5_hash_skb_data() net/802/mrp: fix lockdep splat tcp: gso: do not generate out of order packets bonding: allow TSO being set on bonding master ipv6: fix possible crashes in ip6_cork_release() ip_gre: fix a possible crash in ipgre_err() tcp: xps: fix reordering issues EunBong Song (1): MIPS: Fix build error for crash_dump.c in 3.10-rc1 Ezequiel Garcia (3): ARM: mvebu: Add a ranges entry to translate devbus childs ARM: mvebu: Fix ranges entry on XP GP board serial: 8250_dw: Add valid clk pointer check Fabio Baltieri (5): mfd: abx500-core: Fix sparse warning mfd: ab8500-sysctrl: Fix sparse warning mfd: ab8500-sysctrl: Set sysctrl_dev during probe mfd: ab8500-sysctrl: Always enable pm_power_off handler mfd: ab8500-sysctrl: Let sysctrl driver work without pdata Fabio Estevam (3): fec: Fix inconsistent lock state fec: Invert the order of function calls in fec_restart() staging: imx-drm: imx-tve: Check the return value of 'regulator_enable()' Felix Fietkau (5): ath9k: fix key allocation error handling for powersave keys mac80211: fix spurious RCU warning and update documentation ath9k: fix aggregation stop/flush handling ath9k: fix rate handling/reporting ath9k: fix draining aggregation tid buffers Florian Fainelli (1): broadcom: add include guards to include/linux/brcmphy.h Gabriel de Perthuis (1): btrfs: don't stop searching after encountering the wrong item Geert Uytterhoeven (14): iscsi-target: Make buf param of iscsit_do_crypto_hash_buf() const void * net/ethernet: NET_CALXEDA_XGMAC should depend on HAS_DMA net/ethernet: STMMAC_ETH should depend on HAS_DMA net/wireless: ATH9K should depend on HAS_DMA net/ethernet: ARM_AT91_ETHER should depend on HAS_DMA net/ethernet: MACB should depend on HAS_DMA caif: CAIF_VIRTIO should depend on HAS_DMA mfd: si476x: Do not use binary constants mfd: cros_ec_spi: Use %z to format pointer differences mfd: si476x: Use get_unaligned_be16() for unaligned be16 loads staging: zcache: Fix incorrect module_param_array types net/wireless: ATH9K should depend on HAS_DMA uio: UIO_DMEM_GENIRQ should depend on HAS_DMA fbdev: FB_GOLDFISH should depend on HAS_DMA Gernot Vormayr (1): drivers/block/xsysace.c: fix id with missing port-number Gomella, Andrew (NIH/NHLBI) [F] (1): USB: ftdi_sio: Add support for Newport CONEX motor drivers Grant Likely (1): Merge branch 'for-next' of git://sources.calxeda.com/kernel/linux into HEAD Greg Kroah-Hartman (3): Merge tag 'fixes-for-v3.10-rc2' of git://git.kernel.org/.../balbi/usb into usb-linus driver core: export subsys_virtual_register Merge tag 'iio-fixes-for-3.10a' of git://git.kernel.org/.../jic23/iio into staging-linus Gregory CLEMENT (3): ARM: mvebu: select ARCH_REQUIRE_GPIOLIB for mvebu platform ARM: plat-orion: Fix num_resources and id for ge10 and ge11 ARM: dts: mvebu: Fix wrong the address reg value for the L2-cache node Guenter Roeck (2): hwmon: (nct6775) Do not create non-existing attributes MAINTAINERS: Add myself as maintainer for LM95234 and TMP401 drivers H Hartley Sweeten (1): staging: comedi: ni_labpc: fix build when VIRT_TO_BUS is not defined Haiyang Zhang (1): MAINTAINERS: update Hyper-V file list Hans Schillstrom (1): netfilter: log: netns NULL ptr bug when calling from conntrack Hans-Christian Egtvedt (2): avr32: move NODES_SHIFT into Kconfig and delete numnodes.h avr32: fix relocation check for signed 18-bit offset Hans-Christoph Schemmel (1): qmi_wwan: Added support for Cinterion's PLxx WWAN Interface Heikki Krogerus (1): serial: 8250_dw: add ACPI ID for Intel BayTrail Heiko Abraham (1): Input: egalax_ts - ABS_MT_POSITION_Y not reported well Heiko Carstens (2): s390/disassembler: prevent endless loop in print_fn_code() s390/ftrace: fix mcount adjustment Helge Deller (8): parisc: implement irq stacks - part 2 (v2) parisc: make default cross compiler search more robust (v3) MPILIB: disable usage of floating point registers on parisc parisc: add rp5470 entry to machine database parisc: use PAGE_SHIFT instead of hardcoded value 12 in pacache.S parisc: add additional parisc git tree to MAINTAINERS file parisc: show number of FPE and unaligned access handler calls in /proc/interrupts parisc: fix irq stack on UP and SMP Hema Prathaban (1): staging: vt6656: use free_netdev instead of kfree Himanshu Madhani (2): qlcnic: Fix missing bracket in module parameter. qlcnic: Fix ethtool supported port status for 83xx Ian Abbott (4): staging: comedi: allow buffer resize if previous resize failed staging: comedi: work without HAS_DMA staging: comedi: ni_mio_common: only do counter commands for ni_pcimio staging: comedi: complain if dma buffer allocation not supported Ilan Peer (2): nl80211: Add wdev identifier to some nl80211 notifications iwlwifi: mvm: Always use SCAN_TYPE_FORCED Imre Deak (2): drm/i915: hsw: fix link training for eDP on port-A wait: fix false timeouts when using wait_event_timeout() Ingo Molnar (1): Merge branch 'rcu/doc' of git://git.kernel.org/.../paulmck/linux-rcu into core/urgent Jaccon Bastiaansen (1): ARM: 7720/1: ARM v6/v7 cmpxchg64 shouldn't clear upper 32 bits of the old/new value Jan Kara (1): jbd,jbd2: fix oops in jbd2_journal_put_journal_head() Jan-Simon Möller (1): ACPI: Fix section to __init. Align with usage in acpixf.h Jani Nikula (1): drm/i915: clear the stolen fb before resuming Jarod Wilson (1): drivers/char/random.c: fix priming of last_data Jason Wang (1): virtio_net: enable napi for all possible queues during open Jean Delvare (3): staging: Swap zram and zsmalloc in Kconfig i2c: i801: Document feature bits in modinfo hwmon: (tmp401) Drop redundant safety on cache lifetime Jean-Christophe PLAGNIOL-VILLARD (3): ARM: at91/trivial: fix model name for SAM9X25-EK ARM: at91: rm9200 fix time support ARM: at91: at91sam9n12: move external irq declatation to DT Jeff Layton (1): cifs: only set ops for inodes in I_NEW state Jeff Mahoney (1): hfs: avoid crash in hfs_bnode_create Jeff Moyer (1): aio: fix io_getevents documentation Jiang Liu (1): avr32: fix building warnings caused by redefinitions of HZ Jingoo Han (1): usb: gadget: s3c-hsotg: pass 'struct usb_request *' to usb_gadget_unmap_request() Jiri Kosina (1): random: fix accounting race condition with lockless irq entropy_count update Jiri Olsa (2): perf: Fix EXIT event notification perf: Factor out auxiliary events notification Jiri Pirko (1): macvlan: fix passthru mode race between dev removal and rx path Jiri Slaby (1): TTY: rocket, fix more no-PCI warnings Joachim Eastwood (1): spi/atmel: fix speed_hz check in atmel_spi_transfer() Joern Engel (2): target: removed unused transport_state flag target: close target_put_sess_cmd() vs. core_tmr_abort_task() race Johan Hovold (7): USB: serial: add wait_until_sent operation USB: serial: add generic wait_until_sent implementation USB: ftdi_sio: clean up get_modem_status USB: ftdi_sio: fix chars_in_buffer overhead USB: io_ti: fix chars_in_buffer overhead USB: ti_usb_3410_5052: fix chars_in_buffer overhead USB: serial: clean up chars_in_buffer Johannes Berg (9): cfg80211: fix wiphy_register error path cfg80211: fix WoWLAN wakeup tracing mac80211: fix IEEE80211_SDATA_DISCONNECT_RESUME mac80211: fix HT beacon-based channel switch handling mac80211: use just spin_lock() in ieee80211_get_tkip_p2k() mac80211: fix AP-mode frame matching cfg80211: fix interface down/disconnect state handling mac80211: report deauth to cfg80211 for local state change cfg80211: fix sending WoWLAN TCP wakeup settings Johannes Weiner (1): mm: memcg: remove incorrect VM_BUG_ON for swap cache pages in uncharge John David Anglin (2): parisc: fix SMP races when updating PTE and TLB entries in entry.S parisc: make interrupt and interruption stack allocation reentrant John Stultz (1): time: Revert ALWAYS_USE_PERSISTENT_CLOCK compile time optimizaitons John W. Linville (3): Merge branch 'master' of git://git.kernel.org/.../linville/wireless into for-davem Merge branch 'for-john' of git://git.kernel.org/.../iwlwifi/iwlwifi-fixes Merge branch 'master' of git://git.kernel.org/.../linville/wireless into for-davem Jon Hunter (1): gpio/omap: ensure gpio context is initialised Jon Mason (8): NTB: variable dereferenced before check NTB: Handle 64bit BAR sizes NTB: Link toggle memory leak NTB: reset tx_index on link toggle NTB: Correctly handle receive buffers of the minimal size NTB: memcpy lockup workaround ntb_netdev: remove from list on exit NTB: Multiple NTB client fix Jonathan Corbet (1): dummy-irq: require the user to specify an IRQ number Joonsoo Kim (1): workqueue: correct handling of the pool spin_lock Josef Ahmad (1): i2c: designware: fix RX FIFO overrun Josef Bacik (4): Btrfs: don't null pointer deref on abort Btrfs: remove warn on in free space cache writeout Btrfs: handle running extent ops with skinny metadata Btrfs: make sure roots are assigned before freeing their nodes Joseph Qi (2): ocfs2: unlock rw lock if inode lock failed ocfs2: goto out_unlock if ocfs2_get_clusters_nocache() failed in ocfs2_fiemap() Josh Wu (1): ARM: at91/sama5: fix incorrect PMC pcr div definition K. Y. Srinivasan (1): Drivers: hv: Fix a bug in get_vp_index() Kees Cook (2): drm: refactor call to request_module drm/radeon: check incoming cliprects pointer Kent Yoder (1): drivers/crypto/nx: Fixes for multiple races and issues Kevin Hilman (1): idle: Fix hlt/nohlt command-line handling in new generic idle Konrad Rzeszutek Wilk (11): x86/microcode: Add local mutex to fix physical CPU hot-add deadlock Merge tag 'v3.10-rc1' into stable/for-linus-3.10 xen/tmem: Cleanup. Remove the parts that say temporary. xen/tmem: Move all of the boot and module parameters to the top of the file. xen/tmem: Split out the different module/boot options. xen/tmem: Fix compile warning. xen/tmem: s/disable_// and change the logic. xen/tmem: Remove the boot options and fold them in the tmem.X parameters. xen/tmem: Remove the usage of 'noselfshrink' and use 'tmem.selfshrink' bool instead. xen/tmem: Remove the usage of '[no|]selfballoon' and use 'tmem.selfballooning' bool instead. xen/tmem: Don't use self[ballooning|shrinking] if frontswap is off. Kyle McMartin (1): score: remove redundant kcore_list entries Lachlan McIlroy (1): ext4: limit group search loop for non-extent files Lan Tianyu (2): ACPI / EC: Restart transaction even when the IBF flag set ACPI / AC: Add sleep quirk for Thinkpad e530 Larry Finger (1): rtlwifi: rtl8188ee: Fix warning when building on big-endian systems Lars-Peter Clausen (2): tty: nwpserial: Pass correct pointer to free_irq() drivers/rtc/rtc-pl031.c: pass correct pointer to free_irq() Laurent Pinchart (2): pinctrl: generic: Fix typos and clarify comments pinctrl/lantiq: Free mapping configs for both pin and groups Lee Jones (12): ARM: 7714/1: mmc: mmci: Ensure return value of regulator_enable() is checked ARM: ux500: Remove duplicated assignment of ab8500_platdata ARM: ux500: Fix incorrect DEBUG UART virtual addresses ARM: ux500: Provide device enumeration number suffix for SMSC911x pinctrl: abx500: Specify failed sub-driver by ID instead of driver_data pinctrl: abx500: Rejiggle platform data and DT initialisation staging: ste_rmi4: Suppress 'ignoring return value of ‘regulator_enable()' warning mfd: ab8500-gpadc: Suppress 'ignoring regulator_enable() return value' warning mfd: ab8500-core: Use the correct driver name when enabling gpio/pinctrl mfd: db8500-prcmu: Supply the pdata_size attribute for db8500-thermal mfd: ab8500-core: Pass GPADC compatible string to MFD core serial: pl011: protect attribute read from NULL platform data struct Leonid Yegoshin (1): mm compaction: fix of improper cache flush in migration code Lespiau, Damien (2): drm: Make the HPD status updates debug logs more readable drm: Only print a debug message when the polled connector has changed Li Zhong (6): powerpc: Fix MAX_STACK_TRACE_ENTRIES too low warning again powerpc: Syscall hooks for context tracking subsystem powerpc: Exception hooks for context tracking subsystem powerpc: Exit user context on notify resume powerpc: Use the new schedule_user API on userspace preemption powerpc: select HAVE_CONTEXT_TRACKING for pSeries Libo Chen (4): usb: ehci-s5p: fix memleak when fallback to pdata usb: isp1760-if: fix memleak when platform_get_resource fail usb: ohci: fix goto wrong tag in err case MIPS: Xway: Fix clk leak Linus Lüssing (1): batman-adv: Fix rcu_barrier() miss due to double call_rcu() in TT code Linus Torvalds (72): Merge tag 'fixes-for-linus' of git://git.kernel.org/.../rusty/linux Merge branch 'drm-next' of git://people.freedesktop.org/~airlied/linux Merge tag 'spi-v3.10-rc1' of git://git.kernel.org/.../broonie/spi Merge git://git.kernel.org/.../davem/net Merge branch 'parisc-for-3.10' of git://git.kernel.org/.../deller/parisc-linux Merge tag 'fixes-for-3.10-rc2-tag' of git://git.kernel.org/.../sstabellini/xen Merge branch 'merge' of git://git.kernel.org/.../benh/powerpc Merge branch 'for-3.10-fixes' of git://git.kernel.org/.../tj/wq Merge tag 'ext4_for_linus_stable' of git://git.kernel.org/.../tytso/ext4 Merge branch 'for-linus' of git://git.kernel.org/.../sage/ceph-client Merge branch 'fixes' of git://git.linaro.org/people/rmk/linux-arm Merge branch 'core-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'timers-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'perf-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'x86-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge tag 'trace-fixes-v3.10-rc1' of git://git.kernel.org/.../rostedt/linux-trace Merge branch 'queue' of git://git.kernel.org/.../nab/target-pending Merge branch 'rcu/urgent' of git://git.kernel.org/.../paulmck/linux-rcu Merge branch 'for-3.10-fixes' of git://git.kernel.org/.../tj/wq Merge branch 'ipmi' (minor ipmi fixes from Corey) Merge tag 'ntb-bugfixes-3.10' of git://github.com/jonmason/ntb Merge tag 'pm+acpi-3.10-rc2' of git://git.kernel.org/.../rafael/linux-pm Merge branch 'drm-next' of git://people.freedesktop.org/~airlied/linux Merge tag 'sound-3.10' of git://git.kernel.org/.../tiwai/sound Merge tag 'arm64-fixes' of git://git.kernel.org/.../cmarinas/linux-aarch64 Merge tag 'kmemleak-fixes' of git://git.kernel.org/.../cmarinas/linux-aarch64 Merge branch 'upstream' of git://git.linux-mips.org/pub/scm/ralf/upstream-linus Merge tag 'devicetree-for-linus' of git://git.secretlab.ca/git/linux Merge branch 'devm_no_resource_check' of git://git.kernel.org/.../wsa/linux Merge branch 'for-linus' of git://git.kernel.org/.../mason/linux-btrfs Merge tag 'dm-3.10-fixes' of git://git.kernel.org/.../agk/linux-dm Merge branch 'rc-fixes' of git://git.kernel.org/.../mmarek/kbuild Merge tag 'pinctrl-fixes-v3.10-2' of git://git.kernel.org/.../linusw/linux-pinctrl x86: Fix bit corruption at CPU resume time Merge tag 'hwmon-for-linus' of git://git.kernel.org/.../groeck/linux-staging Merge tag 'stable/for-linus-3.10-rc1-tag' of git://git.kernel.org/.../konrad/xen Linux 3.10-rc2 Merge branch 'drm-fixes' of git://people.freedesktop.org/~airlied/linux Merge branch 'drm-radeon-sun-hainan' of git://people.freedesktop.org/~airlied/linux Merge branch 'for-linus' of git://git.kernel.org/.../s390/linux Merge branch 'i2c/for-current' of git://git.kernel.org/.../wsa/linux Merge branch 'leds-fixes-3.10' of git://git.kernel.org/.../cooloney/linux-leds Merge tag 'fixes-for-linus' of git://git.kernel.org/.../rusty/linux Merge tag 'mfd-fixes-3.10-1' of git://git.kernel.org/.../sameo/mfd-fixes Merge branch 'upstream' of git://git.linux-mips.org/pub/scm/ralf/upstream-linus Merge branch 'for-linus' of git://git.kernel.org/.../egtvedt/linux-avr32 Merge git://git.kernel.org/pub/scm/virt/kvm/kvm Merge tag 'usb-3.10-rc2' of git://git.kernel.org/.../gregkh/usb Merge tag 'char-misc-3.10-rc2' of git://git.kernel.org/.../gregkh/char-misc Merge tag 'driver-core-3.10-rc2' of git://git.kernel.org/.../gregkh/driver-core Merge tag 'staging-3.10-rc2' of git://git.kernel.org/.../gregkh/staging Merge tag 'tty-3.10-rc2' of git://git.kernel.org/.../gregkh/tty Merge branch 'for-linus' of git://git.kernel.org/.../dtor/input Merge tag 'pci-v3.10-fixes-2' of git://git.kernel.org/.../helgaas/pci Merge tag 'gpio-fixes-v3.10-1' of git://git.kernel.org/.../linusw/linux-gpio Merge git://git.kernel.org/.../davem/net Merge branch 'merge' of git://git.kernel.org/.../benh/powerpc Merge git://git.kernel.org/.../steve/gfs2-3.0-fixes Merge branch 'for-next' of git://git.samba.org/sfrench/cifs-2.6 Merge tag 'trace-fixes-v3.10-rc2' of git://git.kernel.org/.../rostedt/linux-trace Merge tag 'fixes-for-linus' of git://git.kernel.org/.../arm/arm-soc Merge branch 'akpm' (incoming from Andrew Morton) Merge branch 'fixes' of git://git.linaro.org/people/rmk/linux-arm Merge tag 'arc-v3.10-fixes' of git://git.kernel.org/.../vgupta/arc Merge branch 'fixes' of git://git.infradead.org/users/vkoul/slave-dma Merge tag 'pm+acpi-3.10-rc3' of git://git.kernel.org/.../rafael/linux-pm Merge tag 'for-v3.10-fixes' of git://git.infradead.org/battery-2.6 Merge tag 'for-linus-v3.10-rc3' of git://oss.sgi.com/xfs/xfs Merge branch 'parisc-for-3.10' of git://git.kernel.org/.../deller/parisc-linux Merge tag 'edac_fixes_for_3.10' of git://git.kernel.org/.../bp/bp Merge tag 'nfs-for-3.10-3' of git://git.linux-nfs.org/projects/trondmy/linux-nfs Linux 3.10-rc3 Linus Walleij (2): mfd: ab8500: Pass AB8500 IRQ to debugfs code by resource ARM: nomadik: fix IRQ assignment for SMC ethernet Lisa Nguyen (3): xen/xenbus: Fixed indentation error in switch case xen/xenbus: Fixed over 80 character limit issue xen: Fixed assignment error in if statement Liu Bo (2): Btrfs: fix off-by-one in fiemap Btrfs: return errno if possible when we fail to allocate memory Lucas Stach (2): clk: tegra: remove USB from clk init table clk: tegra: add ac97 controller clock Ludovic Desroches (1): ARM: at91: dts: request only spi cs-gpios used on sama5d3x cpu module Luiz Capitulino (1): package: Makefile: unbreak binrpm-pkg target Malcolm Priestley (1): staging: vt6656: [bug] Fix missing spin lock in iwctl_siwpower. Manfred Spraul (1): ipc/sem.c: Fix missing wakeups in do_smart_update_queue() Manish Chopra (2): qlcnic: Fix setting MAC address qlcnic: Fix bug in diagnostics test reset recovery path Manjunathappa, Prakash (1): spi/davinci: fix module build error Marc Dietrich (3): staging: nvec: add missing module aliases staging: nvec: implement unregistering of notifiers staging: nvec: cleanup childs on remove Marc Dionne (1): workqueue: Make schedule_work() available again to non GPL modules Marek Lindner (2): batman-adv: check proto length before accessing proto string buffer batman-adv: check return value of pskb_trim_rcsum() Marek Vasut (1): staging: video: imx: Select VIDEOMODE_HELPERS for parallel display Mark Brown (11): Merge branch 'spi/merge' of git://git.secretlab.ca/git/linux into spi-fix-grant Merge remote-tracking branch 'spi/fix/atmel' into spi-linus Merge remote-tracking branch 'spi/fix/grant' into spi-linus ASoC: wm5110: Add missing speaker initialisation ASoC: wm8994: Fix reporting of accessory removal on WM8958 Merge remote-tracking branch 'asoc/fix/arizona' into asoc-linus Merge remote-tracking branch 'asoc/fix/compress' into asoc-linus Merge remote-tracking branch 'asoc/fix/cs42l52' into asoc-linus Merge remote-tracking branch 'asoc/fix/davinci' into asoc-linus Merge remote-tracking branch 'asoc/fix/max98090' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linus Martin Hundebøll (1): batman-adv: Avoid double freeing of bat_counters Martin Michlmayr (1): Kirkwood: Enable PCIe port 1 on QNAP TS-11x/TS-21x Martin Schwidefsky (1): s390/smp: fix cpu re-scan vs. cpu state Masami Hiramatsu (5): tracing/kprobes: Use rcu_dereference_raw for tp->files tracing/kprobes: Fix a sparse warning for incorrect type in assignment tracing/kprobes: Make print_*probe_event static tracing: Return -EBUSY when event_enable_func() fails to get module kprobes/mips: Fix to check double free of insn slot Matthijs Kooijman (2): MIPS: ralink: use the dwc2 driver for the rt305x USB controller staging: dwc2: Fix dma-enabled platform devices using a default dma_mask Matwey V. Kornilov (2): tty: mxser: fix usage of opmode_ioaddr tty: mxser: Fix build warning introduced by dfc7b837c7f9 (Re: linux-next: build warning after merge of the tty.current tree) Maxime Ripard (3): ARM: sunxi: Fix Mini X-plus device tree build ARM: sunxi: select ARCH_REQUIRE_GPIOLIB gpio: mxs: Use set and clear capabilities of the gpio controller Miao Xie (10): Btrfs: don't abort the current transaction if there is no enough space for inode cache Btrfs: don't use global block reservation for inode cache truncation Btrfs: optimize the error handle of use_block_rsv() Btrfs: don't steal the reserved space from the global reserve if their space type is different Btrfs: update the global reserve if it is empty Btrfs: fix accessing a freed tree root Btrfs: fix unprotected root node of the subvolume's inode rb-tree Btrfs: pause the space balance when remounting to R/O Btrfs: remove BUG_ON() in btrfs_read_fs_tree_no_radix() Btrfs: don't invoke btrfs_invalidate_inodes() in the spin lock context Michael Chan (1): tg3: Fix data corruption on 5725 with TSO Michael Ellerman (2): powerpc: Make CONFIG_RTAS_PROC depend on CONFIG_PROC_FS powerpc: Context switch more PMU related SPRs Michael Grzeschik (1): usb: otg: mxs-phy: add missing type to usb_phy Michael Neuling (3): powerpc/perf: Move BHRB code into CONFIG_PPC64 region powerpc/pmu: Fix order of interpreting BHRB target entries powerpc/perf: Fix setting of "to" addresses for BHRB Michael S. Tsirkin (1): virtio_console: fix uapi header Michał Mirosław (1): staging/iio/mxs-lradc: fix preenable for multiple buffers Mika Westerberg (2): i2c: designware: always clear interrupts before enabling them i2c: designware: add Intel BayTrail ACPI ID Nandita Dukkipati (1): tcp: bug fix in proportional rate reduction. Naveen Krishna Chatradhi (1): iio: exynos_adc: fix wrong structure extration in suspend and resume Nicholas Bellinger (5): iscsi-target: Fix NULL pointer dereference in iscsit_send_reject target/rd: Add ramdisk bit for NULLIO operation iscsi-target: Fix typos in RDMAEXTENSIONS macro usage MAINTAINERS: Update target git tree URL target/iblock: Fix WCE=1 + DPOFUA=1 backend WRITE regression Nicolas Ferre (2): ARM: at91/trivial: typo in GEM compatible string net/macb: fix ISR clear-on-write behavior only for some SoC Nicolas Pitre (1): ARM: 7715/1: MCPM: adapt to GIC changes after upstream merge Nicolas Schichan (4): ASoC: cs42l52: microphone bias is controlled by IFACE_CTL2 register. ASoC: cs42l52: fix bogus shifts in "Speaker Volume" and "PCM Mixer Volume" controls. ASoC: cs42l52: fix master playback mute mask. ASoC: cs42l52: fix default value for MASTERA_VOL. Niels Ole Salscheider (2): drm/radeon: Remove superfluous variable drm/radeon: Fix VRAM size calculation for VRAM >= 4GB Nishanth Menon (1): cpufreq: cpufreq-cpu0: defer probe when regulator is not ready Nithin Sujir (3): tg3: Skip powering down function 0 on certain serdes devices tg3: Ensure boot code has completed initialization before accessing hardware tg3: Update version to 3.132 OGAWA Hirofumi (1): fat: fix possible overflow for fat_clusters Oleg Nesterov (1): usermodehelper: check subprocess_info->path != NULL Oliver Neukum (1): USB: reset resume quirk needed by a hub Olof Johansson (10): Merge tag 'fixes-3.10-1' of git://git.infradead.org/users/jcooper/linux into fixes Merge tag 'at91-fixes' of git://github.com/at91linux/linux-at91 into fixes Merge tag 'omap-for-v3.10-rc1/fixes' of git://git.kernel.org/.../tmlind/linux-omap into fixes Merge tag 'fixes-3.10-2' of git://git.infradead.org/users/jcooper/linux into fixes Merge tag 'imx-fixes-3.10' of git://git.linaro.org/people/shawnguo/linux-2.6 into fixes Merge tag 'ux500-arm-soc-v3.10-fixes' of git://git.kernel.org/.../linusw/linux-stericsson into fixes Merge tag 'omap-fixes-a-for-3.10-rc' of git://git.kernel.org/.../pjw/omap-pending into fixes Merge tag 'renesas-boards-marzen-fixes-for-v3.10' of git://git.kernel.org/.../horms/renesas into fixes Merge tag 'at91-fixes' of git://github.com/at91linux/linux-at91 into fixes Merge tag 'sunxi-fixes-for-3.10' of git://github.com/mripard/linux into fixes Or Gerlitz (1): net/mlx4_core: Add missing report on VST and spoof-checking dev caps Pablo Neira Ayuso (3): netfilter: nf_{log,queue}: fix compilation without CONFIG_PROC_FS netfilter: update MAINTAINERS file netfilter: xt_TCPOPTSTRIP: fix possible mangling beyond packet boundary Paolo Bonzini (1): KVM: take over co-maintainership from Marcelo, fix MAINTAINERS entry Paul Bolle (1): usb: phy: remove CONFIG_USB_OTG_UTILS once more Paul E. McKenney (2): kthread: Document ways of reducing OS jitter due to per-CPU kthreads rcu: Fix comparison sense in rcu_needs_cpu() Paul Moore (1): netlabel: improve domain mapping validation Paul Walmsley (1): ARM: OMAP2+: Add missing CONFIG_ARCH_MULTI_V6=y to omap2plus_defconfig Paul Zimmerman (1): staging: dwc2: remove compile warning for USB_DWC2_TRACK_MISSED_SOFS Peter Huewe (1): parisc/superio: Use module_pci_driver to register driver Peter Hurley (1): tty/vt: Fix vc_deallocate() lock order Peter Oberparleiter (1): s390/cio: add channel ID sysfs attribute Petko Manolov (1): drivers: net: usb: rtl8150: concurrent URB bugfix Petri Gynther (1): emac: Fix EMAC soft reset on 460EX/GT Philipp Zabel (1): ARM i.MX6q: Fix periph_clk2_sel and periph2_clk2_sel clocks Ping Cheng (3): Input: wacom - add three new display tablets Input: wacom - add a few new styli for Cintiq series Input: wacom - add an eraser to DTH2242/DTK2241 Rafael J. Wysocki (10): ACPI / PM: Move processor suspend/resume to syscore_ops ACPI / LPSS: register clock device for Lynxpoint DMA properly Merge branch 'pm-cpufreq' Merge branch 'acpi-fixes' Merge branch 'pm-fixes' Merge branch 'pm-cpufreq' Merge branch 'acpi-fixes' ACPI / PM: Allow device power states to be used for CONFIG_PM unset Merge branch 'acpi-fixes' Merge branch 'pm-cpufreq' Rafał Bilski (1): cpufreq / e_powersaver: Fix linker error when ACPI processor is a module Rafał Miłecki (1): bcma: add more core IDs Rahul Sharma (1): of/documentation: move video device bindings to a common place Rajendra Nayak (2): ARM: OMAP2+: hwmod: Fix sidle programming in _enable_sysc()/_idle_sysc() ARM: OMAP2+: hwmod: Add a new flag to handle SIDLE in SWSUP only in active Rajesh Borundia (2): qlcnic: Fix mailbox response handling. qlcnic: Fix validation of link event command. Ralf Baechle (13): MIPS: Rewrite pfn_valid to work in modules, too. Add include dependencies to <linux/printk.h>. MIPS: clock.h: Remove declaration of cpu_wait. MIPS: Consolidate idle loop / WAIT instruction support in a single file. MIPS: Idle: cleaup SMTC idle hook as per Linux coding style. MIPS: Idle: Consistently reformat inline assembler. MIPS: Idle: Make call of function pointer readable. MIPS: Idle: Re-enable irqs at the end of r3081, au1k and loongson2 cpu_wait. MIPS: Idle: Don't call local_irq_disable() in cpu_wait() implementations. MIPS: Idle: Consolidate all declarations in <asm/idle.h>. MIPS: Idle: Do address fiddlery in helper functions. MIPS: Idle: Break r4k_wait into two functions and fix it. mm: Fix virt_to_page() warning Randy Dunlap (7): lp8788-charger: Fix kconfig dependency staging: sep: fix driver build and kconfig lib: make iovec obj instead of lib GFS2: fix DLM depends to fix build errors linux/kernel.h: fix kernel-doc warning auditfilter.c: fix kernel-doc warnings mm/memory_hotplug.c: fix printk format warnings Richard Genoud (1): spi/spi-atmel: BUG: fix doesn' support 16 bits transfers using PIO Rob Herring (1): Merge tag 'tegra-for-3.10-dtc-cpp-chroot-std-headers' of git://git.kernel.org/.../swarren/linux-tegra into for-next Robert Jarzmik (1): usb: phy: Fix NULL pointer exception during usb_get_phy Robert Jennings (1): powerpc: Bring all threads online prior to migration/hibernation Robert P. J. Day (2): Correct typo "supperspeed" to "superspeed". Add a couple kernel-doc lines to prevent warnings. Rony Efraim (1): net/mlx4: Strengthen VLAN tags/priorities enforcement in VST mode Russell King (3): ARM: finally enable IRQ time accounting config NET: mv643xx_eth: avoid lockdep dump on interface down i2c: mv64xxx: work around signals causing I2C transactions to be aborted Rusty Russell (2): lguest: clear cached last cpu when guest_set_pgd() called. Hoist memcpy_fromiovec/memcpy_toiovec into lib/ Ryusuke Konishi (1): nilfs2: fix issue of nilfs_set_page_dirty() for page at EOF boundary Sachin Kamat (12): usb: gadget: atmel_usba_udc: Remove redundant platform_set_drvdata() usb: gadget: bcm63xx_udc: Remove redundant platform_set_drvdata() usb: gadget: dummy_hcd: Remove redundant platform_set_drvdata() usb: gadget: f_uac2: Remove redundant platform_set_drvdata() usb: gadget: imx_udc: Remove redundant platform_set_drvdata() usb: gadget: pxa25x_udc: Remove redundant platform_set_drvdata() usb: gadget: s3c2410_udc: Remove redundant platform_set_drvdata() usb: phy: ab8500-usb: Remove redundant platform_set_drvdata() usb: phy: gpio-vbus-usb: Remove redundant platform_set_drvdata() usb: phy: mv-usb: Remove redundant platform_set_drvdata() usb: phy: mxs-usb: Remove redundant platform_set_drvdata() usb: phy: nop: Remove redundant platform_set_drvdata() Samuel Ortiz (1): mei: bus: Reset event_cb when disabling a device Sanjay Lal (2): KVM/MIPS32: Move include/asm/kvm.h => include/uapi/asm/kvm.h since it is a user visible API. KVM/MIPS32: Wrap calls to gfn_to_pfn() with srcu_read_lock/unlock() Santosh Shilimkar (4): ARM: OMAP2+: hwmod-data: UART IP needs software control to manage sidle modes ARM: OMAP2+: serial: Remove the un-used slave idle hooks SERIAL: OMAP: Remove the slave idle handling from the driver ARM: OMAP2+: hwmod: Remove sysc slave idle and auto idle apis Sarveshwar Bandi (1): be2net: Avoid double insertion of vlan tags. Sasha Levin (2): rcu: Don't allocate bootmem from rcu_init() mac80211_hwsim: correctly register the platform driver Schemmel Hans-Christoph (1): USB: Blacklisted Cinterion's PLxx WWAN Interface Scott Wood (2): powerpc: hard_irq_disable(): Call trace_hardirqs_off after disabling powerpc/booke64: Fix kernel hangs at kernel_dbg_exc Sebastian Hesselbarth (2): ARM: 7711/1: dove: fix Dove cpu type from V7 to PJ4 ARM: plat-orion: add missing ehci include to common.h Sergei Shtylyov (2): <linux/of_platform.h>: fix compilation warnings with DT disabled 3c59x: fix PCI resource management Shahed Shaikh (2): qlcnic: Fix ethtool strings qlcnic: Fix updating netdev->features Shawn Guo (4): net: fec: enable hardware checksum only on imx6q-fec ARM: imx: replicate the diagnostic register of boot cpu into secondary cores ARM: imx: fix typo in gpu3d_shader_sels net: fec: use a more proper compatible string for MVF type device Shlomo Pongratz (1): iscsi-target: Fix processing of OOO commands Shuah Khan (1): PM: Avoid calling kfree() under spinlock in dev_pm_put_subsys_data() Simon Horman (1): ARM: shmobile: marzen: Use error values in usb_power_* Sony Chacko (3): qlcnic: Fix reset recovery after transmit timeout qlcnic: Return proper error codes from probe failure paths qlcnic: remove netdev->trans_start updates within the driver Sridhar Samudrala (1): vxlan: Update vxlan fdb 'used' field after each usage Srinivas Kandagatla (1): ARM: vt8500: Add missing NULL terminator in dt_compat Srivatsa S. Bhat (2): rcu/idle: Wrap cpu-idle poll mode within rcu_idle_enter/exit cpufreq: Preserve sysfs files across suspend/resume Stanislaw Gruszka (2): ath5k: do not reschedule tx_complete_work on stop iwl4965: workaround connection regression on passive channel Stefan Behrens (3): Btrfs: don't allow device replace on RAID5/RAID6 Btrfs: fix possible memory leak in replace_path() Btrfs: explicitly use global_block_rsv for quota_tree Stefano Stabellini (3): xen/arm: initialize pm functions later xen/arm: do not handle VCPUOP_register_vcpu_info failures xen/arm: rename xen_secondary_init and run it on every online cpu Stephen Boyd (1): ARM: smp: Drop RCU_NONIDLE usage in cpu_die() Stephen Warren (2): USB: set device dma_mask without reference to global data drivers/video: implement a simple framebuffer driver Steven Capper (1): ARM: 7727/1: remove the .vm_mm value from gate_vma Steven Rostedt (2): kmemleak: Scan all allocated, writeable and not executable module sections kmemleak: No need for scanning specific module sections Steven Rostedt (Red Hat) (1): tracing: Fix leaks of filter preds Steven Whitehouse (1): GFS2: Fix typo in gfs2_log_end_write loop Sujith Manoharan (4): ath9k: Fix beacon reconfiguration ath9k: Update initvals for AR9565 ath9k: Fix crash on module unload ath9k_hw: Enable manual peak calibration for AR9485 Sukanto Ghosh (1): arm64: mm: Fix operands of clz in __flush_dcache_all Tang Yuantian (1): of/base: release the node correctly in of_parse_phandle_with_args() Tejun Heo (2): workqueue: workqueue_congested() shouldn't translate WORK_CPU_UNBOUND into node number workqueue: don't perform NUMA-aware allocations on offline nodes in wq_numa_init() Teppo Kotilainen (1): usb: option: Add Telewell TW-LTE 4G Thadeu Lima de Souza Cascardo (1): qlge: fix dma map leak when the last chunk is not allocated Theodore Ts'o (1): ext4: revert "ext4: use io_end for multiple bios" Thomas Gleixner (5): Merge branch 'linus' into core/urgent tick: Use zalloc_cpumask_var for allocating offstack cpumasks x86: Fix idle consolidation fallout tick: Cleanup NOHZ per cpu data on cpu down tick: Don't invoke tick_nohz_stop_sched_tick() if the cpu is offline Thomas Petazzoni (2): ARM: mvebu: do not duplicate the mpic alias arm: mvebu: fix the 'ranges' property to handle PCIe Thommy Jakobsson (1): B43: Handle DMA RX descriptor underrun Timo Teräs (2): leds: leds-gpio: reserve gpio before using it xfrm: properly handle invalid states as an error Tirupathi Reddy (1): timer: Don't reinitialize the cpu base lock during CPU_UP_PREPARE Tomas Winkler (1): mei: fix out of array access to me clients array Tomasz Figa (1): drivers/rtc/rtc-max8998.c: check for pdata presence before dereferencing Tony Lindgren (1): ARM: dts: Fix musb interrupt for device tree booting Tony Prisk (1): pinctrl: vt8500: Fix incorrect data in WM8750 pinctrl table Tony Wu (2): MIPS: Fix sibling call handling in get_frame_info MIPS: Extract schedule_mfi info from __schedule Trond Myklebust (4): SUNRPC: Fix a bug in gss_create_upcall SUNRPC: Faster detection if gssd is actually running SUNRPC: Convert auth_gss pipe detection to work in namespaces SUNRPC: Prevent an rpc_task wakeup race Ulf Hansson (1): mfd: db8500-prcmu: Update stored DSI PLL divider value Uwe Kleine-König (1): Documentation/devicetree: make semantic of initrd-end more explicit Vaibhav Hiremath (1): ARM: AM33XX: Add missing .clkdm_name to clkdiv32k_ick clock Vasant Hegde (1): powerpc/rtas_flash: Fix validate_flash buffer overflow issue Ville Syrjälä (3): drm/i915: Fix pipe enabled mask for pipe C in WM calculations drm: Add kernel-doc for drm_fb_helper_funcs->initial_config drm: Remove pointless '-' characters from drm_fb_helper documentation Vineet Gupta (5): ARC: [mm] Prevent stray dcache lines after__sync_icache_dcach() ARC: copy_(to|from)_user() to honor usermode-access permissions ARC: Brown paper bag bug in macro for checking cache color ARC: Use enough bits for determining page's cache color ARC: lazy dcache flush broke gdb in non-aliasing configs Vinod Koul (2): ASoC: wm8994: use the correct pointer to get the control value ASoC: wm8994: check for array index returned Viresh Kumar (12): cpufreq: ARM big LITTLE: Select PM_OPP cpufreq: ARM big LITTLE DT: Return correct transition latency cpufreq: ARM big LITTLE DT: Return CPUFREQ_ETERNAL if clock-latency isn't found cpufreq: ARM big LITTLE: Move cpu_to_cluster() to arm_big_little.h cpufreq: ARM big LITTLE: Improve print message cpufreq: governors: Fix CPUFREQ_GOV_POLICY_{INIT|EXIT} notifiers cpufreq: Issue CPUFREQ_GOV_POLICY_EXIT notifier before dropping policy refcount cpufreq: cpufreq-cpu0: Free parent node for error cases cpufreq: ARM big LITTLE: Fix Kconfig entries cpufreq: Drop rwsem lock around CPUFREQ_GOV_POLICY_EXIT cpufreq: arm_big_little_dt: Register driver only if DT has valid data cpufreq: arm_big_little_dt: Instantiate as platform_driver Vivek Gautam (1): usb: dwc3: Fix compilation break when building with USB_DWC3_DUAL_ROLE=y Wang Shilong (1): Btrfs: fix possible memory leak in the find_parent_nodes() Wang YanQing (1): TTY: Fix tty miss restart after we turn off flow-control Wei WANG (1): USB: usb-stor: realtek_cr: Fix compile error Wei Yang (1): be2net: bug fix on returning an invalid nic descriptor Wei Yongjun (20): hwmon: fix error return code in abituguru_probe() bna: add missing iounmap() on error in bnad_init() cpufreq / intel_pstate: use vzalloc() instead of vmalloc()/memset(0) pinctrl: single: fix error return code in pcs_parse_one_pinctrl_entry() usb: gadget: zero: fix error return code in zero_bind() usb: musb: omap2430: add missing platform_device_put() on error in omap2430_probe() usb: musb: dsps: fix error return code in dsps_create_musb_pdev() usb: gadget: s3c2410_udc: fix error return code in s3c2410_udc_probe() usb: gadget: r8a66597-udc: fix error return code in r8a66597_probe() usb: gadget: m66592-udc: fix error return code in m66592_probe() usb: gadget: fusb300_udc: fix error return code in fusb300_probe() usb: gadget: dummy_hcd: fix error return code in init() usb: gadget: fix error return code in configfs_composite_bind() ARM: OMAP: fix error return code in omap1_system_dma_init() gianfar: add missing iounmap() on error in gianfar_ptp_probe() serial: samsung: add missing platform_driver_unregister() when module exit tty: serial: mpc5xxx: fix error handing in mpc52xx_uart_init() serial: mcf: missing uart_unregister_driver() on error in mcf_init() TTY: ehv_bytechan: add missing platform_driver_unregister() when module exit staging:iio:light:tsl2x7x: fix the error handling in tsl2x7x_probe() Will Deacon (3): arm64: debug: clear mdscr_el1 instead of taking the OS lock arm64: debug: fix mdscr.ss check when enabling debug exceptions ARM: 7729/1: vfp: ensure VFP_arch is non-zero when VFP is not supported Wolfram Sang (36): cpufreq / kirkwood: don't check resource with devm_ioremap_resource drivers/ata: don't check resource with devm_ioremap_resource drivers/char/hw_random: don't check resource with devm_ioremap_resource usb: gadget: don't check resource with devm_ioremap_resource usb: phy: don't check resource with devm_ioremap_resource drivers/cpufreq: don't check resource with devm_ioremap_resource drivers/dma: don't check resource with devm_ioremap_resource drivers/gpio: don't check resource with devm_ioremap_resource drivers/gpu/drm/exynos: don't check resource with devm_ioremap_resource drivers/gpu/host1x/drm: don't check resource with devm_ioremap_resource drivers/i2c/busses: don't check resource with devm_ioremap_resource drivers/memory: don't check resource with devm_ioremap_resource drivers/mfd: don't check resource with devm_ioremap_resource drivers/misc: don't check resource with devm_ioremap_resource drivers/mtd/nand: don't check resource with devm_ioremap_resource drivers/net/ethernet/renesas: don't check resource with devm_ioremap_resource drivers/pinctrl: don't check resource with devm_ioremap_resource drivers/pwm: don't check resource with devm_ioremap_resource drivers/rtc: don't check resource with devm_ioremap_resource drivers/spi: don't check resource with devm_ioremap_resource drivers/staging/dwc2: don't check resource with devm_ioremap_resource drivers/staging/nvec: don't check resource with devm_ioremap_resource drivers/thermal: don't check resource with devm_ioremap_resource drivers/usb/chipidea: don't check resource with devm_ioremap_resource drivers/usb/gadget: don't check resource with devm_ioremap_resource drivers/usb/host: don't check resource with devm_ioremap_resource drivers/usb/phy: don't check resource with devm_ioremap_resource drivers/video/omap2: don't check resource with devm_ioremap_resource drivers/video/omap2/dss: don't check resource with devm_ioremap_resource drivers/w1/masters: don't check resource with devm_ioremap_resource drivers/watchdog: don't check resource with devm_ioremap_resource arch/arm/mach-tegra: don't check resource with devm_ioremap_resource arch/arm/plat-samsung: don't check resource with devm_ioremap_resource arch/mips/lantiq/xway: don't check resource with devm_ioremap_resource sound/soc/fsl: don't check resource with devm_ioremap_resource sound/soc/kirkwood: don't check resource with devm_ioremap_resource Xiao Guangrong (1): mm: mmu_notifier: re-fix freed page still mapped in secondary MMU Xiong Zhou (2): bq27x00: Fix I2C dependency in KConfig staging: android: logger: use kuid_t instead of uid_t Yan, Zheng (1): ext4: fix fio regression Yinghai Lu (1): PCI: acpiphp: Re-enumerate devices when host bridge receives Bus Check Yuval Mintz (1): bnx2x: Fix GSO for 57710/57711 chips Zhang Rui (2): PM: Documentation update for freeze state PM / Documentation: remove inaccurate suspend/hibernate transition lantency statement Zhang Yanfei (1): x86/mm: Add missing comments for initial kernel direct mapping dyoung(a)redhat.com (1): driver core: print sysfs attribute name when warning about bogus permissions françois romieu (1): r8169: fix offloaded tx checksum for small packets. jean-philippe francois (1): ARM: OMAP: fix __init section mismatch for _enable_preprogram nikolay(a)redhat.com (4): bonding: fix set mode race conditions bonding: replace %x with %pI4 for IPv4 addresses bonding: arp_ip_count and arp_targets can be wrong bonding: fix multiple 3ad mode sysfs race conditions salina(a)us.ibm.com (1): Char: lp, protect LPGETSTATUS with port_mutex stephen hemminger (1): 8139cp: reset BQL when ring tx ring cleared wang, biao (1): klist: del waiter from klist_remove_waiters before wakeup waitting process Documentation/devicetree/bindings/net/macb.txt | 2 +- .../{drm/exynos/hdmi.txt => video/exynos_hdmi.txt} | 0 .../hdmiddc.txt => video/exynos_hdmiddc.txt} | 0 .../hdmiphy.txt => video/exynos_hdmiphy.txt} | 0 .../exynos/mixer.txt => video/exynos_mixer.txt} | 0 .../bindings/video/simple-framebuffer.txt | 25 + Documentation/devicetree/usage-model.txt | 8 +- Documentation/kernel-parameters.txt | 21 + Documentation/kernel-per-CPU-kthreads.txt | 202 +++++ Documentation/power/devices.txt | 15 +- Documentation/power/interface.txt | 4 +- Documentation/power/notifiers.txt | 6 +- Documentation/power/states.txt | 30 +- Documentation/rapidio/rapidio.txt | 128 ++- Documentation/rapidio/sysfs.txt | 17 + MAINTAINERS | 43 +- Makefile | 2 +- arch/Kconfig | 3 + arch/arc/boot/dts/abilis_tb100_dvk.dts | 2 +- arch/arc/boot/dts/abilis_tb101_dvk.dts | 2 +- arch/arc/boot/dts/abilis_tb10x.dtsi | 6 +- arch/arc/include/asm/cacheflush.h | 6 +- arch/arc/include/asm/page.h | 9 - arch/arc/include/asm/pgtable.h | 26 +- arch/arc/include/asm/tlb.h | 2 +- arch/arc/mm/cache_arc700.c | 23 +- arch/arc/mm/tlb.c | 3 +- arch/arc/mm/tlbex.S | 6 +- arch/arc/plat-tb10x/tb10x.c | 26 - arch/arm/Kconfig | 3 +- arch/arm/Makefile | 2 +- arch/arm/boot/dts/Makefile | 4 +- arch/arm/boot/dts/armada-370-xp.dtsi | 3 +- arch/arm/boot/dts/armada-370.dtsi | 7 +- arch/arm/boot/dts/armada-xp-gp.dts | 3 + arch/arm/boot/dts/armada-xp-openblocks-ax3-4.dts | 3 + arch/arm/boot/dts/armada-xp.dtsi | 2 +- arch/arm/boot/dts/at91sam9260.dtsi | 2 +- arch/arm/boot/dts/at91sam9n12.dtsi | 1 + arch/arm/boot/dts/at91sam9x25ek.dts | 2 +- arch/arm/boot/dts/omap3.dtsi | 2 +- arch/arm/boot/dts/sama5d3.dtsi | 12 +- arch/arm/boot/dts/sama5d3xcm.dtsi | 4 + arch/arm/boot/dts/ste-nomadik-s8815.dts | 12 +- arch/arm/boot/dts/sun4i-a10-mini-xplus.dts | 4 +- arch/arm/common/mcpm_platsmp.c | 3 - arch/arm/configs/omap1_defconfig | 1 - arch/arm/configs/omap2plus_defconfig | 2 +- arch/arm/configs/tegra_defconfig | 3 +- arch/arm/crypto/sha1-armv4-large.S | 2 +- arch/arm/include/asm/cmpxchg.h | 8 +- arch/arm/include/debug/ux500.S | 6 +- arch/arm/kernel/process.c | 1 - arch/arm/kernel/smp.c | 2 +- arch/arm/mach-at91/at91rm9200_time.c | 7 +- arch/arm/mach-at91/at91sam9n12.c | 6 - arch/arm/mach-at91/include/mach/at91_pmc.h | 6 +- arch/arm/mach-imx/clk-imx6q.c | 11 +- arch/arm/mach-imx/headsmp.S | 12 + arch/arm/mach-imx/platsmp.c | 14 + arch/arm/mach-kirkwood/common.c | 6 - arch/arm/mach-kirkwood/ts219-setup.c | 2 +- arch/arm/mach-mvebu/Kconfig | 1 + arch/arm/mach-mvebu/armada-370-xp.c | 7 - arch/arm/mach-omap1/dma.c | 1 + arch/arm/mach-omap2/cclock33xx_data.c | 26 +- arch/arm/mach-omap2/omap_hwmod.c | 113 +-- arch/arm/mach-omap2/omap_hwmod.h | 7 +- arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c | 3 + arch/arm/mach-omap2/omap_hwmod_33xx_data.c | 6 + arch/arm/mach-omap2/omap_hwmod_3xxx_data.c | 4 + arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 6 +- arch/arm/mach-omap2/serial.c | 31 - arch/arm/mach-orion5x/common.c | 7 - arch/arm/mach-shmobile/board-marzen.c | 6 +- arch/arm/mach-sunxi/Kconfig | 1 + arch/arm/mach-tegra/tegra2_emc.c | 5 - arch/arm/mach-ux500/Kconfig | 1 + arch/arm/mach-ux500/board-mop500.c | 6 +- arch/arm/mach-ux500/cpu-db8500.c | 6 +- arch/arm/mach-ux500/setup.h | 2 +- arch/arm/mach-vt8500/vt8500.c | 1 + arch/arm/plat-orion/common.c | 12 +- arch/arm/plat-orion/include/plat/common.h | 1 + arch/arm/plat-samsung/adc.c | 5 - arch/arm/vfp/entry.S | 2 +- arch/arm/xen/enlighten.c | 33 +- arch/arm64/Kconfig | 2 - arch/arm64/include/asm/assembler.h | 2 +- arch/arm64/kernel/debug-monitors.c | 2 - arch/arm64/kernel/early_printk.c | 5 +- arch/arm64/kernel/setup.c | 12 +- arch/arm64/mm/cache.S | 2 +- arch/arm64/mm/proc.S | 3 +- arch/avr32/Kconfig | 5 + arch/avr32/include/asm/Kbuild | 1 + arch/avr32/include/asm/numnodes.h | 7 - arch/avr32/include/asm/param.h | 9 - arch/avr32/include/uapi/asm/Kbuild | 1 + arch/avr32/include/uapi/asm/param.h | 18 - arch/avr32/kernel/module.c | 2 +- arch/mips/alchemy/board-gpr.c | 1 + arch/mips/alchemy/common/time.c | 1 + arch/mips/ath79/setup.c | 1 + arch/mips/cobalt/reset.c | 1 + arch/mips/configs/db1000_defconfig | 1 - arch/mips/configs/db1235_defconfig | 1 - arch/mips/configs/lemote2f_defconfig | 1 - arch/mips/include/asm/clock.h | 2 - arch/mips/include/asm/idle.h | 23 + arch/mips/include/asm/io.h | 2 +- arch/mips/include/asm/kvm_host.h | 2 +- arch/mips/include/asm/mmu_context.h | 95 +- arch/mips/include/asm/page.h | 17 +- arch/mips/include/asm/processor.h | 1 - arch/mips/include/{ => uapi}/asm/kvm.h | 0 arch/mips/include/uapi/asm/unistd.h | 5 +- arch/mips/kernel/Makefile | 2 +- arch/mips/kernel/cpu-probe.c | 198 ----- arch/mips/kernel/crash_dump.c | 1 + arch/mips/kernel/genex.S | 8 +- arch/mips/kernel/idle.c | 244 +++++ arch/mips/kernel/kprobes.c | 5 +- arch/mips/kernel/proc.c | 1 + arch/mips/kernel/process.c | 53 +- arch/mips/kernel/scall64-64.S | 1 + arch/mips/kernel/smp.c | 1 + arch/mips/kernel/smtc.c | 15 +- arch/mips/kernel/traps.c | 15 +- arch/mips/kvm/kvm_mips_emul.c | 29 +- arch/mips/kvm/kvm_tlb.c | 61 +- arch/mips/lantiq/xway/gptu.c | 6 +- arch/mips/lib/dump_tlb.c | 5 +- arch/mips/lib/r3k_dump_tlb.c | 7 +- arch/mips/loongson/common/reset.c | 1 + arch/mips/loongson1/common/reset.c | 1 + arch/mips/mm/tlb-r3k.c | 20 +- arch/mips/mm/tlb-r4k.c | 2 +- arch/mips/mm/tlb-r8k.c | 2 +- arch/mips/mm/tlbex.c | 79 -- arch/mips/netlogic/xlp/setup.c | 1 + arch/mips/netlogic/xlr/setup.c | 1 + arch/mips/pmcs-msp71xx/msp_prom.c | 2 +- arch/mips/pmcs-msp71xx/msp_setup.c | 1 + arch/mips/ralink/dts/rt3050.dtsi | 10 + arch/mips/ralink/dts/rt3052_eval.dts | 4 + arch/mips/txx9/generic/setup.c | 1 + arch/mips/vr41xx/common/pmu.c | 1 + arch/mips/wrppmc/reset.c | 1 + arch/parisc/Kconfig | 2 +- arch/parisc/Makefile | 21 +- arch/parisc/include/asm/assembly.h | 1 - arch/parisc/include/asm/hardirq.h | 10 +- arch/parisc/include/asm/processor.h | 18 - arch/parisc/kernel/entry.S | 174 ++-- arch/parisc/kernel/hardware.c | 1 + arch/parisc/kernel/irq.c | 122 ++- arch/parisc/kernel/pacache.S | 12 +- arch/parisc/kernel/traps.c | 1 + arch/parisc/kernel/unaligned.c | 3 + arch/parisc/mm/init.c | 4 +- arch/powerpc/Kconfig.debug | 23 + arch/powerpc/configs/ps3_defconfig | 1 - arch/powerpc/include/asm/context_tracking.h | 10 + arch/powerpc/include/asm/firmware.h | 4 +- arch/powerpc/include/asm/hw_irq.h | 5 +- arch/powerpc/include/asm/opal.h | 5 +- arch/powerpc/include/asm/pci-bridge.h | 2 + arch/powerpc/include/asm/pgalloc-64.h | 2 +- arch/powerpc/include/asm/processor.h | 6 + arch/powerpc/include/asm/pte-hash64-64k.h | 2 +- arch/powerpc/include/asm/rtas.h | 2 + arch/powerpc/include/asm/thread_info.h | 7 +- arch/powerpc/include/asm/udbg.h | 1 + arch/powerpc/kernel/asm-offsets.c | 6 + arch/powerpc/kernel/cpu_setup_power.S | 8 +- arch/powerpc/kernel/entry_32.S | 2 - arch/powerpc/kernel/entry_64.S | 33 +- arch/powerpc/kernel/exceptions-64e.S | 8 +- arch/powerpc/kernel/machine_kexec_64.c | 4 + arch/powerpc/kernel/misc_32.S | 11 + arch/powerpc/kernel/misc_64.S | 11 + arch/powerpc/kernel/pci-common.c | 12 +- arch/powerpc/kernel/pci_64.c | 10 + arch/powerpc/kernel/pci_dn.c | 8 + arch/powerpc/kernel/ppc_ksyms.c | 3 +- arch/powerpc/kernel/process.c | 8 + arch/powerpc/kernel/ptrace.c | 5 + arch/powerpc/kernel/rtas.c | 113 +++ arch/powerpc/kernel/rtas_flash.c | 10 +- arch/powerpc/kernel/signal.c | 7 +- arch/powerpc/kernel/traps.c | 80 +- arch/powerpc/kernel/udbg.c | 3 + arch/powerpc/mm/fault.c | 41 +- arch/powerpc/mm/hash_utils_64.c | 36 +- arch/powerpc/mm/init_64.c | 3 +- arch/powerpc/perf/core-book3s.c | 280 +++--- arch/powerpc/platforms/Kconfig | 2 +- arch/powerpc/platforms/powernv/Kconfig | 1 + arch/powerpc/platforms/powernv/opal.c | 30 +- arch/powerpc/platforms/powernv/pci-ioda.c | 36 +- arch/powerpc/platforms/powernv/pci.c | 18 +- arch/powerpc/platforms/powernv/pci.h | 2 + arch/powerpc/platforms/powernv/powernv.h | 2 + arch/powerpc/platforms/powernv/setup.c | 16 +- arch/powerpc/platforms/powernv/smp.c | 62 +- arch/powerpc/platforms/pseries/Kconfig | 1 + arch/powerpc/platforms/pseries/msi.c | 75 +- arch/powerpc/platforms/pseries/suspend.c | 22 + arch/powerpc/platforms/wsp/ics.c | 2 +- arch/powerpc/sysdev/Makefile | 2 + arch/powerpc/sysdev/ehv_pic.c | 2 +- arch/powerpc/sysdev/mpic.c | 2 +- arch/powerpc/sysdev/udbg_memcons.c | 105 +++ arch/powerpc/sysdev/xics/ics-opal.c | 2 +- arch/s390/Kconfig | 1 - arch/s390/include/asm/ftrace.h | 12 +- arch/s390/include/asm/page.h | 20 +- arch/s390/include/asm/pgtable.h | 4 +- arch/s390/kernel/dis.c | 2 + arch/s390/kernel/ftrace.c | 9 +- arch/s390/kernel/mcount.S | 2 + arch/s390/kernel/mcount64.S | 2 + arch/s390/kernel/smp.c | 2 +- arch/s390/mm/pgtable.c | 3 +- arch/score/mm/init.c | 2 - arch/x86/Kconfig | 1 - arch/x86/kernel/head64.c | 2 +- arch/x86/kernel/microcode_intel_early.c | 5 +- arch/x86/kernel/process.c | 5 +- arch/x86/mm/init.c | 19 +- arch/x86/pci/mrst.c | 10 +- drivers/acpi/Makefile | 3 +- drivers/acpi/ac.c | 33 + drivers/acpi/acpi_lpss.c | 26 +- drivers/acpi/csrt.c | 159 ---- drivers/acpi/device_pm.c | 126 +-- drivers/acpi/ec.c | 4 +- drivers/acpi/internal.h | 1 - drivers/acpi/pci_root.c | 4 +- drivers/acpi/processor_driver.c | 8 +- drivers/acpi/processor_idle.c | 29 +- drivers/acpi/scan.c | 4 +- drivers/acpi/video.c | 8 + drivers/acpi/video_detect.c | 8 + drivers/ata/pata_ep93xx.c | 5 - drivers/base/bus.c | 1 + drivers/base/core.c | 6 +- drivers/base/power/common.c | 12 +- drivers/bcma/scan.c | 2 + drivers/block/brd.c | 4 +- drivers/block/rbd.c | 935 ++++++++++++-------- drivers/block/xsysace.c | 3 +- drivers/char/hw_random/mxc-rnga.c | 6 - drivers/char/hw_random/omap-rng.c | 5 - drivers/char/ipmi/ipmi_bt_sm.c | 4 +- drivers/char/ipmi/ipmi_devintf.c | 14 +- drivers/char/ipmi/ipmi_msghandler.c | 3 +- drivers/char/ipmi/ipmi_si_intf.c | 16 +- drivers/char/lp.c | 3 + drivers/char/random.c | 54 +- drivers/char/ttyprintk.c | 2 +- drivers/clk/tegra/clk-tegra20.c | 11 +- drivers/clk/x86/clk-lpt.c | 15 +- drivers/cpufreq/Kconfig | 2 +- drivers/cpufreq/Kconfig.arm | 15 +- drivers/cpufreq/Kconfig.x86 | 2 +- drivers/cpufreq/arm_big_little.c | 7 +- drivers/cpufreq/arm_big_little.h | 5 + drivers/cpufreq/arm_big_little_dt.c | 94 +- drivers/cpufreq/cpufreq-cpu0.c | 27 +- drivers/cpufreq/cpufreq.c | 19 +- drivers/cpufreq/cpufreq_governor.c | 11 +- drivers/cpufreq/cpufreq_governor.h | 1 + drivers/cpufreq/cpufreq_ondemand.c | 1 - drivers/cpufreq/cpufreq_stats.c | 7 +- drivers/cpufreq/intel_pstate.c | 123 +-- drivers/cpufreq/kirkwood-cpufreq.c | 4 - drivers/cpufreq/loongson2_cpufreq.c | 2 + drivers/crypto/nx/nx-aes-cbc.c | 1 + drivers/crypto/nx/nx-aes-ecb.c | 1 + drivers/crypto/nx/nx-aes-gcm.c | 2 +- drivers/crypto/nx/nx-sha256.c | 8 +- drivers/crypto/nx/nx-sha512.c | 7 +- drivers/crypto/nx/nx.c | 38 +- drivers/dma/acpi-dma.c | 172 +++- drivers/dma/tegra20-apb-dma.c | 5 - drivers/edac/amd64_edac_inj.c | 4 +- drivers/gpio/Kconfig | 2 +- drivers/gpio/gpio-langwell.c | 17 +- drivers/gpio/gpio-ml-ioh.c | 3 +- drivers/gpio/gpio-mvebu.c | 5 - drivers/gpio/gpio-mxs.c | 3 +- drivers/gpio/gpio-omap.c | 48 +- drivers/gpio/gpio-pch.c | 3 +- drivers/gpio/gpio-sch.c | 6 +- drivers/gpio/gpio-tegra.c | 5 - drivers/gpio/gpio-viperboard.c | 3 +- drivers/gpu/drm/drm_crtc.c | 5 + drivers/gpu/drm/drm_crtc_helper.c | 27 +- drivers/gpu/drm/drm_drv.c | 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drivers/w1/masters/omap_hdq.c | 5 - drivers/watchdog/ath79_wdt.c | 5 - drivers/watchdog/davinci_wdt.c | 5 - drivers/watchdog/imx2_wdt.c | 5 - drivers/xen/Kconfig | 7 +- drivers/xen/balloon.c | 3 +- drivers/xen/privcmd.c | 2 +- drivers/xen/tmem.c | 87 +- drivers/xen/xen-selfballoon.c | 47 +- drivers/xen/xenbus/xenbus_dev_backend.c | 21 +- fs/aio.c | 7 +- fs/btrfs/backref.c | 3 +- fs/btrfs/check-integrity.c | 2 +- fs/btrfs/ctree.c | 4 +- fs/btrfs/ctree.h | 8 +- fs/btrfs/delayed-ref.h | 1 + fs/btrfs/dev-replace.c | 5 + fs/btrfs/disk-io.c | 52 +- fs/btrfs/extent-tree.c | 94 +- fs/btrfs/extent_io.c | 138 +-- fs/btrfs/extent_io.h | 2 + fs/btrfs/free-space-cache.c | 43 +- fs/btrfs/free-space-cache.h | 2 + fs/btrfs/inode-map.c | 8 +- fs/btrfs/inode.c | 81 +- fs/btrfs/ioctl.c | 10 +- fs/btrfs/raid56.c | 2 +- fs/btrfs/relocation.c | 7 +- fs/btrfs/scrub.c | 10 +- fs/btrfs/super.c | 1 + fs/btrfs/volumes.c | 54 +- fs/btrfs/volumes.h | 20 + fs/cifs/inode.c | 3 +- fs/ext4/ext4.h | 8 +- fs/ext4/extents.c | 9 +- fs/ext4/extents_status.c | 17 +- fs/ext4/extents_status.h | 3 +- fs/ext4/file.c | 4 +- fs/ext4/inode.c | 85 +- fs/ext4/mballoc.c | 6 +- fs/ext4/page-io.c | 121 +-- fs/fat/inode.c | 15 +- fs/gfs2/Kconfig | 2 +- fs/gfs2/lops.c | 2 +- fs/gfs2/quota.c | 4 +- fs/gfs2/rgrp.c | 9 +- fs/hfs/bnode.c | 6 +- fs/nfs/callback_proc.c | 2 +- fs/nfs/callback_xdr.c | 2 +- fs/nfs/nfs4client.c | 2 +- fs/nfs/nfs4proc.c | 2 +- fs/nfs/nfs4session.c | 4 +- fs/nfs/nfs4session.h | 13 +- fs/nfs/nfs4state.c | 15 +- fs/nilfs2/inode.c | 27 +- fs/ocfs2/extent_map.c | 2 +- fs/ocfs2/file.c | 2 +- fs/xfs/xfs_aops.c | 19 + fs/xfs/xfs_attr_leaf.c | 27 +- fs/xfs/xfs_buf.c | 2 +- fs/xfs/xfs_da_btree.c | 7 +- fs/xfs/xfs_dir2_leaf.c | 2 +- fs/xfs/xfs_extfree_item.c | 5 +- fs/xfs/xfs_log_cil.c | 2 +- fs/xfs/xfs_vnodeops.c | 4 +- include/acpi/acpi_bus.h | 40 +- include/acpi/acpiosxf.h | 2 +- include/acpi/processor.h | 10 +- include/drm/drmP.h | 3 +- include/drm/drm_fb_helper.h | 15 +- include/drm/drm_os_linux.h | 9 - include/drm/drm_pciids.h | 6 + include/linux/acpi_dma.h | 4 + include/linux/bcma/bcma.h | 5 +- include/linux/brcmphy.h | 5 + include/linux/journal-head.h | 8 +- include/linux/kernel.h | 5 +- include/linux/kref.h | 33 + include/linux/mfd/abx500/ab8500.h | 2 - include/linux/mlx4/qp.h | 29 +- include/linux/netdevice.h | 11 + include/linux/of_platform.h | 5 +- include/linux/pci-acpi.h | 2 + include/linux/pinctrl/pinconf-generic.h | 12 +- include/linux/platform_data/clk-lpss.h | 5 + include/linux/platform_data/serial-omap.h | 2 - include/linux/printk.h | 1 + include/linux/rio.h | 18 +- include/linux/rio_drv.h | 1 + include/linux/socket.h | 2 - include/linux/spi/spi.h | 4 +- include/linux/time.h | 4 - include/linux/uio.h | 3 + include/linux/usb/gadget.h | 5 +- include/linux/usb/serial.h | 4 + include/linux/vt_kern.h | 2 +- include/linux/wait.h | 16 +- include/net/mac80211.h | 12 +- include/net/netfilter/nf_log.h | 3 +- include/net/netfilter/nfnetlink_log.h | 3 +- include/net/sock.h | 12 + include/target/target_core_base.h | 5 - include/trace/events/ext4.h | 4 +- include/uapi/linux/virtio_console.h | 2 +- ipc/sem.c | 27 +- kernel/auditfilter.c | 3 - kernel/cpu/idle.c | 2 + kernel/events/core.c | 240 ++--- kernel/kmod.c | 5 + kernel/module.c | 21 +- kernel/rcutree_plugin.h | 4 +- kernel/time/Kconfig | 5 - kernel/time/tick-broadcast.c | 10 +- kernel/time/tick-sched.c | 3 +- kernel/timer.c | 2 +- kernel/trace/trace_events.c | 4 +- kernel/trace/trace_events_filter.c | 4 + kernel/trace/trace_kprobe.c | 53 +- kernel/workqueue.c | 19 +- lib/Makefile | 2 +- lib/iovec.c | 53 ++ lib/klist.c | 2 +- lib/mpi/longlong.h | 5 +- mm/huge_memory.c | 7 +- mm/memcontrol.c | 14 +- mm/memory_hotplug.c | 9 +- mm/migrate.c | 2 +- mm/mmu_notifier.c | 79 +- mm/page_alloc.c | 2 +- mm/pagewalk.c | 70 +- net/802/mrp.c | 4 +- net/batman-adv/distributed-arp-table.c | 13 + net/batman-adv/main.c | 19 +- net/batman-adv/network-coding.c | 8 +- net/batman-adv/originator.c | 16 + net/batman-adv/originator.h | 1 + net/batman-adv/soft-interface.c | 1 + net/batman-adv/translation-table.c | 7 +- net/bridge/netfilter/ebt_log.c | 11 +- net/bridge/netfilter/ebt_ulog.c | 18 +- net/ceph/osd_client.c | 5 +- net/core/iovec.c | 50 -- net/core/sock.c | 12 - net/ipv4/ip_gre.c | 3 +- net/ipv4/ip_output.c | 2 +- net/ipv4/netfilter/ipt_ULOG.c | 13 +- net/ipv4/tcp.c | 29 +- net/ipv4/tcp_input.c | 23 +- net/ipv4/tcp_output.c | 10 +- net/ipv6/ip6_gre.c | 2 + net/ipv6/ip6_output.c | 2 +- net/ipv6/tcp_ipv6.c | 12 + net/ipv6/udp.c | 13 +- net/ipv6/udp_impl.h | 2 + net/ipv6/udplite.c | 2 +- net/ipv6/xfrm6_policy.c | 4 +- net/irda/irlap_frame.c | 2 +- net/mac80211/ieee80211_i.h | 1 + net/mac80211/mlme.c | 61 +- net/mac80211/rate.c | 9 +- net/mac80211/rx.c | 3 + net/mac80211/tkip.c | 4 +- net/mac80211/util.c | 7 + net/netfilter/nf_log.c | 7 +- net/netfilter/nfnetlink_log.c | 6 +- net/netfilter/nfnetlink_queue_core.c | 2 + net/netfilter/xt_LOG.c | 13 +- net/netfilter/xt_NFLOG.c | 3 +- net/netfilter/xt_TCPOPTSTRIP.c | 17 +- net/netlabel/netlabel_domainhash.c | 69 ++ net/sunrpc/auth_gss/auth_gss.c | 62 +- net/sunrpc/netns.h | 4 + net/sunrpc/rpc_pipe.c | 5 + net/sunrpc/sched.c | 8 +- net/wireless/core.c | 17 +- net/wireless/nl80211.c | 4 + net/wireless/sme.c | 3 +- net/wireless/trace.h | 23 +- net/xfrm/xfrm_output.c | 1 + scripts/package/Makefile | 2 +- sound/soc/codecs/cs42l52.c | 8 +- sound/soc/codecs/cs42l52.h | 2 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/wm5110.c | 4 +- sound/soc/codecs/wm8994.c | 12 +- sound/soc/davinci/davinci-mcasp.c | 7 +- sound/soc/fsl/imx-ssi.c | 6 - sound/soc/kirkwood/kirkwood-i2s.c | 5 - sound/soc/soc-compress.c | 8 +- tools/perf/scripts/python/net_dropmonitor.py | 39 +- tools/testing/selftests/Makefile | 1 - tools/testing/selftests/soft-dirty/Makefile | 10 - tools/testing/selftests/soft-dirty/soft-dirty.c | 114 --- 841 files changed, 9100 insertions(+), 5717 deletions(-) rename Documentation/devicetree/bindings/{drm/exynos/hdmi.txt => video/exynos_hdmi.txt} (100%) rename Documentation/devicetree/bindings/{drm/exynos/hdmiddc.txt => video/exynos_hdmiddc.txt} (100%) rename Documentation/devicetree/bindings/{drm/exynos/hdmiphy.txt => video/exynos_hdmiphy.txt} (100%) rename Documentation/devicetree/bindings/{drm/exynos/mixer.txt => video/exynos_mixer.txt} (100%) create mode 100644 Documentation/devicetree/bindings/video/simple-framebuffer.txt create mode 100644 Documentation/kernel-per-CPU-kthreads.txt delete mode 100644 arch/avr32/include/asm/numnodes.h delete mode 100644 arch/avr32/include/asm/param.h delete mode 100644 arch/avr32/include/uapi/asm/param.h create mode 100644 arch/mips/include/asm/idle.h rename arch/mips/include/{ => uapi}/asm/kvm.h (100%) create mode 100644 arch/mips/kernel/idle.c create mode 100644 arch/powerpc/include/asm/context_tracking.h create mode 100644 arch/powerpc/sysdev/udbg_memcons.c delete mode 100644 drivers/acpi/csrt.c create mode 100644 drivers/staging/zcache/ramster/ramster-howto.txt create mode 100644 drivers/video/simplefb.c create mode 100644 lib/iovec.c delete mode 100644 tools/testing/selftests/soft-dirty/Makefile delete mode 100644 tools/testing/selftests/soft-dirty/soft-dirty.c

2 1

[alsa-devel] [PATCH 0/6] Convert JZ4740 to dmaengine
by Lars-Peter Clausen 29 May '13

29 May '13

Hi, This series replaces the custom JZ4740 DMA API with a dmaengine driver. This is done in 3 steps: 1) Add a dmaengine driver which wraps the custom JZ4740 DMA API 2) Update all users of the JZ4740 DMA API to use dmaengine instead 3) Remove the custom API and move all direct hardware access to the dmaengine driver. The first two patches in the series also make sure that the clock of the DMA core is enabled. Since the patches in this series depend on each other I'd prefer if they could all go through the DMA tree. - Lars Lars-Peter Clausen (4): dma: Add a jz4740 dmaengine driver MIPS: jz4740: Register jz4740 DMA device ASoC: jz4740: Use the generic dmaengine PCM driver MIPS: jz4740: Remove custom DMA API Maarten ter Huurne (2): MIPS: jz4740: Correct clock gate bit for DMA controller MIPS: jz4740: Acquire and enable DMA controller clock arch/mips/include/asm/mach-jz4740/dma.h | 56 --- arch/mips/include/asm/mach-jz4740/platform.h | 1 + arch/mips/jz4740/Makefile | 2 +- arch/mips/jz4740/board-qi_lb60.c | 1 + arch/mips/jz4740/clock.c | 2 +- arch/mips/jz4740/dma.c | 287 ------------- arch/mips/jz4740/platform.c | 21 + drivers/dma/Kconfig | 6 + drivers/dma/Makefile | 1 + drivers/dma/dma-jz4740.c | 616 +++++++++++++++++++++++++++ sound/soc/jz4740/Kconfig | 1 + sound/soc/jz4740/jz4740-i2s.c | 48 +-- sound/soc/jz4740/jz4740-pcm.c | 310 +------------- sound/soc/jz4740/jz4740-pcm.h | 20 - 14 files changed, 675 insertions(+), 697 deletions(-) delete mode 100644 arch/mips/jz4740/dma.c create mode 100644 drivers/dma/dma-jz4740.c delete mode 100644 sound/soc/jz4740/jz4740-pcm.h -- 1.8.2.1

4 14

[alsa-devel] [PATCH] ASoC: sgtl5000: Fix VAG_POWER enabling/disabling order
by Marek Vasut 28 May '13

28 May '13

The VAG_POWER must be enabled after all other bits in CHIP_ANA_POWER and disabled before any other bit in CHIP_ANA_POWER. See the SGTL5000 datasheet (Table 31, BIT 7, page 42-43). Failing to follow this order will result in ugly loud "POP" noise at the end of playback. To achieve such order, use the _PRE and _POST DAPM widgets to trigger the power_vag_event, where the event type check has to be fixed accordingly as well. Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Dong Aisheng <dong.aisheng(a)linaro.org> Cc: Eric Nelson <eric.nelson(a)boundarydevices.com> Cc: Fabio Estevam <fabio.estevam(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/sgtl5000.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 92bbfec..2068c36 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -151,15 +151,15 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { switch (event) { - case SND_SOC_DAPM_PRE_PMU: + case SND_SOC_DAPM_POST_PMU: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); break; - case SND_SOC_DAPM_POST_PMD: + case SND_SOC_DAPM_PRE_PMD: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, 0); - msleep(400); + mdelay(400); break; default: break; @@ -217,12 +217,11 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { 0, SGTL5000_CHIP_DIG_POWER, 1, 0), - SND_SOC_DAPM_SUPPLY("VAG_POWER", SGTL5000_CHIP_ANA_POWER, 7, 0, - power_vag_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), + + SND_SOC_DAPM_PRE("VAG_POWER_PRE", power_vag_event), + SND_SOC_DAPM_POST("VAG_POWER_POST", power_vag_event), }; /* routes for sgtl5000 */ @@ -230,16 +229,13 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ - {"ADC", NULL, "VAG_POWER"}, {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ - {"DAC", NULL, "VAG_POWER"}, {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ - {"LINE_IN", NULL, "VAG_POWER"}, {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */ {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */ -- 1.7.10.4

3 6

[alsa-devel] [PATCH] ASoC: cs42l52: fix default value for MASTERA_VOL.
by Nicolas Schichan 28 May '13

28 May '13

The default register value for MASTERA_VOL is 0x00, the same as MASTERB_VOL. Signed-off-by: Nicolas Schichan <nschichan(a)freebox.fr> --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 8465c1fa..030f53c 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -86,7 +86,7 @@ static const struct reg_default cs42l52_reg_defaults[] = { { CS42L52_BEEP_VOL, 0x00 }, /* r1D Beep Volume off Time */ { CS42L52_BEEP_TONE_CTL, 0x00 }, /* r1E Beep Tone Cfg. */ { CS42L52_TONE_CTL, 0x00 }, /* r1F Tone Ctl */ - { CS42L52_MASTERA_VOL, 0x88 }, /* r20 Master A Volume */ + { CS42L52_MASTERA_VOL, 0x00 }, /* r20 Master A Volume */ { CS42L52_MASTERB_VOL, 0x00 }, /* r21 Master B Volume */ { CS42L52_HPA_VOL, 0x00 }, /* r22 Headphone A Volume */ { CS42L52_HPB_VOL, 0x00 }, /* r23 Headphone B Volume */ -- 1.7.10.4

3 2

Re: [alsa-devel] [PATCH] ASoC: fsi: fixup sparse errors
by Mark Brown 28 May '13

28 May '13

On Fri, May 24, 2013 at 12:30:07AM -0700, Kuninori Morimoto wrote: > > ping ? Don't top post and don't send contentless pings.

3 4

[alsa-devel] [PATCH] ASoC: ux500: Ensure consistent configuration between DAIs
by Fabio Baltieri 28 May '13

28 May '13

Current implementation of mop500_ab8500 allows for inconsistent sample rate and channel count configuration between the playback and recording interfaces, through in the hardware the two MSP controllers share common clock and frame sync signals. This patch adds the necessary code to ensure that the two device are configure consistently. The check is added at machine driver level, as how to lock DAI configuration depend of the actual hardware implementation. Signed-off-by: Fabio Baltieri <fabio.baltieri(a)linaro.org> --- sound/soc/ux500/mop500_ab8500.c | 35 +++++++++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index 5e0f146..7e923ec 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -16,6 +16,7 @@ #include <linux/device.h> #include <linux/io.h> #include <linux/clk.h> +#include <linux/mutex.h> #include <sound/soc.h> #include <sound/soc-dapm.h> @@ -44,6 +45,12 @@ static unsigned int tx_slots = DEF_TX_SLOTS; static unsigned int rx_slots = DEF_RX_SLOTS; +/* Configuration consistency parameters */ +static DEFINE_MUTEX(mop500_ab8500_params_lock); +static unsigned long mop500_ab8500_usage; +static int mop500_ab8500_rate; +static int mop500_ab8500_channels; + /* Clocks */ static const char * const enum_mclk[] = { "SYSCLK", @@ -231,6 +238,21 @@ static int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, substream->name, substream->number); + /* Ensure configuration consistency between DAIs */ + mutex_lock(&mop500_ab8500_params_lock); + if (mop500_ab8500_usage) { + if (mop500_ab8500_rate != params_rate(params) || + mop500_ab8500_channels != params_channels(params)) { + mutex_unlock(&mop500_ab8500_params_lock); + return -EBUSY; + } + } else { + mop500_ab8500_rate = params_rate(params); + mop500_ab8500_channels = params_channels(params); + } + __set_bit(cpu_dai->id, &mop500_ab8500_usage); + mutex_unlock(&mop500_ab8500_params_lock); + channels = params_channels(params); switch (params_format(params)) { @@ -329,9 +351,22 @@ static int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, return 0; } +static int mop500_ab8500_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + + mutex_lock(&mop500_ab8500_params_lock); + __clear_bit(cpu_dai->id, &mop500_ab8500_usage); + mutex_unlock(&mop500_ab8500_params_lock); + + return 0; +} + struct snd_soc_ops mop500_ab8500_ops[] = { { .hw_params = mop500_ab8500_hw_params, + .hw_free = mop500_ab8500_hw_free, .startup = mop500_ab8500_startup, .shutdown = mop500_ab8500_shutdown, } -- 1.8.2

2 1

[alsa-devel] [PATCH] ARM: dts: wm8994: Add wm8994 regulator support on smdk5250.
by Padmavathi Venna 28 May '13

28 May '13

This patch adds the required regulator supplies and properties for wm8994 codec on smdk5250 board. Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> --- arch/arm/boot/dts/exynos5250-smdk5250.dts | 53 +++++++++++++++++++++++++++- 1 files changed, 51 insertions(+), 2 deletions(-) diff --git a/arch/arm/boot/dts/exynos5250-smdk5250.dts b/arch/arm/boot/dts/exynos5250-smdk5250.dts index f2a025e..42fc79e 100644 --- a/arch/arm/boot/dts/exynos5250-smdk5250.dts +++ b/arch/arm/boot/dts/exynos5250-smdk5250.dts @@ -37,6 +37,46 @@ }; }; + avdd2:fixed-regulator@0 { + compatible = "regulator-fixed"; + regulator-name = "avdd2-supply"; + regulator-min-microvolt = <1800000>; + regulator-max-microvolt = <1800000>; + regulator-always-on; + }; + + cpvdd:fixed-regulator@1 { + compatible = "regulator-fixed"; + regulator-name = "cpvdd-supply"; + regulator-min-microvolt = <1800000>; + regulator-max-microvolt = <1800000>; + regulator-always-on; + }; + + dbvdd:fixed-regulator@2 { + compatible = "regulator-fixed"; + regulator-name = "dbvdd-supply"; + regulator-min-microvolt = <3300000>; + regulator-max-microvolt = <3300000>; + regulator-always-on; + }; + + spkvdd1:fixed-regulator@3 { + compatible = "regulator-fixed"; + regulator-name = "spkvdd1-supply"; + regulator-min-microvolt = <5000000>; + regulator-max-microvolt = <5000000>; + regulator-always-on; + }; + + spkvdd2:fixed-regulator@4 { + compatible = "regulator-fixed"; + regulator-name = "spkvdd2-supply"; + regulator-min-microvolt = <5000000>; + regulator-max-microvolt = <5000000>; + regulator-always-on; + }; + i2c@12C70000 { samsung,i2c-sda-delay = <100>; samsung,i2c-max-bus-freq = <20000>; @@ -47,8 +87,17 @@ }; wm8994: wm8994@1a { - compatible = "wlf,wm8994"; - reg = <0x1a>; + compatible = "wlf,wm8994"; + reg = <0x1a>; + + gpio-controller; + #gpio-cells = <2>; + + AVDD2-supply = <&avdd2>; + CPVDD-supply = <&cpvdd>; + DBVDD-supply = <&dbvdd>; + SPKVDD1-supply = <&spkvdd1>; + SPKVDD2-supply = <&spkvdd2>; }; }; -- 1.7.4.4

2 1

[alsa-devel] [PATCH] ASoC: wm8994: use the correct pointer to get the control value
by Vinod Koul 28 May '13

28 May '13

Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index f1c54af..49b2b00 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1032,7 +1032,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = codec->control_data; + struct wm8994 *control = wm8994->wm8994; int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; int i; int dac; -- 1.7.0.4

2 1

[alsa-devel] [PATCH] ALSA: hda - Enable mic-mute LED on more HP laptops
by Takashi Iwai 28 May '13

28 May '13

The newer HP laptops have SSID 103c:20xx and 103c:21xx, and these usually have the mic-mute LED on Fn-F8. Let's enable it, too. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1d9d642..9b6cb27 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2233,6 +2233,10 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { "HP Folio", STAC_92HD83XXX_HP_MIC_LED), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x1900, "HP", STAC_92HD83XXX_HP_MIC_LED), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x2000, + "HP", STAC_92HD83XXX_HP_MIC_LED), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x2100, + "HP", STAC_92HD83XXX_HP_MIC_LED), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3388, "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3389, -- 1.8.2.3

1 0

[alsa-devel] [PATCH V3 0/4] clk: Samsung: audss: Register audio subsytem clocks using common clk framework
by Padmavathi Venna 28 May '13

28 May '13

Samsung S5PV210 and Exynos SoC has a separate subsystem for audio. This subsystem has a internal clock controller which controls i2s0 and pcm0 clocks. This patch series adds the Samsung audio subsytem clock to the common clock framework and provides the I2S controllers clock information in the dtsi file. This patch series is made based on Kukjin Kim for-next branch Changes since V2: - Removed s5pv210 compatible name from driver as it is not yet supported which is different from Exynos series audio subsystem clock conroller. - Removed clkdev lookup support and added alias names in the i2s0 controller node. Changes since V1: - Reworked on all review comments by Sylwester Nawrocki - Added a header file for all clock indexes as requested by Sylwester - Added different compatible names for s5pv210, exynos4 and exynos5 - Registered the pcm clocks with common clock framework Padmavathi Venna (4): ARM: samsung: use #include for all device trees clk: samsung: register audio subsystem clocks using common clock framework ARM: dts: add Exynos audio subsystem clock controller node ARM: dts: add clock provider information for i2s controllers in Exynos5250 .../bindings/clock/clk-samsung-audss.txt | 64 ++++++++++ arch/arm/boot/dts/exynos4.dtsi | 2 +- arch/arm/boot/dts/exynos4210-origen.dts | 2 +- arch/arm/boot/dts/exynos4210-smdkv310.dts | 2 +- arch/arm/boot/dts/exynos4210-trats.dts | 2 +- arch/arm/boot/dts/exynos4210-universal_c210.dts | 2 +- arch/arm/boot/dts/exynos4210.dtsi | 4 +- arch/arm/boot/dts/exynos4212.dtsi | 2 +- arch/arm/boot/dts/exynos4412-odroidx.dts | 2 +- arch/arm/boot/dts/exynos4412-origen.dts | 2 +- arch/arm/boot/dts/exynos4412-smdk4412.dts | 2 +- arch/arm/boot/dts/exynos4412.dtsi | 2 +- arch/arm/boot/dts/exynos4x12.dtsi | 4 +- arch/arm/boot/dts/exynos5250-arndale.dts | 2 +- arch/arm/boot/dts/exynos5250-smdk5250.dts | 2 +- arch/arm/boot/dts/exynos5250-snow.dts | 4 +- arch/arm/boot/dts/exynos5250.dtsi | 23 +++- arch/arm/boot/dts/exynos5440-sd5v1.dts | 2 +- arch/arm/boot/dts/exynos5440-ssdk5440.dts | 2 +- arch/arm/boot/dts/exynos5440.dtsi | 2 +- arch/arm/boot/dts/s3c2416-smdk2416.dts | 2 +- arch/arm/boot/dts/s3c2416.dtsi | 4 +- arch/arm/boot/dts/s3c24xx.dtsi | 2 +- drivers/clk/samsung/Makefile | 1 + drivers/clk/samsung/clk-samsung-audss.c | 133 ++++++++++++++++++++ include/dt-bindings/clk/samsung-audss-clk.h | 25 ++++ 26 files changed, 269 insertions(+), 27 deletions(-) create mode 100644 Documentation/devicetree/bindings/clock/clk-samsung-audss.txt create mode 100644 drivers/clk/samsung/clk-samsung-audss.c create mode 100644 include/dt-bindings/clk/samsung-audss-clk.h -- 1.7.4.4

1 4

[alsa-devel] Problem opening default device in haswell system using aplay
by Stefano Panella 28 May '13

28 May '13

Hi everyone, sorry if this email has been sent twice... ( I sent it once before subscribing) I am working on some haswell systems running Ubuntu 12.04 and I have a problem if I run aplay -D sysdefault. when I do: cat /proc/asound/pcm I get: 00-03: HDMI 0 : HDMI 0 : playback 1 00-07: HDMI 1 : HDMI 1 : playback 1 00-08: HDMI 2 : HDMI 2 : playback 1 01-00: ALC662 rev3 Analog : ALC662 rev3 Analog : playback 1 : capture 1 01-02: ALC662 rev3 Analog : ALC662 rev3 Analog : capture 1 If I do: aplay -L | grep sysdefault I get: sysdefault:CARD=PCH where PCH is card number 1 (id = 1) on most of my systems (non haswell laptops) the default card is number 0 If I run aplay -D sysdefault I get: ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave aplay: main:682: audio open error: No such file or directory if I strace it I get: open("/dev/snd/pcmC0D0p", O_RDWR|O_NONBLOCK|O_CLOEXEC) = -1 ENOENT (No such file or directory) If I run: aplay -D hw:PCH,0 all works fine... Any idea why hw:PCH,0 works and sysdefault does not? I guess there could be a problem in the mapping of devices in hw systems? Has anybody else observed a similar issue? Or has this been a problem in the past already fixed upstream? If so, where was it fixed (kernel, lib, tools, utils, etc.) ? Thanks for all your help on this. In case this is a real problem I am willing to help debugging it and submit patches. Stefano

3 4

[alsa-devel] [PATCH] Remove an obsolete macro in configure.in
by olivier＠trillion01.com 28 May '13

28 May '13

From: Olivier Langlois <olivier(a)trillion01.com> Replaced AM_CONFIG_HEADER with AC_CONFIG_HEADERS lano1106@whippet2 ~/dev/alsa-lib $ ./gitcompile aclocal: warning: autoconf input should be named 'configure.ac', not 'configure.in' configure.in:56: error: 'AM_CONFIG_HEADER': this macro is obsolete. You should use the 'AC_CONFIG_HEADERS' macro instead. /usr/share/aclocal-1.13/obsolete-err.m4:12: AM_CONFIG_HEADER is expanded from... configure.in:56: the top level autom4te: /usr/bin/m4 failed with exit status: 1 aclocal: error: echo failed with exit status: 1 Signed-off-by: Olivier Langlois <olivier(a)trillion01.com> diff --git a/configure.in b/configure.in index ed3fe04..207e154 100644 --- a/configure.in +++ b/configure.in @@ -53,7 +53,7 @@ CC_NOUNDEFINED dnl Checks for header files. AC_HEADER_STDC -AM_CONFIG_HEADER(include/config.h) +AC_CONFIG_HEADERS(include/config.h) dnl Checks for typedefs, structures, and compiler characteristics. AC_C_CONST -- 1.8.2.3

4 5

[alsa-devel] [PATCH] ASoC: wm8994: Ensure lambda is zeroed for WM8994
by Mark Brown 27 May '13

27 May '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 5f51127..f5eb644 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1,3 +1,4 @@ +#define DEBUG /* * wm8994.c -- WM8994 ALSA SoC Audio driver * @@ -2078,6 +2079,7 @@ static int wm8994_get_fll_config(struct wm8994 *control, struct fll_div *fll, /* Move down to proper range now rounding is done */ fll->k = K / 10; + fll->lambda = 0; pr_debug("N=%x K=%x\n", fll->n, fll->k); break; @@ -3129,9 +3131,7 @@ static int wm8994_codec_suspend(struct snd_soc_codec *codec) static int wm8994_codec_resume(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = wm8994->wm8994; int i, ret; - unsigned int val, mask; for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { if (!wm8994->fll_suspend[i].out) @@ -3986,7 +3986,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case 0: case 1: wm8994->hubs.no_series_update = 1; - wm8994->hubs.no_runtime_dcs = true; + //wm8994->hubs.no_runtime_dcs = true; break; default: break; -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: bells: Hookup DMICs for Bells
by Charles Keepax 27 May '13

27 May '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/bells.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index ceed466..29e2468 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -350,8 +350,16 @@ static struct snd_soc_codec_conf bells_codec_conf[] = { }, }; +static struct snd_soc_dapm_widget bells_widgets[] = { + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + static struct snd_soc_dapm_route bells_routes[] = { { "Sub CLK_SYS", NULL, "OPCLK" }, + + { "DMIC", NULL, "MICBIAS2" }, + { "IN2L", NULL, "DMIC" }, + { "IN2R", NULL, "DMIC" }, }; static struct snd_soc_card bells_cards[] = { @@ -365,6 +373,8 @@ static struct snd_soc_card bells_cards[] = { .late_probe = bells_late_probe, + .dapm_widgets = bells_widgets, + .num_dapm_widgets = ARRAY_SIZE(bells_widgets), .dapm_routes = bells_routes, .num_dapm_routes = ARRAY_SIZE(bells_routes), @@ -383,6 +393,8 @@ static struct snd_soc_card bells_cards[] = { .late_probe = bells_late_probe, + .dapm_widgets = bells_widgets, + .num_dapm_widgets = ARRAY_SIZE(bells_widgets), .dapm_routes = bells_routes, .num_dapm_routes = ARRAY_SIZE(bells_routes), @@ -401,6 +413,8 @@ static struct snd_soc_card bells_cards[] = { .late_probe = bells_late_probe, + .dapm_widgets = bells_widgets, + .num_dapm_widgets = ARRAY_SIZE(bells_widgets), .dapm_routes = bells_routes, .num_dapm_routes = ARRAY_SIZE(bells_routes), -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: wm8994: Fix reporting of accessory removal on WM8958
by Mark Brown 27 May '13

27 May '13

During recent refactoring the code to report removal when MICDET reports an absent microphone was removed, causing problems for systems which rely solely on the MICDET for this functionality. Restore it. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 0bcbd12..5f51127 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3856,6 +3856,11 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) dev_dbg(codec->dev, "Ignoring removed jack\n"); return IRQ_HANDLED; } + } else if (!(reg & WM8958_MICD_STS)) { + snd_soc_jack_report(wm8994->micdet[0].jack, 0, + SND_JACK_MECHANICAL | SND_JACK_HEADSET | + wm8994->btn_mask); + goto out; } if (wm8994->mic_detecting) -- 1.7.10.4

1 0

[alsa-devel] Haswell: Ensuring HDA codec pins refer to physical outputs
by David Henningsson 27 May '13

27 May '13

Hi, I want to take this problem up again, because it's important we get this right. The HDA driver assumes that a codec pin widget node always refers to the same physical output. With Haswell, it seems like this is not guaranteed to be true. I would like to see this fixed on the graphics side. If not, I don't know how to work around it on the audio side. The problems that occur on the audio side are: 1) Some BIOSes set default pin config. E g, if the machine has a single HDMI out, it can set two of the codec pins to "not connected" and let the third remain "jack". As a result, the HDA driver will ignore the two codec pins and only enable the third. As such, HDMI audio will not work correctly, unless it's the third codec pin that is connected to the physical output. 2) Saving and restoring mutes, volumes etc is done on a per-pin basis. E g, imagine that a user has a dual monitor setup and always wants audio output from the left side monitor, and keep the right side monitor silent. If it is not reliable which codec pin refers to which physical output, one day suddenly the sound might come out on the right side monitor instead. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

2 2

[alsa-devel] [PATCH] alsactl init: ca0106 file was omitted
by Gabriel M. Beddingfield 27 May '13

27 May '13

The file alsactl/init/00main includes the line: CARDINFO{driver}=="CA0106", INCLUDE="ca0106", GOTO="init_end" However, the file alsactl/init/ca0106 is not making it into the release tarballs (nor to user installations). This patch adds the ca0106 file to the build. Signed-off-by: Gabriel M. Beddingfield <gabrbedd(a)gmail.com> --- CAVEAT EMPTOR: I found this issue while helping someone in the #alsa channel. I have no way to test this patch. alsactl/init/Makefile.am | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/alsactl/init/Makefile.am b/alsactl/init/Makefile.am index 9e2b83d..64dac41 100644 --- a/alsactl/init/Makefile.am +++ b/alsactl/init/Makefile.am @@ -1,7 +1,7 @@ init_files = \ 00main default help info test \ - hda + hda ca0106 EXTRA_DIST = $(init_files) alsainitdir = $(datadir)/alsa/init alsainit_DATA = $(init_files) -- 1.7.9.5

2 1

[alsa-devel] [PATCH: TinyCompress 1/2] add utils file with rate conversion helper
by Vinod Koul 26 May '13

26 May '13

Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- Android.mk | 2 +- include/tinycompress/tinycompress.h | 2 + makefile.linux | 2 +- utils.c | 96 +++++++++++++++++++++++++++++++++++ 4 files changed, 100 insertions(+), 2 deletions(-) create mode 100644 utils.c diff --git a/Android.mk b/Android.mk index b9c52e2..104dfef 100644 --- a/Android.mk +++ b/Android.mk @@ -2,7 +2,7 @@ LOCAL_PATH:= $(call my-dir) include $(CLEAR_VARS) LOCAL_C_INCLUDES:= $(LOCAL_PATH)/include -LOCAL_SRC_FILES:= compress.c +LOCAL_SRC_FILES:= compress.c utils.c LOCAL_MODULE := libtinycompress LOCAL_SHARED_LIBRARIES:= libcutils libutils LOCAL_MODULE_TAGS := optional diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 9867afc..e947322 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -267,4 +267,6 @@ const char *compress_get_error(struct compress *compress); #define SNDRV_PCM_RATE_176400 (1<<11) /* 176400Hz */ #define SNDRV_PCM_RATE_192000 (1<<12) /* 192000Hz */ +/* utility functions */ +unsigned int compress_get_alsa_rate(unsigned int rate); #endif diff --git a/makefile.linux b/makefile.linux index 60d6c04..677878d 100644 --- a/makefile.linux +++ b/makefile.linux @@ -2,7 +2,7 @@ LIB = libtinycompress BIN = cplay VER = 0.1 LIBSRC = compress.c -SRC = cplay.c +SRC = cplay.c utils.c LIBOBJ = ${LIBSRC:.c=.o} OBJ = ${SRC:.c=.o} diff --git a/utils.c b/utils.c new file mode 100644 index 0000000..fa37f23 --- /dev/null +++ b/utils.c @@ -0,0 +1,96 @@ +/* + * BSD LICENSE + * + * tinycompress utility functions + * Copyright (c) 2011-2013, Intel Corporation + * All rights reserved. + * + * Author: Vinod Koul <vinod.koul(a)intel.com> + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions are met: + * + * Redistributions of source code must retain the above copyright notice, + * this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * Neither the name of Intel Corporation nor the names of its contributors + * may be used to endorse or promote products derived from this software + * without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + * LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF + * THE POSSIBILITY OF SUCH DAMAGE. + * + * LGPL LICENSE + * + * tinycompress utility functions + * Copyright (c) 2011-2013, Intel Corporation + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU Lesser General Public License, + * version 2.1, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public + * License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to + * the Free Software Foundation, Inc., + * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + */ +#include <stdio.h> +#include <stdlib.h> +#include <stdbool.h> +#include <linux/types.h> +#include <sys/time.h> +#define __force +#define __bitwise +#define __user +#include "tinycompress/tinycompress.h" + + +unsigned int compress_get_alsa_rate(unsigned int rate) +{ + switch (rate) { + case 5512: + return SNDRV_PCM_RATE_5512; + case 8000: + return SNDRV_PCM_RATE_8000; + case 11025: + return SNDRV_PCM_RATE_11025; + case 16000: + return SNDRV_PCM_RATE_16000; + case 220500: + return SNDRV_PCM_RATE_22050; + case 32000: + return SNDRV_PCM_RATE_32000; + case 44100: + return SNDRV_PCM_RATE_44100; + case 48000: + return SNDRV_PCM_RATE_48000; + case 64000: + return SNDRV_PCM_RATE_64000; + case 88200: + return SNDRV_PCM_RATE_88200; + case 96000: + return SNDRV_PCM_RATE_96000; + case 176400: + return SNDRV_PCM_RATE_176400; + case 192000: + return SNDRV_PCM_RATE_192000; + default: + return 0; + } +} -- 1.7.0.4

1 2

[alsa-devel] [PATCH: TinyCompress] fix error reporting in tinycompress
by Vinod Koul 26 May '13

26 May '13

From: Lakshmi N Vinnakota <lakshmi.n.vinnakota(a)intel.com> The compress API are expected to return a negative value on an error with errno indicating the type of error. But in some cases, this is not happening. The fix is the oops function always returns -1 and the actual error number is determined by the global variable errno. Signed-off-by: Lakshmi N Vinnakota <lakshmi.n.vinnakota(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- compress.c | 61 ++++++++++++++++++++++++++++++----------------------------- 1 files changed, 31 insertions(+), 30 deletions(-) diff --git a/compress.c b/compress.c index 8753758..7f56c78 100644 --- a/compress.c +++ b/compress.c @@ -102,10 +102,11 @@ static int oops(struct compress *compress, int e, const char *fmt, ...) va_end(ap); sz = strlen(compress->error); - if (errno) - snprintf(compress->error + sz, COMPR_ERR_MAX - sz, - ": %s", strerror(e)); - return e; + snprintf(compress->error + sz, COMPR_ERR_MAX - sz, + ": %s", strerror(e)); + errno = e; + + return -1; } const char *compress_get_error(struct compress *compress) @@ -151,27 +152,27 @@ static bool _is_codec_supported(struct compress *compress, struct compr_config * } } if (codec == false) { - oops(compress, -ENXIO, "this codec is not supported"); + oops(compress, ENXIO, "this codec is not supported"); return false; } if (config->fragment_size < caps->min_fragment_size) { - oops(compress, -EINVAL, "requested fragment size %d is below min supported %d", + oops(compress, EINVAL, "requested fragment size %d is below min supported %d", config->fragment_size, caps->min_fragment_size); return false; } if (config->fragment_size > caps->max_fragment_size) { - oops(compress, -EINVAL, "requested fragment size %d is above max supported %d", + oops(compress, EINVAL, "requested fragment size %d is above max supported %d", config->fragment_size, caps->max_fragment_size); return false; } if (config->fragments < caps->min_fragments) { - oops(compress, -EINVAL, "requested fragments %d are below min supported %d", + oops(compress, EINVAL, "requested fragments %d are below min supported %d", config->fragments, caps->min_fragments); return false; } if (config->fragments > caps->max_fragments) { - oops(compress, -EINVAL, "requested fragments %d are above max supported %d", + oops(compress, EINVAL, "requested fragments %d are above max supported %d", config->fragments, caps->max_fragments); return false; } @@ -219,7 +220,7 @@ struct compress *compress_open(unsigned int card, unsigned int device, char fn[256]; if (!config) { - oops(&bad_compress, -EINVAL, "passed bad config"); + oops(&bad_compress, EINVAL, "passed bad config"); return &bad_compress; } @@ -242,7 +243,7 @@ struct compress *compress_open(unsigned int card, unsigned int device, compress->flags = flags; if (!((flags & COMPRESS_OUT) || (flags & COMPRESS_IN))) { - oops(&bad_compress, -EINVAL, "can't deduce device direction from given flags"); + oops(&bad_compress, EINVAL, "can't deduce device direction from given flags"); goto config_fail; } @@ -315,12 +316,12 @@ int compress_get_hpointer(struct compress *compress, __u64 time; if (!is_compress_ready(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); if (ioctl(compress->fd, SNDRV_COMPRESS_AVAIL, &kavail)) return oops(compress, errno, "cannot get avail"); if (0 == kavail.tstamp.sampling_rate) - return oops(compress, -ENODATA, "sample rate unknown"); + return oops(compress, ENODATA, "sample rate unknown"); *avail = (unsigned int)kavail.avail; time = kavail.tstamp.pcm_io_frames / kavail.tstamp.sampling_rate; tstamp->tv_sec = time; @@ -335,7 +336,7 @@ int compress_get_tstamp(struct compress *compress, struct snd_compr_tstamp ktstamp; if (!is_compress_ready(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); if (ioctl(compress->fd, SNDRV_COMPRESS_TSTAMP, &ktstamp)) return oops(compress, errno, "cannot get tstamp"); @@ -355,9 +356,9 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size const unsigned int frag_size = compress->config->fragment_size; if (!(compress->flags & COMPRESS_IN)) - return oops(compress, -EINVAL, "Invalid flag set"); + return oops(compress, EINVAL, "Invalid flag set"); if (!is_compress_ready(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); fds.fd = compress->fd; fds.events = POLLOUT; @@ -383,7 +384,7 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size continue; } if (fds.revents & POLLERR) { - return oops(compress, -EIO, "poll returned error!"); + return oops(compress, EIO, "poll returned error!"); } } /* write avail bytes */ @@ -415,9 +416,9 @@ int compress_read(struct compress *compress, void *buf, unsigned int size) const unsigned int frag_size = compress->config->fragment_size; if (!(compress->flags & COMPRESS_OUT)) - return oops(compress, -EINVAL, "Invalid flag set"); + return oops(compress, EINVAL, "Invalid flag set"); if (!is_compress_ready(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); fds.fd = compress->fd; fds.events = POLLIN; @@ -440,7 +441,7 @@ int compress_read(struct compress *compress, void *buf, unsigned int size) continue; } if (fds.revents & POLLERR) { - return oops(compress, -EIO, "poll returned error!"); + return oops(compress, EIO, "poll returned error!"); } } /* read avail bytes */ @@ -466,7 +467,7 @@ int compress_read(struct compress *compress, void *buf, unsigned int size) int compress_start(struct compress *compress) { if (!is_compress_ready(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); if (ioctl(compress->fd, SNDRV_COMPRESS_START)) return oops(compress, errno, "cannot start the stream"); compress->running = 1; @@ -477,7 +478,7 @@ int compress_start(struct compress *compress) int compress_stop(struct compress *compress) { if (!is_compress_running(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); if (ioctl(compress->fd, SNDRV_COMPRESS_STOP)) return oops(compress, errno, "cannot stop the stream"); return 0; @@ -486,7 +487,7 @@ int compress_stop(struct compress *compress) int compress_pause(struct compress *compress) { if (!is_compress_running(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); if (ioctl(compress->fd, SNDRV_COMPRESS_PAUSE)) return oops(compress, errno, "cannot pause the stream"); return 0; @@ -502,7 +503,7 @@ int compress_resume(struct compress *compress) int compress_drain(struct compress *compress) { if (!is_compress_running(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); if (ioctl(compress->fd, SNDRV_COMPRESS_DRAIN)) return oops(compress, errno, "cannot drain the stream"); return 0; @@ -511,10 +512,10 @@ int compress_drain(struct compress *compress) int compress_partial_drain(struct compress *compress) { if (!is_compress_running(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); if (!compress->next_track) - return oops(compress, -EPERM, "next track not signalled"); + return oops(compress, EPERM, "next track not signalled"); if (ioctl(compress->fd, SNDRV_COMPRESS_PARTIAL_DRAIN)) return oops(compress, errno, "cannot drain the stream\n"); compress->next_track = 0; @@ -524,10 +525,10 @@ int compress_partial_drain(struct compress *compress) int compress_next_track(struct compress *compress) { if (!is_compress_running(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); if (!compress->gapless_metadata) - return oops(compress, -EPERM, "metadata not set"); + return oops(compress, EPERM, "metadata not set"); if (ioctl(compress->fd, SNDRV_COMPRESS_NEXT_TRACK)) return oops(compress, errno, "cannot set next track\n"); compress->next_track = 1; @@ -542,14 +543,14 @@ int compress_set_gapless_metadata(struct compress *compress, int version; if (!is_compress_ready(compress)) - return oops(compress, -ENODEV, "device not ready"); + return oops(compress, ENODEV, "device not ready"); version = get_compress_version(compress); if (version <= 0) return -1; if (version < SNDRV_PROTOCOL_VERSION(0, 1, 1)) - return oops(compress, -ENXIO, "gapless apis not supported in kernel"); + return oops(compress, ENXIO, "gapless apis not supported in kernel"); metadata.key = SNDRV_COMPRESS_ENCODER_PADDING; metadata.value[0] = mdata->encoder_padding; -- 1.7.0.4

1 1

[alsa-devel] How to specify data format if playback/capture were different ?
by Kuninori Morimoto 25 May '13

25 May '13

Hi Mark, and ALSA ML Now, I'm creating ak4554 AD/DA. My question is about data format. If my understand is correct, we can specify cpu/codec deivce data format by using snd_soc_dai_set_fmt(dai, SND_SOC_DAIFMT_xxx); But, ak4554 case, data formats are playback : SND_SOC_DAIFMT_RIGHT_J capture : SND_SOC_DAIFMT_LEFT_J and, it can't exchange this data format. CPU driver should set these information for ak4554 when playback/capture. Then, how to specify these data format on ALSA SoC ? Best regards --- Kuninori Morimoto

3 8

[alsa-devel] [PATCH 0/6] More code cleanup for ASoC ux500
by Fabio Baltieri 25 May '13

25 May '13

Hi Mark, this series contains more various code cleanup patches that I put together while working on other stuff this drivers. The first patch is actually a resend, as Linus Walleij agreed to drop that code in the original thread (https://lkml.org/lkml/2013/5/17/487). The others are just a bunch of sparse fixes and generic code cleanup and dead code removal. Thanks, Fabio Fabio Baltieri (6): ASoC: ux500: Drop pinctrl sleep support ASoC: ab8500-codec: Move codec ops on a separate structure ASoC: ux500: Drop dangling struct i2s_controller ASoC: ux500: Drop unused code from msp headers ASoC: ux500: Add missing mop500_ab8500.h include ASoC: ux500: Drop redundant msp id enumerations sound/soc/codecs/ab8500-codec.c | 19 ++++------- sound/soc/ux500/mop500_ab8500.c | 1 + sound/soc/ux500/ux500_msp_dai.h | 2 -- sound/soc/ux500/ux500_msp_i2s.c | 75 ++--------------------------------------- sound/soc/ux500/ux500_msp_i2s.h | 59 ++------------------------------ 5 files changed, 12 insertions(+), 144 deletions(-) -- 1.8.2

3 9

[alsa-devel] hdmi output (switching sample format/rates)
by Sander Jansen 25 May '13

25 May '13

In my audio application when I sometimes need to switch to a different sample format and/or rate, I usually leave the audio device open and merely reconfigure the device to the new parameters. This works with all the regular pcm devices, but it fails when I use the "hdmi" device. It seemed to just silently fail and no audio output is heard. As far as I know, no error condition is triggered (but I'll admit, my knowledge of the alsa api is rather limited). My question, Is this a bug or is it supposed to work this way and if so are there ways to detect whether a devices needs to be reopened or not. Thanks, Sander

2 2

[alsa-devel] [PATCH v2] ASoC: codecs: adau1701: add DT bindings
by Daniel Mack 25 May '13

25 May '13

Apart from pure matching, the bindings also support setting the the reset gpio line. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- Lars, thanks for the quick review. Here's v2 of that patch. .../devicetree/bindings/sound/adi,adau1701.txt | 23 ++++++++++++++ sound/soc/codecs/adau1701.c | 35 +++++++++++++++++++++- 2 files changed, 57 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau1701.txt diff --git a/Documentation/devicetree/bindings/sound/adi,adau1701.txt b/Documentation/devicetree/bindings/sound/adi,adau1701.txt new file mode 100644 index 0000000..3afeda7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau1701.txt @@ -0,0 +1,23 @@ +Analog Devices ADAU1701 + +Required properties: + + - compatible: Should contain "adi,adau1701" + - reg: The i2c address. Value depends on the state of ADDR0 + and ADDR1, as wired in hardware. + +Optional properties: + + - reset-gpio: A GPIO spec to define which pin is connected to the + chip's !RESET pin. If specified, the driver will + assert a hardware reset at probe time. + +Examples: + + i2c_bus { + adau1701@34 { + compatible = "adi,adau1701"; + reg = <0x34>; + reset-gpio = <&gpio 23 0>; + }; + }; diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 95e1677..3fc1763 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -13,6 +13,9 @@ #include <linux/i2c.h> #include <linux/delay.h> #include <linux/slab.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -452,6 +455,14 @@ static struct snd_soc_dai_driver adau1701_dai = { .symmetric_rates = 1, }; +#ifdef CONFIG_OF +static const struct of_device_id adau1701_dt_ids[] = { + { .compatible = "adi,adau1701", }, + { } +}; +MODULE_DEVICE_TABLE(of, adau1701_dt_ids); +#endif + static int adau1701_probe(struct snd_soc_codec *codec) { int ret; @@ -494,12 +505,33 @@ static int adau1701_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct adau1701 *adau1701; + struct device *dev = &client->dev; + int gpio_nreset = -EINVAL; int ret; - adau1701 = devm_kzalloc(&client->dev, sizeof(*adau1701), GFP_KERNEL); + adau1701 = devm_kzalloc(dev, sizeof(*adau1701), GFP_KERNEL); if (!adau1701) return -ENOMEM; + if (dev->of_node) { + gpio_nreset = of_get_named_gpio(dev->of_node, "reset-gpio", 0); + if (gpio_nreset < 0 && gpio_nreset != -ENOENT) + return gpio_nreset; + } + + if (gpio_is_valid(gpio_nreset)) { + ret = devm_gpio_request_one(dev, gpio_nreset, GPIOF_OUT_INIT_LOW, + "ADAU1701 Reset"); + if (ret < 0) + return ret; + + /* minimum reset time is 20ns */ + udelay(1); + gpio_set_value(gpio_nreset, 1); + /* power-up time may be as long as 85ms */ + mdelay(85); + } + i2c_set_clientdata(client, adau1701); ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv, &adau1701_dai, 1); @@ -522,6 +554,7 @@ static struct i2c_driver adau1701_i2c_driver = { .driver = { .name = "adau1701", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(adau1701_dt_ids), }, .probe = adau1701_i2c_probe, .remove = adau1701_i2c_remove, -- 1.8.1.4

3 2

[alsa-devel] [PATCH v2] ASoC: Add ssm2518 support
by Lars-Peter Clausen 25 May '13

25 May '13

This patch adds a ASoC CODEC driver for the SSM2516. The SSM2516 is a stereo Class-D audio amplifier with an I2S interface for audio in and a built-in dynamic range control processor. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Power off the codec in the I2C probe function * Allow a sysclk of 0 which won't but any constraints on the supported sample rates. This can be used by systems which are able to change the sysclk on demand. * Remove {} around multi-line single statements --- .../devicetree/bindings/sound/ssm2518.txt | 20 + include/linux/platform_data/ssm2518.h | 22 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2518.c | 856 +++++++++++++++++++++ sound/soc/codecs/ssm2518.h | 20 + 6 files changed, 924 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ssm2518.txt create mode 100644 include/linux/platform_data/ssm2518.h create mode 100644 sound/soc/codecs/ssm2518.c create mode 100644 sound/soc/codecs/ssm2518.h diff --git a/Documentation/devicetree/bindings/sound/ssm2518.txt b/Documentation/devicetree/bindings/sound/ssm2518.txt new file mode 100644 index 0000000..59381a7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ssm2518.txt @@ -0,0 +1,20 @@ +SSM2518 audio amplifier + +This device supports I2C only. + +Required properties: + - compatible : Must be "adi,ssm2518" + - reg : the I2C address of the device. This will either be 0x34 (ADDR pin low) + or 0x35 (ADDR pin high) + +Optional properties: + - gpios : GPIO connected to the nSD pin. If the property is not present it is + assumed that the nSD pin is hardwired to always on. + +Example: + + ssm2518: ssm2518@34 { + compatible = "adi,ssm2518"; + reg = <0x34>; + gpios = <&gpio 5 0>; + }; diff --git a/include/linux/platform_data/ssm2518.h b/include/linux/platform_data/ssm2518.h new file mode 100644 index 0000000..9a8e3ea --- /dev/null +++ b/include/linux/platform_data/ssm2518.h @@ -0,0 +1,22 @@ +/* + * SSM2518 amplifier audio driver + * + * Copyright 2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#ifndef __LINUX_PLATFORM_DATA_SSM2518_H__ +#define __LINUX_PLATFORM_DATA_SSM2518_H__ + +/** + * struct ssm2518_platform_data - Platform data for the ssm2518 driver + * @enable_gpio: GPIO connected to the nSD pin. Set to -1 if the nSD pin is + * hardwired. + */ +struct ssm2518_platform_data { + int enable_gpio; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00..d76609a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -60,6 +60,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF + select SND_SOC_SSM2518 if I2C select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI select SND_SOC_STA32X if I2C select SND_SOC_STA529 if I2C @@ -313,6 +314,9 @@ config SND_SOC_SN95031 config SND_SOC_SPDIF tristate +config SND_SOC_SSM2518 + tristate + config SND_SOC_SSM2602 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dae0aa6..7204c24 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -52,6 +52,7 @@ snd-soc-si476x-objs := si476x.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o +snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o snd-soc-sta529-objs := sta529.o @@ -176,6 +177,7 @@ obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o +obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c new file mode 100644 index 0000000..3139a1b --- /dev/null +++ b/sound/soc/codecs/ssm2518.c @@ -0,0 +1,856 @@ +/* + * SSM2518 amplifier audio driver + * + * Copyright 2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/platform_data/ssm2518.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "ssm2518.h" + +#define SSM2518_REG_POWER1 0x00 +#define SSM2518_REG_CLOCK 0x01 +#define SSM2518_REG_SAI_CTRL1 0x02 +#define SSM2518_REG_SAI_CTRL2 0x03 +#define SSM2518_REG_CHAN_MAP 0x04 +#define SSM2518_REG_LEFT_VOL 0x05 +#define SSM2518_REG_RIGHT_VOL 0x06 +#define SSM2518_REG_MUTE_CTRL 0x07 +#define SSM2518_REG_FAULT_CTRL 0x08 +#define SSM2518_REG_POWER2 0x09 +#define SSM2518_REG_DRC_1 0x0a +#define SSM2518_REG_DRC_2 0x0b +#define SSM2518_REG_DRC_3 0x0c +#define SSM2518_REG_DRC_4 0x0d +#define SSM2518_REG_DRC_5 0x0e +#define SSM2518_REG_DRC_6 0x0f +#define SSM2518_REG_DRC_7 0x10 +#define SSM2518_REG_DRC_8 0x11 +#define SSM2518_REG_DRC_9 0x12 + +#define SSM2518_POWER1_RESET BIT(7) +#define SSM2518_POWER1_NO_BCLK BIT(5) +#define SSM2518_POWER1_MCS_MASK (0xf << 1) +#define SSM2518_POWER1_MCS_64FS (0x0 << 1) +#define SSM2518_POWER1_MCS_128FS (0x1 << 1) +#define SSM2518_POWER1_MCS_256FS (0x2 << 1) +#define SSM2518_POWER1_MCS_384FS (0x3 << 1) +#define SSM2518_POWER1_MCS_512FS (0x4 << 1) +#define SSM2518_POWER1_MCS_768FS (0x5 << 1) +#define SSM2518_POWER1_MCS_100FS (0x6 << 1) +#define SSM2518_POWER1_MCS_200FS (0x7 << 1) +#define SSM2518_POWER1_MCS_400FS (0x8 << 1) +#define SSM2518_POWER1_SPWDN BIT(0) + +#define SSM2518_CLOCK_ASR BIT(0) + +#define SSM2518_SAI_CTRL1_FMT_MASK (0x3 << 5) +#define SSM2518_SAI_CTRL1_FMT_I2S (0x0 << 5) +#define SSM2518_SAI_CTRL1_FMT_LJ (0x1 << 5) +#define SSM2518_SAI_CTRL1_FMT_RJ_24BIT (0x2 << 5) +#define SSM2518_SAI_CTRL1_FMT_RJ_16BIT (0x3 << 5) + +#define SSM2518_SAI_CTRL1_SAI_MASK (0x7 << 2) +#define SSM2518_SAI_CTRL1_SAI_I2S (0x0 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_2 (0x1 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_4 (0x2 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_8 (0x3 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_16 (0x4 << 2) +#define SSM2518_SAI_CTRL1_SAI_MONO (0x5 << 2) + +#define SSM2518_SAI_CTRL1_FS_MASK (0x3) +#define SSM2518_SAI_CTRL1_FS_8000_12000 (0x0) +#define SSM2518_SAI_CTRL1_FS_16000_24000 (0x1) +#define SSM2518_SAI_CTRL1_FS_32000_48000 (0x2) +#define SSM2518_SAI_CTRL1_FS_64000_96000 (0x3) + +#define SSM2518_SAI_CTRL2_BCLK_INTERAL BIT(7) +#define SSM2518_SAI_CTRL2_LRCLK_PULSE BIT(6) +#define SSM2518_SAI_CTRL2_LRCLK_INVERT BIT(5) +#define SSM2518_SAI_CTRL2_MSB BIT(4) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_MASK (0x3 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_32 (0x0 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_24 (0x1 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_16 (0x2 << 2) +#define SSM2518_SAI_CTRL2_BCLK_INVERT BIT(1) + +#define SSM2518_CHAN_MAP_RIGHT_SLOT_OFFSET 4 +#define SSM2518_CHAN_MAP_RIGHT_SLOT_MASK 0xf0 +#define SSM2518_CHAN_MAP_LEFT_SLOT_OFFSET 0 +#define SSM2518_CHAN_MAP_LEFT_SLOT_MASK 0x0f + +#define SSM2518_MUTE_CTRL_ANA_GAIN BIT(5) +#define SSM2518_MUTE_CTRL_MUTE_MASTER BIT(0) + +#define SSM2518_POWER2_APWDN BIT(0) + +#define SSM2518_DAC_MUTE BIT(6) +#define SSM2518_DAC_FS_MASK 0x07 +#define SSM2518_DAC_FS_8000 0x00 +#define SSM2518_DAC_FS_16000 0x01 +#define SSM2518_DAC_FS_32000 0x02 +#define SSM2518_DAC_FS_64000 0x03 +#define SSM2518_DAC_FS_128000 0x04 + +struct ssm2518 { + struct regmap *regmap; + bool right_j; + + unsigned int sysclk; + const struct snd_pcm_hw_constraint_list *constraints; + + int enable_gpio; +}; + +static const struct reg_default ssm2518_reg_defaults[] = { + { 0x00, 0x05 }, + { 0x01, 0x00 }, + { 0x02, 0x02 }, + { 0x03, 0x00 }, + { 0x04, 0x10 }, + { 0x05, 0x40 }, + { 0x06, 0x40 }, + { 0x07, 0x81 }, + { 0x08, 0x0c }, + { 0x09, 0x99 }, + { 0x0a, 0x7c }, + { 0x0b, 0x5b }, + { 0x0c, 0x57 }, + { 0x0d, 0x89 }, + { 0x0e, 0x8c }, + { 0x0f, 0x77 }, + { 0x10, 0x26 }, + { 0x11, 0x1c }, + { 0x12, 0x97 }, +}; + +static const DECLARE_TLV_DB_MINMAX_MUTE(ssm2518_vol_tlv, -7125, 2400); +static const DECLARE_TLV_DB_SCALE(ssm2518_compressor_tlv, -3400, 200, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_expander_tlv, -8100, 300, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_noise_gate_tlv, -9600, 300, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_post_drc_tlv, -2400, 300, 0); + +static const DECLARE_TLV_DB_RANGE(ssm2518_limiter_tlv, + 0, 7, TLV_DB_SCALE_ITEM(-2200, 200, 0), + 7, 15, TLV_DB_SCALE_ITEM(-800, 100, 0), +); + +static const char * const ssm2518_drc_peak_detector_attack_time_text[] = { + "0 ms", "0.1 ms", "0.19 ms", "0.37 ms", "0.75 ms", "1.5 ms", "3 ms", + "6 ms", "12 ms", "24 ms", "48 ms", "96 ms", "192 ms", "384 ms", + "768 ms", "1536 ms", +}; + +static const char * const ssm2518_drc_peak_detector_release_time_text[] = { + "0 ms", "1.5 ms", "3 ms", "6 ms", "12 ms", "24 ms", "48 ms", "96 ms", + "192 ms", "384 ms", "768 ms", "1536 ms", "3072 ms", "6144 ms", + "12288 ms", "24576 ms" +}; + +static const char * const ssm2518_drc_hold_time_text[] = { + "0 ms", "0.67 ms", "1.33 ms", "2.67 ms", "5.33 ms", "10.66 ms", + "21.32 ms", "42.64 ms", "85.28 ms", "170.56 ms", "341.12 ms", + "682.24 ms", "1364 ms", +}; + +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_attack_time_enum, + SSM2518_REG_DRC_2, 4, ssm2518_drc_peak_detector_attack_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_release_time_enum, + SSM2518_REG_DRC_2, 0, ssm2518_drc_peak_detector_release_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_attack_time_enum, + SSM2518_REG_DRC_6, 4, ssm2518_drc_peak_detector_attack_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_decay_time_enum, + SSM2518_REG_DRC_6, 0, ssm2518_drc_peak_detector_release_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_hold_time_enum, + SSM2518_REG_DRC_7, 4, ssm2518_drc_hold_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_noise_gate_hold_time_enum, + SSM2518_REG_DRC_7, 0, ssm2518_drc_hold_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_rms_averaging_time_enum, + SSM2518_REG_DRC_9, 0, ssm2518_drc_peak_detector_release_time_text); + +static const struct snd_kcontrol_new ssm2518_snd_controls[] = { + SOC_SINGLE("Playback De-emphasis Switch", SSM2518_REG_MUTE_CTRL, + 4, 1, 0), + SOC_DOUBLE_R_TLV("Master Playback Volume", SSM2518_REG_LEFT_VOL, + SSM2518_REG_RIGHT_VOL, 0, 0xff, 1, ssm2518_vol_tlv), + SOC_DOUBLE("Master Playback Switch", SSM2518_REG_MUTE_CTRL, 2, 1, 1, 1), + + SOC_SINGLE("Amp Low Power Mode Switch", SSM2518_REG_POWER2, 4, 1, 0), + SOC_SINGLE("DAC Low Power Mode Switch", SSM2518_REG_POWER2, 3, 1, 0), + + SOC_SINGLE("DRC Limiter Switch", SSM2518_REG_DRC_1, 5, 1, 0), + SOC_SINGLE("DRC Compressor Switch", SSM2518_REG_DRC_1, 4, 1, 0), + SOC_SINGLE("DRC Expander Switch", SSM2518_REG_DRC_1, 3, 1, 0), + SOC_SINGLE("DRC Noise Gate Switch", SSM2518_REG_DRC_1, 2, 1, 0), + SOC_DOUBLE("DRC Switch", SSM2518_REG_DRC_1, 0, 1, 1, 0), + + SOC_SINGLE_TLV("DRC Limiter Threshold Volume", + SSM2518_REG_DRC_3, 4, 15, 1, ssm2518_limiter_tlv), + SOC_SINGLE_TLV("DRC Compressor Lower Threshold Volume", + SSM2518_REG_DRC_3, 0, 15, 1, ssm2518_compressor_tlv), + SOC_SINGLE_TLV("DRC Expander Upper Threshold Volume", SSM2518_REG_DRC_4, + 4, 15, 1, ssm2518_expander_tlv), + SOC_SINGLE_TLV("DRC Noise Gate Threshold Volume", + SSM2518_REG_DRC_4, 0, 15, 1, ssm2518_noise_gate_tlv), + SOC_SINGLE_TLV("DRC Upper Output Threshold Volume", + SSM2518_REG_DRC_5, 4, 15, 1, ssm2518_limiter_tlv), + SOC_SINGLE_TLV("DRC Lower Output Threshold Volume", + SSM2518_REG_DRC_5, 0, 15, 1, ssm2518_noise_gate_tlv), + SOC_SINGLE_TLV("DRC Post Volume", SSM2518_REG_DRC_8, + 2, 15, 1, ssm2518_post_drc_tlv), + + SOC_ENUM("DRC Peak Detector Attack Time", + ssm2518_drc_peak_detector_attack_time_enum), + SOC_ENUM("DRC Peak Detector Release Time", + ssm2518_drc_peak_detector_release_time_enum), + SOC_ENUM("DRC Attack Time", ssm2518_drc_attack_time_enum), + SOC_ENUM("DRC Decay Time", ssm2518_drc_decay_time_enum), + SOC_ENUM("DRC Hold Time", ssm2518_drc_hold_time_enum), + SOC_ENUM("DRC Noise Gate Hold Time", + ssm2518_drc_noise_gate_hold_time_enum), + SOC_ENUM("DRC RMS Averaging Time", ssm2518_drc_rms_averaging_time_enum), +}; + +static const struct snd_soc_dapm_widget ssm2518_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DACL", "HiFi Playback", SSM2518_REG_POWER2, 1, 1), + SND_SOC_DAPM_DAC("DACR", "HiFi Playback", SSM2518_REG_POWER2, 2, 1), + + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), +}; + +static const struct snd_soc_dapm_route ssm2518_routes[] = { + { "OUTL", NULL, "DACL" }, + { "OUTR", NULL, "DACR" }, +}; + +struct ssm2518_mcs_lut { + unsigned int rate; + const unsigned int *sysclks; +}; + +static const unsigned int ssm2518_sysclks_2048000[] = { + 2048000, 4096000, 8192000, 12288000, 16384000, 24576000, + 3200000, 6400000, 12800000, 0 +}; + +static const unsigned int ssm2518_sysclks_2822000[] = { + 2822000, 5644800, 11289600, 16934400, 22579200, 33868800, + 4410000, 8820000, 17640000, 0 +}; + +static const unsigned int ssm2518_sysclks_3072000[] = { + 3072000, 6144000, 12288000, 16384000, 24576000, 38864000, + 4800000, 9600000, 19200000, 0 +}; + +static const struct ssm2518_mcs_lut ssm2518_mcs_lut[] = { + { 8000, ssm2518_sysclks_2048000, }, + { 11025, ssm2518_sysclks_2822000, }, + { 12000, ssm2518_sysclks_3072000, }, + { 16000, ssm2518_sysclks_2048000, }, + { 24000, ssm2518_sysclks_3072000, }, + { 22050, ssm2518_sysclks_2822000, }, + { 32000, ssm2518_sysclks_2048000, }, + { 44100, ssm2518_sysclks_2822000, }, + { 48000, ssm2518_sysclks_3072000, }, + { 96000, ssm2518_sysclks_3072000, }, +}; + +static const unsigned int ssm2518_rates_2048000[] = { + 8000, 16000, 32000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_2048000 = { + .list = ssm2518_rates_2048000, + .count = ARRAY_SIZE(ssm2518_rates_2048000), +}; + +static const unsigned int ssm2518_rates_2822000[] = { + 11025, 22050, 44100, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_2822000 = { + .list = ssm2518_rates_2822000, + .count = ARRAY_SIZE(ssm2518_rates_2822000), +}; + +static const unsigned int ssm2518_rates_3072000[] = { + 12000, 24000, 48000, 96000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_3072000 = { + .list = ssm2518_rates_3072000, + .count = ARRAY_SIZE(ssm2518_rates_3072000), +}; + +static const unsigned int ssm2518_rates_12288000[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 96000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_12288000 = { + .list = ssm2518_rates_12288000, + .count = ARRAY_SIZE(ssm2518_rates_12288000), +}; + +static unsigned int ssm2518_lookup_mcs(struct ssm2518 *ssm2518, + unsigned int rate) +{ + const unsigned int *sysclks = NULL; + int i; + + for (i = 0; i < ARRAY_SIZE(ssm2518_mcs_lut); i++) { + if (ssm2518_mcs_lut[i].rate == rate) { + sysclks = ssm2518_mcs_lut[i].sysclks; + break; + } + } + + if (!sysclks) + return -EINVAL; + + for (i = 0; sysclks[i]; i++) { + if (sysclks[i] == ssm2518->sysclk) + return i; + } + + return -EINVAL; +} + +static int ssm2518_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + unsigned int ctrl1, ctrl1_mask; + int mcs; + int ret; + + mcs = ssm2518_lookup_mcs(ssm2518, rate); + if (mcs < 0) + return mcs; + + ctrl1_mask = SSM2518_SAI_CTRL1_FS_MASK; + + if (rate >= 8000 && rate <= 12000) + ctrl1 = SSM2518_SAI_CTRL1_FS_8000_12000; + else if (rate >= 16000 && rate <= 24000) + ctrl1 = SSM2518_SAI_CTRL1_FS_16000_24000; + else if (rate >= 32000 && rate <= 48000) + ctrl1 = SSM2518_SAI_CTRL1_FS_32000_48000; + else if (rate >= 64000 && rate <= 96000) + ctrl1 = SSM2518_SAI_CTRL1_FS_64000_96000; + else + return -EINVAL; + + if (ssm2518->right_j) { + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_16BIT; + break; + case SNDRV_PCM_FORMAT_S24_LE: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_24BIT; + break; + default: + return -EINVAL; + } + ctrl1_mask |= SSM2518_SAI_CTRL1_FMT_MASK; + } + + /* Disable auto samplerate detection */ + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_CLOCK, + SSM2518_CLOCK_ASR, SSM2518_CLOCK_ASR); + if (ret < 0) + return ret; + + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, + ctrl1_mask, ctrl1); + if (ret < 0) + return ret; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_MCS_MASK, mcs << 1); +} + +static int ssm2518_mute(struct snd_soc_dai *dai, int mute) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int val; + + if (mute) + val = SSM2518_MUTE_CTRL_MUTE_MASTER; + else + val = 0; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_MUTE_CTRL, + SSM2518_MUTE_CTRL_MUTE_MASTER, val); +} + +static int ssm2518_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl1 = 0, ctrl2 = 0; + bool invert_fclk; + int ret; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + invert_fclk = false; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl2 |= SSM2518_SAI_CTRL2_BCLK_INVERT; + invert_fclk = false; + break; + case SND_SOC_DAIFMT_NB_IF: + invert_fclk = true; + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl2 |= SSM2518_SAI_CTRL2_BCLK_INVERT; + invert_fclk = true; + break; + default: + return -EINVAL; + } + + ssm2518->right_j = false; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_LJ; + invert_fclk = !invert_fclk; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_24BIT; + ssm2518->right_j = true; + invert_fclk = !invert_fclk; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_PULSE; + ctrl1 |= SSM2518_SAI_CTRL1_FMT_I2S; + invert_fclk = false; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_PULSE; + ctrl1 |= SSM2518_SAI_CTRL1_FMT_LJ; + invert_fclk = false; + break; + default: + return -EINVAL; + } + + if (invert_fclk) + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_INVERT; + + ret = regmap_write(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, ctrl1); + if (ret) + return ret; + + return regmap_write(ssm2518->regmap, SSM2518_REG_SAI_CTRL2, ctrl2); +} + +static int ssm2518_set_power(struct ssm2518 *ssm2518, bool enable) +{ + int ret = 0; + + if (!enable) { + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_SPWDN, SSM2518_POWER1_SPWDN); + regcache_mark_dirty(ssm2518->regmap); + } + + if (gpio_is_valid(ssm2518->enable_gpio)) + gpio_set_value(ssm2518->enable_gpio, enable); + + regcache_cache_only(ssm2518->regmap, !enable); + + if (enable) { + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_SPWDN | SSM2518_POWER1_RESET, 0x00); + regcache_sync(ssm2518->regmap); + } + + return ret; +} + +static int ssm2518_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + ret = ssm2518_set_power(ssm2518, true); + break; + case SND_SOC_BIAS_OFF: + ret = ssm2518_set_power(ssm2518, false); + break; + } + + if (ret) + return ret; + + codec->dapm.bias_level = level; + + return 0; +} + +static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int width) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl1, ctrl2; + int left_slot, right_slot; + int ret; + + if (slots == 0) + return regmap_update_bits(ssm2518->regmap, + SSM2518_REG_SAI_CTRL1, SSM2518_SAI_CTRL1_SAI_MASK, + SSM2518_SAI_CTRL1_SAI_I2S); + + if (tx_mask == 0 || tx_mask != 0) + return -EINVAL; + + if (slots == 1) { + if (tx_mask != 1) + return -EINVAL; + left_slot = 0; + right_slot = 0; + } else { + /* We assume the left channel < right channel */ + left_slot = ffs(tx_mask); + tx_mask &= ~(1 << tx_mask); + if (tx_mask == 0) { + right_slot = left_slot; + } else { + right_slot = ffs(tx_mask); + tx_mask &= ~(1 << tx_mask); + } + } + + if (tx_mask != 0 || left_slot >= slots || right_slot >= slots) + return -EINVAL; + + switch (width) { + case 16: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_16; + break; + case 24: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_24; + break; + case 32: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_32; + break; + default: + return -EINVAL; + } + + switch (slots) { + case 1: + ctrl1 = SSM2518_SAI_CTRL1_SAI_MONO; + break; + case 2: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_2; + break; + case 4: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_4; + break; + case 8: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_8; + break; + case 16: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_16; + break; + default: + return -EINVAL; + } + + ret = regmap_write(ssm2518->regmap, SSM2518_REG_CHAN_MAP, + (left_slot << SSM2518_CHAN_MAP_LEFT_SLOT_OFFSET) | + (right_slot << SSM2518_CHAN_MAP_RIGHT_SLOT_OFFSET)); + if (ret) + return ret; + + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, + SSM2518_SAI_CTRL1_SAI_MASK, ctrl1); + if (ret) + return ret; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL2, + SSM2518_SAI_CTRL2_SLOT_WIDTH_MASK, ctrl2); +} + +static int ssm2518_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + + if (ssm2518->constraints) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, ssm2518->constraints); + + return 0; +} + +#define SSM2518_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static const struct snd_soc_dai_ops ssm2518_dai_ops = { + .startup = ssm2518_startup, + .hw_params = ssm2518_hw_params, + .digital_mute = ssm2518_mute, + .set_fmt = ssm2518_set_dai_fmt, + .set_tdm_slot = ssm2518_set_tdm_slot, +}; + +static struct snd_soc_dai_driver ssm2518_dai = { + .name = "ssm2518-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SSM2518_FORMATS, + }, + .ops = &ssm2518_dai_ops, +}; + +static int ssm2518_probe(struct snd_soc_codec *codec) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = ssm2518->regmap; + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + return ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int ssm2518_remove(struct snd_soc_codec *codec) +{ + ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (clk_id != SSM2518_SYSCLK) + return -EINVAL; + + switch (source) { + case SSM2518_SYSCLK_SRC_MCLK: + val = 0; + break; + case SSM2518_SYSCLK_SRC_BCLK: + /* In this case the bitclock is used as the system clock, and + * the bitclock signal needs to be connected to the MCLK pin and + * the BCLK pin is left unconnected */ + val = SSM2518_POWER1_NO_BCLK; + break; + default: + return -EINVAL; + } + + switch (freq) { + case 0: + ssm2518->constraints = NULL; + break; + case 2048000: + case 4096000: + case 8192000: + case 3200000: + case 6400000: + case 12800000: + ssm2518->constraints = &ssm2518_constraints_2048000; + break; + case 2822000: + case 5644800: + case 11289600: + case 16934400: + case 22579200: + case 33868800: + case 4410000: + case 8820000: + case 17640000: + ssm2518->constraints = &ssm2518_constraints_2822000; + break; + case 3072000: + case 6144000: + case 38864000: + case 4800000: + case 9600000: + case 19200000: + ssm2518->constraints = &ssm2518_constraints_3072000; + break; + case 12288000: + case 16384000: + case 24576000: + ssm2518->constraints = &ssm2518_constraints_12288000; + break; + default: + return -EINVAL; + } + + ssm2518->sysclk = freq; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_NO_BCLK, val); +} + +static struct snd_soc_codec_driver ssm2518_codec_driver = { + .probe = ssm2518_probe, + .remove = ssm2518_remove, + .set_bias_level = ssm2518_set_bias_level, + .set_sysclk = ssm2518_set_sysclk, + .idle_bias_off = true, + + .controls = ssm2518_snd_controls, + .num_controls = ARRAY_SIZE(ssm2518_snd_controls), + .dapm_widgets = ssm2518_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ssm2518_dapm_widgets), + .dapm_routes = ssm2518_routes, + .num_dapm_routes = ARRAY_SIZE(ssm2518_routes), +}; + +static bool ssm2518_register_volatile(struct device *dev, unsigned int reg) +{ + return false; +} + +static const struct regmap_config ssm2518_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .max_register = SSM2518_REG_DRC_9, + .volatile_reg = ssm2518_register_volatile, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ssm2518_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ssm2518_reg_defaults), +}; + +static int ssm2518_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ssm2518_platform_data *pdata = i2c->dev.platform_data; + struct ssm2518 *ssm2518; + int ret; + + ssm2518 = devm_kzalloc(&i2c->dev, sizeof(*ssm2518), GFP_KERNEL); + if (ssm2518 == NULL) + return -ENOMEM; + + if (pdata) { + ssm2518->enable_gpio = pdata->enable_gpio; + } else if (i2c->dev.of_node) { + ssm2518->enable_gpio = of_get_gpio(i2c->dev.of_node, 0); + if (ssm2518->enable_gpio < 0 && ssm2518->enable_gpio != -ENOENT) + return ssm2518->enable_gpio; + } else { + ssm2518->enable_gpio = -1; + } + + if (gpio_is_valid(ssm2518->enable_gpio)) { + ret = devm_gpio_request_one(&i2c->dev, ssm2518->enable_gpio, + GPIOF_OUT_INIT_HIGH, "SSM2518 nSD"); + if (ret) + return ret; + } + + i2c_set_clientdata(i2c, ssm2518); + + ssm2518->regmap = devm_regmap_init_i2c(i2c, &ssm2518_regmap_config); + if (IS_ERR(ssm2518->regmap)) + return PTR_ERR(ssm2518->regmap); + + /* + * The reset bit is obviously volatile, but we need to be able to cache + * the other bits in the register, so we can't just mark the whole + * register as volatile. Since this is the only place where we'll ever + * touch the reset bit just bypass the cache for this operation. + */ + regcache_cache_bypass(ssm2518->regmap, true); + ret = regmap_write(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_RESET); + regcache_cache_bypass(ssm2518->regmap, false); + if (ret) + return ret; + + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER2, + SSM2518_POWER2_APWDN, 0x00); + if (ret) + return ret; + + ret = ssm2518_set_power(ssm2518, false); + if (ret) + return ret; + + return snd_soc_register_codec(&i2c->dev, &ssm2518_codec_driver, + &ssm2518_dai, 1); +} + +static int ssm2518_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ssm2518_i2c_ids[] = { + { "ssm2518", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ssm2518_i2c_ids); + +static struct i2c_driver ssm2518_driver = { + .driver = { + .name = "ssm2518", + .owner = THIS_MODULE, + }, + .probe = ssm2518_i2c_probe, + .remove = ssm2518_i2c_remove, + .id_table = ssm2518_i2c_ids, +}; +module_i2c_driver(ssm2518_driver); + +MODULE_DESCRIPTION("ASoC SSM2518 driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2518.h b/sound/soc/codecs/ssm2518.h new file mode 100644 index 0000000..62511d8 --- /dev/null +++ b/sound/soc/codecs/ssm2518.h @@ -0,0 +1,20 @@ +/* + * SSM2518 amplifier audio driver + * + * Copyright 2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#ifndef __SND_SOC_CODECS_SSM2518_H__ +#define __SND_SOC_CODECS_SSM2518_H__ + +#define SSM2518_SYSCLK 0 + +enum ssm2518_sysclk_src { + SSM2518_SYSCLK_SRC_MCLK = 0, + SSM2518_SYSCLK_SRC_BCLK = 1, +}; + +#endif -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: wm5110: Correct DSP4R Mixer control name
by Charles Keepax 25 May '13

25 May '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index c00480b..ba38f06 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -190,7 +190,7 @@ ARIZONA_MIXER_CONTROLS("DSP2R", ARIZONA_DSP2RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP3L", ARIZONA_DSP3LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP3R", ARIZONA_DSP3RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP4L", ARIZONA_DSP4LMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DSP5R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DSP4R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), -- 1.7.2.5

2 1

[alsa-devel] Handling HDA FIFO error
by Daniel Drake 25 May '13

25 May '13

Hi, On the OLPC XO-1.5 (VIA VX855 SoC with HDA) we sometimes see a problem where audio playback fails. Easy to reproduce by playing a short wav file in a loop, it will fail after a minute or two. The playback app gets confused here: snd_pcm_wait() returns success, but snd_pcm_writei() returns EAGAIN. Digging into the HDA code, I can see that normally, when the stream gets started, we get interrupts with SD_STS as 0x24. In the failure case, as soon as we start the stream, we get a flood of interrupts with SD_STS 0x8 - which indicates FIFO error. Disabling interrupt generation upon FIFO error does not help much - it avoids the flood of interrupts, but doesn't change the fact that audio playing apps hang or loop infinitely. I have tried the various position_fix options with no luck. Are there any things I could check that would explain why a FIFO error might be occurring? After a quick read of the HDA spec I don't really understand what this FIFO actually is and what it does, how does it relate to the BDL, CORB and RIRB? Failing that... any suggestions for how to improve the hda_intel driver to handle FIFO error gracefully? Right now there is no code to handle that. Thanks Daniel

3 3

[alsa-devel] [PATCH] ASoC: Add support for BCM2708
by Florian Meier 24 May '13

24 May '13

This driver adds support for digital audio (I2S) for the BCM2708 SoC that is used by the Raspberry Pi. External audio codecs can be connected to the Raspberry Pi via P5 header. It relies on cyclic DMA engine support for BCM2708 that is not included in this patch. The currently supported audio codecs are TLV320AIC23 and WM8731. Signed-off-by: Florian Meier <florian.meier(a)koalo.de> --- A Raspberry Pi kernel featuring this driver can be found at https://github.com/koalo/linux rpi-3.8.y-asocdev You can find more information about how to use it at http://blog.koalo.de/2013/05/i2s-support-for-raspberry-pi.html sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/bcm2708/Kconfig | 29 ++ sound/soc/bcm2708/Makefile | 12 + sound/soc/bcm2708/bcm2708-i2s.c | 964 +++++++++++++++++++++++++++++++++++++++ sound/soc/bcm2708/bcm2708-pcm.c | 303 ++++++++++++ sound/soc/bcm2708/bcm2708-pcm.h | 40 ++ sound/soc/bcm2708/rpi-mbed.c | 113 +++++ sound/soc/bcm2708/rpi-proto.c | 131 ++++++ 9 files changed, 1594 insertions(+) create mode 100644 sound/soc/bcm2708/Kconfig create mode 100644 sound/soc/bcm2708/Makefile create mode 100644 sound/soc/bcm2708/bcm2708-i2s.c create mode 100644 sound/soc/bcm2708/bcm2708-pcm.c create mode 100644 sound/soc/bcm2708/bcm2708-pcm.h create mode 100644 sound/soc/bcm2708/rpi-mbed.c create mode 100644 sound/soc/bcm2708/rpi-proto.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 9e675c7..d3852f8 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_GENERIC_DMAENGINE_PCM # All the supported SoCs source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" +source "sound/soc/bcm2708/Kconfig" source "sound/soc/blackfin/Kconfig" source "sound/soc/cirrus/Kconfig" source "sound/soc/davinci/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 197b6ae..eef1a7b 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -14,6 +14,7 @@ obj-$(CONFIG_SND_SOC) += codecs/ obj-$(CONFIG_SND_SOC) += generic/ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ +obj-$(CONFIG_SND_SOC) += bcm2708/ obj-$(CONFIG_SND_SOC) += blackfin/ obj-$(CONFIG_SND_SOC) += cirrus/ obj-$(CONFIG_SND_SOC) += davinci/ diff --git a/sound/soc/bcm2708/Kconfig b/sound/soc/bcm2708/Kconfig new file mode 100644 index 0000000..f47c2c0 --- /dev/null +++ b/sound/soc/bcm2708/Kconfig @@ -0,0 +1,29 @@ +config SND_BCM2708_SOC_I2S + tristate + +config SND_BCM2708_SOC + tristate "SoC Audio support for the Broadcom BCM2708 I2S module" + depends on MACH_BCM2708 + select SND_SOC_DMAENGINE_PCM + select DMA_BCM2708 + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M if you want to add support for codecs attached to + the BCM2708 I2S interface. You will also need + to select the audio interfaces to support below. + +config SND_BCM2708_SOC_RPI_CODEC_MBED + tristate "Support for AudioCODEC for mbed (TLV320AIC32B)" + depends on SND_BCM2708_SOC + select SND_BCM2708_SOC_I2S + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for AudioCODEC for mbed (TLV320AIC32B) + +config SND_BCM2708_SOC_RPI_CODEC_PROTO + tristate "Support for Audio Codec Board - PROTO (WM8731)" + depends on SND_BCM2708_SOC + select SND_BCM2708_SOC_I2S + select SND_SOC_WM8731 + help + Say Y if you want to add support for Audio Codec Board - PROTO (WM8731) diff --git a/sound/soc/bcm2708/Makefile b/sound/soc/bcm2708/Makefile new file mode 100644 index 0000000..284d76d --- /dev/null +++ b/sound/soc/bcm2708/Makefile @@ -0,0 +1,12 @@ +# BCM2708 Platform Support +snd-soc-bcm2708-objs := bcm2708-pcm.o +snd-soc-bcm2708-i2s-objs := bcm2708-i2s.o +snd-soc-rpi-mbed-objs := rpi-mbed.o +snd-soc-rpi-proto-objs := rpi-proto.o + +obj-$(CONFIG_SND_BCM2708_SOC) += snd-soc-bcm2708.o +obj-$(CONFIG_SND_BCM2708_SOC_I2S) += snd-soc-bcm2708-i2s.o + +# BCM2708 Machine Support +obj-$(CONFIG_SND_BCM2708_SOC_RPI_CODEC_MBED) += snd-soc-rpi-mbed.o +obj-$(CONFIG_SND_BCM2708_SOC_RPI_CODEC_PROTO) += snd-soc-rpi-proto.o diff --git a/sound/soc/bcm2708/bcm2708-i2s.c b/sound/soc/bcm2708/bcm2708-i2s.c new file mode 100644 index 0000000..a8e995f --- /dev/null +++ b/sound/soc/bcm2708/bcm2708-i2s.c @@ -0,0 +1,964 @@ +/* + * ALSA SoC I2S Audio Layer for Broadcom BCM2708 SoC + * + * Author: Florian Meier, <florian.meier(a)koalo.de> + * Copyright 2013 + * + * based on + * Raspberry Pi PCM I2S ALSA Driver + * Copyright (c) by Phil Poole 2013 + * + * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor + * Vladimir Barinov, <vbarinov(a)embeddedalley.com> + * Copyright (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> + * + * OMAP ALSA SoC DAI driver using McBSP port + * Copyright (C) 2008 Nokia Corporation + * Contact: Jarkko Nikula <jarkko.nikula(a)bitmer.com> + * Peter Ujfalusi <peter.ujfalusi(a)ti.com> + * + * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver + * Author: Timur Tabi <timur(a)freescale.com> + * Copyright 2007-2010 Freescale Semiconductor, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License 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/device.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "bcm2708-pcm.h" + +/* Clock registers */ +#define BCM2708_CLK_PCMCTL_REG 0x00 +#define BCM2708_CLK_PCMDIV_REG 0x04 + +/* Clock register settings */ +#define BCM2708_CLK_PASSWD (0x5a000000) +#define BCM2708_CLK_MASH(v) ((v) << 9) +#define BCM2708_CLK_FLIP (1 << 8) +#define BCM2708_CLK_BUSY (1 << 7) +#define BCM2708_CLK_KILL (1 << 5) +#define BCM2708_CLK_ENAB (1 << 4) +#define BCM2708_CLK_SRC(v) (v) + +#define BCM2708_CLK_DIVI(v) ((v) << 12) +#define BCM2708_CLK_DIVF(v) (v) + +enum { + BCM2708_CLK_MASH_0 = 0, + BCM2708_CLK_MASH_1, + BCM2708_CLK_MASH_2, + BCM2708_CLK_MASH_3, +}; + +enum { + BCM2708_CLK_SRC_GND = 0, + BCM2708_CLK_SRC_OSC, + BCM2708_CLK_SRC_DBG0, + BCM2708_CLK_SRC_DBG1, + BCM2708_CLK_SRC_PLLA, + BCM2708_CLK_SRC_PLLC, + BCM2708_CLK_SRC_PLLD, + BCM2708_CLK_SRC_HDMI, +}; + +/* Most clocks are not useable (freq = 0) */ +static const unsigned int bcm2708_clk_freq[BCM2708_CLK_SRC_HDMI+1] = { + [BCM2708_CLK_SRC_GND] = 0, + [BCM2708_CLK_SRC_OSC] = 19200000, + [BCM2708_CLK_SRC_DBG0] = 0, + [BCM2708_CLK_SRC_DBG1] = 0, + [BCM2708_CLK_SRC_PLLA] = 0, + [BCM2708_CLK_SRC_PLLC] = 0, + [BCM2708_CLK_SRC_PLLD] = 500000000, + [BCM2708_CLK_SRC_HDMI] = 0, +}; + + +/* I2S registers */ +#define BCM2708_I2S_CS_A_REG 0x00 +#define BCM2708_I2S_FIFO_A_REG 0x04 +#define BCM2708_I2S_MODE_A_REG 0x08 +#define BCM2708_I2S_RXC_A_REG 0x0c +#define BCM2708_I2S_TXC_A_REG 0x10 +#define BCM2708_I2S_DREQ_A_REG 0x14 +#define BCM2708_I2S_INTEN_A_REG 0x18 +#define BCM2708_I2S_INTSTC_A_REG 0x1c +#define BCM2708_I2S_GRAY_REG 0x20 + +/* I2S register settings */ +#define BCM2708_I2S_STBY (1 << 25) +#define BCM2708_I2S_SYNC (1 << 24) +#define BCM2708_I2S_RXSEX (1 << 23) +#define BCM2708_I2S_RXF (1 << 22) +#define BCM2708_I2S_TXE (1 << 21) +#define BCM2708_I2S_RXD (1 << 20) +#define BCM2708_I2S_TXD (1 << 19) +#define BCM2708_I2S_RXR (1 << 18) +#define BCM2708_I2S_TXW (1 << 17) +#define BCM2708_I2S_CS_RXERR (1 << 16) +#define BCM2708_I2S_CS_TXERR (1 << 15) +#define BCM2708_I2S_RXSYNC (1 << 14) +#define BCM2708_I2S_TXSYNC (1 << 13) +#define BCM2708_I2S_DMAEN (1 << 9) +#define BCM2708_I2S_RXTHR(v) ((v) << 7) +#define BCM2708_I2S_TXTHR(v) ((v) << 5) +#define BCM2708_I2S_RXCLR (1 << 4) +#define BCM2708_I2S_TXCLR (1 << 3) +#define BCM2708_I2S_TXON (1 << 2) +#define BCM2708_I2S_RXON (1 << 1) +#define BCM2708_I2S_EN (1) + +#define BCM2708_I2S_CLKDIS (1 << 28) +#define BCM2708_I2S_PDMN (1 << 27) +#define BCM2708_I2S_PDME (1 << 26) +#define BCM2708_I2S_FRXP (1 << 25) +#define BCM2708_I2S_FTXP (1 << 24) +#define BCM2708_I2S_CLKM (1 << 23) +#define BCM2708_I2S_CLKI (1 << 22) +#define BCM2708_I2S_FSM (1 << 21) +#define BCM2708_I2S_FSI (1 << 20) +#define BCM2708_I2S_FLEN(v) ((v) << 10) +#define BCM2708_I2S_FSLEN(v) (v) + +#define BCM2708_I2S_CHWEX (1 << 15) +#define BCM2708_I2S_CHEN (1 << 14) +#define BCM2708_I2S_CHPOS(v) ((v) << 4) +#define BCM2708_I2S_CHWID(v) (v) +#define BCM2708_I2S_CH1(v) ((v) << 16) +#define BCM2708_I2S_CH2(v) (v) + +#define BCM2708_I2S_TX_PANIC(v) ((v) << 24) +#define BCM2708_I2S_RX_PANIC(v) ((v) << 16) +#define BCM2708_I2S_TX(v) ((v) << 8) +#define BCM2708_I2S_RX(v) (v) + +#define BCM2708_I2S_INT_RXERR (1 << 3) +#define BCM2708_I2S_INT_TXERR (1 << 2) +#define BCM2708_I2S_INT_RXR (1 << 1) +#define BCM2708_I2S_INT_TXW (1 << 0) + +/* I2S DMA interface */ +#define BCM2708_I2S_FIFO_PHYSICAL_ADDR 0x7E203004 +#define BCM2708_DMA_DREQ_PCM_TX 2 +#define BCM2708_DMA_DREQ_PCM_RX 3 + +/* General device struct */ +struct bcm2708_i2s_dev { + struct device *dev; + struct bcm2708_pcm_dma_data dma_params[2]; + void __iomem *i2s_base; + void __iomem *clk_base; + unsigned int fmt; + struct snd_pcm_substream *first_stream; + struct snd_pcm_substream *second_stream; +}; + +static inline void bcm2708_i2s_write_reg(struct bcm2708_i2s_dev *dev, + int reg, u32 val) +{ + dev_dbg(dev->dev, "I2S write to register %p = %x\n", + dev->clk_base + reg, val); + __raw_writel(val, dev->i2s_base + reg); +} + +static inline u32 bcm2708_i2s_read_reg(struct bcm2708_i2s_dev *dev, int reg) +{ + return __raw_readl(dev->i2s_base + reg); +} + +static inline void bcm2708_i2s_set_bits(struct bcm2708_i2s_dev *dev, + int reg, u32 val) +{ + u32 oldval = __raw_readl(dev->i2s_base + reg); + bcm2708_i2s_write_reg(dev, reg, oldval | val); +} + +static inline void bcm2708_i2s_clear_bits(struct bcm2708_i2s_dev *dev, + int reg, u32 val) +{ + u32 oldval = __raw_readl(dev->i2s_base + reg); + bcm2708_i2s_write_reg(dev, reg, oldval & ~val); +} + +static inline void bcm2708_clk_write_reg(struct bcm2708_i2s_dev *dev, + int reg, u32 val) +{ + dev_dbg(dev->dev, "PCM clk write to register %p = %x\n", + dev->clk_base + reg, val); + __raw_writel(val, dev->clk_base + reg); +} + +static inline u32 bcm2708_clk_read_reg(struct bcm2708_i2s_dev *dev, int reg) +{ + return __raw_readl(dev->clk_base + reg); +} + +static void bcm2708_i2s_start_clock(struct bcm2708_i2s_dev *dev) +{ + /* + * Start the clock if in master mode. + */ + unsigned int master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + if (master == SND_SOC_DAIFMT_CBS_CFS + || master == SND_SOC_DAIFMT_CBS_CFM) { + unsigned int clkreg = bcm2708_clk_read_reg(dev, + BCM2708_CLK_PCMCTL_REG); + bcm2708_clk_write_reg(dev, BCM2708_CLK_PCMCTL_REG, + BCM2708_CLK_PASSWD | clkreg | BCM2708_CLK_ENAB); + } +} + +static void bcm2708_i2s_stop_clock(struct bcm2708_i2s_dev *dev) +{ + int timeout = 1000; + + /* stop clock */ + unsigned int clkreg = bcm2708_clk_read_reg(dev, BCM2708_CLK_PCMCTL_REG); + bcm2708_clk_write_reg(dev, BCM2708_CLK_PCMCTL_REG, ~(BCM2708_CLK_ENAB) + & (BCM2708_CLK_PASSWD | clkreg)); + + /* wait for the BUSY flag going down */ + while ((bcm2708_clk_read_reg(dev, BCM2708_CLK_PCMCTL_REG) + & BCM2708_CLK_BUSY) + && timeout > 0) { + timeout--; + } + + if (timeout <= 0) { + /* KILL the clock */ + dev_err(dev->dev, "I2S clock didn't stop. Kill the clock!\n"); + bcm2708_clk_write_reg(dev, BCM2708_CLK_PCMCTL_REG, + BCM2708_CLK_KILL | BCM2708_CLK_PASSWD + | clkreg); + } +} + +static void bcm2708_i2s_clear_fifos(struct bcm2708_i2s_dev *dev) +{ + int timeout = 1000; + unsigned int syncval; + + /* Backup the current state */ + unsigned int i2s_active_state = bcm2708_i2s_read_reg(dev, + BCM2708_I2S_CS_A_REG) + & (BCM2708_I2S_RXON | BCM2708_I2S_TXON); + + unsigned int clk_active_state = bcm2708_clk_read_reg(dev, + BCM2708_CLK_PCMCTL_REG) & BCM2708_CLK_ENAB; + + /* Start clock if not running */ + if (!clk_active_state) { + unsigned int clkreg = bcm2708_clk_read_reg(dev, + BCM2708_CLK_PCMCTL_REG); + bcm2708_clk_write_reg(dev, BCM2708_CLK_PCMCTL_REG, + BCM2708_CLK_PASSWD | clkreg | BCM2708_CLK_ENAB); + } + + /* Stop I2S module */ + bcm2708_i2s_clear_bits(dev, BCM2708_I2S_CS_A_REG, BCM2708_I2S_RXON + | BCM2708_I2S_TXON); + + /* + * Clear the FIFOs + * Requires at least 2 PCM clock cycles to take effect + */ + bcm2708_i2s_set_bits(dev, BCM2708_I2S_CS_A_REG, BCM2708_I2S_RXCLR + | BCM2708_I2S_TXCLR); + + /* Wait for 2 PCM clock cycles */ + + /* + * Toggle the SYNC flag - after 2 PCM clock cycles it can be read back + * FIXME: This does not seem to work for slave mode! + */ + syncval = bcm2708_i2s_read_reg(dev, BCM2708_I2S_CS_A_REG) + & BCM2708_I2S_SYNC; + if (syncval) + bcm2708_i2s_clear_bits(dev, BCM2708_I2S_CS_A_REG, + BCM2708_I2S_SYNC); + else + bcm2708_i2s_set_bits(dev, BCM2708_I2S_CS_A_REG, + BCM2708_I2S_SYNC); + + /* Wait for the SYNC flag changing it's state */ + while (((bcm2708_i2s_read_reg(dev, BCM2708_I2S_CS_A_REG) + & BCM2708_I2S_SYNC) == syncval) + && timeout > 0) { + timeout--; + } + + if (timeout <= 0) + dev_err(dev->dev, "I2S SYNC error!\n"); + + /* Stop clock if it was not running before */ + if (!clk_active_state) + bcm2708_i2s_stop_clock(dev); + + /* Restore I2S state */ + bcm2708_i2s_set_bits(dev, BCM2708_I2S_CS_A_REG, i2s_active_state); +} + +static void bcm2708_i2s_start(struct bcm2708_i2s_dev *dev, + struct snd_pcm_substream *substream) +{ + bcm2708_i2s_start_clock(dev); + + /* + * Enable the stream. + */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + bcm2708_i2s_set_bits(dev, BCM2708_I2S_CS_A_REG, + BCM2708_I2S_RXON); + } else { + bcm2708_i2s_set_bits(dev, BCM2708_I2S_CS_A_REG, + BCM2708_I2S_TXON); + } +} + +static void bcm2708_i2s_stop(struct bcm2708_i2s_dev *dev, + struct snd_pcm_substream *substream) +{ + unsigned int still_running; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + bcm2708_i2s_clear_bits(dev, BCM2708_I2S_CS_A_REG, + BCM2708_I2S_RXON); + } else { + bcm2708_i2s_clear_bits(dev, BCM2708_I2S_CS_A_REG, + BCM2708_I2S_TXON); + } + + still_running = bcm2708_i2s_read_reg(dev, BCM2708_I2S_CS_A_REG) + & (BCM2708_I2S_TXON | BCM2708_I2S_RXON); + + /* Stop also the clock when not SND_SOC_DAIFMT_CONT */ + if (!still_running && !(dev->fmt & SND_SOC_DAIFMT_CONT)) + bcm2708_i2s_stop_clock(dev); + +} + +static int bcm2708_i2s_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct bcm2708_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + dev->fmt = fmt; + return 0; +} + +static bool bcm2708_i2s_check_other_stream(struct bcm2708_i2s_dev *dev, + struct snd_pcm_substream *substream, + struct snd_pcm_substream *other_stream, + struct snd_pcm_hw_params *params) +{ + if (other_stream->runtime->format && + (other_stream->runtime->format != params_format(params))) { + dev_err(dev->dev, + "Sample formats of streams are different. %i (%s) != %i (%s) Initialization failed!\n", + other_stream->runtime->format, + (other_stream->stream == + SNDRV_PCM_STREAM_PLAYBACK ? + "playback" : "capture"), + params_format(params), + (substream->stream == + SNDRV_PCM_STREAM_PLAYBACK ? + "playback" : "capture")); + return false; + } + + if (other_stream->runtime->rate && + (other_stream->runtime->rate != params_rate(params))) { + dev_err(dev->dev, + "Sampling rates of streams are different. %i (%s) != %i (%s) Initialization failed!\n", + other_stream->runtime->rate, + (other_stream->stream == + SNDRV_PCM_STREAM_PLAYBACK ? + "playback" : "capture"), + params_rate(params), + (substream->stream == + SNDRV_PCM_STREAM_PLAYBACK ? + "playback" : "capture")); + return false; + } + + return true; +} + +static int bcm2708_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct bcm2708_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + unsigned int sampling_rate = params_rate(params); + unsigned int data_length, data_delay, half_frame; + unsigned int ch1pos, ch2pos, mode, format; + unsigned int mash = BCM2708_CLK_MASH_1; + unsigned int divi, divf, target_frequency; + int clk_src = -1; + unsigned int master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + unsigned int bit_master = (master == SND_SOC_DAIFMT_CBS_CFS + || master == SND_SOC_DAIFMT_CBS_CFM); + + unsigned int frame_master = (master == SND_SOC_DAIFMT_CBS_CFS + || master == SND_SOC_DAIFMT_CBM_CFS); + + /* Ensure, that both streams have the same settings */ + struct snd_pcm_substream *other_stream = dev->first_stream; + if (other_stream == substream) + other_stream = dev->second_stream; + + if (other_stream != NULL + && !bcm2708_i2s_check_other_stream(dev, + substream, other_stream, params)) + return -EINVAL; + + /* + * If a stream is already enabled, + * the registers are already set properly. + */ + if (bcm2708_i2s_read_reg(dev, BCM2708_I2S_CS_A_REG) + & (BCM2708_I2S_TXON | BCM2708_I2S_RXON)) + return 0; + + /* + * Adjust the data length according to the format. + * We prefill the half frame length with an integer + * divider of 2400 as explained at the clock settings. + * Maybe it is overwritten there, if the Integer mode + * does not apply. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + data_length = 16; + half_frame = 20; + break; + case SNDRV_PCM_FORMAT_S32_LE: + data_length = 32; + half_frame = 40; + break; + default: + return -EINVAL; + } + + /* + * Clock Settings + * + * The target frequency of the bit clock is + * sampling rate * frame length + * + * Integer mode: + * Sampling rates that are multiples of 8000 kHz + * can be driven by the oscillator of 19.2 MHz + * with an integer divider as long as the frame length + * is an integer divider of 19200000/8000=2400 as set up above. + * This is no longer possible if the sampling rate + * is too high (e.g. 192 kHz), because the oscillator is too slow. + * + * MASH mode: + * For all other sampling rates, it is not possible to + * have an integer divider. Approximate the clock + * with the MASH module that induces a slight frequency + * variance. To minimize that it is best to have the fastest + * clock here. That is PLLD with 500 MHz. + */ + target_frequency = sampling_rate*half_frame*2; + clk_src = BCM2708_CLK_SRC_OSC; + mash = BCM2708_CLK_MASH_0; + + if (bcm2708_clk_freq[clk_src] % target_frequency == 0 + && bit_master && frame_master) { + divi = bcm2708_clk_freq[clk_src]/target_frequency; + divf = 0; + } else { + uint64_t dividend; + + /* + * Overwrite half frame length, because the + * above trick is not needed. + * This is fixed, because a bit clock of 64*fs + * seems to be what most codecs want. + * Is it necessary to have this dynamic? + */ + half_frame = 32; + target_frequency = sampling_rate*half_frame*2; + + clk_src = BCM2708_CLK_SRC_PLLD; + mash = BCM2708_CLK_MASH_1; + + dividend = bcm2708_clk_freq[clk_src]; + dividend *= 1024; + do_div(dividend, target_frequency); + divi = dividend / 1024; + divf = dividend % 1024; + } + + /* Set clock divider */ + bcm2708_clk_write_reg(dev, BCM2708_CLK_PCMDIV_REG, BCM2708_CLK_PASSWD + | BCM2708_CLK_DIVI(divi) + | BCM2708_CLK_DIVF(divf)); + + /* Setup clock, but don't start it yet */ + bcm2708_clk_write_reg(dev, BCM2708_CLK_PCMCTL_REG, BCM2708_CLK_PASSWD + | BCM2708_CLK_MASH(mash) + | BCM2708_CLK_SRC(clk_src)); + + + /* Setup the frame format */ + format = BCM2708_I2S_CHEN; + + if (data_length > 24) + format |= BCM2708_I2S_CHWEX; + + format |= BCM2708_I2S_CHWID((data_length-8)&0xf); + + switch (dev->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + data_delay = 1; + break; + default: + /* TODO + * Others are possible but are not implemented at the moment. + */ + dev_err(dev->dev, "%s:bad format\n", __func__); + return -EINVAL; + } + + ch1pos = data_delay; + ch2pos = half_frame+data_delay; + + switch (params_channels(params)) { + case 2: + format = BCM2708_I2S_CH1(format) | BCM2708_I2S_CH2(format); + format |= BCM2708_I2S_CH1(BCM2708_I2S_CHPOS(ch1pos)); + format |= BCM2708_I2S_CH2(BCM2708_I2S_CHPOS(ch2pos)); + break; + default: + return -EINVAL; + } + + /* + * Set format for both streams. + * We cannot set another frame length + * (and therefore word length) anyway, + * so the format will be the same. + */ + bcm2708_i2s_write_reg(dev, BCM2708_I2S_RXC_A_REG, format); + bcm2708_i2s_write_reg(dev, BCM2708_I2S_TXC_A_REG, format); + + + /* Setup the I2S mode */ + mode = 0; + + if (data_length <= 16) { + /* + * Use frame packed mode (2 channels per 32 bit word) + * We cannot set another frame length in the second stream + * (and therefore word length) anyway, + * so the format will be the same. + */ + mode |= BCM2708_I2S_FTXP | BCM2708_I2S_FRXP; + } + + mode |= BCM2708_I2S_FLEN(half_frame*2-1); + mode |= BCM2708_I2S_FSLEN(half_frame); + + /* master or slave? */ + switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* cpu is master */ + break; + case SND_SOC_DAIFMT_CBM_CFS: + /* + * codec is bit clock master + * cpu is frame master + */ + mode |= BCM2708_I2S_CLKM; + break; + case SND_SOC_DAIFMT_CBS_CFM: + /* + * codec is frame master + * cpu is bit clock master + */ + mode |= BCM2708_I2S_FSM; + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* codec is master */ + mode |= BCM2708_I2S_CLKM; + mode |= BCM2708_I2S_FSM; + break; + default: + dev_err(dev->dev, "%s:bad master\n", __func__); + return -EINVAL; + } + + /* + * Invert clocks? + * + * The BCM approach seems to be inverted to the classical I2S approach. + */ + switch (dev->fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* none - therefore, both for BCM */ + mode |= BCM2708_I2S_CLKI; + mode |= BCM2708_I2S_FSI; + break; + case SND_SOC_DAIFMT_IB_IF: + /* both - therefore, none for BCM*/ + break; + case SND_SOC_DAIFMT_NB_IF: + /* + * invert only frame sync - therefore, + * invert only bit clock for BCM + */ + mode |= BCM2708_I2S_CLKI; + break; + case SND_SOC_DAIFMT_IB_NF: + /* + * invert only bit clock - therefore, + * invert only frame sync for BCM + */ + mode |= BCM2708_I2S_FSI; + break; + default: + return -EINVAL; + } + + bcm2708_i2s_write_reg(dev, BCM2708_I2S_MODE_A_REG, mode); + + + /* Setup the DMA parameters */ + bcm2708_i2s_set_bits(dev, BCM2708_I2S_CS_A_REG, BCM2708_I2S_RXTHR(1) + | BCM2708_I2S_TXTHR(1) + | BCM2708_I2S_DMAEN); + + bcm2708_i2s_write_reg(dev, BCM2708_I2S_DREQ_A_REG, + BCM2708_I2S_TX_PANIC(0x10) + | BCM2708_I2S_RX_PANIC(0x30) + | BCM2708_I2S_TX(0x30) + | BCM2708_I2S_RX(0x20)); + + /* Clear FIFOs */ + bcm2708_i2s_clear_fifos(dev); + + return 0; +} + + +static int bcm2708_i2s_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct bcm2708_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + unsigned int cs_reg; + + bcm2708_i2s_start_clock(dev); + + /* + * Clear both FIFOs if the one that should be started + * is not empty at the moment. This should only happen + * after overrun. Otherwise, hw_params would have cleared + * the FIFO. + */ + cs_reg = bcm2708_i2s_read_reg(dev, BCM2708_I2S_CS_A_REG); + + switch (substream->stream) { + case SNDRV_PCM_STREAM_PLAYBACK: + if (cs_reg & BCM2708_I2S_TXE) + break; + case SNDRV_PCM_STREAM_CAPTURE: + if (!(cs_reg & BCM2708_I2S_RXD)) + break; + + bcm2708_i2s_clear_fifos(dev); + } + + return 0; +} + +static int bcm2708_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct bcm2708_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + bcm2708_i2s_start(dev, substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + bcm2708_i2s_stop(dev, substream); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int bcm2708_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct bcm2708_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + if (!dev->first_stream) { + dev->first_stream = substream; + + /* should this still be running stop it */ + bcm2708_i2s_stop_clock(dev); + + /* enable PCM block */ + bcm2708_i2s_set_bits(dev, BCM2708_I2S_CS_A_REG, BCM2708_I2S_EN); + + /* + * Disable STBY + * Requires at least 4 PCM clock cycles to take effect + */ + bcm2708_i2s_set_bits(dev, BCM2708_I2S_CS_A_REG, + BCM2708_I2S_STBY); + } else { + struct snd_pcm_runtime *first_runtime = + dev->first_stream->runtime; + + /* + * This is the second stream open, so we need to impose + * sample size and sampling rate constraints. + * This is because frame length and clock cannot be specified + * seperately. + * + * Note that this can cause a race condition if the + * second stream is opened before the first stream is + * fully initialized. We provide some protection by + * checking to make sure the first stream is + * initialized, but it's not perfect. ALSA sometimes + * re-initializes the driver with a different sample + * rate or size. If the second stream is opened + * before the first stream has received its final + * parameters, then the second stream may be + * constrained to the wrong sample rate or size. + * + * We will continue in case of failure and recheck the + * constraint in hw_params. + */ + if (!first_runtime->format) { + dev_err(substream->pcm->card->dev, + "Set format in %s stream first! " + "Initialization may fail.\n", + substream->stream == + SNDRV_PCM_STREAM_PLAYBACK + ? "capture" : "playback"); + } else { + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_FORMAT, + first_runtime->format, + first_runtime->format); + } + + if (!first_runtime->rate) { + dev_err(substream->pcm->card->dev, + "Set sampling rate in %s stream first! " + "Initialization may fail!\n", + substream->stream == + SNDRV_PCM_STREAM_PLAYBACK + ? "capture" : "playback"); + } else { + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + first_runtime->rate, + first_runtime->rate); + } + + dev->second_stream = substream; + } + + snd_soc_dai_set_dma_data(dai, substream, + &dev->dma_params[substream->stream]); + + return 0; +} + +static void bcm2708_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct bcm2708_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + bcm2708_i2s_stop(dev, substream); + + if (dev->first_stream == substream) + dev->first_stream = dev->second_stream; + + dev->second_stream = NULL; + + /* If both streams are stopped, disable module and clock */ + if (!dev->first_stream) { + /* Disable the module */ + bcm2708_i2s_clear_bits(dev, BCM2708_I2S_CS_A_REG, + BCM2708_I2S_EN); + + /* + * Stopping clock is necessary, because stop does + * not stop the clock when SND_SOC_DAIFMT_CONT + */ + bcm2708_i2s_stop_clock(dev); + } +} + +static const struct snd_soc_dai_ops bcm2708_i2s_dai_ops = { + .startup = bcm2708_i2s_startup, + .shutdown = bcm2708_i2s_shutdown, + .prepare = bcm2708_i2s_prepare, + .trigger = bcm2708_i2s_trigger, + .hw_params = bcm2708_i2s_hw_params, + .set_fmt = bcm2708_i2s_set_dai_fmt, +}; + +static struct snd_soc_dai_driver bcm2708_i2s_dai = { + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S32_LE + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S32_LE + }, + .ops = &bcm2708_i2s_dai_ops, + .symmetric_rates = 1 +}; + +static void bcm2708_i2s_setup_gpio(void) +{ + /* + * This is the common way to handle the GPIO pins for + * the Raspberry Pi. + * TODO Better way would be to handle + * this in the device tree! + */ +#define INP_GPIO(g) *(gpio+((g)/10)) &= ~(7<<(((g)%10)*3)) +#define SET_GPIO_ALT(g,a) *(gpio+(((g)/10))) |= (((a)<=3?(a)+4:(a)==4?3:2)<<(((g)%10)*3)) + + unsigned int *gpio; + int pin; + gpio = ioremap(GPIO_BASE, SZ_16K); + + /* SPI is on GPIO 7..11 */ + for (pin = 28; pin <= 31; pin++) { + INP_GPIO(pin); /* set mode to GPIO input first */ + SET_GPIO_ALT(pin, 2); /* set mode to ALT 0 */ + } +#undef INP_GPIO +#undef SET_GPIO_ALT +} + +static int bcm2708_i2s_probe(struct platform_device *pdev) +{ + struct bcm2708_i2s_dev *dev; + int i; + void __iomem *base[2]; + + /* request both ioareas */ + for (i = 0; i <= 1; i++) { + struct resource *mem, *ioarea; + mem = platform_get_resource(pdev, IORESOURCE_MEM, i); + if (!mem) { + dev_err(&pdev->dev, "I2S probe: Memory resource could not be found.\n"); + return -ENODEV; + } + + ioarea = devm_request_mem_region(&pdev->dev, mem->start, + resource_size(mem), + pdev->name); + if (!ioarea) { + dev_err(&pdev->dev, "I2S probe: Memory region already claimed.\n"); + return -EBUSY; + } + + base[i] = devm_ioremap(&pdev->dev, mem->start, + resource_size(mem)); + if (!base[i]) { + dev_err(&pdev->dev, "I2S probe: ioremap failed.\n"); + return -ENOMEM; + } + } + + dev = devm_kzalloc(&pdev->dev, sizeof(struct bcm2708_i2s_dev), + GFP_KERNEL); + if (!dev) { + dev_err(&pdev->dev, "I2S probe: kzalloc failed.\n"); + return -ENOMEM; + } + + dev->i2s_base = base[0]; + dev->clk_base = base[1]; + + bcm2708_i2s_setup_gpio(); + + /* Set the appropriate DMA parameters */ + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].port_addr = + (dma_addr_t)BCM2708_I2S_FIFO_PHYSICAL_ADDR; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].port_addr = + (dma_addr_t)BCM2708_I2S_FIFO_PHYSICAL_ADDR; + + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_req = + BCM2708_DMA_DREQ_PCM_TX; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_req = + BCM2708_DMA_DREQ_PCM_RX; + + /* Store the pdev */ + dev->dev = &pdev->dev; + dev_set_drvdata(&pdev->dev, dev); + + return snd_soc_register_dai(&pdev->dev, &bcm2708_i2s_dai); +} + +static int bcm2708_i2s_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&pdev->dev); + return 0; +} + +static struct platform_driver bcm2708_i2s_driver = { + .probe = bcm2708_i2s_probe, + .remove = bcm2708_i2s_remove, + .driver = { + .name = "bcm2708-i2s", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(bcm2708_i2s_driver); + +MODULE_AUTHOR("Florian Meier"); +MODULE_DESCRIPTION("BCM2708 I2S Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/bcm2708/bcm2708-pcm.c b/sound/soc/bcm2708/bcm2708-pcm.c new file mode 100644 index 0000000..4b3e688 --- /dev/null +++ b/sound/soc/bcm2708/bcm2708-pcm.c @@ -0,0 +1,303 @@ +/* + * ALSA PCM interface for Broadcom BCM2708 SoC + * + * Author: Florian Meier, <florian.meier(a)koalo.de> + * Copyright 2013 + * + * based on + * ALSA PCM interface for the OMAP SoC + * Copyright (C) 2008 Nokia Corporation + * Contact: Jarkko Nikula <jarkko.nikula(a)bitmer.com> + * Peter Ujfalusi <peter.ujfalusi(a)ti.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + * TODO This should be transferred to the generic DMA engine driver + * as soon as the official BCM2708 kernel migrates to 3.10. + * Otherwise, it would be difficult to test. + */ + +#include <linux/dma-mapping.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/bcm2708-dmaengine.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/dmaengine_pcm.h> +#include <sound/soc.h> + +#include "bcm2708-pcm.h" + +static const struct snd_pcm_hardware bcm2708_pcm_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_JOINT_DUPLEX | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .period_bytes_min = 32, + .period_bytes_max = 64 * 1024, + .periods_min = 2, + .periods_max = 255, + .buffer_bytes_max = 128 * 1024, +}; + +static int bcm2708_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct bcm2708_pcm_dma_data *dma_data; + struct bcm2708_dma_slave_config config; + struct dma_chan *chan; + int err = 0; + + /* get DMA data (e.g. FIFO address and DREQ) */ + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + /* + * There seems to be no use for bufferless + * transfer with this SoC. + */ + if (!dma_data) + return 0; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + chan = snd_dmaengine_pcm_get_chan(substream); + if (!chan) + return -EINVAL; + + /* fills in addr_width and direction */ + err = snd_hwparams_to_dma_slave_config(substream, params, &config.cfg); + if (err) + return err; + + /* Override the *_dma addr_width if requested by the DAI driver */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + config.cfg.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + else + config.cfg.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + + config.cfg.src_addr = dma_data->port_addr; + config.cfg.dst_addr = dma_data->port_addr; + config.cfg.src_maxburst = dma_data->packet_size; + config.cfg.dst_maxburst = dma_data->packet_size; + config.sync_dreq = dma_data->dma_req; + + return dmaengine_slave_config(chan, &config.cfg); +} + +static int bcm2708_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + return 0; +} + +static int bcm2708_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct bcm2708_pcm_dma_data *dma_data; + int ret = 0; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + break; + default: + ret = -EINVAL; + } + + if (ret == 0) + ret = snd_dmaengine_pcm_trigger(substream, cmd); + + return ret; +} + +static snd_pcm_uframes_t bcm2708_pcm_pointer( + struct snd_pcm_substream *substream) +{ + return snd_dmaengine_pcm_pointer(substream); +} + +static int bcm2708_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct bcm2708_pcm_dma_data *dma_data; + int ret; + + snd_soc_set_runtime_hwparams(substream, &bcm2708_pcm_hardware); + + /* Ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + return snd_dmaengine_pcm_open(substream, NULL, NULL); +} + +static int bcm2708_pcm_close(struct snd_pcm_substream *substream) +{ + snd_dmaengine_pcm_close(substream); + return 0; +} + +static int bcm2708_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops bcm2708_pcm_ops = { + .open = bcm2708_pcm_open, + .close = bcm2708_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = bcm2708_pcm_hw_params, + .hw_free = bcm2708_pcm_hw_free, + .trigger = bcm2708_pcm_trigger, + .pointer = bcm2708_pcm_pointer, + .mmap = bcm2708_pcm_mmap, +}; + +static u64 bcm2708_pcm_dmamask = DMA_BIT_MASK(64); + +static int bcm2708_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, + int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = bcm2708_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} + +static void bcm2708_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static int bcm2708_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &bcm2708_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(64); + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + ret = bcm2708_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = bcm2708_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + +out: + /* free preallocated buffers in case of error */ + if (ret) + bcm2708_pcm_free_dma_buffers(pcm); + + return ret; +} + +static struct snd_soc_platform_driver bcm2708_soc_platform = { + .ops = &bcm2708_pcm_ops, + .pcm_new = bcm2708_pcm_new, + .pcm_free = bcm2708_pcm_free_dma_buffers, +}; + +static int bcm2708_pcm_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, + &bcm2708_soc_platform); +} + +static int bcm2708_pcm_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver bcm2708_pcm_driver = { + .driver = { + .name = "bcm2708-pcm-audio", + .owner = THIS_MODULE, + }, + + .probe = bcm2708_pcm_probe, + .remove = bcm2708_pcm_remove, +}; + +module_platform_driver(bcm2708_pcm_driver); + +MODULE_AUTHOR("Florian Meier"); +MODULE_DESCRIPTION("BCM2708 PCM DMA module"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:bcm2708-pcm-audio"); diff --git a/sound/soc/bcm2708/bcm2708-pcm.h b/sound/soc/bcm2708/bcm2708-pcm.h new file mode 100644 index 0000000..e6a6acb --- /dev/null +++ b/sound/soc/bcm2708/bcm2708-pcm.h @@ -0,0 +1,40 @@ +/* + * ALSA PCM interface for Broadcom BCM2708 SoC + * + * Author: Florian Meier, <florian.meier(a)koalo.de> + * Copyright 2013 + * + * based on + * ALSA PCM interface for the OMAP SoC + * Copyright (C) 2008 Nokia Corporation + * Contact: Jarkko Nikula <jarkko.nikula(a)bitmer.com> + * Peter Ujfalusi <peter.ujfalusi(a)ti.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef __BCM2708_PCM_H__ +#define __BCM2708_PCM_H__ + +struct snd_pcm_substream; + +struct bcm2708_pcm_dma_data { + char *name; /* stream identifier */ + int dma_req; /* DMA request line */ + unsigned long port_addr; /* transmit/receive register */ + int packet_size; +}; + +#endif diff --git a/sound/soc/bcm2708/rpi-mbed.c b/sound/soc/bcm2708/rpi-mbed.c new file mode 100644 index 0000000..cb8d25f --- /dev/null +++ b/sound/soc/bcm2708/rpi-mbed.c @@ -0,0 +1,113 @@ +/* + * ASoC driver for mbed AudioCODEC (with a TLV320AIC23B) + * connected to a Raspberry Pi + * + * Author: Florian Meier, <florian.meier(a)koalo.de> + * Copyright 2013 + * + * 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/module.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> + +#include "../codecs/tlv320aic23.h" + +static int snd_rpi_mbed_init(struct snd_soc_pcm_runtime *rtd) +{ + return 0; +} + +static int snd_rpi_mbed_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int sysclk; + + sysclk = 12000000; /* this is fixed on this board */ + + /* set tlv320aic23 sysclk */ + snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, 0); + + return 0; +} + +/* machine stream operations */ +static struct snd_soc_ops snd_rpi_mbed_ops = { + .hw_params = snd_rpi_mbed_hw_params, +}; + +static struct snd_soc_dai_link snd_rpi_mbed_dai[] = { +{ + .name = "TLV320AIC23", + .stream_name = "TLV320AIC23 HiFi", + .cpu_dai_name = "bcm2708-i2s.0", + .codec_dai_name = "tlv320aic23-hifi", + .platform_name = "bcm2708-pcm-audio.0", + .codec_name = "tlv320aic23-codec.1-001b", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ops = &snd_rpi_mbed_ops, + .init = snd_rpi_mbed_init, +}, +}; + +/* audio machine driver */ +static struct snd_soc_card snd_rpi_mbed = { + .name = "snd_rpi_mbed", + .dai_link = snd_rpi_mbed_dai, + .num_links = ARRAY_SIZE(snd_rpi_mbed_dai), +}; + +static int snd_rpi_mbed_probe(struct platform_device *pdev) +{ + int ret = 0; + + snd_rpi_mbed.dev = &pdev->dev; + ret = snd_soc_register_card(&snd_rpi_mbed); + if (ret) { + dev_err(&pdev->dev, + "snd_soc_register_card() failed: %d\n", ret); + } + + return ret; +} + + +static int snd_rpi_mbed_remove(struct platform_device *pdev) +{ + return snd_soc_unregister_card(&snd_rpi_mbed); +} + +static struct platform_driver snd_rpi_mbed_driver = { + .driver = { + .name = "snd-rpi-mbed", + .owner = THIS_MODULE, + }, + .probe = snd_rpi_mbed_probe, + .remove = snd_rpi_mbed_remove, +}; + +module_platform_driver(snd_rpi_mbed_driver); + +MODULE_AUTHOR("Florian Meier"); +MODULE_DESCRIPTION("ASoC Driver for Raspberry Pi connected to mbed AudioCODEC"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/bcm2708/rpi-proto.c b/sound/soc/bcm2708/rpi-proto.c new file mode 100644 index 0000000..25be85a --- /dev/null +++ b/sound/soc/bcm2708/rpi-proto.c @@ -0,0 +1,131 @@ +/* + * ASoC driver for PROTO AudioCODEC (with a WM8731) + * connected to a Raspberry Pi + * + * Author: Florian Meier, <florian.meier(a)koalo.de> + * Copyright 2013 + * + * 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/module.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> + +#include "../codecs/wm8731.h" + +static const unsigned int wm8731_rates_12288000[] = { + 8000, 32000, 48000, 96000, +}; + +static struct snd_pcm_hw_constraint_list wm8731_constraints_12288000 = { + .list = wm8731_rates_12288000, + .count = ARRAY_SIZE(wm8731_rates_12288000), +}; + +static int snd_rpi_proto_startup(struct snd_pcm_substream *substream) +{ + /* Setup constraints, because there is a 12.288 MHz XTAL on the board */ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &wm8731_constraints_12288000); + return 0; +} + +static int snd_rpi_proto_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int sysclk = 12288000; /* This is fixed on this board */ + + /* Set proto sysclk */ + int ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, + sysclk, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(substream->pcm->dev, + "Failed to set WM8731 SYSCLK: %d\n", ret); + return ret; + } + + return 0; +} + +/* machine stream operations */ +static struct snd_soc_ops snd_rpi_proto_ops = { + .startup = snd_rpi_proto_startup, + .hw_params = snd_rpi_proto_hw_params, +}; + +static struct snd_soc_dai_link snd_rpi_proto_dai[] = { +{ + .name = "WM8731", + .stream_name = "WM8731 HiFi", + .cpu_dai_name = "bcm2708-i2s.0", + .codec_dai_name = "wm8731-hifi", + .platform_name = "bcm2708-pcm-audio.0", + .codec_name = "wm8731.1-001a", + .dai_fmt = SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ops = &snd_rpi_proto_ops, +}, +}; + +/* audio machine driver */ +static struct snd_soc_card snd_rpi_proto = { + .name = "snd_rpi_proto", + .dai_link = snd_rpi_proto_dai, + .num_links = ARRAY_SIZE(snd_rpi_proto_dai), +}; + +static int snd_rpi_proto_probe(struct platform_device *pdev) +{ + int ret = 0; + + snd_rpi_proto.dev = &pdev->dev; + ret = snd_soc_register_card(&snd_rpi_proto); + if (ret) { + dev_err(&pdev->dev, + "snd_soc_register_card() failed: %d\n", ret); + } + + return ret; +} + + +static int snd_rpi_proto_remove(struct platform_device *pdev) +{ + return snd_soc_unregister_card(&snd_rpi_proto); +} + +static struct platform_driver snd_rpi_proto_driver = { + .driver = { + .name = "snd-rpi-proto", + .owner = THIS_MODULE, + }, + .probe = snd_rpi_proto_probe, + .remove = snd_rpi_proto_remove, +}; + +module_platform_driver(snd_rpi_proto_driver); + +MODULE_AUTHOR("Florian Meier"); +MODULE_DESCRIPTION("ASoC Driver for Raspberry Pi connected to PROTO board (WM8731)"); +MODULE_LICENSE("GPL"); -- 1.7.9.5

4 9

[alsa-devel] [PATCH alsa-lib] Allow specifying the max number of cards
by Takashi Iwai 24 May '13

24 May '13

Add --with-max-cards option to specify the max number of cards in configure script, when the support for more than 32 cards is required. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- configure.in | 14 ++++++++++++++ src/confmisc.c | 2 +- src/control/cards.c | 6 +++--- src/control/control_hw.c | 2 +- src/hwdep/hwdep_hw.c | 2 +- 5 files changed, 20 insertions(+), 6 deletions(-) diff --git a/configure.in b/configure.in index 35b8e84..3d394fc 100644 --- a/configure.in +++ b/configure.in @@ -632,6 +632,20 @@ for t in $CTL_PLUGIN_LIST; do fi done +dnl Max number of cards +AC_MSG_CHECKING(for max number of cards) +AC_ARG_WITH(max-cards, + AS_HELP_STRING([--with-max-cards], [Specify the max number of cards (default = 32)]), + [ max_cards="$withval" ], [ max_cards="32" ]) +AC_MSG_RESULT([$max_cards]) + +if test "$max_cards" -lt 1; then + AC_ERROR([Invalid max cards $max_cards]) +elif test "$max_cards" -gt 256; then + AC_ERROR([Invalid max cards $max_cards]) +fi +AC_DEFINE_UNQUOTED(SND_MAX_CARDS, $max_cards, [Max number of cards]) + dnl Make a symlink for inclusion of alsa/xxx.h if test ! -L "$srcdir"/include/alsa ; then echo "Making a symlink include/alsa" diff --git a/src/confmisc.c b/src/confmisc.c index 80b0027..af686be 100644 --- a/src/confmisc.c +++ b/src/confmisc.c @@ -668,7 +668,7 @@ int snd_determine_driver(int card, char **driver) char *res = NULL; int err; - assert(card >= 0 && card <= 32); + assert(card >= 0 && card <= SND_MAX_CARDS); err = open_ctl(card, &ctl); if (err < 0) { SNDERR("could not open control for card %i", card); diff --git a/src/control/cards.c b/src/control/cards.c index b528e33..5d7376c 100644 --- a/src/control/cards.c +++ b/src/control/cards.c @@ -103,7 +103,7 @@ int snd_card_next(int *rcard) return -EINVAL; card = *rcard; card = card < 0 ? 0 : card + 1; - for (; card < 32; card++) { + for (; card < SND_MAX_CARDS; card++) { if (snd_card_load(card)) { *rcard = card; return 0; @@ -134,7 +134,7 @@ int snd_card_get_index(const char *string) (isdigit(*string) && isdigit(*(string + 1)) && *(string + 2) == 0)) { if (sscanf(string, "%i", &card) != 1) return -EINVAL; - if (card < 0 || card > 31) + if (card < 0 || card >= SND_MAX_CARDS) return -EINVAL; err = snd_card_load1(card); if (err >= 0) @@ -143,7 +143,7 @@ int snd_card_get_index(const char *string) } if (string[0] == '/') /* device name */ return snd_card_load2(string); - for (card = 0; card < 32; card++) { + for (card = 0; card < SND_MAX_CARDS; card++) { #ifdef SUPPORT_ALOAD if (! snd_card_load(card)) continue; diff --git a/src/control/control_hw.c b/src/control/control_hw.c index 90c4ba7..148097f 100644 --- a/src/control/control_hw.c +++ b/src/control/control_hw.c @@ -382,7 +382,7 @@ int snd_ctl_hw_open(snd_ctl_t **handle, const char *name, int card, int mode) *handle = NULL; - if (CHECK_SANITY(card < 0 || card >= 32)) { + if (CHECK_SANITY(card < 0 || card >= SND_MAX_CARDS)) { SNDMSG("Invalid card index %d", card); return -EINVAL; } diff --git a/src/hwdep/hwdep_hw.c b/src/hwdep/hwdep_hw.c index e4fcdc3..4314e32 100644 --- a/src/hwdep/hwdep_hw.c +++ b/src/hwdep/hwdep_hw.c @@ -112,7 +112,7 @@ int snd_hwdep_hw_open(snd_hwdep_t **handle, const char *name, int card, int devi *handle = NULL; - if (card < 0 || card >= 32) + if (card < 0 || card >= SND_MAX_CARDS) return -EINVAL; sprintf(filename, SNDRV_FILE_HWDEP, card, device); fd = snd_open_device(filename, mode); -- 1.8.2.3

1 0

[alsa-devel] [PATCH 1/2] ALSA: Fix the default suffix string with high card number
by Takashi Iwai 24 May '13

24 May '13

ALSA core tries to add a suffix as "_1" automatically when the given id string conflicts. The current code assumes implicitly that the max card number is 16 so that the single hex "_X" suffix can be put. However, with the dynamic device management, the card can be at most 32, so it can put even a non-hex character there. Also, when the max card number is increased in future, this would result in worse. This patch rewrites the code to add the suffix string in a simpler (thus cleaner) way. It can support up to three digits, so it should suffice for most requirements. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/init.c | 30 +++++++++++++----------------- 1 file changed, 13 insertions(+), 17 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 6ef0640..ed4a481 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -549,7 +549,6 @@ static void snd_card_set_id_no_lock(struct snd_card *card, const char *src, const char *nid) { int len, loops; - bool with_suffix; bool is_default = false; char *id; @@ -565,26 +564,23 @@ static void snd_card_set_id_no_lock(struct snd_card *card, const char *src, is_default = true; } - with_suffix = false; + len = strlen(id); for (loops = 0; loops < SNDRV_CARDS; loops++) { + char *spos; + char sfxstr[5]; /* "_012" */ + int sfxlen; + if (card_id_ok(card, id)) return; /* OK */ - len = strlen(id); - if (!with_suffix) { - /* add the "_X" suffix */ - char *spos = id + len; - if (len > sizeof(card->id) - 3) - spos = id + sizeof(card->id) - 3; - strcpy(spos, "_1"); - with_suffix = true; - } else { - /* modify the existing suffix */ - if (id[len - 1] != '9') - id[len - 1]++; - else - id[len - 1] = 'A'; - } + /* Add _XYZ suffix */ + sprintf(sfxstr, "_%X", loops + 1); + sfxlen = strlen(sfxstr); + if (len + sfxlen >= sizeof(card->id)) + spos = id + sizeof(card->id) - sfxlen - 1; + else + spos = id + len; + strcpy(spos, sfxstr); } /* fallback to the default id */ if (!is_default) { -- 1.8.2.3

1 1

[alsa-devel] [PATCH] ALSA: Constify the snd_pcm_substream struct ops field
by Lars-Peter Clausen 24 May '13

24 May '13

The ops field of the snd_pcm_substream struct is never modified inside the ALSA core. Making it const allows drivers to declare their snd_pcm_ops struct as const. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/pcm.h | 5 +++-- sound/core/pcm_lib.c | 3 ++- 2 files changed, 5 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index b48792f..84b10f9 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -384,7 +384,7 @@ struct snd_pcm_substream { unsigned int dma_buf_id; size_t dma_max; /* -- hardware operations -- */ - struct snd_pcm_ops *ops; + const struct snd_pcm_ops *ops; /* -- runtime information -- */ struct snd_pcm_runtime *runtime; /* -- timer section -- */ @@ -871,7 +871,8 @@ const unsigned char *snd_pcm_format_silence_64(snd_pcm_format_t format); int snd_pcm_format_set_silence(snd_pcm_format_t format, void *buf, unsigned int frames); snd_pcm_format_t snd_pcm_build_linear_format(int width, int unsigned, int big_endian); -void snd_pcm_set_ops(struct snd_pcm * pcm, int direction, struct snd_pcm_ops *ops); +void snd_pcm_set_ops(struct snd_pcm * pcm, int direction, + const struct snd_pcm_ops *ops); void snd_pcm_set_sync(struct snd_pcm_substream *substream); int snd_pcm_lib_interleave_len(struct snd_pcm_substream *substream); int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream, diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 41b3dfe..82bb029 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -568,7 +568,8 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) * * Sets the given PCM operators to the pcm instance. */ -void snd_pcm_set_ops(struct snd_pcm *pcm, int direction, struct snd_pcm_ops *ops) +void snd_pcm_set_ops(struct snd_pcm *pcm, int direction, + const struct snd_pcm_ops *ops) { struct snd_pcm_str *stream = &pcm->streams[direction]; struct snd_pcm_substream *substream; -- 1.8.0

2 1

[alsa-devel] My ALSA amidicat program
by Josh Lehan 24 May '13

24 May '13

Hello! A few years ago, we talked on the alsa-devel mailing list. I had written a program "amidicat" which was well received. It's been a while, and I just now remembered it again. http://mailman.alsa-project.org/pipermail/alsa-devel/2010-June/028446.html You mentioned that such a program might be useful to have in alsa-utils. Is this still the case? If so, I would be honored to get my program in there. http://mailman.alsa-project.org/pipermail/alsa-devel/2010-January/024413.ht… Thanks! Josh Lehan

2 1

[alsa-devel] [PATCH] ucm: Fix segfault when device argument is missing from _sw{dev, mod}
by Juho Hämäläinen 24 May '13

24 May '13

Signed-off-by: Juho Hämäläinen <juho.hamalainen(a)tieto.com> --- src/ucm/main.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/src/ucm/main.c b/src/ucm/main.c index d38b7fa..74e1530 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -1649,6 +1649,10 @@ int snd_use_case_set(snd_use_case_mgr_t *uc_mgr, } else { str = NULL; } + if (!str) { + err = -EINVAL; + goto __end; + } if (check_identifier(identifier, "_swdev")) err = switch_device(uc_mgr, str, value); else if (check_identifier(identifier, "_swmod")) -- 1.7.10.4

2 3

[alsa-devel] [PATCH] ASoC: wm5110: Correct DSP4R Mixer control name
by Charles Keepax 24 May '13

24 May '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index c00480b..ba38f06 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -190,7 +190,7 @@ ARIZONA_MIXER_CONTROLS("DSP2R", ARIZONA_DSP2RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP3L", ARIZONA_DSP3LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP3R", ARIZONA_DSP3RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP4L", ARIZONA_DSP4LMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DSP5R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DSP4R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), -- 1.7.2.5

1 0

[alsa-devel] [PATCH 4/4 V2] ALSA: hda - Continue probe in work context to avoid request_module deadlock
by Wang Xingchao 24 May '13

24 May '13

There's deadlock when request_module(i915) in azx_probe. It looks like: device_lock(audio pci device) -> azx_probe -> module_request (or symbol_request) -> modprobe (userspace) -> i915 init -> drm_pci_init -> pci_register_driver -> bus_add_driver -> driver_attach -> which in turn tries all locks on pci bus, and when it tries the one on the audio device, it will deadlock. This patch introduce a work to store remaining probe stuff, and let request_module run in safe work context. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/hda_i915.c | 13 ++++-- sound/pci/hda/hda_intel.c | 105 +++++++++++++++++++++++++++------------------- 2 files changed, 71 insertions(+), 47 deletions(-) diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c index 76c13d5..7547b20 100644 --- a/sound/pci/hda/hda_i915.c +++ b/sound/pci/hda/hda_i915.c @@ -42,13 +42,18 @@ int hda_i915_init(void) { int err = 0; - get_power = symbol_request(i915_request_power_well); + get_power = symbol_get(i915_request_power_well); if (!get_power) { - snd_printk(KERN_WARNING "hda-i915: get_power symbol get fail\n"); - return -ENODEV; + request_module("i915"); + get_power = symbol_get(i915_request_power_well); + if (!get_power) { + snd_printk(KERN_WARNING "hda-i915: get_power symbol get fail\n"); + return -ENODEV; + } + snd_printdd("hda-i915: get_power symbol get successful\n"); } - put_power = symbol_request(i915_release_power_well); + put_power = symbol_get(i915_release_power_well); if (!put_power) { symbol_put(i915_request_power_well); get_power = NULL; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f20a88c..1bc7c3b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -76,6 +76,7 @@ static int probe_only[SNDRV_CARDS]; static int jackpoll_ms[SNDRV_CARDS]; static bool single_cmd; static int enable_msi = -1; +static int dev; #ifdef CONFIG_SND_HDA_PATCH_LOADER static char *patch[SNDRV_CARDS]; #endif @@ -542,6 +543,8 @@ struct azx { /* for pending irqs */ struct work_struct irq_pending_work; + struct delayed_work probe_work; + /* reboot notifier (for mysterious hangup problem at power-down) */ struct notifier_block reboot_notifier; @@ -3670,58 +3673,22 @@ static void azx_firmware_cb(const struct firmware *fw, void *context) } #endif -static int azx_probe(struct pci_dev *pci, - const struct pci_device_id *pci_id) +static void azx_probe_work(struct work_struct *work) { - static int dev; - struct snd_card *card; - struct azx *chip; + struct azx *chip = + container_of(work, struct azx, probe_work.work); + struct snd_card *card = chip->card; + struct pci_dev *pci = chip->pci; bool probe_now; int err; - if (dev >= SNDRV_CARDS) - return -ENODEV; - if (!enable[dev]) { - dev++; - return -ENOENT; - } - - err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); - if (err < 0) { - snd_printk(KERN_ERR "hda-intel: Error creating card!\n"); - return err; - } - - snd_card_set_dev(card, &pci->dev); - - err = azx_create(card, pci, dev, pci_id->driver_data, &chip); - if (err < 0) - goto out_free; - card->private_data = chip; - - pci_set_drvdata(pci, card); - - err = register_vga_switcheroo(chip); - if (err < 0) { - snd_printk(KERN_ERR SFX - "%s: Error registering VGA-switcheroo client\n", pci_name(pci)); - goto out_free; - } - - if (check_hdmi_disabled(pci)) { - snd_printk(KERN_INFO SFX "%s: VGA controller is disabled\n", - pci_name(pci)); - snd_printk(KERN_INFO SFX "%s: Delaying initialization\n", pci_name(pci)); - chip->disabled = true; - } - /* Request power well for Haswell HDA controller and codec */ if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { #ifdef CONFIG_SND_HDA_I915 err = hda_i915_init(); if (err < 0) { snd_printk(KERN_ERR SFX "Error request power-well from i915\n"); - return err; + goto out_free; } hda_display_power(true); #else @@ -3760,7 +3727,7 @@ static int azx_probe(struct pci_dev *pci, dev++; complete_all(&chip->probe_wait); - return 0; + return; out_free_power: if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { hda_display_power(false); @@ -3769,6 +3736,58 @@ out_free_power: out_free: snd_card_free(card); pci_set_drvdata(pci, NULL); +} + +static int azx_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + struct snd_card *card; + struct azx *chip; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) { + snd_printk(KERN_ERR "hda-intel: Error creating card!\n"); + return err; + } + + snd_card_set_dev(card, &pci->dev); + + err = azx_create(card, pci, dev, pci_id->driver_data, &chip); + if (err < 0) + goto out_free; + card->private_data = chip; + + pci_set_drvdata(pci, card); + + err = register_vga_switcheroo(chip); + if (err < 0) { + snd_printk(KERN_ERR SFX + "%s: Error registering VGA-switcheroo client\n", pci_name(pci)); + goto out_free; + } + + if (check_hdmi_disabled(pci)) { + snd_printk(KERN_INFO SFX "%s: VGA controller is disabled\n", + pci_name(pci)); + snd_printk(KERN_INFO SFX "%s: Delaying initialization\n", pci_name(pci)); + chip->disabled = true; + } + + /* continue probing in work context as may trigger request module */ + INIT_DELAYED_WORK(&chip->probe_work, azx_probe_work); + schedule_delayed_work(&chip->probe_work, 0); + return 0; +out_free: + snd_card_free(card); + pci_set_drvdata(pci, NULL); return err; } -- 1.8.1.2

4 6

Re: [alsa-devel] [PATCH 00/33] devm improvement series, part 1, take 2
by Thierry Reding 23 May '13

23 May '13

On Thu, May 16, 2013 at 01:15:28PM +0200, Wolfram Sang wrote: > Lately, I have been experimenting how to improve the devm interface to make > writing device drivers easier and less error prone while also getting rid of > its subtle issues. I think it has more potential but still needs work and > definately conistency, especiall in its usage. > > The first thing I come up with is a low hanging fruit regarding > devm_ioremap_resouce(). This function already checks if the passed resource is > valid and gives an error message if not. So, we can remove similar checks from > the drivers and get rid of a bit of code and a number of inconsistent error > strings. Sorry for jumping in so late. I generally like the idea. One small inconvenience is that devm_ioremap_resource() returns -EINVAL if res == NULL, which means that drivers will now also return -EINVAL in cases where no resource was returned. Typically drivers handle this by returning something like -ENODEV, -ENXIO, -ENOENT. Some do return -EINVAL but perhaps having a separate error code (and maybe error message as well) for a missing resource would be helpful. Doing this would be rather easy now that you've paved the way by making devm_ioremap_resource() usage consistent across drivers. Thierry

1 0

[alsa-devel] [PATCH 0/4 V5] Power-well API implementation for Haswell
by Wang Xingchao 23 May '13

23 May '13

Hi all, This is V5 and here're some changes notes: change from V4-->V5: - fix reference count bug - new patch on general runtime pm support for audio pci device - new patch to avoid request_module() deadlock change between V3-->V4: - add new structure i915_power_well - initialize drm_device pointer at module init time - change function name change between V2-->V3: - make SND_HDA_I915 selectable - use snd_printdd to output message - add return error code check - use symbol_request to replace symbol_get - release power_well at azx_free - some typo fixes changes between V1-->V2: - use reference count to track power-well usage - remove external module, compiled into snd-hda-intel instead - manage symbols and module loading properly - remove IS_HSW macro, use flag instead - remove audio callback for gfx driver to avoid dependency - split whole patch into two pieces for easy review - more typo fixes Wang Xingchao (4): i915/drm: Add private api for power well usage ALSA: hda - Add power-welll support for haswell HDA ALSA: hda - Fix runtime PM check ALSA: hda - Continue probe in work context to avoid request_module deadlock drivers/gpu/drm/i915/i915_dma.c | 6 ++ drivers/gpu/drm/i915/i915_drv.h | 12 ++++ drivers/gpu/drm/i915/intel_drv.h | 4 ++ drivers/gpu/drm/i915/intel_pm.c | 92 ++++++++++++++++++++++++--- include/drm/i915_powerwell.h | 36 +++++++++++ sound/pci/hda/Kconfig | 10 +++ sound/pci/hda/Makefile | 3 + sound/pci/hda/hda_i915.c | 75 ++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 35 +++++++++++ sound/pci/hda/hda_intel.c | 132 ++++++++++++++++++++++++++++----------- 10 files changed, 360 insertions(+), 45 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.8.1.2

4 12

[alsa-devel] [PATCH 1/2] ASoC: codecs: adau1701: refactor firmware loading function
by Daniel Mack 23 May '13

23 May '13

Pass a struct i2c_client * to adau1701_load_firmware directly to make the code more readable. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/adau1701.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index dafdbe8..95e1677 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -180,9 +180,9 @@ static unsigned int adau1701_read(struct snd_soc_codec *codec, unsigned int reg) return value; } -static int adau1701_load_firmware(struct snd_soc_codec *codec) +static int adau1701_load_firmware(struct i2c_client *client) { - return process_sigma_firmware(codec->control_data, ADAU1701_FIRMWARE); + return process_sigma_firmware(client, ADAU1701_FIRMWARE); } static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec, @@ -455,10 +455,11 @@ static struct snd_soc_dai_driver adau1701_dai = { static int adau1701_probe(struct snd_soc_codec *codec) { int ret; + struct i2c_client *client = to_i2c_client(codec->dev); - codec->control_data = to_i2c_client(codec->dev); + codec->control_data = client; - ret = adau1701_load_firmware(codec); + ret = adau1701_load_firmware(client); if (ret) dev_warn(codec->dev, "Failed to load firmware\n"); -- 1.8.1.4

3 4

[alsa-devel] [PATCH 0/4] CS42L52 fixes.
by Nicolas Schichan 23 May '13

23 May '13

Hi, In the following emails, you will find the fixes I currently have for the CS42L52 driver. The fixes are: - Set both Master Volume A & B init value to 0x88 (otherwise left channel and right channel master volume would not be the same after init). - Use IFACE_CTL2 register for Microphone Bias enum (IFACE_CTL1 is clearly not controling microphone bias according to the datasheet). - Various controls would use a bogus shift value, resulting in only the high order bits being updated when changing them. - The wrong mask was used when muting/unmuting the codec, resutling in the wrong part of the PB_CTL1 register being cleared, especially the headphone analog gain part. All those patches are against the latest linus tree. Regards, -- Nicolas Schichan

4 15

[alsa-devel] [PATCH] pcm: Fix typo in documentation: PREPARE → PREPARED.
by Jordi Mallach 23 May '13

23 May '13

Signed-off-by: Jordi Mallach <jordi(a)debian.org> --- src/pcm/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 438fb3f..7ec1f0e 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -437,7 +437,7 @@ to #SND_PCM_STATE_SETUP if successfully finishes, otherwise the state #SND_PCM_STATE_OPEN is entered. When it is brought to SETUP state, this function automatically -calls #snd_pcm_prepare() function to bring to the PREPARE state +calls #snd_pcm_prepare() function to bring to the PREPARED state as below. \par snd_pcm_prepare -- 1.7.10.4

2 1

[alsa-devel] [PATCH alsa-utils] Require alsa-lib 1.0.27.
by Jordi Mallach 23 May '13

23 May '13

Both aplay and speaker-test use the new snd_pcm_abort() function, which was introduced in alsa-lib 1.0.27. Signed-off-by: Jordi Mallach <jordi(a)debian.org> --- configure.in | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.in b/configure.in index aae3e66..0023e4d 100644 --- a/configure.in +++ b/configure.in @@ -32,7 +32,7 @@ AC_PROG_MKDIR_P AC_PROG_LN_S AC_PROG_SED PKG_PROG_PKG_CONFIG -AM_PATH_ALSA(1.0.24) +AM_PATH_ALSA(1.0.27) if test "x$enable_alsatest" = "xyes"; then AC_CHECK_FUNC([snd_ctl_elem_add_enumerated], , [AC_ERROR([No user enum control support in alsa-lib])]) -- 1.7.10.4

2 1

[alsa-devel] [PATCH] usb-6fire: Modify firmware version check
by Torsten Schenk 23 May '13

23 May '13

Check only the uppermost 16 bits instead of the whole 32 bits of the version information. Do this because all firmware version tested with this version information worked correctly and the strict check causes problems for several users. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c --- a/sound/usb/6fire/firmware.c 2013-05-23 13:32:39.432545897 +0200 +++ b/sound/usb/6fire/firmware.c 2013-05-23 13:33:21.998193012 +0200 @@ -42,8 +42,8 @@ 0x94, 0x01, 0x5c, 0x02 /* alt 3: 404 EP2 and 604 EP6 (25 fpp) */ }; -static const u8 known_fw_versions[][4] = { - { 0x03, 0x01, 0x0b, 0x00 } +static const u8 known_fw_versions[][2] = { + { 0x03, 0x01 } }; struct ihex_record { @@ -343,7 +343,7 @@ int i; for (i = 0; i < ARRAY_SIZE(known_fw_versions); i++) - if (!memcmp(version, known_fw_versions + i, 4)) + if (!memcmp(version, known_fw_versions + i, 2)) return 0; snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. "

2 1

[alsa-devel] [PATCH 01/11] ALSA: pxa2xx-ac97: Remove redundant platform_set_drvdata()
by Sachin Kamat 23 May '13

23 May '13

Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> Cc: Eric Miao <eric.y.miao(a)gmail.com> --- sound/arm/pxa2xx-ac97.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index ec54be4..ce431e6 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -230,7 +230,6 @@ static int pxa2xx_ac97_remove(struct platform_device *dev) if (card) { snd_card_free(card); - platform_set_drvdata(dev, NULL); pxa2xx_ac97_hw_remove(dev); } -- 1.7.9.5

3 12

[alsa-devel] [PATCH] ASoC: fsl: expand the size of the name in fsl_ssi_private struct
by Nicolin Chen 23 May '13

23 May '13

strcpy(ssi_private->name, p) in probe() sets "name" by "p", gotten from dts, while the length of "p", if the devicetree node name of SSI is commonly set, would always be larger than 1 char size, so need a larger size for "name". Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2f2d837..b6a5f94 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -152,5 +152,5 @@ struct fsl_ssi_private { } stats; - char name[1]; + char name[32]; }; -- 1.7.1

3 2

[alsa-devel] [TINYCOMPRESS][PATCH 1/1] crec: Initial version of a compressed capture utility
by Richard Fitzgerald 23 May '13

23 May '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- Android.mk | 10 ++ crec.c | 339 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ makefile.linux | 20 ++- 3 files changed, 362 insertions(+), 7 deletions(-) create mode 100644 crec.c diff --git a/Android.mk b/Android.mk index b9c52e2..5baccaf 100644 --- a/Android.mk +++ b/Android.mk @@ -20,3 +20,13 @@ LOCAL_MODULE_TAGS := optional include $(BUILD_EXECUTABLE) +include $(CLEAR_VARS) + +LOCAL_C_INCLUDES:= $(LOCAL_PATH)/include +LOCAL_SRC_FILES:= crec.c +LOCAL_MODULE := crec +LOCAL_SHARED_LIBRARIES:= libcutils libutils libtinycompress +LOCAL_MODULE_TAGS := optional + +include $(BUILD_EXECUTABLE) + diff --git a/crec.c b/crec.c new file mode 100644 index 0000000..4f13878 --- /dev/null +++ b/crec.c @@ -0,0 +1,339 @@ +/* + * BSD LICENSE + * + * tinyplay command line player for compress audio offload in alsa + * Copyright (c) 2011-2012, Intel Corporation + * All rights reserved. + * + * Author: Vinod Koul <vinod.koul(a)linux.intel.com> + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions are met: + * + * Redistributions of source code must retain the above copyright notice, + * this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * Neither the name of Intel Corporation nor the names of its contributors + * may be used to endorse or promote products derived from this software + * without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + * LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF + * THE POSSIBILITY OF SUCH DAMAGE. + * + * LGPL LICENSE + * + * tinyplay command line player for compress audio offload in alsa + * Copyright (c) 2011-2012, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU Lesser General Public License, + * version 2.1, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public + * License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to + * the Free Software Foundation, Inc., + * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include <stdint.h> +#include <linux/types.h> +#include <fcntl.h> +#include <errno.h> +#include <unistd.h> +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <signal.h> +#include <stdbool.h> +#include <getopt.h> +#include <sys/time.h> +#define __force +#define __bitwise +#define __user +#include "sound/compress_params.h" +#include "tinycompress/tinycompress.h" + +static int verbose; + +static void usage(void) +{ + fprintf(stderr, "usage: crec [OPTIONS] filename\n" + "-c\tcard number\n" + "-d\tdevice node\n" + "-b\tbuffer size\n" + "-f\tfragments\n\n" + "-v\tverbose mode\n" + "-h\tPrints this help list\n\n" + "Example:\n" + "\tcrec -c 1 -d 2 test.raw\n" + "\tcrec -f 5 test.raw\n"); + + exit(EXIT_FAILURE); +} + +static int sample_rate_to_alsa(unsigned int *alsa_rate, int rate) +{ + switch (rate) { + case 5512: + *alsa_rate = SNDRV_PCM_RATE_5512; + return 0; + case 8000: + *alsa_rate = SNDRV_PCM_RATE_8000; + return 0; + case 11025: + *alsa_rate = SNDRV_PCM_RATE_11025; + return 0; + case 16000: + *alsa_rate = SNDRV_PCM_RATE_16000; + return 0; + case 220500: + *alsa_rate = SNDRV_PCM_RATE_22050; + return 0; + case 32000: + *alsa_rate = SNDRV_PCM_RATE_32000; + return 0; + case 44100: + *alsa_rate = SNDRV_PCM_RATE_44100; + return 0; + case 48000: + *alsa_rate = SNDRV_PCM_RATE_48000; + return 0; + case 64000: + *alsa_rate = SNDRV_PCM_RATE_64000; + return 0; + case 88200: + *alsa_rate = SNDRV_PCM_RATE_88200; + return 0; + case 96000: + *alsa_rate = SNDRV_PCM_RATE_96000; + return 0; + case 176400: + *alsa_rate = SNDRV_PCM_RATE_176400; + return 0; + case 192000: + *alsa_rate = SNDRV_PCM_RATE_192000; + return 0; + default: + return -EINVAL; + } +} + +static int alsa_to_sample_rate(unsigned int alsa_rate) +{ + switch (alsa_rate) { + case SNDRV_PCM_RATE_5512: + return 5512; + case SNDRV_PCM_RATE_8000: + return 8000; + case SNDRV_PCM_RATE_11025: + return 11025; + case SNDRV_PCM_RATE_16000: + return 16000; + case SNDRV_PCM_RATE_22050: + return 22050; + case SNDRV_PCM_RATE_32000: + return 32000; + case SNDRV_PCM_RATE_44100: + return 44100; + case SNDRV_PCM_RATE_48000: + return 48000; + case SNDRV_PCM_RATE_64000: + return 64000; + case SNDRV_PCM_RATE_88200: + return 88200; + case SNDRV_PCM_RATE_96000: + return 96000; + case SNDRV_PCM_RATE_176400: + return 176400; + case SNDRV_PCM_RATE_192000: + return 192000; + default: + return -EINVAL; + } +} + +static int print_time(struct compress *compress) +{ + unsigned int avail; + struct timespec tstamp; + + if (compress_get_hpointer(compress, &avail, &tstamp) != 0) { + fprintf(stderr, "Error querying timestamp\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + return -1; + } else + fprintf(stderr, "DSP played %jd.%jd\n", (intmax_t)tstamp.tv_sec, (intmax_t)tstamp.tv_nsec*1000); + return 0; +} + +void capture_samples(char *name, unsigned int card, unsigned int device, + unsigned long buffer_size, unsigned int frag, unsigned int rate) +{ + struct compr_config config; + struct snd_codec codec; + struct compress *compress; + FILE *file; + char *buffer; + int size, num_read, wrote; + + if (verbose) + printf("%s: entry\n", __func__); + file = fopen(name, "wb"); + if (!file) { + fprintf(stderr, "Unable to open file '%s'\n", name); + exit(EXIT_FAILURE); + } + + codec.id = SND_AUDIOCODEC_PCM; + codec.ch_in = 1; + codec.ch_out = 1; + if (sample_rate_to_alsa(&codec.sample_rate, rate) != 0) { + fprintf(stderr, "unknown sample rate %d\n", rate); + goto FILE_EXIT; + } + + codec.bit_rate = 0; + codec.rate_control = 0; + codec.profile = 0; + codec.level = 0; + codec.ch_mode = 0; + codec.format = 0; + if ((buffer_size != 0) && (frag != 0)) { + config.fragment_size = buffer_size/frag; + config.fragments = frag; + } else { + /* use driver defaults */ + config.fragment_size = 0; + config.fragments = 0; + } + config.codec = &codec; + + compress = compress_open(card, device, COMPRESS_OUT, &config); + if (!compress || !is_compress_ready(compress)) { + fprintf(stderr, "Unable to open Compress device %d:%d\n", + card, device); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto FILE_EXIT; + }; + if (verbose) + printf("%s: Opened compress device\n", __func__); + size = config.fragment_size; + buffer = malloc(size * config.fragments); + if (!buffer) { + fprintf(stderr, "Unable to allocate %d bytes\n", size); + goto COMP_EXIT; + } + + printf("Recording file %s On Card %u device %u, with buffer of %lu bytes\n", + name, card, device, buffer_size); + printf("Format %u Channels %u, %u Hz\n", + codec.id, codec.ch_out, + alsa_to_sample_rate(codec.sample_rate)); + + compress_start(compress); + if (verbose) + printf("%s: Capturing audio NOW!!!\n", __func__); + + do { + num_read = compress_read(compress, buffer, size); + if (num_read > 0) { + wrote = fwrite(buffer, 1, num_read, file); + if (wrote < 0) { + fprintf(stderr, "Error writing output file\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto BUF_EXIT; + } + if (wrote != num_read) { + /* TODO: Buffer pointer needs to be set here */ + fprintf(stderr, "DSP supplied %d, we wrote %d\n", num_read, wrote); + } + if (verbose) { + print_time(compress); + printf("%s: read %d\n", __func__, num_read); + } + } + } while (num_read > 0); + + if (verbose) + printf("%s: exit success\n", __func__); + free(buffer); + fclose(file); + compress_close(compress); + return; +BUF_EXIT: + free(buffer); +COMP_EXIT: + compress_close(compress); +FILE_EXIT: + fclose(file); + if (verbose) + printf("%s: exit failure\n", __func__); + exit(EXIT_FAILURE); +} + +int main(int argc, char **argv) +{ + char *file; + unsigned long buffer_size = 0; + int c; + unsigned int card = 0, device = 0, frag = 0, rate = 16000; + + + if (argc < 2) + usage(); + + verbose = 0; + while ((c = getopt(argc, argv, "rhvb:f:c:d:")) != -1) { + switch (c) { + case 'h': + usage(); + break; + case 'b': + buffer_size = strtol(optarg, NULL, 0); + break; + case 'f': + frag = strtol(optarg, NULL, 10); + break; + case 'c': + card = strtol(optarg, NULL, 10); + break; + case 'd': + device = strtol(optarg, NULL, 10); + break; + case 'r': + rate = strtol(optarg, NULL, 10); + break; + case 'v': + verbose = 1; + break; + default: + exit(EXIT_FAILURE); + } + } + if (optind >= argc) + usage(); + + file = argv[optind]; + + capture_samples(file, card, device, buffer_size, frag, rate); + + fprintf(stderr, "Finish capturing... Close Normally\n"); + exit(EXIT_SUCCESS); +} + diff --git a/makefile.linux b/makefile.linux index 60d6c04..9404fd8 100644 --- a/makefile.linux +++ b/makefile.linux @@ -1,10 +1,13 @@ LIB = libtinycompress -BIN = cplay VER = 0.1 LIBSRC = compress.c -SRC = cplay.c +PLAYBIN = cplay +RECBIN = crec +PLAYSRC = cplay.c +RECSRC = crec.c LIBOBJ = ${LIBSRC:.c=.o} -OBJ = ${SRC:.c=.o} +PLAYOBJ = ${PLAYSRC:.c=.o} +RECOBJ = ${RECSRC:.c=.o} CC = gcc CROSS_COMPILE = @@ -13,23 +16,26 @@ CFLAGS += -std=c99 -Wall -Wextra -Wunused -DVERSION=\"${VER}\" -I./include LDFLAGS += -L. -ltinycompress LIBLDFLAGS = -lasound -all: libtinycompress cplay +all: libtinycompress cplay crec $(LIB): ${LIBOBJ} @echo " LD "$@ @${CROSS_COMPILE}${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} -$(BIN): ${OBJ} ${LIB} +$(PLAYBIN): ${PLAYOBJ} ${LIB} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${OBJ} ${LDFLAGS} + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${PLAYOBJ} ${LDFLAGS} +$(RECBIN): ${RECOBJ} ${LIB} + @echo " LD "$@ + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${RECOBJ} ${LDFLAGS} %.o: %.c @echo " CC "$< @${CROSS_COMPILE}${CC} ${CFLAGS} -c -fPIC -o $@ $< clean: - @rm -rf ${BIN} ${OBJ} ${LIB}.so ${LIBOBJ} + @rm -rf ${BIN} ${PLAYOBJ} ${RECOBJ} ${LIB}.so ${LIBOBJ} .PHONY: all clean -- 1.7.2.5

3 2

[alsa-devel] [PATCH] ASoC: fsl: expand the size of the name in fsl_ssi_private struct
by Nicolin Chen 23 May '13

23 May '13

strcpy(ssi_private->name, p) in probe() sets "name" by "p", gotten from dts, while the length of "p", if the devicetree node name of SSI is commonly set, would always be larger than 1 char size, so need a larger size for "name". Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2f2d837..b6a5f94 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -152,5 +152,5 @@ struct fsl_ssi_private { } stats; - char name[1]; + char name[32]; }; -- 1.7.1

1 0

[alsa-devel] [TINYCOMPRESS][PATCH 1/1] compress: fix hpointer error when no sample rate
by Richard Fitzgerald 23 May '13

23 May '13

Fixes the oops() in compress_get_hpointer() to return -ENODATA instead of errno if the sample rate is zero. The value in errno here is not related to the reason we are returning an error. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- compress.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/compress.c b/compress.c index e80e598..abd0799 100644 --- a/compress.c +++ b/compress.c @@ -315,7 +315,7 @@ int compress_get_hpointer(struct compress *compress, if (ioctl(compress->fd, SNDRV_COMPRESS_AVAIL, &kavail)) return oops(compress, errno, "cannot get avail"); if (0 == kavail.tstamp.sampling_rate) - return oops(compress, errno, "invalid paramter"); + return oops(compress, -ENODATA, "sample rate unknown"); *avail = (unsigned int)kavail.avail; time = kavail.tstamp.pcm_io_frames / kavail.tstamp.sampling_rate; tstamp->tv_sec = time; -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: Fix early event callback list iteration
by Lars-Peter Clausen 23 May '13

23 May '13

The power_list field is used when adding a widget to a power sequence list. Use the same field when iterating the list using list_for_each_entry, otherwise we'll see undefined behavior. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 071579b..3507346 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1738,11 +1738,11 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) &async_domain); async_synchronize_full_domain(&async_domain); - list_for_each_entry(w, &down_list, list) { + list_for_each_entry(w, &down_list, power_list) { dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMD); } - list_for_each_entry(w, &up_list, list) { + list_for_each_entry(w, &up_list, power_list) { dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMU); } -- 1.8.2.1

2 1

[alsa-devel] [PATCH] ASoC: Add ssm2518 support
by Lars-Peter Clausen 23 May '13

23 May '13

This patch adds a ASoC CODEC driver for the SSM2516. The SSM2516 is a stereo Class-D audio amplifier with an I2S interface for audio in and a built-in dynamic range control processor. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- .../devicetree/bindings/sound/ssm2518.txt | 20 + include/linux/platform_data/ssm2518.h | 22 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2518.c | 845 +++++++++++++++++++++ sound/soc/codecs/ssm2518.h | 20 + 6 files changed, 913 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ssm2518.txt create mode 100644 include/linux/platform_data/ssm2518.h create mode 100644 sound/soc/codecs/ssm2518.c create mode 100644 sound/soc/codecs/ssm2518.h diff --git a/Documentation/devicetree/bindings/sound/ssm2518.txt b/Documentation/devicetree/bindings/sound/ssm2518.txt new file mode 100644 index 0000000..59381a7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ssm2518.txt @@ -0,0 +1,20 @@ +SSM2518 audio amplifier + +This device supports I2C only. + +Required properties: + - compatible : Must be "adi,ssm2518" + - reg : the I2C address of the device. This will either be 0x34 (ADDR pin low) + or 0x35 (ADDR pin high) + +Optional properties: + - gpios : GPIO connected to the nSD pin. If the property is not present it is + assumed that the nSD pin is hardwired to always on. + +Example: + + ssm2518: ssm2518@34 { + compatible = "adi,ssm2518"; + reg = <0x34>; + gpios = <&gpio 5 0>; + }; diff --git a/include/linux/platform_data/ssm2518.h b/include/linux/platform_data/ssm2518.h new file mode 100644 index 0000000..9a8e3ea --- /dev/null +++ b/include/linux/platform_data/ssm2518.h @@ -0,0 +1,22 @@ +/* + * SSM2518 amplifier audio driver + * + * Copyright 2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#ifndef __LINUX_PLATFORM_DATA_SSM2518_H__ +#define __LINUX_PLATFORM_DATA_SSM2518_H__ + +/** + * struct ssm2518_platform_data - Platform data for the ssm2518 driver + * @enable_gpio: GPIO connected to the nSD pin. Set to -1 if the nSD pin is + * hardwired. + */ +struct ssm2518_platform_data { + int enable_gpio; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00..d76609a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -60,6 +60,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF + select SND_SOC_SSM2518 if I2C select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI select SND_SOC_STA32X if I2C select SND_SOC_STA529 if I2C @@ -313,6 +314,9 @@ config SND_SOC_SN95031 config SND_SOC_SPDIF tristate +config SND_SOC_SSM2518 + tristate + config SND_SOC_SSM2602 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dae0aa6..7204c24 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -52,6 +52,7 @@ snd-soc-si476x-objs := si476x.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o +snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o snd-soc-sta529-objs := sta529.o @@ -176,6 +177,7 @@ obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o +obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c new file mode 100644 index 0000000..5874ba5 --- /dev/null +++ b/sound/soc/codecs/ssm2518.c @@ -0,0 +1,845 @@ +/* + * SSM2518 amplifier audio driver + * + * Copyright 2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/platform_data/ssm2518.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "ssm2518.h" + +#define SSM2518_REG_POWER1 0x00 +#define SSM2518_REG_CLOCK 0x01 +#define SSM2518_REG_SAI_CTRL1 0x02 +#define SSM2518_REG_SAI_CTRL2 0x03 +#define SSM2518_REG_CHAN_MAP 0x04 +#define SSM2518_REG_LEFT_VOL 0x05 +#define SSM2518_REG_RIGHT_VOL 0x06 +#define SSM2518_REG_MUTE_CTRL 0x07 +#define SSM2518_REG_FAULT_CTRL 0x08 +#define SSM2518_REG_POWER2 0x09 +#define SSM2518_REG_DRC_1 0x0a +#define SSM2518_REG_DRC_2 0x0b +#define SSM2518_REG_DRC_3 0x0c +#define SSM2518_REG_DRC_4 0x0d +#define SSM2518_REG_DRC_5 0x0e +#define SSM2518_REG_DRC_6 0x0f +#define SSM2518_REG_DRC_7 0x10 +#define SSM2518_REG_DRC_8 0x11 +#define SSM2518_REG_DRC_9 0x12 + +#define SSM2518_POWER1_RESET BIT(7) +#define SSM2518_POWER1_NO_BCLK BIT(5) +#define SSM2518_POWER1_MCS_MASK (0xf << 1) +#define SSM2518_POWER1_MCS_64FS (0x0 << 1) +#define SSM2518_POWER1_MCS_128FS (0x1 << 1) +#define SSM2518_POWER1_MCS_256FS (0x2 << 1) +#define SSM2518_POWER1_MCS_384FS (0x3 << 1) +#define SSM2518_POWER1_MCS_512FS (0x4 << 1) +#define SSM2518_POWER1_MCS_768FS (0x5 << 1) +#define SSM2518_POWER1_MCS_100FS (0x6 << 1) +#define SSM2518_POWER1_MCS_200FS (0x7 << 1) +#define SSM2518_POWER1_MCS_400FS (0x8 << 1) +#define SSM2518_POWER1_SPWDN BIT(0) + +#define SSM2518_CLOCK_ASR BIT(0) + +#define SSM2518_SAI_CTRL1_FMT_MASK (0x3 << 5) +#define SSM2518_SAI_CTRL1_FMT_I2S (0x0 << 5) +#define SSM2518_SAI_CTRL1_FMT_LJ (0x1 << 5) +#define SSM2518_SAI_CTRL1_FMT_RJ_24BIT (0x2 << 5) +#define SSM2518_SAI_CTRL1_FMT_RJ_16BIT (0x3 << 5) + +#define SSM2518_SAI_CTRL1_SAI_MASK (0x7 << 2) +#define SSM2518_SAI_CTRL1_SAI_I2S (0x0 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_2 (0x1 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_4 (0x2 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_8 (0x3 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_16 (0x4 << 2) +#define SSM2518_SAI_CTRL1_SAI_MONO (0x5 << 2) + +#define SSM2518_SAI_CTRL1_FS_MASK (0x3) +#define SSM2518_SAI_CTRL1_FS_8000_12000 (0x0) +#define SSM2518_SAI_CTRL1_FS_16000_24000 (0x1) +#define SSM2518_SAI_CTRL1_FS_32000_48000 (0x2) +#define SSM2518_SAI_CTRL1_FS_64000_96000 (0x3) + +#define SSM2518_SAI_CTRL2_BCLK_INTERAL BIT(7) +#define SSM2518_SAI_CTRL2_LRCLK_PULSE BIT(6) +#define SSM2518_SAI_CTRL2_LRCLK_INVERT BIT(5) +#define SSM2518_SAI_CTRL2_MSB BIT(4) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_MASK (0x3 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_32 (0x0 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_24 (0x1 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_16 (0x2 << 2) +#define SSM2518_SAI_CTRL2_BCLK_INVERT BIT(1) + +#define SSM2518_CHAN_MAP_RIGHT_SLOT_OFFSET 4 +#define SSM2518_CHAN_MAP_RIGHT_SLOT_MASK 0xf0 +#define SSM2518_CHAN_MAP_LEFT_SLOT_OFFSET 0 +#define SSM2518_CHAN_MAP_LEFT_SLOT_MASK 0x0f + +#define SSM2518_MUTE_CTRL_ANA_GAIN BIT(5) +#define SSM2518_MUTE_CTRL_MUTE_MASTER BIT(0) + +#define SSM2518_POWER2_APWDN BIT(0) + +#define SSM2518_DAC_MUTE BIT(6) +#define SSM2518_DAC_FS_MASK 0x07 +#define SSM2518_DAC_FS_8000 0x00 +#define SSM2518_DAC_FS_16000 0x01 +#define SSM2518_DAC_FS_32000 0x02 +#define SSM2518_DAC_FS_64000 0x03 +#define SSM2518_DAC_FS_128000 0x04 + +struct ssm2518 { + struct regmap *regmap; + bool right_j; + + unsigned int sysclk; + const struct snd_pcm_hw_constraint_list *constraints; + + int enable_gpio; +}; + +static const struct reg_default ssm2518_reg_defaults[] = { + { 0x00, 0x05 }, + { 0x01, 0x00 }, + { 0x02, 0x02 }, + { 0x03, 0x00 }, + { 0x04, 0x10 }, + { 0x05, 0x40 }, + { 0x06, 0x40 }, + { 0x07, 0x81 }, + { 0x08, 0x0c }, + { 0x09, 0x99 }, + { 0x0a, 0x7c }, + { 0x0b, 0x5b }, + { 0x0c, 0x57 }, + { 0x0d, 0x89 }, + { 0x0e, 0x8c }, + { 0x0f, 0x77 }, + { 0x10, 0x26 }, + { 0x11, 0x1c }, + { 0x12, 0x97 }, +}; + +static const DECLARE_TLV_DB_MINMAX_MUTE(ssm2518_vol_tlv, -7125, 2400); +static const DECLARE_TLV_DB_SCALE(ssm2518_compressor_tlv, -3400, 200, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_expander_tlv, -8100, 300, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_noise_gate_tlv, -9600, 300, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_post_drc_tlv, -2400, 300, 0); + +static const DECLARE_TLV_DB_RANGE(ssm2518_limiter_tlv, + 0, 7, TLV_DB_SCALE_ITEM(-2200, 200, 0), + 7, 15, TLV_DB_SCALE_ITEM(-800, 100, 0), +); + +static const char * const ssm2518_drc_peak_detector_attack_time_text[] = { + "0 ms", "0.1 ms", "0.19 ms", "0.37 ms", "0.75 ms", "1.5 ms", "3 ms", + "6 ms", "12 ms", "24 ms", "48 ms", "96 ms", "192 ms", "384 ms", + "768 ms", "1536 ms", +}; + +static const char * const ssm2518_drc_peak_detector_release_time_text[] = { + "0 ms", "1.5 ms", "3 ms", "6 ms", "12 ms", "24 ms", "48 ms", "96 ms", + "192 ms", "384 ms", "768 ms", "1536 ms", "3072 ms", "6144 ms", + "12288 ms", "24576 ms" +}; + +static const char * const ssm2518_drc_hold_time_text[] = { + "0 ms", "0.67 ms", "1.33 ms", "2.67 ms", "5.33 ms", "10.66 ms", + "21.32 ms", "42.64 ms", "85.28 ms", "170.56 ms", "341.12 ms", + "682.24 ms", "1364 ms", +}; + +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_attack_time_enum, + SSM2518_REG_DRC_2, 4, ssm2518_drc_peak_detector_attack_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_release_time_enum, + SSM2518_REG_DRC_2, 0, ssm2518_drc_peak_detector_release_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_attack_time_enum, + SSM2518_REG_DRC_6, 4, ssm2518_drc_peak_detector_attack_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_decay_time_enum, + SSM2518_REG_DRC_6, 0, ssm2518_drc_peak_detector_release_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_hold_time_enum, + SSM2518_REG_DRC_7, 4, ssm2518_drc_hold_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_noise_gate_hold_time_enum, + SSM2518_REG_DRC_7, 0, ssm2518_drc_hold_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_rms_averaging_time_enum, + SSM2518_REG_DRC_9, 0, ssm2518_drc_peak_detector_release_time_text); + +static const struct snd_kcontrol_new ssm2518_snd_controls[] = { + SOC_SINGLE("Playback De-emphasis Switch", SSM2518_REG_MUTE_CTRL, + 4, 1, 0), + SOC_DOUBLE_R_TLV("Master Playback Volume", SSM2518_REG_LEFT_VOL, + SSM2518_REG_RIGHT_VOL, 0, 0xff, 1, ssm2518_vol_tlv), + SOC_DOUBLE("Master Playback Switch", SSM2518_REG_MUTE_CTRL, 2, 1, 1, 1), + + SOC_SINGLE("Amp Low Power Mode Switch", SSM2518_REG_POWER2, 4, 1, 0), + SOC_SINGLE("DAC Low Power Mode Switch", SSM2518_REG_POWER2, 3, 1, 0), + + SOC_SINGLE("DRC Limiter Switch", SSM2518_REG_DRC_1, 5, 1, 0), + SOC_SINGLE("DRC Compressor Switch", SSM2518_REG_DRC_1, 4, 1, 0), + SOC_SINGLE("DRC Expander Switch", SSM2518_REG_DRC_1, 3, 1, 0), + SOC_SINGLE("DRC Noise Gate Switch", SSM2518_REG_DRC_1, 2, 1, 0), + SOC_DOUBLE("DRC Switch", SSM2518_REG_DRC_1, 0, 1, 1, 0), + + SOC_SINGLE_TLV("DRC Limiter Threshold Volume", + SSM2518_REG_DRC_3, 4, 15, 1, ssm2518_limiter_tlv), + SOC_SINGLE_TLV("DRC Compressor Lower Threshold Volume", + SSM2518_REG_DRC_3, 0, 15, 1, ssm2518_compressor_tlv), + SOC_SINGLE_TLV("DRC Expander Upper Threshold Volume", SSM2518_REG_DRC_4, + 4, 15, 1, ssm2518_expander_tlv), + SOC_SINGLE_TLV("DRC Noise Gate Threshold Volume", + SSM2518_REG_DRC_4, 0, 15, 1, ssm2518_noise_gate_tlv), + SOC_SINGLE_TLV("DRC Upper Output Threshold Volume", + SSM2518_REG_DRC_5, 4, 15, 1, ssm2518_limiter_tlv), + SOC_SINGLE_TLV("DRC Lower Output Threshold Volume", + SSM2518_REG_DRC_5, 0, 15, 1, ssm2518_noise_gate_tlv), + SOC_SINGLE_TLV("DRC Post Volume", SSM2518_REG_DRC_8, + 2, 15, 1, ssm2518_post_drc_tlv), + + SOC_ENUM("DRC Peak Detector Attack Time", + ssm2518_drc_peak_detector_attack_time_enum), + SOC_ENUM("DRC Peak Detector Release Time", + ssm2518_drc_peak_detector_release_time_enum), + SOC_ENUM("DRC Attack Time", ssm2518_drc_attack_time_enum), + SOC_ENUM("DRC Decay Time", ssm2518_drc_decay_time_enum), + SOC_ENUM("DRC Hold Time", ssm2518_drc_hold_time_enum), + SOC_ENUM("DRC Noise Gate Hold Time", + ssm2518_drc_noise_gate_hold_time_enum), + SOC_ENUM("DRC RMS Averaging Time", ssm2518_drc_rms_averaging_time_enum), +}; + +static const struct snd_soc_dapm_widget ssm2518_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DACL", "HiFi Playback", SSM2518_REG_POWER2, 1, 1), + SND_SOC_DAPM_DAC("DACR", "HiFi Playback", SSM2518_REG_POWER2, 2, 1), + + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), +}; + +static const struct snd_soc_dapm_route ssm2518_routes[] = { + { "OUTL", NULL, "DACL" }, + { "OUTR", NULL, "DACR" }, +}; + +struct ssm2518_mcs_lut { + unsigned int rate; + const unsigned int *sysclks; +}; + +static const unsigned int ssm2518_sysclks_2048000[] = { + 2048000, 4096000, 8192000, 12288000, 16384000, 24576000, + 3200000, 6400000, 12800000, 0 +}; + +static const unsigned int ssm2518_sysclks_2822000[] = { + 2822000, 5644800, 11289600, 16934400, 22579200, 33868800, + 4410000, 8820000, 17640000, 0 +}; + +static const unsigned int ssm2518_sysclks_3072000[] = { + 3072000, 6144000, 12288000, 16384000, 24576000, 38864000, + 4800000, 9600000, 19200000, 0 +}; + +static const struct ssm2518_mcs_lut ssm2518_mcs_lut[] = { + { 8000, ssm2518_sysclks_2048000, }, + { 11025, ssm2518_sysclks_2822000, }, + { 12000, ssm2518_sysclks_3072000, }, + { 16000, ssm2518_sysclks_2048000, }, + { 24000, ssm2518_sysclks_3072000, }, + { 22050, ssm2518_sysclks_2822000, }, + { 32000, ssm2518_sysclks_2048000, }, + { 44100, ssm2518_sysclks_2822000, }, + { 48000, ssm2518_sysclks_3072000, }, + { 96000, ssm2518_sysclks_3072000, }, +}; + +static const unsigned int ssm2518_rates_2048000[] = { + 8000, 16000, 32000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_2048000 = { + .list = ssm2518_rates_2048000, + .count = ARRAY_SIZE(ssm2518_rates_2048000), +}; + +static const unsigned int ssm2518_rates_2822000[] = { + 11025, 22050, 44100, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_2822000 = { + .list = ssm2518_rates_2822000, + .count = ARRAY_SIZE(ssm2518_rates_2822000), +}; + +static const unsigned int ssm2518_rates_3072000[] = { + 12000, 24000, 48000, 96000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_3072000 = { + .list = ssm2518_rates_3072000, + .count = ARRAY_SIZE(ssm2518_rates_3072000), +}; + +static const unsigned int ssm2518_rates_12288000[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 96000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_12288000 = { + .list = ssm2518_rates_12288000, + .count = ARRAY_SIZE(ssm2518_rates_12288000), +}; + +static unsigned int ssm2518_lookup_mcs(struct ssm2518 *ssm2518, + unsigned int rate) +{ + const unsigned int *sysclks = NULL; + int i; + + for (i = 0; i < ARRAY_SIZE(ssm2518_mcs_lut); i++) { + if (ssm2518_mcs_lut[i].rate == rate) { + sysclks = ssm2518_mcs_lut[i].sysclks; + break; + } + } + if (!sysclks) + return -EINVAL; + + for (i = 0; sysclks[i]; i++) { + if (sysclks[i] == ssm2518->sysclk) + return i; + } + return -EINVAL; +} + +static int ssm2518_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + unsigned int ctrl1, ctrl1_mask; + int mcs; + int ret; + + mcs = ssm2518_lookup_mcs(ssm2518, rate); + if (mcs < 0) + return mcs; + + ctrl1_mask = SSM2518_SAI_CTRL1_FS_MASK; + + if (rate >= 8000 && rate <= 12000) + ctrl1 = SSM2518_SAI_CTRL1_FS_8000_12000; + else if (rate >= 16000 && rate <= 24000) + ctrl1 = SSM2518_SAI_CTRL1_FS_16000_24000; + else if (rate >= 32000 && rate <= 48000) + ctrl1 = SSM2518_SAI_CTRL1_FS_32000_48000; + else if (rate >= 64000 && rate <= 96000) + ctrl1 = SSM2518_SAI_CTRL1_FS_64000_96000; + else + return -EINVAL; + + if (ssm2518->right_j) { + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_16BIT; + break; + case SNDRV_PCM_FORMAT_S24_LE: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_24BIT; + break; + default: + return -EINVAL; + } + ctrl1_mask |= SSM2518_SAI_CTRL1_FMT_MASK; + } + + /* Disable auto samplerate detection */ + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_CLOCK, + SSM2518_CLOCK_ASR, SSM2518_CLOCK_ASR); + if (ret < 0) + return ret; + + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, + ctrl1_mask, ctrl1); + if (ret < 0) + return ret; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_MCS_MASK, mcs << 1); +} + +static int ssm2518_mute(struct snd_soc_dai *dai, int mute) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int val; + + if (mute) + val = SSM2518_MUTE_CTRL_MUTE_MASTER; + else + val = 0; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_MUTE_CTRL, + SSM2518_MUTE_CTRL_MUTE_MASTER, val); +} + +static int ssm2518_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl1 = 0, ctrl2 = 0; + bool invert_fclk; + int ret; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + invert_fclk = false; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl2 |= SSM2518_SAI_CTRL2_BCLK_INVERT; + invert_fclk = false; + break; + case SND_SOC_DAIFMT_NB_IF: + invert_fclk = true; + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl2 |= SSM2518_SAI_CTRL2_BCLK_INVERT; + invert_fclk = true; + break; + default: + return -EINVAL; + } + + ssm2518->right_j = false; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_LJ; + invert_fclk = !invert_fclk; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_24BIT; + ssm2518->right_j = true; + invert_fclk = !invert_fclk; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_PULSE; + ctrl1 |= SSM2518_SAI_CTRL1_FMT_I2S; + invert_fclk = false; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_PULSE; + ctrl1 |= SSM2518_SAI_CTRL1_FMT_LJ; + invert_fclk = false; + break; + default: + return -EINVAL; + } + + if (invert_fclk) + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_INVERT; + + ret = regmap_write(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, ctrl1); + if (ret) + return ret; + return regmap_write(ssm2518->regmap, SSM2518_REG_SAI_CTRL2, ctrl2); +} + +static int ssm2518_set_power(struct ssm2518 *ssm2518, bool enable) +{ + int ret = 0; + + if (!enable) { + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_SPWDN, SSM2518_POWER1_SPWDN); + regcache_mark_dirty(ssm2518->regmap); + } + + if (gpio_is_valid(ssm2518->enable_gpio)) + gpio_set_value(ssm2518->enable_gpio, enable); + + regcache_cache_only(ssm2518->regmap, !enable); + + if (enable) { + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_SPWDN | SSM2518_POWER1_RESET, 0x00); + regcache_sync(ssm2518->regmap); + } + + return ret; +} + +static int ssm2518_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + ret = ssm2518_set_power(ssm2518, true); + break; + case SND_SOC_BIAS_OFF: + ret = ssm2518_set_power(ssm2518, false); + break; + } + + if (ret) + return ret; + + codec->dapm.bias_level = level; + return 0; +} + +static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int width) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl1, ctrl2; + int left_slot, right_slot; + int ret; + + if (slots == 0) { + return regmap_update_bits(ssm2518->regmap, + SSM2518_REG_SAI_CTRL1, SSM2518_SAI_CTRL1_SAI_MASK, + SSM2518_SAI_CTRL1_SAI_I2S); + } + + if (tx_mask == 0 || tx_mask != 0) + return -EINVAL; + + if (slots == 1) { + if (tx_mask != 1) + return -EINVAL; + left_slot = 0; + right_slot = 0; + } else { + /* We assume the left channel < right channel */ + left_slot = ffs(tx_mask); + tx_mask &= ~(1 << tx_mask); + if (tx_mask == 0) { + right_slot = left_slot; + } else { + right_slot = ffs(tx_mask); + tx_mask &= ~(1 << tx_mask); + } + } + + if (tx_mask != 0 || left_slot >= slots || right_slot >= slots) + return -EINVAL; + + switch (width) { + case 16: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_16; + break; + case 24: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_24; + break; + case 32: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_32; + break; + default: + return -EINVAL; + } + + switch (slots) { + case 1: + ctrl1 = SSM2518_SAI_CTRL1_SAI_MONO; + break; + case 2: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_2; + break; + case 4: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_4; + break; + case 8: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_8; + break; + case 16: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_16; + break; + default: + return -EINVAL; + } + + ret = regmap_write(ssm2518->regmap, SSM2518_REG_CHAN_MAP, + (left_slot << SSM2518_CHAN_MAP_LEFT_SLOT_OFFSET) | + (right_slot << SSM2518_CHAN_MAP_RIGHT_SLOT_OFFSET)); + if (ret) + return ret; + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, + SSM2518_SAI_CTRL1_SAI_MASK, ctrl1); + if (ret) + return ret; + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL2, + SSM2518_SAI_CTRL2_SLOT_WIDTH_MASK, ctrl2); +} + +static int ssm2518_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + + if (ssm2518->constraints) { + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, ssm2518->constraints); + } + + return 0; +} + +#define SSM2518_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static const struct snd_soc_dai_ops ssm2518_dai_ops = { + .startup = ssm2518_startup, + .hw_params = ssm2518_hw_params, + .digital_mute = ssm2518_mute, + .set_fmt = ssm2518_set_dai_fmt, + .set_tdm_slot = ssm2518_set_tdm_slot, +}; + +static struct snd_soc_dai_driver ssm2518_dai = { + .name = "ssm2518-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SSM2518_FORMATS, + }, + .ops = &ssm2518_dai_ops, +}; + +static int ssm2518_probe(struct snd_soc_codec *codec) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = ssm2518->regmap; + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER2, + SSM2518_POWER2_APWDN, 0x00); + if (ret) + return ret; + + return ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int ssm2518_remove(struct snd_soc_codec *codec) +{ + ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (clk_id != SSM2518_SYSCLK) + return -EINVAL; + + switch (source) { + case SSM2518_SYSCLK_SRC_MCLK: + val = 0; + break; + case SSM2518_SYSCLK_SRC_BCLK: + /* In this case the bitclock is used as the system clock, and + * the bitclock signal needs to be connected to the MCLK pin and + * the BCLK pin is left unconnected */ + val = SSM2518_POWER1_NO_BCLK; + break; + default: + return -EINVAL; + } + + switch (freq) { + case 2048000: + case 4096000: + case 8192000: + case 3200000: + case 6400000: + case 12800000: + ssm2518->constraints = &ssm2518_constraints_2048000; + break; + case 2822000: + case 5644800: + case 11289600: + case 16934400: + case 22579200: + case 33868800: + case 4410000: + case 8820000: + case 17640000: + ssm2518->constraints = &ssm2518_constraints_2822000; + break; + case 3072000: + case 6144000: + case 38864000: + case 4800000: + case 9600000: + case 19200000: + ssm2518->constraints = &ssm2518_constraints_3072000; + break; + case 12288000: + case 16384000: + case 24576000: + ssm2518->constraints = &ssm2518_constraints_12288000; + break; + default: + return -EINVAL; + } + + ssm2518->sysclk = freq; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_NO_BCLK, val); +} + +static struct snd_soc_codec_driver ssm2518_codec_driver = { + .probe = ssm2518_probe, + .remove = ssm2518_remove, + .set_bias_level = ssm2518_set_bias_level, + .set_sysclk = ssm2518_set_sysclk, + .idle_bias_off = true, + + .controls = ssm2518_snd_controls, + .num_controls = ARRAY_SIZE(ssm2518_snd_controls), + .dapm_widgets = ssm2518_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ssm2518_dapm_widgets), + .dapm_routes = ssm2518_routes, + .num_dapm_routes = ARRAY_SIZE(ssm2518_routes), +}; + +static bool ssm2518_register_volatile(struct device *dev, unsigned int reg) +{ + return false; +} + +static const struct regmap_config ssm2518_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .max_register = SSM2518_REG_DRC_9, + .volatile_reg = ssm2518_register_volatile, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ssm2518_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ssm2518_reg_defaults), +}; + +static int ssm2518_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ssm2518_platform_data *pdata = i2c->dev.platform_data; + struct ssm2518 *ssm2518; + int ret; + + ssm2518 = devm_kzalloc(&i2c->dev, sizeof(*ssm2518), GFP_KERNEL); + if (ssm2518 == NULL) + return -ENOMEM; + + if (pdata) { + ssm2518->enable_gpio = pdata->enable_gpio; + } else if (i2c->dev.of_node) { + ssm2518->enable_gpio = of_get_gpio(i2c->dev.of_node, 0); + if (ssm2518->enable_gpio == -EPROBE_DEFER) + return -EPROBE_DEFER; + } else { + ssm2518->enable_gpio = -1; + } + + if (gpio_is_valid(ssm2518->enable_gpio)) { + ret = devm_gpio_request_one(&i2c->dev, ssm2518->enable_gpio, + GPIOF_OUT_INIT_HIGH, "SSM2518 nSD"); + if (ret) + return ret; + } + + i2c_set_clientdata(i2c, ssm2518); + + ssm2518->regmap = devm_regmap_init_i2c(i2c, &ssm2518_regmap_config); + if (IS_ERR(ssm2518->regmap)) + return PTR_ERR(ssm2518->regmap); + + /* + * The reset bit is obviously volatile, but we need to be able to cache + * the other bits in the register, so we can't just mark the whole + * register as volatile. Since this is the only place where we'll ever + * touch the reset bit just bypass the cache for this operation. + */ + regcache_cache_bypass(ssm2518->regmap, true); + ret = regmap_write(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_RESET); + if (ret) + return ret; + regcache_cache_bypass(ssm2518->regmap, false); + + return snd_soc_register_codec(&i2c->dev, &ssm2518_codec_driver, + &ssm2518_dai, 1); +} + +static int ssm2518_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ssm2518_i2c_ids[] = { + { "ssm2518", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ssm2518_i2c_ids); + +static struct i2c_driver ssm2518_driver = { + .driver = { + .name = "ssm2518", + .owner = THIS_MODULE, + }, + .probe = ssm2518_i2c_probe, + .remove = ssm2518_i2c_remove, + .id_table = ssm2518_i2c_ids, +}; +module_i2c_driver(ssm2518_driver); + +MODULE_DESCRIPTION("ASoC SSM2518 driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2518.h b/sound/soc/codecs/ssm2518.h new file mode 100644 index 0000000..62511d8 --- /dev/null +++ b/sound/soc/codecs/ssm2518.h @@ -0,0 +1,20 @@ +/* + * SSM2518 amplifier audio driver + * + * Copyright 2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#ifndef __SND_SOC_CODECS_SSM2518_H__ +#define __SND_SOC_CODECS_SSM2518_H__ + +#define SSM2518_SYSCLK 0 + +enum ssm2518_sysclk_src { + SSM2518_SYSCLK_SRC_MCLK = 0, + SSM2518_SYSCLK_SRC_BCLK = 1, +}; + +#endif -- 1.8.0

2 3

[alsa-devel] [PATCH] ALSA: ASoC: add codec driver for TI TAS5086
by Daniel Mack 22 May '13

22 May '13

This patch adds a driver for TI's TA5086 6-channel PWM processor. This chip has a very unusual register layout, specifically because the registers are of unequal size, and multi-byte registers require bulk writes to take effect. Regmap does not support these kind of mappings. Currently, the driver does not touch any of the registers >= 0x20, so it doesn't matter, because the register map is mapped to an 8-bit array. In case more features will be added in the future that require access to higher registers, the entire regmap H/W I/O routines have to be open-coded. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- .../devicetree/bindings/sound/ti,tas5086.txt | 32 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5086.c | 587 +++++++++++++++++++++ 4 files changed, 625 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tas5086.txt create mode 100644 sound/soc/codecs/tas5086.c diff --git a/Documentation/devicetree/bindings/sound/ti,tas5086.txt b/Documentation/devicetree/bindings/sound/ti,tas5086.txt new file mode 100644 index 0000000..8ea4f5b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas5086.txt @@ -0,0 +1,32 @@ +Texas Instruments TAS5086 6-channel PWM Processor + +Required properties: + + - compatible: Should contain "ti,tas5086". + - reg: The i2c address. Should contain <0x1b>. + +Optional properties: + + - reset-gpio: A GPIO spec to define which pin is connected to the + chip's !RESET pin. If specified, the driver will + assert a hardware reset at probe time. + + - ti,charge-period: This property should contain the time in microseconds + that closely matches the external single-ended + split-capacitor charge period. The hardware chip + waits for this period of time before starting the + PWM signals. This helps reduce pops and clicks. + + When not specified, the hardware default of 1300ms + is retained. + +Examples: + + i2c_bus { + tas5086@1b { + compatible = "ti,tas5086"; + reg = <0x1b>; + reset-gpio = <&gpio 23 0>; + ti,charge-period = <156000>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 45b7256..86b3524 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -63,6 +63,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STA32X if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS + select SND_SOC_TAS5086 if I2C select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC32X4 if I2C @@ -320,6 +321,9 @@ config SND_SOC_STA529 config SND_SOC_STAC9766 tristate +config SND_SOC_TAS5086 + tristate + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6a3b3c3..8077bc2 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o +snd-soc-tas5086-objs := tas5086.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -177,6 +178,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o +obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c new file mode 100644 index 0000000..cd3c267 --- /dev/null +++ b/sound/soc/codecs/tas5086.c @@ -0,0 +1,587 @@ +/* + * TAS5086 ASoC codec driver + * + * Copyright (c) 2013 Daniel Mack <zonque(a)gmail.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * TODO: + * - implement DAPM and input muxing + * - implement modulation limit + * - implement non-default PWM start + * + * Note that this chip has a very unusual register layout, specifically + * because the registers are of unequal size, and multi-byte registers + * require bulk writes to take effect. Regmap does not support that kind + * of devices. + * + * Currently, the driver does not touch any of the registers >= 0x20, so + * it doesn't matter because the entire map can be accessed as 8-bit + * array. In case more features will be added in the future + * that require access to higher registers, the entire regmap H/W I/O + * routines have to be open-coded. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/spi/spi.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#define TAS5086_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE) + +#define TAS5086_PCM_RATES (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) + +/* + * TAS5086 registers + */ +#define TAS5086_CLOCK_CONTROL 0x00 /* Clock control register */ +#define TAS5086_CLOCK_RATE(val) (val << 5) +#define TAS5086_CLOCK_RATE_MASK (0x7 << 5) +#define TAS5086_CLOCK_RATIO(val) (val << 2) +#define TAS5086_CLOCK_RATIO_MASK (0x7 << 2) +#define TAS5086_CLOCK_SCLK_RATIO_48 (1 << 1) +#define TAS5086_CLOCK_VALID (1 << 0) + +#define TAS5086_DEV_ID 0x01 /* Device ID register */ +#define TAS5086_ERROR_STATUS 0x02 /* Error status register */ +#define TAS5086_SYS_CONTROL_1 0x03 /* System control register 1 */ +#define TAS5086_SERIAL_DATA_IF 0x04 /* Serial data interface register */ +#define TAS5086_SYS_CONTROL_2 0x05 /* System control register 2 */ +#define TAS5086_SOFT_MUTE 0x06 /* Soft mute register */ +#define TAS5086_MASTER_VOL 0x07 /* Master volume */ +#define TAS5086_CHANNEL_VOL(X) (0x08 + (X)) /* Channel 1-6 volume */ +#define TAS5086_VOLUME_CONTROL 0x09 /* Volume control register */ +#define TAS5086_MOD_LIMIT 0x10 /* Modulation limit register */ +#define TAS5086_PWM_START 0x18 /* PWM start register */ +#define TAS5086_SURROUND 0x19 /* Surround register */ +#define TAS5086_SPLIT_CAP_CHARGE 0x1a /* Split cap charge period register */ +#define TAS5086_OSC_TRIM 0x1b /* Oscillator trim register */ +#define TAS5086_BKNDERR 0x1c + +#define TAS5086_DEEMPH_MASK 0x03 + +/* + * Default TAS5086 power-up configuration + */ +static const struct reg_default tas5086_reg_defaults[] = { + { 0x00, 0x6c }, + { 0x01, 0x03 }, + { 0x02, 0x00 }, + { 0x03, 0xa0 }, + { 0x04, 0x05 }, + { 0x05, 0x60 }, + { 0x06, 0x00 }, + { 0x07, 0xff }, + { 0x08, 0x30 }, + { 0x09, 0x30 }, + { 0x0a, 0x30 }, + { 0x0b, 0x30 }, + { 0x0c, 0x30 }, + { 0x0d, 0x30 }, + { 0x0e, 0xb1 }, + { 0x0f, 0x00 }, + { 0x10, 0x02 }, + { 0x11, 0x00 }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, + { 0x14, 0x00 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, + { 0x18, 0x3f }, + { 0x19, 0x00 }, + { 0x1a, 0x18 }, + { 0x1b, 0x82 }, + { 0x1c, 0x05 }, +}; + +static bool tas5086_accessible_reg(struct device *dev, unsigned int reg) +{ + return !((reg == 0x0f) || (reg >= 0x11 && reg <= 0x17)); +} + +static bool tas5086_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == TAS5086_DEV_ID || + reg == TAS5086_ERROR_STATUS; +} + +static bool tas5086_writeable_reg(struct device *dev, unsigned int reg) +{ + return tas5086_accessible_reg(dev, reg) && (reg != TAS5086_DEV_ID); +} + +struct tas5086_private { + struct regmap *regmap; + unsigned int mclk, sclk; + unsigned int format; + bool deemph; + /* Current sample rate for de-emphasis control */ + int rate; + /* GPIO driving Reset pin, if any */ + int gpio_nreset; +}; + +static int tas5086_deemph[] = { 0, 32000, 44100, 48000 }; + +static int tas5086_set_deemph(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int i, val = 0; + + if (priv->deemph) + for (i = 0; i < ARRAY_SIZE(tas5086_deemph); i++) + if (tas5086_deemph[i] == priv->rate) + val = i; + + return regmap_update_bits(priv->regmap, TAS5086_SYS_CONTROL_1, + TAS5086_DEEMPH_MASK, val); +} + +static int tas5086_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = priv->deemph; + + return 0; +} + +static int tas5086_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->deemph = ucontrol->value.enumerated.item[0]; + + return tas5086_set_deemph(codec); +} + + +static int tas5086_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case 0: /* MCLK */ + priv->mclk = freq; + break; + case 1: /* SCLK */ + priv->sclk = freq; + break; + } + + return 0; +} + +static int tas5086_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + /* The TAS5086 can only be slave to all clocks */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Invalid clocking mode\n"); + return -EINVAL; + } + + /* we need to refer to the data format from hw_params() */ + priv->format = format; + + return 0; +} + +static const int tas5086_sample_rates[] = { + 32000, 38000, 44100, 48000, 88200, 96000, 176400, 192000 +}; + +static const int tas5086_ratios[] = { + 64, 128, 192, 256, 384, 512 +}; + +static int index_in_array(const int *array, int len, int needle) +{ + int i; + + for (i = 0; i < len; i++) + if (array[i] == needle) + return i; + + return -ENOENT; +} + +static int tas5086_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int ret; + + priv->rate = params_rate(params); + + /* Look up the sample rate and refer to the offset in the list */ + val = index_in_array(tas5086_sample_rates, + ARRAY_SIZE(tas5086_sample_rates), priv->rate); + + if (val < 0) { + dev_err(codec->dev, "Invalid sample rate\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_RATE_MASK, + TAS5086_CLOCK_RATE(val)); + if (ret < 0) + return ret; + + /* MCLK / Fs ratio */ + val = index_in_array(tas5086_ratios, ARRAY_SIZE(tas5086_ratios), + priv->mclk / priv->rate); + if (val < 0) { + dev_err(codec->dev, "Inavlid MCLK / Fs ratio\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_RATIO_MASK, + TAS5086_CLOCK_RATIO(val)); + if (ret < 0) + return ret; + + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_SCLK_RATIO_48, + (priv->sclk == 48 * priv->rate) ? + TAS5086_CLOCK_SCLK_RATIO_48 : 0); + if (ret < 0) + return ret; + + /* + * The chip has a very unituitive register mapping and muxes information + * about data format and sample depth into the same register, but not on + * a logical bit-boundary. Hence, we have to refer to the format passed + * in the set_dai_fmt() callback and set up everything from here. + * + * First, determine the 'base' value, using the format ... + */ + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + val = 0x00; + break; + case SND_SOC_DAIFMT_I2S: + val = 0x03; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = 0x06; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + /* ... then add the offset for the sample bit depth. */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + val += 0; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val += 1; + break; + case SNDRV_PCM_FORMAT_S24_3LE: + val += 2; + break; + default: + dev_err(codec->dev, "Invalid bit width\n"); + return -EINVAL; + }; + + ret = regmap_write(priv->regmap, TAS5086_SERIAL_DATA_IF, val); + if (ret < 0) + return ret; + + /* clock is considered valid now */ + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_VALID, TAS5086_CLOCK_VALID); + if (ret < 0) + return ret; + + return tas5086_set_deemph(codec); +} + +static int tas5086_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + return regmap_write(priv->regmap, TAS5086_SOFT_MUTE, + mute ? 0x3f : 0x00); +} + +/* TAS5086 controls */ +static const DECLARE_TLV_DB_SCALE(tas5086_dac_tlv, -10350, 50, 1); + +static const struct snd_kcontrol_new tas5086_controls[] = { + SOC_SINGLE_TLV("Master Playback Volume", TAS5086_MASTER_VOL, + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 1/2 Playback Volume", + TAS5086_CHANNEL_VOL(0), TAS5086_CHANNEL_VOL(1), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 3/4 Playback Volume", + TAS5086_CHANNEL_VOL(2), TAS5086_CHANNEL_VOL(3), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 5/6 Playback Volume", + TAS5086_CHANNEL_VOL(4), TAS5086_CHANNEL_VOL(5), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0, + tas5086_get_deemph, tas5086_put_deemph), +}; + +static const struct snd_soc_dai_ops tas5086_dai_ops = { + .hw_params = tas5086_hw_params, + .set_sysclk = tas5086_set_dai_sysclk, + .set_fmt = tas5086_set_dai_fmt, + .digital_mute = tas5086_digital_mute, +}; + +static struct snd_soc_dai_driver tas5086_dai = { + .name = "tas5086-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 6, + .rates = TAS5086_PCM_RATES, + .formats = TAS5086_PCM_FORMATS, + }, + .ops = &tas5086_dai_ops, +}; + +#ifdef CONFIG_PM +static int tas5086_soc_suspend(struct snd_soc_codec *codec) +{ + return 0; +} + +static int tas5086_soc_resume(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + /* Restore codec state */ + return regcache_sync(priv->regmap); +} +#else +#define tas5086_soc_suspend NULL +#define tas5086_soc_resume NULL +#endif /* CONFIG_PM */ + +#ifdef CONFIG_OF +static const struct of_device_id tas5086_dt_ids[] = { + { .compatible = "ti,tas5086", }, + { } +}; +MODULE_DEVICE_TABLE(of, tas5086_dt_ids); +#endif + +/* charge period values in microseconds */ +static const int tas5086_charge_period[] = { + 13000, 16900, 23400, 31200, 41600, 54600, 72800, 96200, + 130000, 156000, 234000, 312000, 416000, 546000, 728000, 962000, + 1300000, 169000, 2340000, 3120000, 4160000, 5460000, 7280000, 9620000, +}; + +static int tas5086_probe(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int charge_period = 1300000; /* hardware default is 1300 ms */ + int gpio_nreset = -EINVAL; + int i, ret; + + if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) { + struct device_node *of_node = codec->dev->of_node; + + gpio_nreset = of_get_named_gpio(of_node, "reset-gpio", 0); + of_property_read_u32(of_node, "ti,charge-period", &charge_period); + } + + if (gpio_nreset >= 0) + if (devm_gpio_request(codec->dev, gpio_nreset, "TAS5086 Reset")) + gpio_nreset = -EINVAL; + if (gpio_nreset >= 0) { + /* Reset codec - minimum assertion time is 400ns */ + gpio_direction_output(gpio_nreset, 0); + udelay(1); + gpio_set_value(gpio_nreset, 1); + + /* Codec needs ~15ms to wake up */ + msleep(15); + } +msleep(300); + priv->gpio_nreset = gpio_nreset; + + /* The TAS5086 always returns 0x03 in its TAS5086_DEV_ID register */ + ret = regmap_read(priv->regmap, TAS5086_DEV_ID, &i); + if (ret < 0) + return ret; + + if (i != 0x3) { + dev_err(codec->dev, + "Failed to identify TAS5086 codec (got %02x)\n", i); + return -ENODEV; + } + + /* lookup and set split-capacitor charge period */ + if (charge_period == 0) { + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); + } else { + i = index_in_array(tas5086_charge_period, + ARRAY_SIZE(tas5086_charge_period), + charge_period); + if (i >= 0) + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, + i + 0x08); + else + dev_warn(codec->dev, + "Invalid split-cap charge period of %d ns.\n", + charge_period); + } + + /* enable factory trim */ + ret = regmap_write(priv->regmap, TAS5086_OSC_TRIM, 0x00); + if (ret < 0) + return ret; + + /* start all channels */ + ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x20); + if (ret < 0) + return ret; + + /* set master volume to 0 dB */ + ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30); + if (ret < 0) + return ret; + + return 0; +} + +static int tas5086_remove(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(priv->gpio_nreset)) + /* Set codec to the reset state */ + gpio_set_value(priv->gpio_nreset, 0); + + return 0; +}; + +static struct snd_soc_codec_driver soc_codec_dev_tas5086 = { + .probe = tas5086_probe, + .remove = tas5086_remove, + .suspend = tas5086_soc_suspend, + .resume = tas5086_soc_resume, + .controls = tas5086_controls, + .num_controls = ARRAY_SIZE(tas5086_controls), +}; + +static const struct i2c_device_id tas5086_i2c_id[] = { + { "tas5086", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas5086_i2c_id); + +static const struct regmap_config tas5086_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(tas5086_reg_defaults), + .reg_defaults = tas5086_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas5086_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .volatile_reg = tas5086_volatile_reg, + .writeable_reg = tas5086_writeable_reg, + .readable_reg = tas5086_accessible_reg, +}; + +static int tas5086_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct tas5086_private *priv; + int ret; + + priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(i2c, &tas5086_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&i2c->dev, "Failed to create regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(i2c, priv); + + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086, + &tas5086_dai, 1); +} + +static int tas5086_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +static struct i2c_driver tas5086_i2c_driver = { + .driver = { + .name = "tas5086", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tas5086_dt_ids), + }, + .id_table = tas5086_i2c_id, + .probe = tas5086_i2c_probe, + .remove = tas5086_i2c_remove, +}; + +static int __init tas5086_modinit(void) +{ + return i2c_add_driver(&tas5086_i2c_driver); +} +module_init(tas5086_modinit); + +static void __exit tas5086_modexit(void) +{ + i2c_del_driver(&tas5086_i2c_driver); +} +module_exit(tas5086_modexit); + +MODULE_AUTHOR("Daniel Mack <zonque(a)gmail.com>"); +MODULE_DESCRIPTION("Texas Instruments TAS5086 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); -- 1.8.1.4

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Re: [alsa-devel] [PATCH 3/3] sound/soc/fsl: remove leftover release_mem_region
by Mark Brown 22 May '13

22 May '13

On Wed, May 22, 2013 at 06:04:00PM +0200, Wolfram Sang wrote: > On Wed, May 22, 2013 at 10:56:28AM -0500, Mark Brown wrote: > > This doesn't apply against current code, can you please regenerate > > against topic/fsl or mainline? > ? Works fine against v3.10-rc2 here. Or is mainline v3.9 for you? To be honest I didn't actually try mainline as I wasn't expecting any conflicts from the branch and it didn't seem urgent as a bug fix.

1 0

[alsa-devel] [PATCH v4] ASoC: dfbmcs320: make the driver common for other BT modules
by Barry Song 22 May '13

22 May '13

DFBM-CS320 is only one of bluetooth modules using CSR bluetooth chips, we don't want everyone to have a seperate codec driver. anyway, the feature of Bluetooth SCO is same on all platforms, so this patch makes the DFBM-CS320 driver become a common BT SCO link driver. Cc: Graeme Gregory <graeme(a)openmoko.org> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- v4: change "select SND_SOC_DFBMCS320" to "select SND_SOC_BT_SCO" for OpenMOKO user as v3 missed this sound/soc/codecs/Kconfig | 4 ++-- sound/soc/codecs/Makefile | 4 ++-- sound/soc/codecs/{dfbmcs320.c => bt-sco.c} | 37 +++++++++++++++++++----------- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/neo1973_wm8753.c | 2 +- 5 files changed, 29 insertions(+), 20 deletions(-) rename sound/soc/codecs/{dfbmcs320.c => bt-sco.c} (53%) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d8c4f3d..386292d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -40,7 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7213 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C - select SND_SOC_DFBMCS320 + select SND_SOC_BT_SCO select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -263,7 +263,7 @@ config SND_SOC_DA732X config SND_SOC_DA9055 tristate -config SND_SOC_DFBMCS320 +config SND_SOC_BT_SCO tristate config SND_SOC_DMIC diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 49ff127..82aa34e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -27,7 +27,7 @@ snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o -snd-soc-dfbmcs320-objs := dfbmcs320.o +snd-soc-bt-sco-objs := bt-sco.o snd-soc-dmic-objs := dmic.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o @@ -154,7 +154,7 @@ obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o -obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o +obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o diff --git a/sound/soc/codecs/dfbmcs320.c b/sound/soc/codecs/bt-sco.c similarity index 53% rename from sound/soc/codecs/dfbmcs320.c rename to sound/soc/codecs/bt-sco.c index 4f4f7f4..a081d9f 100644 --- a/sound/soc/codecs/dfbmcs320.c +++ b/sound/soc/codecs/bt-sco.c @@ -1,5 +1,5 @@ /* - * Driver for the DFBM-CS320 bluetooth module + * Driver for generic Bluetooth SCO link * Copyright 2011 Lars-Peter Clausen <lars(a)metafoo.de> * * This program is free software; you can redistribute it and/or modify it @@ -15,8 +15,8 @@ #include <sound/soc.h> -static struct snd_soc_dai_driver dfbmcs320_dai = { - .name = "dfbmcs320-pcm", +static struct snd_soc_dai_driver bt_sco_dai = { + .name = "bt-sco-pcm", .playback = { .channels_min = 1, .channels_max = 1, @@ -31,32 +31,41 @@ static struct snd_soc_dai_driver dfbmcs320_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_dfbmcs320; +static struct snd_soc_codec_driver soc_codec_dev_bt_sco; -static int dfbmcs320_probe(struct platform_device *pdev) +static int bt_sco_probe(struct platform_device *pdev) { - return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_dfbmcs320, - &dfbmcs320_dai, 1); + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_bt_sco, + &bt_sco_dai, 1); } -static int dfbmcs320_remove(struct platform_device *pdev) +static int bt_sco_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; } -static struct platform_driver dfmcs320_driver = { +static struct platform_device_id bt_sco_driver_ids[] = { + { + .name = "dfbmcs320", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids); + +static struct platform_driver bt_sco_driver = { .driver = { - .name = "dfbmcs320", + .name = "bt-sco", .owner = THIS_MODULE, }, - .probe = dfbmcs320_probe, - .remove = dfbmcs320_remove, + .probe = bt_sco_probe, + .remove = bt_sco_remove, + .id_table = bt_sco_driver_ids, }; -module_platform_driver(dfmcs320_driver); +module_platform_driver(bt_sco_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); -MODULE_DESCRIPTION("ASoC DFBM-CS320 bluethooth module driver"); +MODULE_DESCRIPTION("ASoC generic bluethooth sco link driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 475fb0d..ae0ea87 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -39,7 +39,7 @@ config SND_SOC_SAMSUNG_NEO1973_WM8753 depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA02 select SND_S3C24XX_I2S select SND_SOC_WM8753 - select SND_SOC_DFBMCS320 + select SND_SOC_SCO help Say Y here to enable audio support for the Openmoko Neo1973 Smartphones. diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index e591c38..807db41 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -373,7 +373,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { { /* Voice via BT */ .name = "Bluetooth", .stream_name = "Voice", - .cpu_dai_name = "dfbmcs320-pcm", + .cpu_dai_name = "bt-sco-pcm", .codec_dai_name = "wm8753-voice", .codec_name = "wm8753.0-001a", .ops = &neo1973_voice_ops, -- 1.8.2.3

3 4

Re: [alsa-devel] [PATCH 3/3] sound/soc/fsl: remove leftover release_mem_region
by Mark Brown 22 May '13

22 May '13

On Wed, May 22, 2013 at 03:21:26PM +0200, Wolfram Sang wrote: > When converting this driver to devm_ioremap_resource, the removal of this now > unneeded function has been forgotten. This doesn't apply against current code, can you please regenerate against topic/fsl or mainline?

1 0

[alsa-devel] [PATCH] ALSA: hda - Enable runtime pm for Haswell
by Wang Xingchao 22 May '13

22 May '13

Haswell doesnot support runtime pm by default. This patch let haswell Display HD-A controller enter runtime suspend, and bring more power saving whith power-well. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bf27693..eb25888 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3755,6 +3755,12 @@ static int azx_probe(struct pci_dev *pci, if (pci_dev_run_wake(pci)) pm_runtime_put_noidle(&pci->dev); + else if (chip->driver_caps + & AZX_DCAPS_I915_POWERWELL) { + /* Haswell doesnot support runtime pm by default */ + pm_runtime_put_noidle(&pci->dev); + pm_runtime_allow(&pci->dev); + } dev++; complete_all(&chip->probe_wait); -- 1.7.9.5

3 4

[alsa-devel] [PATCH] ASoC: davinci: fix sample rotation
by Daniel Mack 21 May '13

21 May '13

McASP serial audio engine needs different rotation values on TX and RX channels. Commit dde109fb462 ("ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback") changed the calculation to fix the playback format, but broke the capture stream by doing it for both TXFMT and RXFMT. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Michal Bachraty <michal.bachraty(a)streamunlimited.com> Cc: stable(a)vger.kernel.org [3.9 only] --- sound/soc/davinci/davinci-mcasp.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 63633e9..8d805ec 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -628,7 +628,8 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, int word_length) { u32 fmt; - u32 rotate = (word_length / 4) & 0x7; + u32 tx_rotate = (word_length / 4) & 0x7; + u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; /* @@ -651,9 +652,9 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXSSZ(fmt), TXSSZ(0x0F)); mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, - TXROT(rotate), TXROT(7)); + TXROT(tx_rotate), TXROT(7)); mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, - RXROT(rotate), RXROT(7)); + RXROT(rx_rotate), RXROT(7)); mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); } -- 1.8.1.4

2 1

[alsa-devel] [PATCH 0/2] Set ab8500-codec dai slots from machine drivers
by Fabio Baltieri 21 May '13

21 May '13

Hi Mark, these two patches rework the slot selection code in ab8500-codec to use the slots masks provided by the machine drivers instead of the hardcoded ones as suggested. The patches replaces some of the previous macros with a parametrized version in the effort of making the actual code compact and readable, and should be applied in order. Also, the second patch drops the hardcoded defintions swapped in -rc2, so you may want to rebase the ux500 topic branch to avoid the conflict, or I can resend the patch based on the topic branch directly. Just let me know. Big thanks to Ola for helping me figuring out the weird ab8500 slot mapping logic. Fabio Fabio Baltieri (2): ASoC: ab8500-codec: Set tx dai slots from tx_mask ASoC: ab8500-codec: Set rx dai slots from rx_mask sound/soc/codecs/ab8500-codec.c | 60 ++++++++++++++++++++++++++--------------- sound/soc/codecs/ab8500-codec.h | 42 +++++++++++++++-------------- 2 files changed, 61 insertions(+), 41 deletions(-) -- 1.8.2

2 5

[alsa-devel] [PATCH 0/2 V4] Power-well API implementation for Haswell
by Wang Xingchao 21 May '13

21 May '13

Hi all, This is V4 and here're some changes notes: change between V3-->V4: - add new structure i915_power_well - initialize drm_device pointer at module init time - change function name change between V2-->V3: - make SND_HDA_I915 selectable - use snd_printdd to output message - add return error code check - use symbol_request to replace symbol_get - release power_well at azx_free - some typo fixes changes between V1-->V2: - use reference count to track power-well usage - remove external module, compiled into snd-hda-intel instead - manage symbols and module loading properly - remove IS_HSW macro, use flag instead - remove audio callback for gfx driver to avoid dependency - split whole patch into two pieces for easy review - more typo fixes Wang Xingchao (2): i915/drm: Add private api for power well usage ALSA: hda - Add power-welll support for haswell HDA drivers/gpu/drm/i915/i915_dma.c | 6 +++ drivers/gpu/drm/i915/i915_drv.h | 12 +++++ drivers/gpu/drm/i915/intel_drv.h | 4 ++ drivers/gpu/drm/i915/intel_pm.c | 98 +++++++++++++++++++++++++++++++++++--- include/drm/i915_powerwell.h | 36 ++++++++++++++ sound/pci/hda/Kconfig | 10 ++++ sound/pci/hda/Makefile | 3 ++ sound/pci/hda/hda_i915.c | 75 +++++++++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 35 ++++++++++++++ sound/pci/hda/hda_intel.c | 41 ++++++++++++++-- 10 files changed, 310 insertions(+), 10 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.7.9.5

3 7

Re: [alsa-devel] [Alsa-devel] add support for syba 7.1 surround sound card
by Clemens Ladisch 21 May '13

21 May '13

good1.2guy(a)gmail.com wrote: > Summary: add support for syba 7.1 surround sound card > Keywords: syba via VT1723 Tremor > Kernel: 3.8.8-203.fc18.x86_64 What are these three lines for? > add support for syba 7.1 surround sound card > actually the card only has 6 dmas, so it is really only 5.1 The website shows a daughterboard. > added code to force_rate setting to not fail EBUSY when the dmas are running. > suggest using force_rate with pulseaudio to avoid loops with error logging. This sounds as if it has nothing to do with this particular SYBA card but affects all VT1723 cards. Please move it into a separate patch. And there should be no reason for a module parameter; if this is a bug in the driver and/or in PulseAudio, it should be just fixed. > Signed-off-by: psmith2004(a)yahoo.com Real names only. Please read Documentation/SubmittingPatches. > +static void syba_quirk_open(struct snd_ice1712 *ice) > +{ > + unsigned long flags; It's common to have an empty line here. > + spin_lock_irqsave(&ice->reg_lock, flags); > + outb(0xff, ICEREG1724(ice, AC97_CFG)); > + udelay(500); > + outb(ice->eeprom.data[ICE_EEP2_ACLINK], ICEREG1724(ice, AC97_CFG)); > + spin_unlock_irqrestore(&ice->reg_lock, flags); > +} What is the reason for this quirk? Is it possible that this might be needed for other cards using a AC'97 codec? Regards, Clemens

1 0

[alsa-devel] what is the difference between SNDRV_PCM_IOCTL_WRITEN_FRAMES and SNDRV_PCM_IOCTL_WRITEI_FRAMES
by Zhang wei 21 May '13

21 May '13

I know the format likes below when use SNDRV_PCM_IOCTL_WRITEI_FRAMES <http://lxr.free-electrons.com/ident?i=SNDRV_PCM_IOCTL_WRITEN_FRAMES> L+R+L+R+L+R... L:a frame left channel data. R:a frame right channel data. Right? But what is the kind of format when use SNDRV_PCM_IOCTL_WRITEN_FRAMES <http://lxr.free-electrons.com/ident?i=SNDRV_PCM_IOCTL_WRITEN_FRAMES> ? Thank you:-) Br zw

2 4

[alsa-devel] [PATCH] ASoC: wm8994: Handle LRCLK inversion for WM8958 and WM1811A
by Mark Brown 20 May '13

20 May '13

On WM8958 and WM1811A separate control of the LRCLK inversion bit is available for the DAC and ADC LRCLKs which for compatibility reasons is done in a new register bit. Since writes to each scheme have no effect on parts using the other just always write to both for simplicity. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> Tested-by: Samreen Nilofer <samreen.nilofer(a)intel.com> --- include/linux/mfd/wm8994/registers.h | 8 ++++++++ sound/soc/codecs/wm8994.c | 14 ++++++++++++++ 2 files changed, 22 insertions(+) diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h index 0535489..db8cef3 100644 --- a/include/linux/mfd/wm8994/registers.h +++ b/include/linux/mfd/wm8994/registers.h @@ -2668,6 +2668,10 @@ /* * R772 (0x304) - AIF1ADC LRCLK */ +#define WM8958_AIF1_LRCLK_INV 0x1000 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_MASK 0x1000 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_SHIFT 12 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_WIDTH 1 /* AIF1_LRCLK_INV */ #define WM8994_AIF1ADC_LRCLK_DIR 0x0800 /* AIF1ADC_LRCLK_DIR */ #define WM8994_AIF1ADC_LRCLK_DIR_MASK 0x0800 /* AIF1ADC_LRCLK_DIR */ #define WM8994_AIF1ADC_LRCLK_DIR_SHIFT 11 /* AIF1ADC_LRCLK_DIR */ @@ -2679,6 +2683,10 @@ /* * R773 (0x305) - AIF1DAC LRCLK */ +#define WM8958_AIF1_LRCLK_INV 0x1000 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_MASK 0x1000 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_SHIFT 12 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_WIDTH 1 /* AIF1_LRCLK_INV */ #define WM8994_AIF1DAC_LRCLK_DIR 0x0800 /* AIF1DAC_LRCLK_DIR */ #define WM8994_AIF1DAC_LRCLK_DIR_MASK 0x0800 /* AIF1DAC_LRCLK_DIR */ #define WM8994_AIF1DAC_LRCLK_DIR_SHIFT 11 /* AIF1DAC_LRCLK_DIR */ diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index cb67603..0bcbd12 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2574,17 +2574,24 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct wm8994 *control = wm8994->wm8994; int ms_reg; int aif1_reg; + int dac_reg; + int adc_reg; int ms = 0; int aif1 = 0; + int lrclk = 0; switch (dai->id) { case 1: ms_reg = WM8994_AIF1_MASTER_SLAVE; aif1_reg = WM8994_AIF1_CONTROL_1; + dac_reg = WM8994_AIF1DAC_LRCLK; + adc_reg = WM8994_AIF1ADC_LRCLK; break; case 2: ms_reg = WM8994_AIF2_MASTER_SLAVE; aif1_reg = WM8994_AIF2_CONTROL_1; + dac_reg = WM8994_AIF1DAC_LRCLK; + adc_reg = WM8994_AIF1ADC_LRCLK; break; default: return -EINVAL; @@ -2603,6 +2610,7 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif1 |= WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x18; break; @@ -2641,12 +2649,14 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; case SND_SOC_DAIFMT_IB_IF: aif1 |= WM8994_AIF1_BCLK_INV | WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; break; case SND_SOC_DAIFMT_IB_NF: aif1 |= WM8994_AIF1_BCLK_INV; break; case SND_SOC_DAIFMT_NB_IF: aif1 |= WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; break; default: return -EINVAL; @@ -2677,6 +2687,10 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) aif1); snd_soc_update_bits(codec, ms_reg, WM8994_AIF1_MSTR, ms); + snd_soc_update_bits(codec, dac_reg, + WM8958_AIF1_LRCLK_INV, lrclk); + snd_soc_update_bits(codec, adc_reg, + WM8958_AIF1_LRCLK_INV, lrclk); return 0; } -- 1.7.10.4

1 0

[alsa-devel] [PATCH v2] ASoC: soc-compress: Send correct stream event for capture start
by Charles Keepax 20 May '13

20 May '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3853f7e..06a8000 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -220,8 +220,12 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, goto err; } - snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_START); + if (cstream->direction == SND_COMPRESS_PLAYBACK) + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, + SND_SOC_DAPM_STREAM_START); + else + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, + SND_SOC_DAPM_STREAM_START); /* cancel any delayed stream shutdown that is pending */ rtd->pop_wait = 0; -- 1.7.2.5

2 1

Re: [alsa-devel] [PATCH] ASoC: max98090: request IRQF_ONESHOT interrupt
by Mark Brown 20 May '13

20 May '13

On Sun, May 19, 2013 at 10:58:07PM -0700, Andrew Bresticker wrote: > request_threaded_irq() rejects calls which both do not specify a handler > (indicating that the primary IRQ handler should be used) and do not set > IRQF_ONESHOT because the combination is unsafe with level-triggered > interrupts. It is safe in this case, though, since max98090 IRQs are Applied, thanks. If request_threaded_irq() is rejecting these that's not ideal - some threaded interrupts can only be generated from within threaded context and therefore can't have primary handlers. Unless it only rejects in the case where the parent IRQ is a hard IRQ of course.

1 0

[alsa-devel] [PATCH 0/2 V3] Power-well API implementation for Haswell
by Wang Xingchao 20 May '13

20 May '13

Hi all, This is V3 and some fixes after Takashi's review. change between V2-->V3: - make SND_HDA_I915 selectable - use snd_printdd to output message - add return error code check - use symbol_request to replace symbol_get - release power_well at azx_free - some typo fixes changes between V1-->V2: - use reference count to track power-well usage - remove external module, compiled into snd-hda-intel instead - manage symbols and module loading properly - remove IS_HSW macro, use flag instead - remove audio callback for gfx driver to avoid dependency - split whole patch into two pieces for easy review - more typo fixes Wang Xingchao (2): drm/915: Add private api for power well usage ALSA: hda - Add power-welll support for haswell HDA drivers/gpu/drm/i915/intel_pm.c | 76 +++++++++++++++++++++++++++++++++++---- include/drm/i915_powerwell.h | 36 +++++++++++++++++++ sound/pci/hda/Kconfig | 10 ++++++ sound/pci/hda/Makefile | 3 ++ sound/pci/hda/hda_i915.c | 75 ++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 35 ++++++++++++++++++ sound/pci/hda/hda_intel.c | 41 +++++++++++++++++++-- 7 files changed, 262 insertions(+), 10 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.7.9.5

3 6

Re: [alsa-devel] [PATCH v2] ASoC: dfbmcs320: make the driver common for other BT modules
by Mark Brown 18 May '13

18 May '13

On Fri, May 17, 2013 at 10:19:37AM +0800, Barry Song wrote: > DFBM-CS320 is only one of bluetooth modules using CSR bluetooth chips, > we don't want everyone to have a seperate codec driver. anyway, the > feature of Bluetooth handsfree is same on all platforms, so this patch > makes the DFBM-CS320 driver become a common BT handsfree driver. Sorry, should have thought of this before but why are we calling this "handsfree"? It's just a generic bluetooth link as far as the host system is concerned, "bt-sco" might be more appropriate? Otherwise this looks OK for ASoC though I'd expect to see a change to registration of the OpenMoko BT device (perhaps I missed it though).

3 4

[alsa-devel] [patch] ASoC: wm8994: missing break in wm8994_get_fll_config()
by Dan Carpenter 18 May '13

18 May '13

Smatch complains that: sound/soc/codecs/wm8994.c:2087 wm8994_get_fll_config() warn: missing break? reassigning 'fll->k' Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 9f32dd8..303d755 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2080,6 +2080,7 @@ static int wm8994_get_fll_config(struct wm8994 *control, struct fll_div *fll, fll->k = K / 10; pr_debug("N=%x K=%x\n", fll->n, fll->k); + break; default: gcd_fll = gcd(freq_out, freq_in);

2 1

[alsa-devel] [PATCH v3] ASoC: dfbmcs320: make the driver common for other BT modules
by Barry Song 18 May '13

18 May '13

DFBM-CS320 is only one of bluetooth modules using CSR bluetooth chips, we don't want everyone to have a seperate codec driver. anyway, the feature of Bluetooth SCO is same on all platforms, so this patch makes the DFBM-CS320 driver become a common BT SCO link driver. Cc: Graeme Gregory <graeme(a)openmoko.org> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- -v3: rename to sco according to Mark's comment sound/soc/codecs/Kconfig | 4 ++-- sound/soc/codecs/Makefile | 4 ++-- sound/soc/codecs/{dfbmcs320.c => bt-sco.c} | 37 +++++++++++++++++++----------- sound/soc/samsung/neo1973_wm8753.c | 2 +- 4 files changed, 28 insertions(+), 19 deletions(-) rename sound/soc/codecs/{dfbmcs320.c => bt-sco.c} (53%) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d8c4f3d..386292d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -40,7 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7213 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C - select SND_SOC_DFBMCS320 + select SND_SOC_BT_SCO select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -263,7 +263,7 @@ config SND_SOC_DA732X config SND_SOC_DA9055 tristate -config SND_SOC_DFBMCS320 +config SND_SOC_BT_SCO tristate config SND_SOC_DMIC diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 49ff127..82aa34e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -27,7 +27,7 @@ snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o -snd-soc-dfbmcs320-objs := dfbmcs320.o +snd-soc-bt-sco-objs := bt-sco.o snd-soc-dmic-objs := dmic.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o @@ -154,7 +154,7 @@ obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o -obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o +obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o diff --git a/sound/soc/codecs/dfbmcs320.c b/sound/soc/codecs/bt-sco.c similarity index 53% rename from sound/soc/codecs/dfbmcs320.c rename to sound/soc/codecs/bt-sco.c index 4f4f7f4..a081d9f 100644 --- a/sound/soc/codecs/dfbmcs320.c +++ b/sound/soc/codecs/bt-sco.c @@ -1,5 +1,5 @@ /* - * Driver for the DFBM-CS320 bluetooth module + * Driver for generic Bluetooth SCO link * Copyright 2011 Lars-Peter Clausen <lars(a)metafoo.de> * * This program is free software; you can redistribute it and/or modify it @@ -15,8 +15,8 @@ #include <sound/soc.h> -static struct snd_soc_dai_driver dfbmcs320_dai = { - .name = "dfbmcs320-pcm", +static struct snd_soc_dai_driver bt_sco_dai = { + .name = "bt-sco-pcm", .playback = { .channels_min = 1, .channels_max = 1, @@ -31,32 +31,41 @@ static struct snd_soc_dai_driver dfbmcs320_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_dfbmcs320; +static struct snd_soc_codec_driver soc_codec_dev_bt_sco; -static int dfbmcs320_probe(struct platform_device *pdev) +static int bt_sco_probe(struct platform_device *pdev) { - return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_dfbmcs320, - &dfbmcs320_dai, 1); + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_bt_sco, + &bt_sco_dai, 1); } -static int dfbmcs320_remove(struct platform_device *pdev) +static int bt_sco_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; } -static struct platform_driver dfmcs320_driver = { +static struct platform_device_id bt_sco_driver_ids[] = { + { + .name = "dfbmcs320", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids); + +static struct platform_driver bt_sco_driver = { .driver = { - .name = "dfbmcs320", + .name = "bt-sco", .owner = THIS_MODULE, }, - .probe = dfbmcs320_probe, - .remove = dfbmcs320_remove, + .probe = bt_sco_probe, + .remove = bt_sco_remove, + .id_table = bt_sco_driver_ids, }; -module_platform_driver(dfmcs320_driver); +module_platform_driver(bt_sco_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); -MODULE_DESCRIPTION("ASoC DFBM-CS320 bluethooth module driver"); +MODULE_DESCRIPTION("ASoC generic bluethooth sco link driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index e591c38..807db41 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -373,7 +373,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { { /* Voice via BT */ .name = "Bluetooth", .stream_name = "Voice", - .cpu_dai_name = "dfbmcs320-pcm", + .cpu_dai_name = "bt-sco-pcm", .codec_dai_name = "wm8753-voice", .codec_name = "wm8753.0-001a", .ops = &neo1973_voice_ops, -- 1.8.2.3

1 0

[alsa-devel] [PATCH 0/6] Second set of fixes for ux500 ASoC drivers
by Fabio Baltieri 18 May '13

18 May '13

Hi, These are some others fixes for various small issues I found while testing the ux500 ASoC driver on -next. Patch 1 adds some missing declarations for AD controls that were causing some weird behaviour in alsamixer, as the default state was outside the declared range. Patch 2 fixes a kernel crash when opening and closing the audio device without sending any data. Patch 3 drops pinctrl code altogether from the driver. The actual implementation is buggy as the pins are only registered to the playback interfaces, which gives a bunch of warnings during kernel startup and also kills the capture interface by setting the shared pins to hi-z mode even if that's still active. As putting those pins in high-z is not really needed and was removed from the internal STE driver anyway, I'm just dropping that code form here as well. In parallel, I'm sending a pinctrl patch to declare those pin as a hog. Patches 4 to 6 fixes some weirdness with time slot usage. After this series the driver seems to work fine for both capture and playback interface (tested on a snowball v11). Thanks, Fabio Fabio Baltieri (6): ASoC: ab8500-codec: Add missing ad_to_slot definitions ASoC: ux500: Do not clear state if already idle ASoC: ux500: Drop pinctrl sleep support ASoC: ux500: Update tx tdm slots configuration ASoC: ux500: Swap even/odd AD slot definitions ASoC: ux500: Use the first two AD slots for capture sound/soc/codecs/ab8500-codec.c | 39 +++++++++++++++------------ sound/soc/codecs/ab8500-codec.h | 36 ++++++++++++------------- sound/soc/ux500/mop500_ab8500.c | 4 +-- sound/soc/ux500/ux500_msp_i2s.c | 58 +++-------------------------------------- sound/soc/ux500/ux500_msp_i2s.h | 6 ----- 5 files changed, 46 insertions(+), 97 deletions(-) -- 1.8.2

4 54

[alsa-devel] [PATCH] ASoC: soc-compress: Send correct stream event for capture start
by Charles Keepax 17 May '13

17 May '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Sorry little bug I missed due to some integration bugs in our setup here. Again this applies onto Takashi's sound for-next branch. Thanks, Charles sound/soc/soc-compress.c | 9 +++++++-- 1 files changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3853f7e..a5107df 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -220,8 +220,13 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, goto err; } - snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_START); + if (cstream->direction == SND_COMPRESS_PLAYBACK) { + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, + SND_SOC_DAPM_STREAM_START); + } else { + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, + SND_SOC_DAPM_STREAM_START); + } /* cancel any delayed stream shutdown that is pending */ rtd->pop_wait = 0; -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: wm8994: Handle LRCLK inversion for WM8958 and WM1811A
by Mark Brown 17 May '13

17 May '13

On WM8958 and WM1811A separate control of the LRCLK inversion bit is available for the DAC and ADC LRCLKs which for compatibility reasons is done in a new register bit. Since writes to each scheme have no effect on parts using the other just always write to both for simplicity. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index ab7ebd9..74238bd 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2573,17 +2573,24 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct wm8994 *control = wm8994->wm8994; int ms_reg; int aif1_reg; + int dac_reg; + int adc_reg; int ms = 0; int aif1 = 0; + int lrclk = 0; switch (dai->id) { case 1: ms_reg = WM8994_AIF1_MASTER_SLAVE; aif1_reg = WM8994_AIF1_CONTROL_1; + dac_reg = WM8994_AIF1DAC_LRCLK; + adc_reg = WM8994_AIF1ADC_LRCLK; break; case 2: ms_reg = WM8994_AIF2_MASTER_SLAVE; aif1_reg = WM8994_AIF2_CONTROL_1; + dac_reg = WM8994_AIF1DAC_LRCLK; + adc_reg = WM8994_AIF1ADC_LRCLK; break; default: return -EINVAL; @@ -2602,6 +2609,7 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif1 |= WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x18; break; @@ -2640,12 +2648,14 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; case SND_SOC_DAIFMT_IB_IF: aif1 |= WM8994_AIF1_BCLK_INV | WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; break; case SND_SOC_DAIFMT_IB_NF: aif1 |= WM8994_AIF1_BCLK_INV; break; case SND_SOC_DAIFMT_NB_IF: aif1 |= WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; break; default: return -EINVAL; @@ -2676,6 +2686,10 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) aif1); snd_soc_update_bits(codec, ms_reg, WM8994_AIF1_MSTR, ms); + snd_soc_update_bits(codec, dac_reg, + WM8958_AIF1_LRCLK_INV, lrclk); + snd_soc_update_bits(codec, adc_reg, + WM8958_AIF1_LRCLK_INV, lrclk); return 0; } -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm5110: Add missing speaker initialisation
by Mark Brown 17 May '13

17 May '13

Add callback to initialise the speaker in the core following the recent changes to handling of integration with the thermal interrupts. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index e77410e..91a7015 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1252,6 +1252,8 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; + arizona_init_spk(codec); + snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); priv->core.arizona->dapm = &codec->dapm; -- 1.7.10.4

1 0

[alsa-devel] module dependency between i915 and snd-hda-intel
by Wang, Xingchao 17 May '13

17 May '13

Hi Takashi, Seems kernel building will determine the module loading sequence, this would fix the dependency for i915 and snd-hda-intel. If we call the i915 module API directly in snd-hda-intel side, kernel building will change the module.dep accordingly like: " kernel/sound/pci/hda/snd-hda-intel.ko: kernel/drivers/gpu/drm/i915/i915.ko kernel/drivers/gpu/dr m/drm_kms_helper.ko kernel/drivers/gpu/drm/drm.ko kernel/drivers/i2c/algos/i2c-algo-bit.ko kerne l/drivers/acpi/video.ko kernel/sound/pci/hda/snd-hda-codec.ko kernel/sound/core/snd-hwdep.ko ker nel/sound/core/snd-pcm.ko kernel/sound/core/snd-timer.ko kernel/sound/core/snd.ko kernel/sound/s oundcore.ko kernel/sound/core/snd-page-alloc.ko " So snd-hda-intel will wait for i915 loading. Even I added i915 into blacklist.conf, it will force load i915.ko. So I think the only bad case is no i915 module built-in, that would cause haswell hda initialize fail, and the patch would output error message. thanks --xingchao

2 6

[alsa-devel] [PATCH 1/3] ASoC: arizona: Provide simple DAI ops for autoconfiguring interfaces
by Mark Brown 17 May '13

17 May '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 7 +++++++ sound/soc/codecs/arizona.h | 1 + 2 files changed, 8 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 90c3db1..4a42041 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1198,6 +1198,13 @@ const struct snd_soc_dai_ops arizona_dai_ops = { }; EXPORT_SYMBOL_GPL(arizona_dai_ops); +const struct snd_soc_dai_ops arizona_simple_dai_ops = { + .startup = arizona_startup, + .hw_params = arizona_hw_params_rate, + .set_sysclk = arizona_dai_set_sysclk, +}; +EXPORT_SYMBOL_GPL(arizona_simple_dai_ops); + int arizona_init_dai(struct arizona_priv *priv, int id) { struct arizona_dai_priv *dai_priv = &priv->dai[id]; diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index af39f10..a1a5dad 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -213,6 +213,7 @@ extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); extern const struct snd_soc_dai_ops arizona_dai_ops; +extern const struct snd_soc_dai_ops arizona_simple_dai_ops; #define ARIZONA_FLL_NAME_LEN 20 -- 1.7.10.4

1 2

[alsa-devel] [PATCH] ASoC: simplify registration of snd-soc-dummy device
by Uwe Kleine-König 17 May '13

17 May '13

Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/soc-utils.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 4b3be6c..29b211e 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -159,15 +159,10 @@ int __init snd_soc_util_init(void) { int ret; - soc_dummy_dev = platform_device_alloc("snd-soc-dummy", -1); - if (!soc_dummy_dev) - return -ENOMEM; - - ret = platform_device_add(soc_dummy_dev); - if (ret != 0) { - platform_device_put(soc_dummy_dev); - return ret; - } + soc_dummy_dev = + platform_device_register_simple("snd-soc-dummy", -1, NULL, 0); + if (IS_ERR(soc_dummy_dev)) + return PTR_ERR(soc_dummy_dev); ret = platform_driver_register(&soc_dummy_driver); if (ret != 0) -- 1.8.2.1

2 1

Re: [alsa-devel] [PATCH v2] ASoC: max98090: add digital mic mux to record path
by Mark Brown 17 May '13

17 May '13

On Thu, May 16, 2013 at 12:03:54PM -0700, Andrew Bresticker wrote: > The max98090 driver currently treats the digital mic enable as a supply > on the record path, causing the digital mic enable to always be turned on > when attempting to record. This is incorrect, however, since the digital Applied, thanks.

1 0

[alsa-devel] [PATCH v2] ALSA: hda - add PCI IDs for Intel BayTrail
by Chew Chiau Ee 17 May '13

17 May '13

From: Chew, Chiau Ee <chiau.ee.chew(a)intel.com> Add HD Audio Device PCI ID for the Intel BayTrail platform. Signed-off-by: Chew, Chiau Ee <chiau.ee.chew(a)intel.com> Signed-off-by: Artem Bityutskiy <artem.bityutskiy(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bcd40ee..fa902a4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3847,6 +3847,9 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* Oaktrail */ { PCI_DEVICE(0x8086, 0x080a), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, + /* BayTrail */ + { PCI_DEVICE(0x8086, 0x0f04), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* ICH */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | -- 1.7.4.4

2 1

Re: [alsa-devel] [PATCH] ALSA: usb-audio: proc: use found syncmaxsize to determine feedback format
by Takashi Iwai 17 May '13

17 May '13

At Thu, 16 May 2013 20:26:17 +0200, Torstein Hegge wrote: > > freqshift is only set for the data endpoint and syncmaxsize is only set > for the sync endpoint. This results in a syncmaxsize of zero used in the > proc output feedback format calculation, which gives a feedback format > incorrectly shown as 8.16 for UAC2 devices. > > As neither the data nor the sync endpoint gives all the relevant > content, output the two combined. > > Also remove the sync_endpoint "packet size" which is always zero > and the sync_endpoint "momentary freq" which is constant. > > Tested with UAC2 async and UAC1 adaptive, not tested with UAC1 async. > > Reported-by: B. Zhang <bb.zhang(a)free.fr> > Signed-off-by: Torstein Hegge <hegge(a)resisty.net> Thanks, applied. Takashi > --- > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: Daniel Mack <zonque(a)gmail.com> > Cc: Clemens Ladisch <clemens(a)ladisch.de> > Cc: Davor Herga <davor.herga(a)guest.arnes.si> > > sound/usb/proc.c | 22 +++++++++++----------- > 1 file changed, 11 insertions(+), 11 deletions(-) > > diff --git a/sound/usb/proc.c b/sound/usb/proc.c > index d218f76..462a5ed 100644 > --- a/sound/usb/proc.c > +++ b/sound/usb/proc.c > @@ -117,21 +117,22 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s > } > > static void proc_dump_ep_status(struct snd_usb_substream *subs, > - struct snd_usb_endpoint *ep, > + struct snd_usb_endpoint *data_ep, > + struct snd_usb_endpoint *sync_ep, > struct snd_info_buffer *buffer) > { > - if (!ep) > + if (!data_ep) > return; > - snd_iprintf(buffer, " Packet Size = %d\n", ep->curpacksize); > + snd_iprintf(buffer, " Packet Size = %d\n", data_ep->curpacksize); > snd_iprintf(buffer, " Momentary freq = %u Hz (%#x.%04x)\n", > subs->speed == USB_SPEED_FULL > - ? get_full_speed_hz(ep->freqm) > - : get_high_speed_hz(ep->freqm), > - ep->freqm >> 16, ep->freqm & 0xffff); > - if (ep->freqshift != INT_MIN) { > - int res = 16 - ep->freqshift; > + ? get_full_speed_hz(data_ep->freqm) > + : get_high_speed_hz(data_ep->freqm), > + data_ep->freqm >> 16, data_ep->freqm & 0xffff); > + if (sync_ep && data_ep->freqshift != INT_MIN) { > + int res = 16 - data_ep->freqshift; > snd_iprintf(buffer, " Feedback Format = %d.%d\n", > - (ep->syncmaxsize > 3 ? 32 : 24) - res, res); > + (sync_ep->syncmaxsize > 3 ? 32 : 24) - res, res); > } > } > > @@ -141,8 +142,7 @@ static void proc_dump_substream_status(struct snd_usb_substream *subs, struct sn > snd_iprintf(buffer, " Status: Running\n"); > snd_iprintf(buffer, " Interface = %d\n", subs->interface); > snd_iprintf(buffer, " Altset = %d\n", subs->altset_idx); > - proc_dump_ep_status(subs, subs->data_endpoint, buffer); > - proc_dump_ep_status(subs, subs->sync_endpoint, buffer); > + proc_dump_ep_status(subs, subs->data_endpoint, subs->sync_endpoint, buffer); > } else { > snd_iprintf(buffer, " Status: Stop\n"); > } > -- > 1.8.3.rc1.377.g7010c6b >

1 0

Re: [alsa-devel] [PATCH] ASoC: dfbmcs320: make the driver common for other BT modules
by Lars-Peter Clausen 16 May '13

16 May '13

On 05/16/2013 08:53 AM, Barry Song wrote: > DFBM-CS320 is only one of bluetooth modules using CSR bluetooth chips, > we don't want everyone to have a seperate codec driver. anyway, the > feature of Bluetooth handsfree is same on all platforms, so this patch > makes the DFBM-CS320 driver become a common BT handsfree driver. [...] > -static struct platform_driver dfmcs320_driver = { > +static struct platform_driver bt_handsfree_driver = { > .driver = { > - .name = "dfbmcs320", > + .name = "bt-handsfree", > .owner = THIS_MODULE, In my opinion it's a good idea to keep a platform_id list of all the supported devices. > }, > - .probe = dfbmcs320_probe, > - .remove = dfbmcs320_remove, > + .probe = bt_handsfree_probe, > + .remove = bt_handsfree_remove, > }; > > -module_platform_driver(dfmcs320_driver); > +module_platform_driver(bt_handsfree_driver);

3 4

[alsa-devel] [PATCH] ASoC: wm5102: Correct OSR control name for EPOUT
by Charles Keepax 16 May '13

16 May '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index cb03cc4..803d1b8 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -814,7 +814,7 @@ SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, SOC_VALUE_ENUM("HPOUT1 OSR", wm5102_hpout_osr[0]), SOC_VALUE_ENUM("HPOUT2 OSR", wm5102_hpout_osr[1]), -SOC_VALUE_ENUM("HPOUT3 OSR", wm5102_hpout_osr[2]), +SOC_VALUE_ENUM("EPOUT OSR", wm5102_hpout_osr[2]), SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), -- 1.7.2.5

2 1

[alsa-devel] Debian alsa-plugins patches
by Jordi Mallach 16 May '13

16 May '13

As previously discussed, I'm posting all Debian patches against alsa-plugins that were pending upstreaming Thanks, Jordi

3 8

[alsa-devel] Only stereo output possible with ATI Tahiti XT HDMI Audio [Radeon HD 7970 Series] [1002:aaa0]
by Olivier Langlois 16 May '13

16 May '13

I have an HDMI chain with a Radeon card on 1 side and a 1080P monitor at the other end with a Pioner SC-09TX A/V Receiver in the middle. Video is transmitted fine. 2 channels pcm plays fine on the receiver. I am totally unable to output sound with 8 channels. lano1106@whippet2 /proc/asound/card0 $ speaker-test -c8 speaker-test 1.0.27 Playback device is default Stream parameters are 48000Hz, S16_LE, 8 channels Using 16 octaves of pink noise Channels count (8) not available for playbacks: Invalid argument Setting of hwparams failed: Invalid argument My neophyte instinct tells me that the problem is interoperability issue between the Radeon Audio controller and the SC-09TX receiver. That would be great to be able to tweak the hda driver to workaround the problem but if I was able to manually program the hda codec module to accept 8 channel LPCM input, that would be good enough for me. My understanding is that in the advent of HDMI handshake failure, the sound source fallbacks to the bare minimum requirement that the HDMI standard requires. I have started to read Intel HDA manual. What I would like to know is: Is it known if my particular HDA device is able to output 48KHz 8 channels LPCM? would it configurable manually? Which HDA verbs should get familiar with if I want to tweak the code to configure the controller manually? Thank you, Olivier lano1106@whippet2 ~ $ lspci -nn | grep Audio 01:00.1 Audio device [0403]: Advanced Micro Devices [AMD] nee ATI Tahiti XT HDMI Audio [Radeon HD 7970 Series] [1002:aaa0] Linux: 3.8.12 alsalib:1.0.27 lano1106@whippet2 /proc/asound/card0 $ dmesg | grep ALSA [ 6.351490] ALSA sound/pci/hda/hda_intel.c:2848 0000:01:00.1: Handle VGA-switcheroo audio client [ 6.351491] ALSA sound/pci/hda/hda_intel.c:3040 0000:01:00.1: Using LPIB position fix [ 6.351493] ALSA sound/pci/hda/hda_intel.c:3157 0000:01:00.1: Force to non-snoop mode [ 6.351619] ALSA sound/pci/hda/hda_intel.c:3282 0000:01:00.1: chipset global capabilities = 0x0 [ 6.354036] ALSA sound/pci/hda/hda_intel.c:1131 0000:01:00.1: codec_mask = 0x1 [ 6.354061] ALSA sound/pci/hda/hda_intel.c:1600 0000:01:00.1: codec #0 probed OK [ 6.354062] ALSA sound/pci/hda/hda_intel.c:1716 0000:01:00.1: Enable sync_write for stable communication [ 6.411369] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=3 Presence_Detect=0 ELD_Valid=0 [ 6.411397] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=5 Presence_Detect=0 ELD_Valid=0 [ 6.411424] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=7 Presence_Detect=0 ELD_Valid=0 [ 6.411453] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=9 Presence_Detect=0 ELD_Valid=0 [ 6.411482] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=11 Presence_Detect=0 ELD_Valid=0 [ 6.411512] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=13 Presence_Detect=0 ELD_Valid=0 [ 29.487577] ALSA sound/pci/hda/patch_hdmi.c:951 HDMI hot plug event: Codec=0 Pin=3 Presence_Detect=1 ELD_Valid=0 [ 29.487593] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=3 Presence_Detect=1 ELD_Valid=1 [ 29.487605] ALSA sound/pci/hda/hda_eld.c:337 HDMI: ELD buf size is 0, force 128 [ 29.487617] ALSA sound/pci/hda/hda_eld.c:356 HDMI: invalid ELD data byte 0 lano1106@whippet2 /proc/asound/card0 $ cat eld#0.0 monitor_present 1 eld_valid 0 lano1106@whippet2 /proc/asound/card0 :( $ cat codec#0 Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100300 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x5]: PCM AC3 Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=1, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 Node 0x03 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="HDMI/DP,pcm=3 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="ELD", index=0, device=3 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=01, enabled=1 Connection: 1 0x02 Node 0x04 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 Node 0x05 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="HDMI/DP,pcm=7 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=1, device=0 Control: name="IEC958 Playback Pro Mask", index=1, device=0 Control: name="IEC958 Playback Default", index=1, device=0 Control: name="IEC958 Playback Switch", index=1, device=0 Control: name="ELD", index=0, device=7 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=02, enabled=1 Connection: 1 0x04 Node 0x06 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 Node 0x07 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="HDMI/DP,pcm=8 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=2, device=0 Control: name="IEC958 Playback Pro Mask", index=2, device=0 Control: name="IEC958 Playback Default", index=2, device=0 Control: name="IEC958 Playback Switch", index=2, device=0 Control: name="ELD", index=0, device=8 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=03, enabled=1 Connection: 1 0x06 Node 0x08 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 Node 0x09 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="HDMI/DP,pcm=9 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=3, device=0 Control: name="IEC958 Playback Pro Mask", index=3, device=0 Control: name="IEC958 Playback Default", index=3, device=0 Control: name="IEC958 Playback Switch", index=3, device=0 Control: name="ELD", index=0, device=9 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=04, enabled=1 Connection: 1 0x08 Node 0x0a [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 Node 0x0b [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="HDMI/DP,pcm=10 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=4, device=0 Control: name="IEC958 Playback Pro Mask", index=4, device=0 Control: name="IEC958 Playback Default", index=4, device=0 Control: name="IEC958 Playback Switch", index=4, device=0 Control: name="ELD", index=0, device=10 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=05, enabled=1 Connection: 1 0x0a Node 0x0c [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 Node 0x0d [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="HDMI/DP,pcm=11 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=5, device=0 Control: name="IEC958 Playback Pro Mask", index=5, device=0 Control: name="IEC958 Playback Default", index=5, device=0 Control: name="IEC958 Playback Switch", index=5, device=0 Control: name="ELD", index=0, device=11 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=06, enabled=1 Connection: 1 0x0c whippet2 /sys/kernel/debug/tracing # cat trace_pipe kworker/u:5-130 [002] ...1 79390.094652: hda_send_cmd: [0:0] val=3f0900 kworker/u:5-130 [002] ...1 79390.094665: hda_get_response: [0:0] val=ffffffff kworker/u:5-130 [002] ...1 79390.094667: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [002] ...1 79390.094677: hda_get_response: [0:0] val=0 kworker/u:5-130 [002] ...1 79390.094679: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [002] ...1 79390.094689: hda_get_response: [0:0] val=0 kworker/u:5-130 [002] ...1 79390.394001: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [002] ...1 79390.394045: hda_get_response: [0:0] val=0 kworker/u:5-130 [002] ...1 79390.394050: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [002] ...1 79390.394071: hda_get_response: [0:0] val=0 kworker/u:5-130 [002] ...1 79390.694227: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [002] ...1 79390.694281: hda_get_response: [0:0] val=0 kworker/u:5-130 [002] ...1 79390.694286: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [002] ...1 79390.694307: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28163.617406: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [000] ...1 28163.617420: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28163.617425: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [000] ...1 28163.617436: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28163.917628: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [000] ...1 28163.917642: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28163.917647: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [000] ...1 28163.917657: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28164.217846: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [000] ...1 28164.217860: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28164.217865: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [000] ...1 28164.217875: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28164.518101: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [000] ...1 28164.518115: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28164.518119: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [000] ...1 28164.518130: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28164.818310: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [000] ...1 28164.818323: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28164.818328: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [000] ...1 28164.818339: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28165.118549: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [000] ...1 28165.118563: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28165.118568: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [000] ...1 28165.118579: hda_get_response: [0:0] val=0

1 2

[alsa-devel] 7.1 surround channels missing on CS46xx
by Ondrej Zary 16 May '13

16 May '13

Hello, I got a Hercules Fortissimo 7.1 sound card (CS4624 + two CS4294) but snd-cs46xx driver supports only 6 channels (front, rear, center/lfe). The surround output (7&8) is not supported - as seen in dsp_spos_scb_lib.c: case DSP_PCM_S71_CHANNEL: /* TODO */ snd_BUG(); break; Tried to add a code for this, mostly copying the center/lfe code but I miss some data: - which slots (ACOSV_SLVx) should be used (maybe 10 and 12?) - what are the channel_disp and fifo_addr values for cs46xx_dsp_create_codec_out_scb() function - what values should S71_MIXER_SCB_ADDR and S71_CODEC_SCB_ADDR be (maybe 0x1b0 and 0x1c0?) -- Ondrej Zary

1 0

[alsa-devel] Debian alsa-tools patches
by Jordi Mallach 16 May '13

16 May '13

As previously discussed, I'm posting all Debian patches against alsa-tools that were pending upstreaming. Please review patch 8, as this one is new and the general state of configure scripts in alsa-tools is less than optimal. I've left out the patch to correct envy24control's configuration directory, as the additional check for the old dir is not implemented. If anyone wants to pick that up, please contact me or fetch the (trivial) patch from Debian sources. Thanks, Jordi

3 15

[alsa-devel] Debian alsa-utils patches
by Jordi Mallach 16 May '13

16 May '13

As previously discussed, I'm posting all Debian patches against alsa-utils that were pending upstreaming. Thanks, Jordi

2 5

[alsa-devel] Debian alsa-lib patches
by Jordi Mallach 16 May '13

16 May '13

As previously discussed, I'm posting all Debian patches against alsa-lib that were pending upstreaming. Thanks, Jordi

2 4

[alsa-devel] Debian alsa-lib patches
by Jordi Mallach 16 May '13

16 May '13

As previously discussed, I'm posting all Debian patches against alsa-oss that were pending upstreaming. Thanks, Jordi [PATCH 1/3] Fix path to libaoss.so. [PATCH 2/3] Show a warning if the alsa wrapper failed. [PATCH 3/3] Add AM_MAINTAINER_MODE([enable]) to configure.in.

2 6

[alsa-devel] [PATCH - pcm_plug] Add details about what pcm_plug does in doc
by olivier＠trillion01.com 16 May '13

16 May '13

From: Olivier Langlois <olivier(a)trillion01.com> Current doc only mention conversion to channels, rate and format. It alsa takes care of access mode mismatch. This is what I was looking for and I had to look the source code to figure out. Signed-off-by: Olivier Langlois <olivier(a)trillion01.com> diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index fa84eaa..8a639af 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -1155,7 +1155,8 @@ int snd_pcm_plug_open(snd_pcm_t **pcmp, \section pcm_plugins_plug Automatic conversion plugin -This plugin converts channels, rate and format on request. +This plugin converts channels, rate and format on request. It will also +implicitly adapt to the slave plugin access mode if different than our own. \code pcm.name { -- 1.8.2.3

1 0

Re: [alsa-devel] [PATCH] ALSA: hda - add PCI IDs for Intel BayTrail
by Takashi Iwai 16 May '13

16 May '13

At Thu, 16 May 2013 13:27:18 +0800, Chew Chiau Ee wrote: > > From: Chew, Chiau Ee <chiau.ee.chew(a)intel.com> > > Add HD Audio Device PCI ID for the Intel BayTrail platform. It's rather the successor of Oaktrail & co, right? Then put after Oaktrail entry, please. thanks, Takashi > Signed-off-by: Chew, Chiau Ee <chiau.ee.chew(a)intel.com> > Signed-off-by: Artem Bityutskiy <artem.bityutskiy(a)linux.intel.com> > --- > sound/pci/hda/hda_intel.c | 3 +++ > 1 files changed, 3 insertions(+), 0 deletions(-) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index bcd40ee..18640d9 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -3811,6 +3811,9 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > /* CPT */ > { PCI_DEVICE(0x8086, 0x1c20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > + /* BayTrail */ > + { PCI_DEVICE(0x8086, 0x0f04), > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > /* PBG */ > { PCI_DEVICE(0x8086, 0x1d20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > -- > 1.7.4.4 >

1 0

[alsa-devel] [PATCH 0/2 V2] Power-well API implementation for Haswell
by Wang Xingchao 16 May '13

16 May '13

Hi all, This is V2 and some fixes after first round review: - use reference count to track power-well usage - remove external module, compiled into snd-hda-intel instead - manage symbols and module loading properly - remove IS_HSW macro, use flag instead - remove audio callback for gfx driver to avoid dependency - split whole patch into two pieces for easy review - more typo fixes Wang Xingchao (2): drm/915: Add private api for power well usage ALSA: hda - Add power-welll support for haswell HDA drivers/gpu/drm/i915/intel_pm.c | 76 +++++++++++++++++++++++++++++--- include/drm/i915_powerwell.h | 36 +++++++++++++++ sound/pci/hda/Kconfig | 8 ++++ sound/pci/hda/Makefile | 3 ++ sound/pci/hda/hda_i915.c | 92 +++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 35 +++++++++++++++ sound/pci/hda/hda_intel.c | 33 ++++++++++++-- 7 files changed, 273 insertions(+), 10 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.7.9.5

3 7

[alsa-devel] [PATCH] ALSA: hda - Add headset mic support for another Dell machine
by David Henningsson 15 May '13

15 May '13

BugLink: https://bugs.launchpad.net/bugs/1180351 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- 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 6bf47f7..59d2e91 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3482,6 +3482,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05c9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05ca, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05de, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05ea, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05eb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.7.9.5

2 1

[alsa-devel] Error after wakeup from suspend?! (HDA Intel PCH)
by Tom 15 May '13

15 May '13

Hi, I woke up my ArchLinux system and pulseaudio is not working. Also I got this message from the commandline: https://gist.github.com/anonymous/5576712 Pulseaudio is running like it would work fine. I can see the volume bars shaking in Pulse Audio Volume Manager. Except, I can't hear any sound. It's like the audio plug is not connected. Hopefully this helps you a bit.

2 1

[alsa-devel] [Fwd: [PATCH] ALSA: snd-aoa: Add a layout entry for PowerBook6, 5]
by Benjamin Herrenschmidt 15 May '13

15 May '13

Acked-by: Benjamin Herrenschmidt <benh(a)kernel.crashing.org>

2 1

[alsa-devel] [PATCH 0/5] ASoC: fsl: imx-pcm driver cleanup
by Shawn Guo 15 May '13

15 May '13

The series removes the use of imx-pcm-audio/imx-fiq-pcm-audio platform device, and deletes imx-pcm driver by moving the functions needed by imx-pcm-fiq driver into imx-pcm-fiq.c. This can be done because imx-pcm-dma has moved to generic dmaengine pcm driver and does not use any helper from imx-pcm driver now. Shawn Guo (5): ASoC: fsl: remove use of imx-pcm-audio from fsl_ssi ASoC: fsl: remove use of imx-pcm-audio from imx-ssi ASoC: fsl: create function imx_pcm_fiq_exit() ASoC: fsl: remove use of imx-fiq-pcm-audio from imx-ssi ASoC: fsl: remove imx-pcm driver sound/soc/fsl/Kconfig | 5 -- sound/soc/fsl/Makefile | 11 +-- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_ssi.c | 13 ++-- sound/soc/fsl/imx-mc13783.c | 2 +- sound/soc/fsl/imx-pcm-dma.c | 2 + sound/soc/fsl/imx-pcm-fiq.c | 92 +++++++++++++++++++++++++ sound/soc/fsl/imx-pcm.c | 145 --------------------------------------- sound/soc/fsl/imx-pcm.h | 10 +-- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/imx-ssi.c | 44 +++--------- sound/soc/fsl/imx-ssi.h | 3 - sound/soc/fsl/mx27vis-aic32x4.c | 2 +- sound/soc/fsl/phycore-ac97.c | 2 +- sound/soc/fsl/wm1133-ev1.c | 2 +- 15 files changed, 122 insertions(+), 215 deletions(-) delete mode 100644 sound/soc/fsl/imx-pcm.c -- 1.7.9.5

2 13

[alsa-devel] ASoC updates for v3.10
by Mark Brown 15 May '13

15 May '13

The following changes since commit f722406faae2d073cc1d01063d1123c35425939e: Linux 3.10-rc1 (2013-05-11 17:14:08 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.10-rc1 for you to fetch changes up to 407a0e18ecfbd7f58f49c307a657b63e6f32544d: Merge remote-tracking branch 'asoc/fix/wm0010' into asoc-linus (2013-05-14 16:53:21 +0400) ---------------------------------------------------------------- ASoC: Fixes for v3.10 A few small driver-specific fixes, none of them especially crippling. ---------------------------------------------------------------- Axel Lin (1): ASoC: da7213: Fix setting dmic_samplephase and dmic_clk_rate Fabio Baltieri (1): ASoC: ux500: Swap even/odd AD slot definitions Mark Brown (3): Merge remote-tracking branch 'asoc/fix/da7213' into asoc-linus Merge remote-tracking branch 'asoc/fix/ux500' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm0010' into asoc-linus Wei Yongjun (1): ASoC: wm0010: fix error return code in wm0010_boot() sound/soc/codecs/ab8500-codec.h | 36 ++++++++++++++++++------------------ sound/soc/codecs/da7213.c | 8 ++++---- sound/soc/codecs/wm0010.c | 1 + 3 files changed, 23 insertions(+), 22 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: hda - Check the activity of the NID to be powered down
by Takashi Iwai 15 May '13

15 May '13

When an inactive path is powered down with spec->power_down_unused flag, we should check the activity of each widget in the path whether it's still referred from any active path. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index ac079f9..ae85bbd2 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -606,6 +606,10 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, return false; } +/* check whether the NID is referred by any active paths */ +#define is_active_nid_for_any(codec, nid) \ + is_active_nid(codec, nid, HDA_OUTPUT, 0) + /* get the default amp value for the target state */ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps, bool enable) @@ -759,7 +763,8 @@ static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path) for (i = 0; i < path->depth; i++) { hda_nid_t nid = path->path[i]; - if (!snd_hda_check_power_state(codec, nid, AC_PWRST_D3)) { + if (!snd_hda_check_power_state(codec, nid, AC_PWRST_D3) && + !is_active_nid_for_any(codec, nid)) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); @@ -4157,7 +4162,7 @@ static unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, return power_state; if (get_wcaps_type(get_wcaps(codec, nid)) >= AC_WID_POWER) return power_state; - if (is_active_nid(codec, nid, HDA_OUTPUT, 0)) + if (is_active_nid_for_any(codec, nid)) return power_state; return AC_PWRST_D3; } -- 1.8.2.1

1 0

[alsa-devel] Segmentation fault during snd_pcm_prepare()
by Shane Loretz 15 May '13

15 May '13

I'm interested in writing an application using alsa. I've chosen to explore the library by writing a program to echo sound data from a microphone back to my speakers. I'm only interested in mono sound. When running this program, the first call to snd_pcm_readi() fails, and a segmentation fault occurs inside of the snd_pcm_prepare() function according to a backtrace from gdb. I suspect I'm not allocating enough space for the pointer capdata, but I don't know how to determine how much space I need. Here is the code // Test out the alsa library // #include <alsa/asoundlib.h> #include <cstdio> #include <unistd.h> bool getMonoDevice(snd_pcm_t *pcm_handle, snd_pcm_stream_t stream, unsigned int rate, int periods, snd_pcm_uframes_t periodsize); int main() { snd_pcm_t *pcm_handle_read; snd_pcm_t *pcm_handle_write; snd_pcm_stream_t stream = SND_PCM_STREAM_CAPTURE; unsigned int rate = 44100; int periods = 2; snd_pcm_uframes_t periodsize = 8192; if (!getMonoDevice(pcm_handle_write, SND_PCM_STREAM_PLAYBACK,rate,periods,periodsize)) { fprintf(stderr, "Error couldn't open speakers\n"); return -1; } if (!getMonoDevice(pcm_handle_read, SND_PCM_STREAM_CAPTURE,rate,periods,periodsize)) { fprintf(stderr, "Error couldn't open microphone\n"); return -1; } //read frames from device //and echo them out unsigned char *capdata; int pcmreturn; int num_frames; int error; capdata = (unsigned char *)malloc(periodsize); num_frames = periodsize * 2; while (true) { while ((pcmreturn = snd_pcm_readi(pcm_handle_read, capdata, num_frames)) < 0) { usleep(10); error = snd_pcm_prepare(pcm_handle_read); if (error < 0) { fprintf(stderr, "Unrecoverable mic error %s\n",snd_strerror(error)); return error; } } while ((pcmreturn = snd_pcm_writei(pcm_handle_write, capdata, num_frames)) < 0) { usleep(10); error = snd_pcm_prepare(pcm_handle_write); if (error < 0) { fprintf(stderr, "Unrecoverable speaker error %s\n",snd_strerror(error)); return error; } } } //stop pcm device and drop pending frames snd_pcm_drop(pcm_handle_read); snd_pcm_drain(pcm_handle_read); snd_pcm_drop(pcm_handle_write); snd_pcm_drain(pcm_handle_write); } bool getMonoDevice(snd_pcm_t *pcm_handle, snd_pcm_stream_t stream, unsigned int rate, int periods, snd_pcm_uframes_t periodsize) { snd_pcm_hw_params_t *hwparams; char *pcm_name; pcm_name = strdup("plughw:0,0"); snd_pcm_hw_params_alloca(&hwparams); if (snd_pcm_open(&pcm_handle, pcm_name, stream, 0) < 0) { fprintf(stderr, "Error opening PCM device %s\n", pcm_name); return false; } if (snd_pcm_hw_params_any(pcm_handle, hwparams) < 0 ) { fprintf(stderr, "Can not configure this PCM device.\n"); return false; } unsigned int exact_rate; int dir; if (snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) { fprintf(stderr, "Error setting access.\n"); return false; } if (snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE) < 0) { fprintf(stderr, "Error setting format.\n"); return false; } exact_rate = rate; if (snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &exact_rate, 0) < 0) { fprintf(stderr, "Error setting rate.\n"); return false; } if (rate != exact_rate) { fprintf(stderr, "The rate %d is not supported by your hardware.\n Using %d Hz instead.\n", rate, exact_rate); } //Configure mono sound if (snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 1) < 0) { fprintf(stderr, "Error setting channels.\n"); return false; } if (snd_pcm_hw_params_set_periods(pcm_handle, hwparams, periods, 0) < 0) { fprintf(stderr, "Error setting periods.\n"); return false; } if (snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, (periodsize * periods) >> 2) < 0 ) { fprintf(stderr, "Error setting buffersize.\n"); return false; } if (snd_pcm_hw_params(pcm_handle, hwparams) < 0) { fprintf(stderr, "Error setting HW params.\n"); return false; } //device is finally configured return true; }

2 1

[alsa-devel] [PATCH] ASoC: remove saarb and tavorevb3 machine drivers
by Paul Bolle 15 May '13

15 May '13

Support for PXA95x was removed in v3.8. This means that the Kconfig symbols MACH_SAARB and MACH_TAVOREVB3 are no longer available. This leaves the SoC Audio support for Marvell Saarb and Marvell Tavor EVB3 unbuildable. Remove these drivers too. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- 0) Basically untested. 1) To be specific: it was commit 49ea7fc094 ("ARM: pxa: remove pxa95x support") that removed MACH_SAARB and MACH_TAVOREVB3. sound/soc/pxa/Kconfig | 20 ----- sound/soc/pxa/Makefile | 4 - sound/soc/pxa/saarb.c | 190 ---------------------------------------------- sound/soc/pxa/tavorevb3.c | 189 --------------------------------------------- 4 files changed, 403 deletions(-) delete mode 100644 sound/soc/pxa/saarb.c delete mode 100644 sound/soc/pxa/tavorevb3.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 4d2e46f..b358094 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -130,26 +130,6 @@ config SND_PXA2XX_SOC_PALM27X Say Y if you want to add support for SoC audio on Palm T|X, T5, E2 or LifeDrive handheld computer. -config SND_SOC_SAARB - tristate "SoC Audio support for Marvell Saarb" - depends on SND_PXA2XX_SOC && MACH_SAARB - select MFD_88PM860X - select SND_PXA_SOC_SSP - select SND_SOC_88PM860X - help - Say Y if you want to add support for SoC audio on the - Marvell Saarb reference platform. - -config SND_SOC_TAVOREVB3 - tristate "SoC Audio support for Marvell Tavor EVB3" - depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 - select MFD_88PM860X - select SND_PXA_SOC_SSP - select SND_SOC_88PM860X - help - Say Y if you want to add support for SoC audio on the - Marvell Saarb reference platform. - config SND_PXA910_SOC tristate "SoC Audio for Marvell PXA910 chip" depends on ARCH_MMP && SND diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index d8a265d..2cff67b 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -23,8 +23,6 @@ snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o -snd-soc-saarb-objs := saarb.o -snd-soc-tavorevb3-objs := tavorevb3.o snd-soc-zylonite-objs := zylonite.o snd-soc-hx4700-objs := hx4700.o snd-soc-magician-objs := magician.o @@ -48,8 +46,6 @@ obj-$(CONFIG_SND_PXA2XX_SOC_HX4700) += snd-soc-hx4700.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o -obj-$(CONFIG_SND_SOC_SAARB) += snd-soc-saarb.o -obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c deleted file mode 100644 index c34146b..0000000 --- a/sound/soc/pxa/saarb.c +++ /dev/null @@ -1,190 +0,0 @@ -/* - * saarb.c -- SoC audio for saarb - * - * Copyright (C) 2010 Marvell International Ltd. - * Haojian Zhuang <haojian.zhuang(a)marvell.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> -#include <linux/clk.h> -#include <linux/i2c.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/jack.h> - -#include <asm/mach-types.h> - -#include "../codecs/88pm860x-codec.h" -#include "pxa-ssp.h" - -static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd); - -static struct platform_device *saarb_snd_device; - -static struct snd_soc_jack hs_jack, mic_jack; - -static struct snd_soc_jack_pin hs_jack_pins[] = { - { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, -}; - -static struct snd_soc_jack_pin mic_jack_pins[] = { - { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, -}; - -/* saarb machine dapm widgets */ -static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Stereophone", NULL), - SND_SOC_DAPM_LINE("Lineout Out 1", NULL), - SND_SOC_DAPM_LINE("Lineout Out 2", NULL), - SND_SOC_DAPM_SPK("Ext Speaker", NULL), - SND_SOC_DAPM_MIC("Ext Mic 1", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Ext Mic 3", NULL), -}; - -/* saarb machine audio map */ -static const struct snd_soc_dapm_route saarb_audio_map[] = { - {"Headset Stereophone", NULL, "HS1"}, - {"Headset Stereophone", NULL, "HS2"}, - - {"Ext Speaker", NULL, "LSP"}, - {"Ext Speaker", NULL, "LSN"}, - - {"Lineout Out 1", NULL, "LINEOUT1"}, - {"Lineout Out 2", NULL, "LINEOUT2"}, - - {"MIC1P", NULL, "Mic1 Bias"}, - {"MIC1N", NULL, "Mic1 Bias"}, - {"Mic1 Bias", NULL, "Ext Mic 1"}, - - {"MIC2P", NULL, "Mic1 Bias"}, - {"MIC2N", NULL, "Mic1 Bias"}, - {"Mic1 Bias", NULL, "Headset Mic 2"}, - - {"MIC3P", NULL, "Mic3 Bias"}, - {"MIC3N", NULL, "Mic3 Bias"}, - {"Mic3 Bias", NULL, "Ext Mic 3"}, -}; - -static int saarb_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int width = snd_pcm_format_physical_width(params_format(params)); - int ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, - PM860X_CLK_DIR_OUT); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); - - return ret; -} - -static struct snd_soc_ops saarb_i2s_ops = { - .hw_params = saarb_i2s_hw_params, -}; - -static struct snd_soc_dai_link saarb_dai[] = { - { - .name = "88PM860x I2S", - .stream_name = "I2S Audio", - .cpu_dai_name = "pxa-ssp-dai.1", - .codec_dai_name = "88pm860x-i2s", - .platform_name = "pxa-pcm-audio", - .codec_name = "88pm860x-codec", - .init = saarb_pm860x_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &saarb_i2s_ops, - }, -}; - -static struct snd_soc_card snd_soc_card_saarb = { - .name = "Saarb", - .owner = THIS_MODULE, - .dai_link = saarb_dai, - .num_links = ARRAY_SIZE(saarb_dai), - - .dapm_widgets = saarb_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(saarb_dapm_widgets), - .dapm_routes = saarb_audio_map, - .num_dapm_routes = ARRAY_SIZE(saarb_audio_map), -}; - -static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* connected pins */ - snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); - snd_soc_dapm_enable_pin(dapm, "Ext Mic 1"); - snd_soc_dapm_enable_pin(dapm, "Ext Mic 3"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); - snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); - - /* Headset jack detection */ - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE - | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, - &hs_jack); - snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); - snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE, - &mic_jack); - snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), - mic_jack_pins); - - /* headphone, microphone detection & headset short detection */ - pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE, - SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2); - pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE); - return 0; -} - -static int __init saarb_init(void) -{ - int ret; - - if (!machine_is_saarb()) - return -ENODEV; - saarb_snd_device = platform_device_alloc("soc-audio", -1); - if (!saarb_snd_device) - return -ENOMEM; - - platform_set_drvdata(saarb_snd_device, &snd_soc_card_saarb); - - ret = platform_device_add(saarb_snd_device); - if (ret) - platform_device_put(saarb_snd_device); - - return ret; -} - -static void __exit saarb_exit(void) -{ - platform_device_unregister(saarb_snd_device); -} - -module_init(saarb_init); -module_exit(saarb_exit); - -MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); -MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c deleted file mode 100644 index 8b5ab8f..0000000 --- a/sound/soc/pxa/tavorevb3.c +++ /dev/null @@ -1,189 +0,0 @@ -/* - * tavorevb3.c -- SoC audio for Tavor EVB3 - * - * Copyright (C) 2010 Marvell International Ltd. - * Haojian Zhuang <haojian.zhuang(a)marvell.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> -#include <linux/clk.h> -#include <linux/i2c.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/jack.h> - -#include <asm/mach-types.h> - -#include "../codecs/88pm860x-codec.h" -#include "pxa-ssp.h" - -static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd); - -static struct platform_device *evb3_snd_device; - -static struct snd_soc_jack hs_jack, mic_jack; - -static struct snd_soc_jack_pin hs_jack_pins[] = { - { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, -}; - -static struct snd_soc_jack_pin mic_jack_pins[] = { - { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, -}; - -/* tavorevb3 machine dapm widgets */ -static const struct snd_soc_dapm_widget evb3_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headset Stereophone", NULL), - SND_SOC_DAPM_LINE("Lineout Out 1", NULL), - SND_SOC_DAPM_LINE("Lineout Out 2", NULL), - SND_SOC_DAPM_SPK("Ext Speaker", NULL), - SND_SOC_DAPM_MIC("Ext Mic 1", NULL), - SND_SOC_DAPM_MIC("Headset Mic 2", NULL), - SND_SOC_DAPM_MIC("Ext Mic 3", NULL), -}; - -/* tavorevb3 machine audio map */ -static const struct snd_soc_dapm_route evb3_audio_map[] = { - {"Headset Stereophone", NULL, "HS1"}, - {"Headset Stereophone", NULL, "HS2"}, - - {"Ext Speaker", NULL, "LSP"}, - {"Ext Speaker", NULL, "LSN"}, - - {"Lineout Out 1", NULL, "LINEOUT1"}, - {"Lineout Out 2", NULL, "LINEOUT2"}, - - {"MIC1P", NULL, "Mic1 Bias"}, - {"MIC1N", NULL, "Mic1 Bias"}, - {"Mic1 Bias", NULL, "Ext Mic 1"}, - - {"MIC2P", NULL, "Mic1 Bias"}, - {"MIC2N", NULL, "Mic1 Bias"}, - {"Mic1 Bias", NULL, "Headset Mic 2"}, - - {"MIC3P", NULL, "Mic3 Bias"}, - {"MIC3N", NULL, "Mic3 Bias"}, - {"Mic3 Bias", NULL, "Ext Mic 3"}, -}; - -static int evb3_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int width = snd_pcm_format_physical_width(params_format(params)); - int ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, - PM860X_CLK_DIR_OUT); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); - return ret; -} - -static struct snd_soc_ops evb3_i2s_ops = { - .hw_params = evb3_i2s_hw_params, -}; - -static struct snd_soc_dai_link evb3_dai[] = { - { - .name = "88PM860x I2S", - .stream_name = "I2S Audio", - .cpu_dai_name = "pxa-ssp-dai.1", - .codec_dai_name = "88pm860x-i2s", - .platform_name = "pxa-pcm-audio", - .codec_name = "88pm860x-codec", - .init = evb3_pm860x_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &evb3_i2s_ops, - }, -}; - -static struct snd_soc_card snd_soc_card_evb3 = { - .name = "Tavor EVB3", - .owner = THIS_MODULE, - .dai_link = evb3_dai, - .num_links = ARRAY_SIZE(evb3_dai), - - .dapm_widgets = evb3_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(evb3_dapm_widgets), - .dapm_routes = evb3_audio_map, - .num_dapm_routes = ARRAY_SIZE(evb3_audio_map), -}; - -static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* connected pins */ - snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); - snd_soc_dapm_enable_pin(dapm, "Ext Mic 1"); - snd_soc_dapm_enable_pin(dapm, "Ext Mic 3"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); - snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); - - /* Headset jack detection */ - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE - | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, - &hs_jack); - snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); - snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE, - &mic_jack); - snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), - mic_jack_pins); - - /* headphone, microphone detection & headset short detection */ - pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE, - SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2); - pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE); - return 0; -} - -static int __init tavorevb3_init(void) -{ - int ret; - - if (!machine_is_tavorevb3()) - return -ENODEV; - evb3_snd_device = platform_device_alloc("soc-audio", -1); - if (!evb3_snd_device) - return -ENOMEM; - - platform_set_drvdata(evb3_snd_device, &snd_soc_card_evb3); - - ret = platform_device_add(evb3_snd_device); - if (ret) - platform_device_put(evb3_snd_device); - - return ret; -} - -static void __exit tavorevb3_exit(void) -{ - platform_device_unregister(evb3_snd_device); -} - -module_init(tavorevb3_init); -module_exit(tavorevb3_exit); - -MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); -MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3"); -MODULE_LICENSE("GPL"); -- 1.7.11.7

4 3

[alsa-devel] [PATCH 01/11] ASoC: ep93xx-i2s: Staticize non exported struct
by Lars-Peter Clausen 15 May '13

15 May '13

The ep93xx_i2s_dma_data struct is not used outside of ep93xx-i2s.c, so make it static. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Ryan Mallon <ryan(a)bluewatersys.com> --- sound/soc/cirrus/ep93xx-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index ef731e6..17ad70b 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -63,7 +63,7 @@ struct ep93xx_i2s_info { void __iomem *regs; }; -struct ep93xx_dma_data ep93xx_i2s_dma_data[] = { +static struct ep93xx_dma_data ep93xx_i2s_dma_data[] = { [SNDRV_PCM_STREAM_PLAYBACK] = { .name = "i2s-pcm-out", .port = EP93XX_DMA_I2S1, -- 1.8.0

3 21

[alsa-devel] Debian patches for ALSA
by Jordi Mallach 15 May '13

15 May '13

Hi! I've been wanting to put some order in the collection of patches we've been building up in Debian due to me neglecting forwarding anything for far too long. Let's try to correct this. I have a number of patches, most of which should be trivial and no-brainers, all updated to 1.0.27. I'm going to outline all the individual patches below. The patch tracker has the patches just as I would send them upstream, but if you prefer git formatted patches, I can go ahead and do that too, if it makes your life easier. Thanks! ALSA-LIB: http://patch-tracker.debian.org/package/alsa-lib/1.0.27-3 =================================================================== - http://patch-tracker.debian.org/patch/series/view/alsa-lib/1.0.27-3/add-mai… add-maintainer-mode.patch is something I'd like to see in all ALSA configure scripts: have maintainer mode enabled, but add the possibility to disable it via --disable-maintainer-mode. See http://blogs.gnome.org/desrt/2011/09/08/am_maintainer_mode-is-not-cool/ for reference. - http://patch-tracker.debian.org/patch/series/view/alsa-lib/1.0.27-3/multiar… The patch header is self explanatory: it fixes the plugin lookup path for Debian/Ubuntu's multiarch paths. - http://patch-tracker.debian.org/patch/series/view/alsa-lib/1.0.27-3/spellin… Little to explain here. ;) ALSA-UTILS: http://patch-tracker.debian.org/package/alsa-utils/1.0.27-2 ======================================================================= - http://patch-tracker.debian.org/patch/series/view/alsa-utils/1.0.27-2/spell… Again, trivial. - http://patch-tracker.debian.org/patch/series/view/alsa-utils/1.0.27-2/fix_m… Even more trivial. - http://patch-tracker.debian.org/patch/series/view/alsa-utils/1.0.27-2/utils… Fixes a build failure when using -Werror=format-security, which is default in Debian and Uubntu's toolchain. - http://patch-tracker.debian.org/patch/series/view/alsa-utils/1.0.27-2/alsa-… This one I want to discuss. We think the After condition is wrong, as it's depending on an optional service (alsa-state might not happen if the config file is not present). Instead, it should be checking for sysinit, to ensure local-fs. - http://patch-tracker.debian.org/patch/series/view/alsa-utils/1.0.27-2/manpa… Extensive formatting fixes for all alsa-utils manpages, plus an English improvement (allows to → allows one to). ALSA-PLUGINS: http://patch-tracker.debian.org/package/alsa-plugins/1.0.27-2 =========================================================================== - http://patch-tracker.debian.org/patch/series/view/alsa-plugins/1.0.27-2/arc… Fix linking to avoid undefined symbols in arcam-av. - http://patch-tracker.debian.org/patch/series/view/alsa-plugins/1.0.27-2/use… The libav changes in 1.0.27 were insufficient: a52 uses av_free(), which is part of libavutil. The patch needs to be completed to look for the correct header files, though, but it's enough to get the package compiling in our setup. ALSA-TOOLS: http://patch-tracker.debian.org/package/alsa-tools/1.0.27-2 - http://patch-tracker.debian.org/patch/series/view/alsa-tools/1.0.27-2/init_… Fix trivial bashisms. - http://patch-tracker.debian.org/patch/series/view/alsa-tools/1.0.27-2/manpa… Spelling fixes. - http://patch-tracker.debian.org/patch/series/view/alsa-tools/1.0.27-2/ld10k… More -Werror=format-security issues. - http://patch-tracker.debian.org/patch/series/view/alsa-tools/1.0.27-2/hda-v… Extend check for sys/io.h for other platforms in Debian that don't support it. Alternatively, the check could be reversed to chef for the architectures currently know to have the header, as stated in the comment. This fixes the build in a number of Debian architectures. - http://patch-tracker.debian.org/patch/series/view/alsa-tools/1.0.27-2/envy2… This one is to be discussed. envy24control currently creates a visible directory in $HOME, which is quite unacceptable. This patch just moves the directory to a dotdir, but an upgrade path might need to be coded. I would just move the thing away. - http://patch-tracker.debian.org/patch/series/view/alsa-tools/1.0.27-2/as10k… For convenience to our build systems, add an alias for clean. - http://patch-tracker.debian.org/patch/series/view/alsa-tools/1.0.27-2/usx2y… While not ready for submission, this one probably reveals a hardcoded path that should be made configurable, if not all distros store firmware in this dir. Not sure about Fedora, at least. /etc/hotplug looks plain old, anyway. ALSA-OSS: http://patch-tracker.debian.org/package/alsa-oss/1.0.25-1 =================================================================== - http://patch-tracker.debian.org/patch/series/view/alsa-oss/1.0.25-1/aoss_wr… Make it obvious aoss is not doing anything useful if ALSA is not present. - http://patch-tracker.debian.org/patch/series/view/alsa-oss/1.0.25-1/aoss_li… Fix path to libaoss on the multiarch case. Thanks, Jordi -- Jordi Mallach Pérez -- Debian developer http://www.debian.org/ jordi(a)sindominio.net jordi(a)debian.org http://www.sindominio.net/ GnuPG public key information available at http://oskuro.net/ -- Jordi Mallach Pérez -- Debian developer http://www.debian.org/ jordi(a)sindominio.net jordi(a)debian.org http://www.sindominio.net/ GnuPG public key information available at http://oskuro.net/

3 3

[alsa-devel] [PATCH] conf/ucm: DAISY-I2S: Set default filter mode for HiFi.
by Dylan Reid 14 May '13

14 May '13

Default to music mode filter for the HiFi use case on the Samsung ARM Chromebook. This mode is better at 44.1k and 48k audio than the "Voice" setting. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- src/conf/ucm/DAISY-I2S/HiFi.conf | 1 + 1 file changed, 1 insertion(+) diff --git a/src/conf/ucm/DAISY-I2S/HiFi.conf b/src/conf/ucm/DAISY-I2S/HiFi.conf index 66b0afa..d8c06ad 100644 --- a/src/conf/ucm/DAISY-I2S/HiFi.conf +++ b/src/conf/ucm/DAISY-I2S/HiFi.conf @@ -11,6 +11,7 @@ SectionVerb { cset "name='Left ADC Mixer MIC2 Switch' on" cset "name='Right ADC Mixer MIC2 Switch' on" cset "name='EQ1 Mode' Default" + cset "name='DAI1 Filter Mode' Music" ] DisableSequence [ ] -- 1.8.1.3.605.g02339dd

2 1

Re: [alsa-devel] [PATCH 1/5] DAISY-I2S: add Samsung ARM Chromebook UCM config from ChromeOS
by Dylan Reid 14 May '13

14 May '13

On Fri, May 10, 2013 at 5:51 AM, Marcin Juszkiewicz <marcin(a)juszkiewicz.com.pl> wrote: > Signed-off-by: Marcin Juszkiewicz <marcin(a)juszkiewicz.com.pl> > --- > DAISY-I2S/DAISY-I2S.conf | 6 +++++ > DAISY-I2S/HiFi.conf | 61 ++++++++++++++++++++++++++++++++++++++++++++++++ > 2 files changed, 67 insertions(+) > create mode 100644 DAISY-I2S/DAISY-I2S.conf > create mode 100644 DAISY-I2S/HiFi.conf > > diff --git a/DAISY-I2S/DAISY-I2S.conf b/DAISY-I2S/DAISY-I2S.conf > new file mode 100644 > index 0000000..0f4b5fc > --- /dev/null > +++ b/DAISY-I2S/DAISY-I2S.conf > @@ -0,0 +1,6 @@ > +Comment "Daisy internal card" > + > +SectionUseCase."HiFi" { > + File "HiFi.conf" > + Comment "Default" > +} > diff --git a/DAISY-I2S/HiFi.conf b/DAISY-I2S/HiFi.conf > new file mode 100644 > index 0000000..66b0afa > --- /dev/null > +++ b/DAISY-I2S/HiFi.conf > @@ -0,0 +1,61 @@ > +SectionVerb { > + EnableSequence [ > + cdev "hw:DAISYI2S" > + cset "name='Left Speaker Mixer Left DAC1 Switch' on" > + cset "name='Right Speaker Mixer Right DAC1 Switch' on" > + cset "name='Left Headphone Mixer Left DAC1 Switch' on" > + cset "name='Right Headphone Mixer Right DAC1 Switch' on" > + cset "name='DMIC1 Left Capture Switch' on" > + cset "name='DMIC1 Right Capture Switch' on" > + cset "name='MIC2 External Mic Switch' on" > + cset "name='Left ADC Mixer MIC2 Switch' on" > + cset "name='Right ADC Mixer MIC2 Switch' on" > + cset "name='EQ1 Mode' Default" The current version has one additional line here: cset "name='DAI1 Filter Mode' Music" Thanks for finding a potential home for this. > + ] > + DisableSequence [ > + ] > +} > + > +SectionDevice."Headphone".0 { > + Value { > + JackName "DAISY-I2S Headphone Jack" > + } > + > + EnableSequence [ > + cdev "hw:DAISYI2S" > + cset "name='EQ1 Switch' off" > + cset "name='Left Headphone Mixer Left DAC1 Switch' on" > + cset "name='Right Headphone Mixer Right DAC1 Switch' on" > + ] > + DisableSequence [ > + cdev "hw:DAISYI2S" > + cset "name='EQ1 Mode' Default" > + cset "name='EQ1 Switch' on" > + cset "name='Left Speaker Mixer Left DAC1 Switch' on" > + cset "name='Right Speaker Mixer Right DAC1 Switch' on" > + ] > +} > + > +SectionDevice."Mic".0 { > + Value { > + JackName "DAISY-I2S Mic Jack" > + } > + > + EnableSequence [ > + cdev "hw:DAISYI2S" > + > + cset "name='DMIC1 Left Capture Switch' off" > + cset "name='DMIC1 Right Capture Switch' off" > + cset "name='Left ADC Mixer MIC2 Switch' on" > + cset "name='Right ADC Mixer MIC2 Switch' on" > + ] > + > + DisableSequence [ > + cdev "hw:DAISYI2S" > + > + cset "name='Left ADC Mixer MIC2 Switch' off" > + cset "name='Right ADC Mixer MIC2 Switch' off" > + cset "name='DMIC1 Left Capture Switch' on" > + cset "name='DMIC1 Right Capture Switch' on" > + ] > +} > -- > 1.8.1.2 > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

2 1

Re: [alsa-devel] UCM profiles taken from Ubuntu
by Liam Girdwood 14 May '13

14 May '13

On Fri, 2013-05-10 at 15:56 +0200, Jordi Mallach wrote: > El dv 10 de 05 de 2013 a les 15:17 +0200, en/na Takashi Iwai va > escriure: > > At Fri, 10 May 2013 14:51:05 +0200, > > Marcin Juszkiewicz wrote: > > > > > > When I bought Samsung ARM Chromebook few months ago I had no idea > > > about UCM profiles and burnt speakers (left is dead, right is > > > resting). Then I learnt, took ones from ChromeOS and added them into > > > Ubuntu so other users will be a bit more safe. > > > > > > Later I had a discussion with some Debian, Ubuntu, Fedora developers > > > about how to solve such problems and how to keep UCM profiles shared > > > between distributions. > > > > > > Liam Girdwood pointed me to not used git tree at ALSA Project server > > > and finally I reminded myself about it. > > > > > > The idea is to collect UCM profiles, keep them in one place and create > > > alsa-ucm package in each distribution. > > Thanks Marcin! This part of the Debian←→Ubuntu delta was something I > wanted to bring up with upstream after the wheezy release. > I'll echo that thought, many thanks Marcin ! > > Would it make sense to put into alsa-lib.git? Since UCM itself was > > already integrated into alsa-lib and there are already tons of > > card-based config files in alsa-lib/src/conf/*, these also fit to a > > similar directory there, I think. > > > > They could be in a separate repo, but UCM config is coupled with UCM > > code, so we may keep in the single repo. > > It makes sense to me, fwiw. > Agreed, the initial intention of the separate repo was to do frequent releases that would be likely out of sync alsa-lib. But I think combining is the best way forward now. Takashi, please feel free to remove the other repo (I think it's mostly empty anyway). Liam

3 2

[alsa-devel] [PATCH v2 8/8] sound/oxygen_io: take msecs_to_jiffies_timeout into use
by Imre Deak 14 May '13

14 May '13

Use msecs_to_jiffies_timeout instead of open-coding the same. Signed-off-by: Imre Deak <imre.deak(a)intel.com> --- sound/pci/oxygen/oxygen_io.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 521eae4..51bfcd0 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -108,7 +108,7 @@ static int oxygen_ac97_wait(struct oxygen *chip, unsigned int mask) wait_event_timeout(chip->ac97_waitqueue, ({ status |= oxygen_read8(chip, OXYGEN_AC97_INTERRUPT_STATUS); status & mask; }), - msecs_to_jiffies(1) + 1); + msecs_to_jiffies_timeout(1)); /* * Check even after a timeout because this function should not require * the AC'97 interrupt to be enabled. -- 1.7.10.4

2 1

[alsa-devel] [patch] ASoC: wm_adsp: memory leak in wm_adsp_create_control()
by Dan Carpenter 14 May '13

14 May '13

There are two return paths which don't kfree(name). Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 1378306..d715c8e 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -836,7 +836,8 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, region_name = "ZM"; break; default: - return -EINVAL; + ret = -EINVAL; + goto err_name; } snprintf(name, PAGE_SIZE, "DSP%d %s %x", @@ -847,7 +848,7 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, if (!strcmp(ctl->name, name)) { if (!ctl->enabled) ctl->enabled = 1; - return 0; + goto found; } } @@ -887,6 +888,7 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, INIT_WORK(&ctl_work->work, wm_adsp_ctl_work); schedule_work(&ctl_work->work); +found: kfree(name); return 0;

3 2

[alsa-devel] [PATCH RFT] ASoC: da7213: Fix setting dmic_samplephase and dmic_clk_rate
by Axel Lin 14 May '13

14 May '13

When set dmic_samplephase and dmic_clk_rate bits for dmic_cfg, current code checks pdata->dmic_data_sel which is wrong. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- This looks like a copy-paste bug. sound/soc/codecs/da7213.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 41230ad..4a6f1da 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1488,17 +1488,17 @@ static int da7213_probe(struct snd_soc_codec *codec) DA7213_DMIC_DATA_SEL_SHIFT); break; } - switch (pdata->dmic_data_sel) { + switch (pdata->dmic_samplephase) { case DA7213_DMIC_SAMPLE_ON_CLKEDGE: case DA7213_DMIC_SAMPLE_BETWEEN_CLKEDGE: - dmic_cfg |= (pdata->dmic_data_sel << + dmic_cfg |= (pdata->dmic_samplephase << DA7213_DMIC_SAMPLEPHASE_SHIFT); break; } - switch (pdata->dmic_data_sel) { + switch (pdata->dmic_clk_rate) { case DA7213_DMIC_CLK_3_0MHZ: case DA7213_DMIC_CLK_1_5MHZ: - dmic_cfg |= (pdata->dmic_data_sel << + dmic_cfg |= (pdata->dmic_clk_rate << DA7213_DMIC_CLK_RATE_SHIFT); break; } -- 1.8.1.2

3 2

[alsa-devel] [PATCH 1/3] ASoC: Move snd_soc_set_runtime_hwparams() to soc-pcm.c
by Lars-Peter Clausen 14 May '13

14 May '13

snd_soc_set_runtime_hwparams() is the only PCM related function that lives in soc-core.c. All other PCM related functions live in soc-pcm.c, so move snd_soc_set_runtime_hwparams() over as well for a bit more consistency. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 23 ----------------------- sound/soc/soc-pcm.c | 23 +++++++++++++++++++++++ 2 files changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 308895a..9d95ef5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2220,29 +2220,6 @@ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, EXPORT_SYMBOL_GPL(snd_soc_test_bits); /** - * snd_soc_set_runtime_hwparams - set the runtime hardware parameters - * @substream: the pcm substream - * @hw: the hardware parameters - * - * Sets the substream runtime hardware parameters. - */ -int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, - const struct snd_pcm_hardware *hw) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - runtime->hw.info = hw->info; - runtime->hw.formats = hw->formats; - runtime->hw.period_bytes_min = hw->period_bytes_min; - runtime->hw.period_bytes_max = hw->period_bytes_max; - runtime->hw.periods_min = hw->periods_min; - runtime->hw.periods_max = hw->periods_max; - runtime->hw.buffer_bytes_max = hw->buffer_bytes_max; - runtime->hw.fifo_size = hw->fifo_size; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); - -/** * snd_soc_cnew - create new control * @_template: control template * @data: control private data diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 73bb8ee..058b403 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -33,6 +33,29 @@ #define DPCM_MAX_BE_USERS 8 +/** + * snd_soc_set_runtime_hwparams - set the runtime hardware parameters + * @substream: the pcm substream + * @hw: the hardware parameters + * + * Sets the substream runtime hardware parameters. + */ +int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, + const struct snd_pcm_hardware *hw) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + runtime->hw.info = hw->info; + runtime->hw.formats = hw->formats; + runtime->hw.period_bytes_min = hw->period_bytes_min; + runtime->hw.period_bytes_max = hw->period_bytes_max; + runtime->hw.periods_min = hw->periods_min; + runtime->hw.periods_max = hw->periods_max; + runtime->hw.buffer_bytes_max = hw->buffer_bytes_max; + runtime->hw.fifo_size = hw->fifo_size; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); + /* DPCM stream event, send event to FE and all active BEs. */ static int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event) -- 1.8.0

3 4

[alsa-devel] [PATCH] ASoC: tegra: always use clk_get() in utility code
by Stephen Warren 14 May '13

14 May '13

From: Stephen Warren <swarren(a)nvidia.com> Now that all of the Tegra device trees have been updated to represent the required audio clocks, remove the compatibility code from the Tegra ASoC utility code, and always use clk_get() rather than clk_get_sys(). Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra_asoc_utils.c | 23 +++++------------------ 1 file changed, 5 insertions(+), 18 deletions(-) diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index 24fb001b..d173880 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -173,7 +173,6 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, struct device *dev) { int ret; - bool new_clocks = false; data->dev = dev; @@ -181,40 +180,28 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; else if (of_machine_is_compatible("nvidia,tegra30")) data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30; - else if (of_machine_is_compatible("nvidia,tegra114")) { + else if (of_machine_is_compatible("nvidia,tegra114")) data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA114; - new_clocks = true; - } else { + else { dev_err(data->dev, "SoC unknown to Tegra ASoC utils\n"); return -EINVAL; } - if (new_clocks) - data->clk_pll_a = clk_get(dev, "pll_a"); - else - data->clk_pll_a = clk_get_sys(NULL, "pll_a"); + data->clk_pll_a = clk_get(dev, "pll_a"); if (IS_ERR(data->clk_pll_a)) { dev_err(data->dev, "Can't retrieve clk pll_a\n"); ret = PTR_ERR(data->clk_pll_a); goto err; } - if (new_clocks) - data->clk_pll_a_out0 = clk_get(dev, "pll_a_out0"); - else - data->clk_pll_a_out0 = clk_get_sys(NULL, "pll_a_out0"); + data->clk_pll_a_out0 = clk_get(dev, "pll_a_out0"); if (IS_ERR(data->clk_pll_a_out0)) { dev_err(data->dev, "Can't retrieve clk pll_a_out0\n"); ret = PTR_ERR(data->clk_pll_a_out0); goto err_put_pll_a; } - if (new_clocks) - data->clk_cdev1 = clk_get(dev, "mclk"); - else if (data->soc == TEGRA_ASOC_UTILS_SOC_TEGRA20) - data->clk_cdev1 = clk_get_sys(NULL, "cdev1"); - else - data->clk_cdev1 = clk_get_sys("extern1", NULL); + data->clk_cdev1 = clk_get(dev, "mclk"); if (IS_ERR(data->clk_cdev1)) { dev_err(data->dev, "Can't retrieve clk cdev1\n"); ret = PTR_ERR(data->clk_cdev1); -- 1.7.10.4

2 1

[alsa-devel] [PATCH 1/8] ASoC: generic-dmaengine-pcm: Add support for half-duplex
by Lars-Peter Clausen 13 May '13

13 May '13

Some platforms which are half-duplex share the same DMA channel between the playback and capture stream. Add support for this to the generic dmaengine PCM driver. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/dmaengine_pcm.h | 5 ++++ sound/soc/soc-generic-dmaengine-pcm.c | 43 ++++++++++++++++++++++++----------- 2 files changed, 35 insertions(+), 13 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index b1d1150..f11c35c 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -91,6 +91,11 @@ void snd_dmaengine_pcm_set_config_from_dai_data( * bytes that are still left to transfer. */ #define SND_DMAENGINE_PCM_FLAG_NO_RESIDUE BIT(2) +/* + * The PCM is half duplex and the DMA channel is shared between capture and + * playback. + */ +#define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3) /** * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index ae0c37e..5fd5ed4 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -29,7 +29,7 @@ struct dmaengine_pcm { struct dma_chan *chan[SNDRV_PCM_STREAM_CAPTURE + 1]; const struct snd_dmaengine_pcm_config *config; struct snd_soc_platform platform; - bool compat; + unsigned int flags; }; static struct dmaengine_pcm *soc_platform_to_pcm(struct snd_soc_platform *p) @@ -128,6 +128,9 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) + return pcm->chan[0]; + if (pcm->config->compat_request_channel) return pcm->config->compat_request_channel(rtd, substream); @@ -148,7 +151,7 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!substream) continue; - if (!pcm->chan[i] && pcm->compat) { + if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd, substream); } @@ -215,6 +218,25 @@ static const char * const dmaengine_pcm_dma_channel_names[] = { [SNDRV_PCM_STREAM_CAPTURE] = "rx", }; +static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, + struct device_node *of_node) +{ + unsigned int i; + + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !of_node) + return; + + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { + pcm->chan[0] = of_dma_request_slave_channel(of_node, "tx_rx"); + pcm->chan[1] = pcm->chan[0]; + } else { + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { + pcm->chan[i] = of_dma_request_slave_channel(of_node, + dmaengine_pcm_dma_channel_names[i]); + } + } +} + /** * snd_dmaengine_pcm_register - Register a dmaengine based PCM device * @dev: The parent device for the PCM device @@ -225,23 +247,15 @@ int snd_dmaengine_pcm_register(struct device *dev, const struct snd_dmaengine_pcm_config *config, unsigned int flags) { struct dmaengine_pcm *pcm; - unsigned int i; pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); if (!pcm) return -ENOMEM; pcm->config = config; + pcm->flags = flags; - if (flags & SND_DMAENGINE_PCM_FLAG_COMPAT) - pcm->compat = true; - - if (!(flags & SND_DMAENGINE_PCM_FLAG_NO_DT) && dev->of_node) { - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - pcm->chan[i] = of_dma_request_slave_channel(dev->of_node, - dmaengine_pcm_dma_channel_names[i]); - } - } + dmaengine_pcm_request_chan_of(pcm, dev->of_node); if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) return snd_soc_add_platform(dev, &pcm->platform, @@ -272,8 +286,11 @@ void snd_dmaengine_pcm_unregister(struct device *dev) pcm = soc_platform_to_pcm(platform); for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - if (pcm->chan[i]) + if (pcm->chan[i]) { dma_release_channel(pcm->chan[i]); + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) + break; + } } snd_soc_remove_platform(platform); -- 1.8.0

3 19

[alsa-devel] [PATCH 1/2] ASoC: sgtl5000: Simplify the handling of VDDD
by Fabio Estevam 13 May '13

13 May '13

sgtl5000 has two power supplies that are mandatory: VDDA and VDDIO. VDDD is an optional supply and can be left unconnected and in this case an internal regulator is used. Current code always use the internal regulator even if an external VDDD supply is provided and the sglt5000 revision is equal or greater than 0x11. It is better to simplify the code to handle VDDD and always turn on the internal regulator since all the production sgtl5000 have revision equal or greater 0x11. While at it, turn on the regulators prior to reading the SGTL5000_CHIP_ID register. Tested on a mx51evk board that has VDDA and VDDIO provided by the PMIC. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 394 ++++++++----------------------------------- 1 file changed, 70 insertions(+), 324 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1c3b20f..2e56d00 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -65,45 +65,12 @@ static const struct reg_default sgtl5000_reg_defaults[] = { enum sgtl5000_regulator_supplies { VDDA, VDDIO, - VDDD, SGTL5000_SUPPLY_NUM }; -/* vddd is optional supply */ static const char *supply_names[SGTL5000_SUPPLY_NUM] = { "VDDA", "VDDIO", - "VDDD" -}; - -#define LDO_CONSUMER_NAME "VDDD_LDO" -#define LDO_VOLTAGE 1200000 - -static struct regulator_consumer_supply ldo_consumer[] = { - REGULATOR_SUPPLY(LDO_CONSUMER_NAME, NULL), -}; - -static struct regulator_init_data ldo_init_data = { - .constraints = { - .min_uV = 1200000, - .max_uV = 1200000, - .valid_modes_mask = REGULATOR_MODE_NORMAL, - .valid_ops_mask = REGULATOR_CHANGE_STATUS, - }, - .num_consumer_supplies = 1, - .consumer_supplies = &ldo_consumer[0], -}; - -/* - * sgtl5000 internal ldo regulator, - * enabled when VDDD not provided - */ -struct ldo_regulator { - struct regulator_desc desc; - struct regulator_dev *dev; - int voltage; - void *codec_data; - bool enabled; }; /* sgtl5000 private structure in codec */ @@ -112,7 +79,7 @@ struct sgtl5000_priv { int master; /* i2s master or not */ int fmt; /* i2s data format */ struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM]; - struct ldo_regulator *ldo; + struct regulator *vddd; struct regmap *regmap; }; @@ -732,157 +699,6 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -#ifdef CONFIG_REGULATOR -static int ldo_regulator_is_enabled(struct regulator_dev *dev) -{ - struct ldo_regulator *ldo = rdev_get_drvdata(dev); - - return ldo->enabled; -} - -static int ldo_regulator_enable(struct regulator_dev *dev) -{ - struct ldo_regulator *ldo = rdev_get_drvdata(dev); - struct snd_soc_codec *codec = (struct snd_soc_codec *)ldo->codec_data; - int reg; - - if (ldo_regulator_is_enabled(dev)) - return 0; - - /* set regulator value firstly */ - reg = (1600 - ldo->voltage / 1000) / 50; - reg = clamp(reg, 0x0, 0xf); - - /* amend the voltage value, unit: uV */ - ldo->voltage = (1600 - reg * 50) * 1000; - - /* set voltage to register */ - snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, - SGTL5000_LINREG_VDDD_MASK, reg); - - snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_LINEREG_D_POWERUP, - SGTL5000_LINEREG_D_POWERUP); - - /* when internal ldo enabled, simple digital power can be disabled */ - snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_LINREG_SIMPLE_POWERUP, - 0); - - ldo->enabled = 1; - return 0; -} - -static int ldo_regulator_disable(struct regulator_dev *dev) -{ - struct ldo_regulator *ldo = rdev_get_drvdata(dev); - struct snd_soc_codec *codec = (struct snd_soc_codec *)ldo->codec_data; - - snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_LINEREG_D_POWERUP, - 0); - - /* clear voltage info */ - snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, - SGTL5000_LINREG_VDDD_MASK, 0); - - ldo->enabled = 0; - - return 0; -} - -static int ldo_regulator_get_voltage(struct regulator_dev *dev) -{ - struct ldo_regulator *ldo = rdev_get_drvdata(dev); - - return ldo->voltage; -} - -static struct regulator_ops ldo_regulator_ops = { - .is_enabled = ldo_regulator_is_enabled, - .enable = ldo_regulator_enable, - .disable = ldo_regulator_disable, - .get_voltage = ldo_regulator_get_voltage, -}; - -static int ldo_regulator_register(struct snd_soc_codec *codec, - struct regulator_init_data *init_data, - int voltage) -{ - struct ldo_regulator *ldo; - struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); - struct regulator_config config = { }; - - ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL); - - if (!ldo) { - dev_err(codec->dev, "failed to allocate ldo_regulator\n"); - return -ENOMEM; - } - - ldo->desc.name = kstrdup(dev_name(codec->dev), GFP_KERNEL); - if (!ldo->desc.name) { - kfree(ldo); - dev_err(codec->dev, "failed to allocate decs name memory\n"); - return -ENOMEM; - } - - ldo->desc.type = REGULATOR_VOLTAGE; - ldo->desc.owner = THIS_MODULE; - ldo->desc.ops = &ldo_regulator_ops; - ldo->desc.n_voltages = 1; - - ldo->codec_data = codec; - ldo->voltage = voltage; - - config.dev = codec->dev; - config.driver_data = ldo; - config.init_data = init_data; - - ldo->dev = regulator_register(&ldo->desc, &config); - if (IS_ERR(ldo->dev)) { - int ret = PTR_ERR(ldo->dev); - - dev_err(codec->dev, "failed to register regulator\n"); - kfree(ldo->desc.name); - kfree(ldo); - - return ret; - } - sgtl5000->ldo = ldo; - - return 0; -} - -static int ldo_regulator_remove(struct snd_soc_codec *codec) -{ - struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); - struct ldo_regulator *ldo = sgtl5000->ldo; - - if (!ldo) - return 0; - - regulator_unregister(ldo->dev); - kfree(ldo->desc.name); - kfree(ldo); - - return 0; -} -#else -static int ldo_regulator_register(struct snd_soc_codec *codec, - struct regulator_init_data *init_data, - int voltage) -{ - dev_err(codec->dev, "this setup needs regulator support in the kernel\n"); - return -EINVAL; -} - -static int ldo_regulator_remove(struct snd_soc_codec *codec) -{ - return 0; -} -#endif - /* * set dac bias * common state changes: @@ -910,6 +726,11 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, sgtl5000->supplies); if (ret) return ret; + if (sgtl5000->vddd) { + ret = regulator_enable(sgtl5000->vddd); + if (ret) + return ret; + } udelay(10); } @@ -917,6 +738,8 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); + if (sgtl5000->vddd) + regulator_disable(sgtl5000->vddd); break; } @@ -1136,7 +959,10 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) vdda = regulator_get_voltage(sgtl5000->supplies[VDDA].consumer); vddio = regulator_get_voltage(sgtl5000->supplies[VDDIO].consumer); - vddd = regulator_get_voltage(sgtl5000->supplies[VDDD].consumer); + if (sgtl5000->vddd) + vddd = regulator_get_voltage(sgtl5000->vddd); + else + vddd = 0; vdda = vdda / 1000; vddio = vddio / 1000; @@ -1198,15 +1024,14 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) * simple digital supply should be clear to get * proper VDDD voltage. */ - if (ana_pwr & SGTL5000_LINEREG_D_POWERUP) - snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINEREG_D_POWERUP, + SGTL5000_LINEREG_D_POWERUP); + + /* when internal ldo enabled, simple digital power can be disabled */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_LINREG_SIMPLE_POWERUP, 0); - else - snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_LINREG_SIMPLE_POWERUP | - SGTL5000_STARTUP_POWERUP, - 0); /* * set ADC/DAC VAG to vdda / 2, @@ -1245,110 +1070,6 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) return 0; } -static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) -{ - struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); - int ret; - - /* set internal ldo to 1.2v */ - ret = ldo_regulator_register(codec, &ldo_init_data, LDO_VOLTAGE); - if (ret) { - dev_err(codec->dev, - "Failed to register vddd internal supplies: %d\n", ret); - return ret; - } - - sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - - if (ret) { - ldo_regulator_remove(codec); - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - return ret; - } - - dev_info(codec->dev, "Using internal LDO instead of VDDD\n"); - return 0; -} - -static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) -{ - int reg; - int ret; - int rev; - int i; - int external_vddd = 0; - struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); - - for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++) - sgtl5000->supplies[i].supply = supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - if (!ret) - external_vddd = 1; - else { - ret = sgtl5000_replace_vddd_with_ldo(codec); - if (ret) - return ret; - } - - ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - if (ret) - goto err_regulator_free; - - /* wait for all power rails bring up */ - udelay(10); - - /* - * workaround for revision 0x11 and later, - * roll back to use internal LDO - */ - - ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); - if (ret) - goto err_regulator_disable; - - rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; - - if (external_vddd && rev >= 0x11) { - /* disable all regulator first */ - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - /* free VDDD regulator */ - regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - - ret = sgtl5000_replace_vddd_with_ldo(codec); - if (ret) - return ret; - - ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - if (ret) - goto err_regulator_free; - - /* wait for all power rails bring up */ - udelay(10); - } - - return 0; - -err_regulator_disable: - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); -err_regulator_free: - regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - if (external_vddd) - ldo_regulator_remove(codec); - return ret; - -} - static int sgtl5000_probe(struct snd_soc_codec *codec) { int ret; @@ -1362,14 +1083,10 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) return ret; } - ret = sgtl5000_enable_regulators(codec); - if (ret) - return ret; - /* power up sgtl5000 */ ret = sgtl5000_set_power_regs(codec); if (ret) - goto err; + return ret; /* enable small pop, introduce 400ms delay in turning off */ snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, @@ -1412,32 +1129,16 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) /* leading to standby state */ ret = sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (ret) - goto err; + return ret; return 0; -err: - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - ldo_regulator_remove(codec); - - return ret; } static int sgtl5000_remove(struct snd_soc_codec *codec) { - struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); - sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - ldo_regulator_remove(codec); - return 0; } @@ -1497,30 +1198,63 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct sgtl5000_priv *sgtl5000; - int ret, reg, rev; + int ret, reg, rev, i; sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), GFP_KERNEL); if (!sgtl5000) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++) + sgtl5000->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(&client->dev, + ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); + if (ret) { + dev_err(&client->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) { + dev_err(&client->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + sgtl5000->vddd = devm_regulator_get(&client->dev, "VDDD"); + if (!IS_ERR(sgtl5000->vddd)) { + ret = regulator_enable(sgtl5000->vddd); + if (ret) { + dev_err(&client->dev, + "Failed to enable VDDD: %d\n", ret); + goto err_bulk; + } + } else { + sgtl5000->vddd = NULL; + } + + /* wait for all power rails bring up */ + udelay(10); + sgtl5000->regmap = devm_regmap_init_i2c(client, &sgtl5000_regmap); if (IS_ERR(sgtl5000->regmap)) { ret = PTR_ERR(sgtl5000->regmap); dev_err(&client->dev, "Failed to allocate regmap: %d\n", ret); - return ret; + goto err_vddd; } /* read chip information */ ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); if (ret) - return ret; + goto err_vddd; if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != SGTL5000_PARTID_PART_ID) { dev_err(&client->dev, "Device with ID register %x is not a sgtl5000\n", reg); - return -ENODEV; + ret = -ENODEV; + goto err_vddd; } rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; @@ -1531,10 +1265,22 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, /* Ensure sgtl5000 will start with sane register values */ ret = sgtl5000_fill_defaults(sgtl5000); if (ret) - return ret; + goto err_vddd; ret = snd_soc_register_codec(&client->dev, &sgtl5000_driver, &sgtl5000_dai, 1); + if (ret) + goto err_vddd; + + return 0; + +err_vddd: + if (sgtl5000->vddd) + regulator_disable(sgtl5000->vddd); +err_bulk: + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + return ret; } -- 1.7.9.5

2 4

Re: [alsa-devel] [PATCH 11/11] sound/oxygen_io: take msecs_to_jiffies_min into use
by Takashi Iwai 13 May '13

13 May '13

At Fri, 10 May 2013 15:13:29 +0300, Imre Deak wrote: > > Use msecs_to_jiffies_min instead of open-coding the same. > > Signed-off-by: Imre Deak <imre.deak(a)intel.com> > --- > sound/pci/oxygen/oxygen_io.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c > index 521eae4..132ecbe 100644 > --- a/sound/pci/oxygen/oxygen_io.c > +++ b/sound/pci/oxygen/oxygen_io.c > @@ -108,7 +108,7 @@ static int oxygen_ac97_wait(struct oxygen *chip, unsigned int mask) > wait_event_timeout(chip->ac97_waitqueue, > ({ status |= oxygen_read8(chip, OXYGEN_AC97_INTERRUPT_STATUS); > status & mask; }), > - msecs_to_jiffies(1) + 1); > + msecs_to_jiffies_min(1)); This would change the behavior, I guess. (Though, I'm not sure whether the original code was intentional.) And, isn't msecs_to_jiffies_min(1) identical with msecs_to_jiffies(1)? thanks, Takashi

3 3

[alsa-devel] [PATCH] sound: Fix make allmodconfig on MIPS correctly
by Paul Bolle 13 May '13

13 May '13

Commit d4702b189c ("sound: Fix make allmodconfig on MIPS") added a (negative) dependency on ISA_DMA_SUPPORT_BROKEN. Since that Kconfig symbol doesn't exist, this dependency will always evaluate to true. Apparently GENERIC_ISA_DMA_SUPPORT_BROKEN was meant to be used here. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Tested only with "git grep". sound/oss/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index 51c4ba9..1a96402 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -250,7 +250,7 @@ config MSND_FIFOSIZE menuconfig SOUND_OSS tristate "OSS sound modules" depends on ISA_DMA_API && VIRT_TO_BUS - depends on !ISA_DMA_SUPPORT_BROKEN + depends on !GENERIC_ISA_DMA_SUPPORT_BROKEN help OSS is the Open Sound System suite of sound card drivers. They make sound programming easier since they provide a common API. Say Y or -- 1.7.11.7

2 1

[alsa-devel] [PATCH -next] ASoC: mid-x86: add missing iounmap() to probe error and remove
by Wei Yongjun 13 May '13

13 May '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Add the missing iounmap() before return from snd_mfld_mc_probe() in the error handling case and to snd_mfld_mc_remove(). Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/mid-x86/mfld_machine.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 78d5825..8206e84 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -397,7 +397,7 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx); if (ret_val) { pr_err("cannot register IRQ\n"); - goto unalloc; + goto unmap; } /* register the soc card */ snd_soc_card_mfld.dev = &pdev->dev; @@ -412,6 +412,8 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) freeirq: free_irq(irq, mc_drv_ctx); +unmap: + iounmap(mc_drv_ctx->int_base); unalloc: kfree(mc_drv_ctx); return ret_val; @@ -424,6 +426,7 @@ static int snd_mfld_mc_remove(struct platform_device *pdev) pr_debug("snd_mfld_mc_remove called\n"); free_irq(platform_get_irq(pdev, 0), mc_drv_ctx); snd_soc_unregister_card(&snd_soc_card_mfld); + iounmap(mc_drv_ctx->int_base); kfree(mc_drv_ctx); return 0; }

2 1

[alsa-devel] [PATCH 0/5] Power-well API implementation for Haswell
by Wang Xingchao 13 May '13

13 May '13

This patchset intended to Add power-well api support for Haswell. For Intel Haswell , “power well” in GPU has impact for both Display HD-A controller and codecs. Gfx driver has power well feature enaled but donot think much about audio side. The issue is if gfx driver disabled power well, audio side would lose power and crash. David helped a initial patch to add external module patch_hdmi_i915, which built in only DRM_I915 and HDMI_CODEC enabled, This patchset was based on David’s work, thanks David! As codec depends on controller’s power, so we removed the pm callback temporary. So only HDA controller driver would such need to request/release power, the codec will always be safe. Gfx driver would shutdown power well in below cases, audio driver would be damaged as losing power totally without any notification. (There would be more cases not covered, welcome more comments here. i can verify the patchset under more cases) - Only eDP port active - HDMI monitor hot plug - System suspend Also there’re some calling from i915 to disable power well from time to time, this patchset would reject those operations if audio using power well, but it will record i915 driver’s request and shut down power if audio release it. Basically, here's the latest working status on my side: I tested the power well feature on Haswell ULT board, there's only one eDP port used. Without this patch and we enabled power well feature, gfx driver will shutdown power well, audio driver will crash. Applied this patch, display audio driver will request power well on before initialize the card. In gfx side, it will enable power well. Also power_save feature in audio side should be enabled, I set "5" second to let codec enter suspend when free for 5s long. Audio controller driver detects all codecs are suspended it will release power well and suspend too. Gfx driver will receive the request and shut down power well. If user trigger some audio operations(cat codec# info), audio controller driver will request power well again... If gfx side decided to disable power well now, while audio is in use, power well would be kept on. (i will send out a documentation which has some logs and analysis there.) Generally audio drvier will request power well on need and release it immediately after suspend. Gfx should make decision at his side. The new introduced module "hda-i915" has dependency on i915, only built in when i915 enabled. So it's safe for call. For non-Haswell platform, power well is no need atm. Even the module is built in wrongly, hda driver will not call the power well api. Also gfx will reject power well request at its side, so it's safe too. This patch could make sure it would not cause damage when snd-hda-intel and i915 module loaded in any order. If i915 module loaded at first, it's safe to get the symbol. If snd-hda-intel loaded first, it will try to load i915 manually and get notification when the power-well api are exported. Wang Xingchao (5): drm/915: Add private api for power well usage ALSA: hda - Add external module hda-i915 for power well ALSA: hda - Power well request/release for hda controller ALSA: hda - Fix module dependency with gfx i915 ALSA/i915: Check power well API existense before calling drivers/gpu/drm/i915/i915_dma.c | 3 + drivers/gpu/drm/i915/intel_drv.h | 2 + drivers/gpu/drm/i915/intel_pm.c | 139 ++++++++++++++++++++++++++++++++++++-- include/drm/i915_powerwell.h | 40 +++++++++++ sound/pci/hda/Kconfig | 13 ++++ sound/pci/hda/Makefile | 6 ++ sound/pci/hda/hda_i915.c | 92 +++++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 15 ++++ sound/pci/hda/hda_intel.c | 22 ++++++ 9 files changed, 325 insertions(+), 7 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.7.9.5

5 21

[alsa-devel] [PATCH v2] ASoC: mxs: move to use generic DMA helper
by Shawn Guo 13 May '13

13 May '13

With mxs-dma converted to generic DMA bindings, let's move mxs-pcm to use it by removing flages SND_DMAENGINE_PCM_FLAG_NO_DT and SND_DMAENGINE_PCM_FLAG_COMPAT. As the result, those mxs custom dma params code can be removed now. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- Changes since v1: * rebase against Mark's topic/mxs branch .../devicetree/bindings/sound/mxs-saif.txt | 17 ++++++++---- sound/soc/mxs/mxs-pcm.c | 18 ------------ sound/soc/mxs/mxs-pcm.h | 7 ----- sound/soc/mxs/mxs-saif.c | 29 +------------------- sound/soc/mxs/mxs-saif.h | 1 - 5 files changed, 12 insertions(+), 60 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mxs-saif.txt b/Documentation/devicetree/bindings/sound/mxs-saif.txt index c37ba61..7ba07a1 100644 --- a/Documentation/devicetree/bindings/sound/mxs-saif.txt +++ b/Documentation/devicetree/bindings/sound/mxs-saif.txt @@ -3,8 +3,11 @@ Required properties: - compatible: Should be "fsl,<chip>-saif" - reg: Should contain registers location and length -- interrupts: Should contain ERROR and DMA interrupts -- fsl,saif-dma-channel: APBX DMA channel for the SAIF +- interrupts: Should contain ERROR interrupt number +- dmas: DMA specifier, consisting of a phandle to DMA controller node + and SAIF DMA channel ID. + Refer to dma.txt and fsl-mxs-dma.txt for details. +- dma-names: Must be "rx-tx". Optional properties: - fsl,saif-master: phandle to the master SAIF. It's only required for @@ -23,14 +26,16 @@ aliases { saif0: saif@80042000 { compatible = "fsl,imx28-saif"; reg = <0x80042000 2000>; - interrupts = <59 80>; - fsl,saif-dma-channel = <4>; + interrupts = <59>; + dmas = <&dma_apbx 4>; + dma-names = "rx-tx"; }; saif1: saif@80046000 { compatible = "fsl,imx28-saif"; reg = <0x80046000 2000>; - interrupts = <58 81>; - fsl,saif-dma-channel = <5>; + interrupts = <58>; + dmas = <&dma_apbx 5>; + dma-names = "rx-tx"; fsl,saif-master = <&saif0>; }; diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index b41fffc..b16abbb 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -49,24 +49,8 @@ static const struct snd_pcm_hardware snd_mxs_hardware = { .fifo_size = 32, }; -static bool filter(struct dma_chan *chan, void *param) -{ - struct mxs_pcm_dma_params *dma_params = param; - - if (!mxs_dma_is_apbx(chan)) - return false; - - if (chan->chan_id != dma_params->chan_num) - return false; - - chan->private = &dma_params->dma_data; - - return true; -} - static const struct snd_dmaengine_pcm_config mxs_dmaengine_pcm_config = { .pcm_hardware = &snd_mxs_hardware, - .compat_filter_fn = filter, .prealloc_buffer_size = 64 * 1024, }; @@ -74,8 +58,6 @@ int mxs_pcm_platform_register(struct device *dev) { return snd_dmaengine_pcm_register(dev, &mxs_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | - SND_DMAENGINE_PCM_FLAG_NO_DT | - SND_DMAENGINE_PCM_FLAG_COMPAT | SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX); } EXPORT_SYMBOL_GPL(mxs_pcm_platform_register); diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h index 3aa918f..bc685b6 100644 --- a/sound/soc/mxs/mxs-pcm.h +++ b/sound/soc/mxs/mxs-pcm.h @@ -19,13 +19,6 @@ #ifndef _MXS_PCM_H #define _MXS_PCM_H -#include <linux/fsl/mxs-dma.h> - -struct mxs_pcm_dma_params { - struct mxs_dma_data dma_data; - int chan_num; -}; - int mxs_pcm_platform_register(struct device *dev); void mxs_pcm_platform_unregister(struct device *dev); diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 71a972f..49d8700 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -26,7 +26,6 @@ #include <linux/clk.h> #include <linux/delay.h> #include <linux/time.h> -#include <linux/fsl/mxs-dma.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -604,8 +603,6 @@ static int mxs_saif_dai_probe(struct snd_soc_dai *dai) struct mxs_saif *saif = dev_get_drvdata(dai->dev); snd_soc_dai_set_drvdata(dai, saif); - dai->playback_dma_data = &saif->dma_param; - dai->capture_dma_data = &saif->dma_param; return 0; } @@ -664,7 +661,7 @@ static irqreturn_t mxs_saif_irq(int irq, void *dev_id) static int mxs_saif_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; - struct resource *iores, *dmares; + struct resource *iores; struct mxs_saif *saif; int ret = 0; struct device_node *master; @@ -719,22 +716,6 @@ static int mxs_saif_probe(struct platform_device *pdev) if (IS_ERR(saif->base)) return PTR_ERR(saif->base); - dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmares) { - /* - * TODO: This is a temporary solution and should be changed - * to use generic DMA binding later when the helplers get in. - */ - ret = of_property_read_u32(np, "fsl,saif-dma-channel", - &saif->dma_param.chan_num); - if (ret) { - dev_err(&pdev->dev, "failed to get dma channel\n"); - return ret; - } - } else { - saif->dma_param.chan_num = dmares->start; - } - saif->irq = platform_get_irq(pdev, 0); if (saif->irq < 0) { ret = saif->irq; @@ -751,14 +732,6 @@ static int mxs_saif_probe(struct platform_device *pdev) return ret; } - saif->dma_param.dma_data.chan_irq = platform_get_irq(pdev, 1); - if (saif->dma_param.dma_data.chan_irq < 0) { - ret = saif->dma_param.dma_data.chan_irq; - dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", - ret); - return ret; - } - platform_set_drvdata(pdev, saif); ret = snd_soc_register_component(&pdev->dev, &mxs_saif_component, diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index 3cb342e..53eaa4b 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -117,7 +117,6 @@ struct mxs_saif { unsigned int mclk_in_use; void __iomem *base; int irq; - struct mxs_pcm_dma_params dma_param; unsigned int id; unsigned int master_id; unsigned int cur_rate; -- 1.7.9.5

2 1

[alsa-devel] ASOC bug? maybe no need to set bias_level to PREPARE.
by Zhang, Ning A 13 May '13

13 May '13

When I try to fix wm1811 standby issue for our customer, I find a strange behavior in ASOC core. here is my test code: snd_soc_dapm_disable_pin(&dapm,"MICBIAS2"); snd_soc_dapm_disable_pin(&dapm,"CLK_SYS"); snd_soc_dapm_sync(&dapm); and add debug info in dapm_pre_sequence_async() and we will find these debug info printed: d->bias_level = 1 d->target_bias_level = 0 in snd_soc_dapm_set_bias_level from the debug log, when STANDBY -> OFF transition, it set bias_level to PREPARE. that's wrong logic. And more there is comment in source code: /* Prepare for a STADDBY->ON or ON->STANDBY transition */ But, the code isn't the same as the comment. ---- >From 58d8994bdc543909e9904b8df0787afc36684283 Mon Sep 17 00:00:00 2001 From: ning.a.zhang <ning.a.zhang(a)intel.com> Date: Fri, 10 May 2013 09:44:06 +0700 Subject: [PATCH] Asoc dapm behavior fix in soc_dapm.c, there is a illogic behavior, when bias_level from STANDBY -> OFF transition. source does not follow its comment. this patch is based on the comment Signed-off-by: ning.a.zhang <ning.a.zhang(a)intel.com> --- sound/soc/soc-dapm.c | 14 ++++++++++++-- 1 files changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d6d9ba2..834ab4c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1486,8 +1486,9 @@ static void dapm_pre_sequence_async(void *data, async_cookie_t cookie) "ASoC: Failed to turn on bias: %d\n", ret); } - /* Prepare for a STADDBY->ON or ON->STANDBY transition */ - if (d->bias_level != d->target_bias_level) { + /* If we're on and we're not supposed to be go into PREPARE */ + if (d->bias_level == SND_SOC_BIAS_ON && + d->target_bias_level != SND_SOC_BIAS_ON) { ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE); if (ret != 0) dev_err(d->dev, @@ -1525,6 +1526,15 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) pm_runtime_put(d->dev); } + if (d->bias_level == SND_SOC_BIAS_STANDBY && + (d->target_bias_level == SND_SOC_BIAS_PREPARE || + d->target_bias_level == SND_SOC_BIAS_ON)) { + ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE); + if (ret != 0) + dev_err(d->dev, "ASoC: Failed to apply prepare bias: %d\n", + ret); + } + /* If we just powered up then move to active bias */ if (d->bias_level == SND_SOC_BIAS_PREPARE && d->target_bias_level == SND_SOC_BIAS_ON) { -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: wm8994: Support EFS mode for FLL
by Mark Brown 13 May '13

13 May '13

Later WM8994 devices support an enhanced accuracy FLL divisor mode called EFS which allows more precise selection of fractional source to output ratios. Support this on relevant devices. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 49 +++++++++++++++++++++++++++++++++------------ 1 file changed, 36 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1eb152c..9f32dd8 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -16,6 +16,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/gcd.h> #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/pm_runtime.h> @@ -2005,15 +2006,16 @@ struct fll_div { u16 outdiv; u16 n; u16 k; + u16 lambda; u16 clk_ref_div; u16 fll_fratio; }; -static int wm8994_get_fll_config(struct fll_div *fll, +static int wm8994_get_fll_config(struct wm8994 *control, struct fll_div *fll, int freq_in, int freq_out) { u64 Kpart; - unsigned int K, Ndiv, Nmod; + unsigned int K, Ndiv, Nmod, gcd_fll; pr_debug("FLL input=%dHz, output=%dHz\n", freq_in, freq_out); @@ -2062,20 +2064,30 @@ static int wm8994_get_fll_config(struct fll_div *fll, Nmod = freq_out % freq_in; pr_debug("Nmod=%d\n", Nmod); - /* Calculate fractional part - scale up so we can round. */ - Kpart = FIXED_FLL_SIZE * (long long)Nmod; + switch (control->type) { + case WM8994: + /* Calculate fractional part - scale up so we can round. */ + Kpart = FIXED_FLL_SIZE * (long long)Nmod; - do_div(Kpart, freq_in); + do_div(Kpart, freq_in); - K = Kpart & 0xFFFFFFFF; + K = Kpart & 0xFFFFFFFF; - if ((K % 10) >= 5) - K += 5; + if ((K % 10) >= 5) + K += 5; - /* Move down to proper range now rounding is done */ - fll->k = K / 10; + /* Move down to proper range now rounding is done */ + fll->k = K / 10; - pr_debug("N=%x K=%x\n", fll->n, fll->k); + pr_debug("N=%x K=%x\n", fll->n, fll->k); + + default: + gcd_fll = gcd(freq_out, freq_in); + + fll->k = (freq_out - (freq_in * fll->n)) / gcd_fll; + fll->lambda = freq_in / gcd_fll; + + } return 0; } @@ -2139,9 +2151,9 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, * analysis bugs spewing warnings. */ if (freq_out) - ret = wm8994_get_fll_config(&fll, freq_in, freq_out); + ret = wm8994_get_fll_config(control, &fll, freq_in, freq_out); else - ret = wm8994_get_fll_config(&fll, wm8994->fll[id].in, + ret = wm8994_get_fll_config(control, &fll, wm8994->fll[id].in, wm8994->fll[id].out); if (ret < 0) return ret; @@ -2186,6 +2198,17 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, WM8994_FLL1_N_MASK, fll.n << WM8994_FLL1_N_SHIFT); + if (fll.lambda) { + snd_soc_update_bits(codec, WM8958_FLL1_EFS_1 + reg_offset, + WM8958_FLL1_LAMBDA_MASK, + fll.lambda); + snd_soc_update_bits(codec, WM8958_FLL1_EFS_2 + reg_offset, + WM8958_FLL1_EFS_ENA, WM8958_FLL1_EFS_ENA); + } else { + snd_soc_update_bits(codec, WM8958_FLL1_EFS_2 + reg_offset, + WM8958_FLL1_EFS_ENA, 0); + } + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, WM8994_FLL1_FRC_NCO | WM8958_FLL1_BYP | WM8994_FLL1_REFCLK_DIV_MASK | -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: mxs: move to use generic DMA helper
by Shawn Guo 13 May '13

13 May '13

With mxs-dma converted to generic DMA bindings, let's move mxs-pcm to use it by removing flages SND_DMAENGINE_PCM_FLAG_NO_DT and SND_DMAENGINE_PCM_FLAG_COMPAT. As the result, those mxs custom dma params code can be removed now. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- .../devicetree/bindings/sound/mxs-saif.txt | 17 ++++++++---- sound/soc/mxs/mxs-pcm.c | 18 ------------ sound/soc/mxs/mxs-pcm.h | 7 ----- sound/soc/mxs/mxs-saif.c | 29 +------------------- sound/soc/mxs/mxs-saif.h | 1 - 5 files changed, 12 insertions(+), 60 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mxs-saif.txt b/Documentation/devicetree/bindings/sound/mxs-saif.txt index c37ba61..7ba07a1 100644 --- a/Documentation/devicetree/bindings/sound/mxs-saif.txt +++ b/Documentation/devicetree/bindings/sound/mxs-saif.txt @@ -3,8 +3,11 @@ Required properties: - compatible: Should be "fsl,<chip>-saif" - reg: Should contain registers location and length -- interrupts: Should contain ERROR and DMA interrupts -- fsl,saif-dma-channel: APBX DMA channel for the SAIF +- interrupts: Should contain ERROR interrupt number +- dmas: DMA specifier, consisting of a phandle to DMA controller node + and SAIF DMA channel ID. + Refer to dma.txt and fsl-mxs-dma.txt for details. +- dma-names: Must be "rx-tx". Optional properties: - fsl,saif-master: phandle to the master SAIF. It's only required for @@ -23,14 +26,16 @@ aliases { saif0: saif@80042000 { compatible = "fsl,imx28-saif"; reg = <0x80042000 2000>; - interrupts = <59 80>; - fsl,saif-dma-channel = <4>; + interrupts = <59>; + dmas = <&dma_apbx 4>; + dma-names = "rx-tx"; }; saif1: saif@80046000 { compatible = "fsl,imx28-saif"; reg = <0x80046000 2000>; - interrupts = <58 81>; - fsl,saif-dma-channel = <5>; + interrupts = <58>; + dmas = <&dma_apbx 5>; + dma-names = "rx-tx"; fsl,saif-master = <&saif0>; }; diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index b41fffc..b16abbb 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -49,24 +49,8 @@ static const struct snd_pcm_hardware snd_mxs_hardware = { .fifo_size = 32, }; -static bool filter(struct dma_chan *chan, void *param) -{ - struct mxs_pcm_dma_params *dma_params = param; - - if (!mxs_dma_is_apbx(chan)) - return false; - - if (chan->chan_id != dma_params->chan_num) - return false; - - chan->private = &dma_params->dma_data; - - return true; -} - static const struct snd_dmaengine_pcm_config mxs_dmaengine_pcm_config = { .pcm_hardware = &snd_mxs_hardware, - .compat_filter_fn = filter, .prealloc_buffer_size = 64 * 1024, }; @@ -74,8 +58,6 @@ int mxs_pcm_platform_register(struct device *dev) { return snd_dmaengine_pcm_register(dev, &mxs_dmaengine_pcm_config, SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | - SND_DMAENGINE_PCM_FLAG_NO_DT | - SND_DMAENGINE_PCM_FLAG_COMPAT | SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX); } EXPORT_SYMBOL_GPL(mxs_pcm_platform_register); diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h index 3aa918f..bc685b6 100644 --- a/sound/soc/mxs/mxs-pcm.h +++ b/sound/soc/mxs/mxs-pcm.h @@ -19,13 +19,6 @@ #ifndef _MXS_PCM_H #define _MXS_PCM_H -#include <linux/fsl/mxs-dma.h> - -struct mxs_pcm_dma_params { - struct mxs_dma_data dma_data; - int chan_num; -}; - int mxs_pcm_platform_register(struct device *dev); void mxs_pcm_platform_unregister(struct device *dev); diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index d31dc52..d8c9a19 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -26,7 +26,6 @@ #include <linux/clk.h> #include <linux/delay.h> #include <linux/time.h> -#include <linux/fsl/mxs-dma.h> #include <linux/pinctrl/consumer.h> #include <sound/core.h> #include <sound/pcm.h> @@ -605,8 +604,6 @@ static int mxs_saif_dai_probe(struct snd_soc_dai *dai) struct mxs_saif *saif = dev_get_drvdata(dai->dev); snd_soc_dai_set_drvdata(dai, saif); - dai->playback_dma_data = &saif->dma_param; - dai->capture_dma_data = &saif->dma_param; return 0; } @@ -665,7 +662,7 @@ static irqreturn_t mxs_saif_irq(int irq, void *dev_id) static int mxs_saif_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; - struct resource *iores, *dmares; + struct resource *iores; struct mxs_saif *saif; struct pinctrl *pinctrl; int ret = 0; @@ -727,22 +724,6 @@ static int mxs_saif_probe(struct platform_device *pdev) if (IS_ERR(saif->base)) return PTR_ERR(saif->base); - dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmares) { - /* - * TODO: This is a temporary solution and should be changed - * to use generic DMA binding later when the helplers get in. - */ - ret = of_property_read_u32(np, "fsl,saif-dma-channel", - &saif->dma_param.chan_num); - if (ret) { - dev_err(&pdev->dev, "failed to get dma channel\n"); - return ret; - } - } else { - saif->dma_param.chan_num = dmares->start; - } - saif->irq = platform_get_irq(pdev, 0); if (saif->irq < 0) { ret = saif->irq; @@ -759,14 +740,6 @@ static int mxs_saif_probe(struct platform_device *pdev) return ret; } - saif->dma_param.dma_data.chan_irq = platform_get_irq(pdev, 1); - if (saif->dma_param.dma_data.chan_irq < 0) { - ret = saif->dma_param.dma_data.chan_irq; - dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", - ret); - return ret; - } - platform_set_drvdata(pdev, saif); ret = snd_soc_register_component(&pdev->dev, &mxs_saif_component, diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index 3cb342e..53eaa4b 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -117,7 +117,6 @@ struct mxs_saif { unsigned int mclk_in_use; void __iomem *base; int irq; - struct mxs_pcm_dma_params dma_param; unsigned int id; unsigned int master_id; unsigned int cur_rate; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: jz4740-i2s: Use clk_prepare_enable/clk_disable_unprepare
by Lars-Peter Clausen 12 May '13

12 May '13

In preparation to switching the jz4740 clk driver to the common clk framework update the clk enable/disable calls to clk_prepare_enable/clk_disable_unprepare. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/jz4740/jz4740-i2s.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 9a12644..12c21bd 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -118,7 +118,7 @@ static int jz4740_i2s_startup(struct snd_pcm_substream *substream, ctrl |= JZ_AIC_CTRL_FLUSH; jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl); - clk_enable(i2s->clk_i2s); + clk_prepare_enable(i2s->clk_i2s); conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); conf |= JZ_AIC_CONF_ENABLE; @@ -140,7 +140,7 @@ static void jz4740_i2s_shutdown(struct snd_pcm_substream *substream, conf &= ~JZ_AIC_CONF_ENABLE; jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); - clk_disable(i2s->clk_i2s); + clk_disable_unprepare(i2s->clk_i2s); } static int jz4740_i2s_trigger(struct snd_pcm_substream *substream, int cmd, @@ -314,10 +314,10 @@ static int jz4740_i2s_suspend(struct snd_soc_dai *dai) conf &= ~JZ_AIC_CONF_ENABLE; jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); - clk_disable(i2s->clk_i2s); + clk_disable_unprepare(i2s->clk_i2s); } - clk_disable(i2s->clk_aic); + clk_disable_unprepare(i2s->clk_aic); return 0; } @@ -327,10 +327,10 @@ static int jz4740_i2s_resume(struct snd_soc_dai *dai) struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); uint32_t conf; - clk_enable(i2s->clk_aic); + clk_prepare_enable(i2s->clk_aic); if (dai->active) { - clk_enable(i2s->clk_i2s); + clk_prepare_enable(i2s->clk_i2s); conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); conf |= JZ_AIC_CONF_ENABLE; @@ -368,7 +368,7 @@ static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai) struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); uint32_t conf; - clk_enable(i2s->clk_aic); + clk_prepare_enable(i2s->clk_aic); jz4740_i2c_init_pcm_config(i2s); @@ -388,7 +388,7 @@ static int jz4740_i2s_dai_remove(struct snd_soc_dai *dai) { struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); - clk_disable(i2s->clk_aic); + clk_disable_unprepare(i2s->clk_aic); return 0; } -- 1.8.2.1

2 1

[alsa-devel] UCM profiles taken from Ubuntu
by Marcin Juszkiewicz 11 May '13

11 May '13

When I bought Samsung ARM Chromebook few months ago I had no idea about UCM profiles and burnt speakers (left is dead, right is resting). Then I learnt, took ones from ChromeOS and added them into Ubuntu so other users will be a bit more safe. Later I had a discussion with some Debian, Ubuntu, Fedora developers about how to solve such problems and how to keep UCM profiles shared between distributions. Liam Girdwood pointed me to not used git tree at ALSA Project server and finally I reminded myself about it. The idea is to collect UCM profiles, keep them in one place and create alsa-ucm package in each distribution.

6 11

[alsa-devel] [PATCH v5 1/2] ASoC: sgtl5000: Read SGTL5000_CHIP_ID in i2c_probe()
by Fabio Estevam 10 May '13

10 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> The usual place for reading chip ID is inside i2c_probe, so move it there and also convert it to regmap. sgtl5000_enable_regulators() needs to read the chip revision, so keep the revision check there. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v4: - Just introduced in this version. sound/soc/codecs/sgtl5000.c | 39 ++++++++++++++++++++++++--------------- 1 file changed, 24 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 327b443..af0c8aa 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1275,7 +1275,7 @@ static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) { - u16 reg; + int reg; int ret; int rev; int i; @@ -1303,23 +1303,17 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) /* wait for all power rails bring up */ udelay(10); - /* read chip information */ - reg = snd_soc_read(codec, SGTL5000_CHIP_ID); - if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != - SGTL5000_PARTID_PART_ID) { - dev_err(codec->dev, - "Device with ID register %x is not a sgtl5000\n", reg); - ret = -ENODEV; - goto err_regulator_disable; - } - - rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; - dev_info(codec->dev, "sgtl5000 revision 0x%x\n", rev); - /* * workaround for revision 0x11 and later, * roll back to use internal LDO */ + + ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); + if (ret) + goto err_regulator_disable; + + rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; + if (external_vddd && rev >= 0x11) { /* disable all regulator first */ regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), @@ -1478,7 +1472,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct sgtl5000_priv *sgtl5000; - int ret; + int ret, reg, rev; sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), GFP_KERNEL); @@ -1492,6 +1486,21 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, return ret; } + /* read chip information */ + ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); + if (ret) + return ret; + + if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != + SGTL5000_PARTID_PART_ID) { + dev_err(&client->dev, + "Device with ID register %x is not a sgtl5000\n", reg); + return -ENODEV; + } + + rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; + dev_info(&client->dev, "sgtl5000 revision 0x%x\n", rev); + i2c_set_clientdata(client, sgtl5000); ret = snd_soc_register_codec(&client->dev, -- 1.7.9.5

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[alsa-devel] [PATCH] snd-es1968: Add radio support for MediaForte M56VAP
by Ondrej Zary 10 May '13

10 May '13

Add support for TEA5757 tuner on MediaForte M56VAP sound+modem+radio card. The GPIO connection type is automatically detected (like snd-fm801 driver). Also add a safety subsystem vendor check to skip radio detection if vendor differs from ESS (so we don't touch GPIOs on laptops). Tested with SF64-PCE2 and M56VAP cards. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/es1968.c | 73 +++++++++++++++++++++++++++++++++++---------------- 1 files changed, 50 insertions(+), 23 deletions(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index a1f32b5..db08641 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -564,6 +564,7 @@ struct es1968 { #ifdef CONFIG_SND_ES1968_RADIO struct v4l2_device v4l2_dev; struct snd_tea575x tea; + unsigned int tea575x_tuner; #endif }; @@ -2557,37 +2558,47 @@ static int snd_es1968_input_register(struct es1968 *chip) bits 1=unmask write to given bit */ #define IO_DIR 8 /* direction register offset from GPIO_DATA bits 0/1=read/write direction */ -/* mask bits for GPIO lines */ -#define STR_DATA 0x0040 /* GPIO6 */ -#define STR_CLK 0x0080 /* GPIO7 */ -#define STR_WREN 0x0100 /* GPIO8 */ -#define STR_MOST 0x0200 /* GPIO9 */ + +/* GPIO to TEA575x maps */ +struct snd_es1968_tea575x_gpio { + u8 data, clk, wren, most; + char *name; +}; + +static struct snd_es1968_tea575x_gpio snd_es1968_tea575x_gpios[] = { + { .data = 6, .clk = 7, .wren = 8, .most = 9, .name = "SF64-PCE2" }, + { .data = 7, .clk = 8, .wren = 6, .most = 10, .name = "M56VAP" }, +}; + +#define get_tea575x_gpio(chip) \ + (&snd_es1968_tea575x_gpios[(chip)->tea575x_tuner]) + static void snd_es1968_tea575x_set_pins(struct snd_tea575x *tea, u8 pins) { struct es1968 *chip = tea->private_data; - unsigned long io = chip->io_port + GPIO_DATA; + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); u16 val = 0; - val |= (pins & TEA575X_DATA) ? STR_DATA : 0; - val |= (pins & TEA575X_CLK) ? STR_CLK : 0; - val |= (pins & TEA575X_WREN) ? STR_WREN : 0; + val |= (pins & TEA575X_DATA) ? (1 << gpio.data) : 0; + val |= (pins & TEA575X_CLK) ? (1 << gpio.clk) : 0; + val |= (pins & TEA575X_WREN) ? (1 << gpio.wren) : 0; - outw(val, io); + outw(val, chip->io_port + GPIO_DATA); } static u8 snd_es1968_tea575x_get_pins(struct snd_tea575x *tea) { struct es1968 *chip = tea->private_data; - unsigned long io = chip->io_port + GPIO_DATA; - u16 val = inw(io); - u8 ret; + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); + u16 val = inw(chip->io_port + GPIO_DATA); + u8 ret = 0; - ret = 0; - if (val & STR_DATA) + if (val & (1 << gpio.data)) ret |= TEA575X_DATA; - if (val & STR_MOST) + if (val & (1 << gpio.most)) ret |= TEA575X_MOST; + return ret; } @@ -2596,13 +2607,18 @@ static void snd_es1968_tea575x_set_direction(struct snd_tea575x *tea, bool outpu struct es1968 *chip = tea->private_data; unsigned long io = chip->io_port + GPIO_DATA; u16 odir = inw(io + IO_DIR); + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); if (output) { - outw(~(STR_DATA | STR_CLK | STR_WREN), io + IO_MASK); - outw(odir | STR_DATA | STR_CLK | STR_WREN, io + IO_DIR); + outw(~((1 << gpio.data) | (1 << gpio.clk) | (1 << gpio.wren)), + io + IO_MASK); + outw(odir | (1 << gpio.data) | (1 << gpio.clk) | (1 << gpio.wren), + io + IO_DIR); } else { - outw(~(STR_CLK | STR_WREN | STR_DATA | STR_MOST), io + IO_MASK); - outw((odir & ~(STR_DATA | STR_MOST)) | STR_CLK | STR_WREN, io + IO_DIR); + outw(~((1 << gpio.clk) | (1 << gpio.wren) | (1 << gpio.data) | (1 << gpio.most)), + io + IO_MASK); + outw((odir & ~((1 << gpio.data) | (1 << gpio.most))) + | (1 << gpio.clk) | (1 << gpio.wren), io + IO_DIR); } } @@ -2772,6 +2788,9 @@ static int snd_es1968_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); #ifdef CONFIG_SND_ES1968_RADIO + /* don't play with GPIOs on laptops */ + if (chip->pci->subsystem_vendor != 0x125d) + goto no_radio; err = v4l2_device_register(&pci->dev, &chip->v4l2_dev); if (err < 0) { snd_es1968_free(chip); @@ -2781,10 +2800,18 @@ static int snd_es1968_create(struct snd_card *card, chip->tea.private_data = chip; chip->tea.radio_nr = radio_nr; chip->tea.ops = &snd_es1968_tea_ops; - strlcpy(chip->tea.card, "SF64-PCE2", sizeof(chip->tea.card)); sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); - if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) - printk(KERN_INFO "es1968: detected TEA575x radio\n"); + for (i = 0; i < ARRAY_SIZE(snd_es1968_tea575x_gpios); i++) { + chip->tea575x_tuner = i; + if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) { + snd_printk(KERN_INFO "es1968: detected TEA575x radio type %s\n", + get_tea575x_gpio(chip)->name); + strlcpy(chip->tea.card, get_tea575x_gpio(chip)->name, + sizeof(chip->tea.card)); + break; + } + } +no_radio: #endif *chip_ret = chip; -- Ondrej Zary

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Re: [alsa-devel] [PATCH v5 2/2] ASoC: sgtl5000: Fix driver probe after reset
by Fabio Estevam 10 May '13

10 May '13

Hi Matt, On Fri, May 10, 2013 at 2:14 PM, Matt Sealey <matt(a)genesi-usa.com> wrote: > Hi Fabio! > > Is there any way to make that failure a little less of a "failure" or > even a warning, since the message below; > >> sgtl5000 0-000a: Using internal LDO instead of VDDD > > Pretty much covers that VDDD was not supplied. I would consider that > you would want to say "VDDD wasn't supplied" in the sense that it is > optional and not give in pdata or dt, and then remark that the LDO was > actually configured.. I don't like it when my kernels boot up and spit > out high numbered errno's for the sake of a completely optional entry > in the design vs. missing and still optional entry in the device tree. Yes, I plan to clean the sgtl5000 log messages. I agree that it is currently too noisy. Will do it after Mark applies this series. Regards, Fabio Estevam

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