June 2013 - Alsa-devel - mailman.alsa-project.org

[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

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[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

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[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.

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[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

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[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

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[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)>] >

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[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.

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[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(-)

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[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

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[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

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[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

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[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(-)

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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

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[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

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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 >

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[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

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