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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