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

[alsa-devel] [PATCH V5] ASoC: fsl: add imx-wm8962 machine driver
by Nicolin Chen 10 Jun '13

10 Jun '13

This is the initial imx-wm8962 device-tree-only machine driver working with fsl_ssi driver. More features can be added on top of it later. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- ChangeLog: v1->v2: * Fixed a typo in probe(): *** data->dai.ops = &imx_hifi_ops, ==> data->dai.ops = &imx_hifi_ops; v2->v3: * Moved Headphone/Microphone jack detection GPIO port to Optional properties. * Use set_bias_level() instead of hw_params/hw_free() to open/close FLL. * Use devm_clk_get() instead of clk_get(). * Dropped clock-frequency grabbing code. * Fixed a typo: *** data->card.late_probe = imx_wm8962_late_probe, *** ==> *** data->card.late_probe = imx_wm8962_late_probe; v3->v4: * Revised binding doc to describe two gpio pins more clearly. v4->v5: * Added return value checking for clk_prepare_enable(). * Revised error message for devm_clk_get() to keep same pattern as above. .../devicetree/bindings/sound/imx-audio-wm8962.txt | 55 ++++ sound/soc/fsl/Kconfig | 12 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8962.c | 340 ++++++++++++++++++++ 4 files changed, 409 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt create mode 100644 sound/soc/fsl/imx-wm8962.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt new file mode 100644 index 0000000..a34b94b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt @@ -0,0 +1,55 @@ +Freescale i.MX audio complex with WM8962 codec + +Required properties: +- compatible : "fsl,imx-audio-wm8962" +- model : The user-visible name of this sound complex +- ssi-controller : The phandle of the i.MX SSI controller +- audio-codec : The phandle of the WM8962 audio codec +- audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names could be power + supplies, WM8962 pins, and the jacks on the board: + + Power supplies: + * Mic Bias + + Board connectors: + * Mic Jack + * Headphone Jack + * Ext Spk + +- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) +- mux-ext-port : The external port of the i.MX audio muxer + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Optional properties: +- hp-det-gpios : The gpio pin to detect plug in/out event that happens to + Headphone jack. + +- mic-det-gpios: The gpio pin to detect plug in/out event that happens to + Micphone jack. + +Example: + +sound { + compatible = "fsl,imx6q-sabresd-wm8962", + "fsl,imx-audio-wm8962"; + model = "wm8962-audio"; + ssi-controller = <&ssi2>; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUTL", + "Headphone Jack", "HPOUTR", + "Ext Spk", "SPKOUTL", + "Ext Spk", "SPKOUTR", + "MICBIAS", "AMIC", + "IN3R", "MICBIAS", + "DMIC", "MICBIAS", + "DMICDAT", "DMIC"; + mux-int-port = <2>; + mux-ext-port = <3>; + hp-det-gpios = <&gpio7 8 1>; /* active low */ + mic-det-gpios = <&gpio1 9 1>; /* active low */ +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 7860cc2..aa43854 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -168,6 +168,18 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8962 + tristate "SoC Audio support for i.MX boards with wm8962" + depends on OF && I2C + select SND_SOC_WM8962 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + help + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8962 codec. + config SND_SOC_IMX_SGTL5000 tristate "SoC Audio support for i.MX boards with sgtl5000" depends on OF && I2C diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 91883f8..d4b4aa8 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -42,6 +42,7 @@ snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o @@ -49,4 +50,5 @@ obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c new file mode 100644 index 0000000..3bc5b9d --- /dev/null +++ b/sound/soc/fsl/imx-wm8962.c @@ -0,0 +1,340 @@ +/* + * Copyright 2013 Freescale Semiconductor, Inc. + * + * Based on imx-sgtl5000.c + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/of_i2c.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <linux/clk.h> +#include <linux/clk-provider.h> +#include <linux/irq.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/pinctrl/consumer.h> + +#include "../codecs/wm8962.h" +#include "imx-audmux.h" + +#define DAI_NAME_SIZE 32 + +struct imx_wm8962_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + struct clk *codec_clk; + unsigned int clk_frequency; +}; + +struct imx_priv { + int hp_gpio; + int hp_active_low; + int hp_status; + int amic_gpio; + int amic_active_low; + int amic_status; + struct platform_device *pdev; + struct snd_pcm_substream *first_stream; + struct snd_pcm_substream *second_stream; +}; +static struct imx_priv card_priv; + +static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static int sample_rate = 44100; +static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; + +static int imx_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + sample_rate = params_rate(params); + sample_format = params_format(params); + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_hifi_hw_params, +}; + +static int imx_wm8962_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + unsigned int pll_out; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + if (sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = sample_rate * 384; + else + pll_out = sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, data->clk_frequency, + pll_out); + if (ret < 0) { + pr_err("Failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_FLL, pll_out, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK: %d\n", ret); + return ret; + } + } + break; + + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_MCLK, data->clk_frequency, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to switch away from FLL: %d\n", + ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL: %d\n", ret); + return ret; + } + } + break; + + default: + break; + } + + dapm->bias_level = level; + + return 0; +} + +static int imx_wm8962_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + data->clk_frequency, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(card->dev, "Failed to set sysclk in %s\n", __func__); + + return ret; +} + +static int imx_wm8962_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct imx_priv *priv = &card_priv; + struct i2c_client *codec_dev; + struct imx_wm8962_data *data; + int int_port, ext_port; + int ret; + + priv->pdev = pdev; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(&pdev->dev, "audmux internal port setup failed\n"); + return ret; + } + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(&pdev->dev, "audmux external port setup failed\n"); + return ret; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + return -EINVAL; + } + + priv->hp_gpio = of_get_named_gpio_flags(np, "hp-det-gpios", 0, + (enum of_gpio_flags *)&priv->hp_active_low); + priv->amic_gpio = of_get_named_gpio_flags(np, "mic-det-gpios", 0, + (enum of_gpio_flags *)&priv->amic_active_low); + + priv->first_stream = NULL; + priv->second_stream = NULL; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); + if (IS_ERR(data->codec_clk)) { + ret = PTR_ERR(data->codec_clk); + dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); + goto fail; + } + + data->clk_frequency = clk_get_rate(data->codec_clk); + ret = clk_prepare_enable(data->codec_clk); + if (ret) { + dev_err(&codec_dev->dev, "failed to enable codec clk: %d\n", ret); + goto fail; + } + + data->dai.name = "HiFi"; + data->dai.stream_name = "HiFi"; + data->dai.codec_dai_name = "wm8962"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.platform_of_node = ssi_np; + data->dai.ops = &imx_hifi_ops; + data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto clk_fail; + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto clk_fail; + data->card.num_links = 1; + data->card.dai_link = &data->dai; + data->card.dapm_widgets = imx_wm8962_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); + + data->card.late_probe = imx_wm8962_late_probe; + data->card.set_bias_level = imx_wm8962_set_bias_level; + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto clk_fail; + } + + platform_set_drvdata(pdev, data); + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; + +clk_fail: + if (!IS_ERR(data->codec_clk)) + clk_disable_unprepare(data->codec_clk); +fail: + if (ssi_np) + of_node_put(ssi_np); + if (codec_np) + of_node_put(codec_np); + + return ret; +} + +static int imx_wm8962_remove(struct platform_device *pdev) +{ + struct imx_wm8962_data *data = platform_get_drvdata(pdev); + + if (!IS_ERR(data->codec_clk)) + clk_disable_unprepare(data->codec_clk); + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_wm8962_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8962", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids); + +static struct platform_driver imx_wm8962_driver = { + .driver = { + .name = "imx-wm8962", + .owner = THIS_MODULE, + .of_match_table = imx_wm8962_dt_ids, + }, + .probe = imx_wm8962_probe, + .remove = imx_wm8962_remove, +}; +module_platform_driver(imx_wm8962_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8962"); -- 1.7.1

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[alsa-devel] [PATCH] ASoC: ssm2518: Fix trivial typo in checking tx_mask and rx_mask values
by Axel Lin 10 Jun '13

10 Jun '13

Otherwise, ssm2518_set_tdm_slot() always returns error if slots != 0. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/ssm2518.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index a76b341..f4883d9 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -538,7 +538,7 @@ static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, SSM2518_REG_SAI_CTRL1, SSM2518_SAI_CTRL1_SAI_MASK, SSM2518_SAI_CTRL1_SAI_I2S); - if (tx_mask == 0 || tx_mask != 0) + if (tx_mask == 0 || rx_mask != 0) return -EINVAL; if (slots == 1) { -- 1.8.1.2

3 2

[alsa-devel] firewire-kernel-streaming branch into main kernel tree?
by Rolf Anderegg 10 Jun '13

10 Jun '13

Hello Clemens, it's been a long time since I last inquired about the snd_dice module provided by your firewire-kernel-streaming branch (http://git.alsa-project.org/?p=alsa-kprivate.git;a=shortlog;h=refs/heads/fi…), which is a good sign I guess. In the meantime we have been making heavy use of the snd_dice module for over a year with all of our Weiss DICE based products (obviously playback only), and it has proven to be extremely stable (currently 3.4.18-rt29, patched with firewire-kernel-streaming). Therefore I have been meaning to ask you for some time now whether you have any interest in merging this firewire-kernel-streaming branch into the main kernel tree (at least as an experimental option, mind you), or are there any major objections? Given that you don't have unlimited time at hand to further develop the snd_dice module, let alone provide capture support in the near future, AND the fact that DICE is likely to see its EOL soon (and replaced by a sophisticated successor by TCAT), I realize that this is probably not a high priority. However from our perspective it would be sad to see these ALSA Firewire proceedings be lost and not be appreciated by a larger audience. With respect to this: * what would be the customary approach to requesting such a merge? * what are your reservations, resp. are there any existing shortcomings that you are yet unhappy with? * have you received any other (negative/positive) feedback from snd_dice users? I know of the Vortexbox community, which has been offering an optional patch to support Firewire plackback devices; as far as I know, they're happy with it. * if you need any backing or assistance, we'd be happy to help out Keen to hear your feedback on this suggestion. Best regards, Rolf Anderegg (Weiss Engineering Ltd.)

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[alsa-devel] [PATCH (alsa-lib)] pcm_hw: Fix buffer overflow in chmap
by David Henningsson 10 Jun '13

10 Jun '13

We can't calculate memory storage in bytes, when we're supposed to store ints in it! Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/pcm/pcm_hw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 196393d..9ff83a9 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -1187,7 +1187,7 @@ static snd_pcm_chmap_t *snd_pcm_hw_get_chmap(snd_pcm_t *pcm) snd_pcm_state_name(FAST_PCM_STATE(hw))); return NULL; } - map = malloc(pcm->channels + 1); + map = malloc(pcm->channels * sizeof(map->pos[0]) + sizeof(*map)); if (!map) return NULL; map->channels = pcm->channels; -- 1.8.1.2

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[alsa-devel] [PATCH 1/2] ASoC: sgtl5000: Use i2c_get_clientdata()
by Fabio Estevam 10 Jun '13

10 Jun '13

We should use i2c_get_clientdata() to get the codec private structure. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 2e0227b..d441559 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1571,11 +1571,7 @@ disable_clk: static int sgtl5000_i2c_remove(struct i2c_client *client) { - struct sgtl5000_priv *sgtl5000; - sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), - GFP_KERNEL); - if (!sgtl5000) - return -ENOMEM; + struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); clk_disable_unprepare(sgtl5000->mclk); -- 1.8.1.2

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[alsa-devel] [PATCH] ASoC: adav80x: Select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI
by Axel Lin 10 Jun '13

10 Jun '13

This driver is useless if both SPI and I2C are not configured. Thus don't build this driver if both SPI and I2C are not configured. This patch silences below build warning if both SPI and I2C are not configured. CC sound/soc/codecs/adav80x.o sound/soc/codecs/adav80x.c:842:12: warning: 'adav80x_bus_probe' defined but not used [-Wunused-function] sound/soc/codecs/adav80x.c:863:12: warning: 'adav80x_bus_remove' defined but not used [-Wunused-function] Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c90d298..6234900 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,7 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 select SND_SOC_ADAU1373 if I2C - select SND_SOC_ADAV80X + select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C -- 1.8.1.2

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[alsa-devel] ASoC: ssm2518: bug in ssm2518_set_tdm_slot
by Axel Lin 10 Jun '13

10 Jun '13

ASoC: ssm2518: bug in ssm2518_set_tdm_slot The code in ssm2518_set_tdm_slot looks wrong: if (tx_mask == 0 || tx_mask != 0) return -EINVAL; So ssm2518_set_tdm_slot() always returns error.

2 1

[alsa-devel] [PATCH 0/4] ASoC: codecs: some more improvements for adau1701
by Daniel Mack 10 Jun '13

10 Jun '13

Here are some more patches that I worked on while bootstrapping a new audio device featuring an ADAU1701 DSP. Daniel Mack (4): ASoC: codecs: adau1701: factor out firmware reset ASoC: codecs: adau1701: allow configuration of PLL mode pins ASoC: codecs: adau1701: switch to direct regmap API usage ASoC: codecs: adau1701: add support for pin muxing .../devicetree/bindings/sound/adi,adau1701.txt | 17 ++ sound/soc/codecs/adau1701.c | 255 ++++++++++++++++++--- sound/soc/codecs/adau1701.h | 4 + 3 files changed, 238 insertions(+), 38 deletions(-) -- 1.8.1.4

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[alsa-devel] [PATCH] ASoC: wm8962: Restore device state after reset in runtime resume
by Mark Brown 10 Jun '13

10 Jun '13

After the device has been reset we need to repeat the same initialisation we do on probe to make sure that the device is in a known state. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8962.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d56dd86..32c2c47 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3723,6 +3723,21 @@ static int wm8962_runtime_resume(struct device *dev) wm8962_reset(wm8962); + /* SYSCLK defaults to on; make sure it is off so we can safely + * write to registers if the device is declocked. + */ + regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2, + WM8962_SYSCLK_ENA, 0); + + /* Ensure we have soft control over all registers */ + regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2, + WM8962_CLKREG_OVD, WM8962_CLKREG_OVD); + + /* Ensure that the oscillator and PLLs are disabled */ + regmap_update_bits(wm8962->regmap, WM8962_PLL2, + WM8962_OSC_ENA | WM8962_PLL2_ENA | WM8962_PLL3_ENA, + 0); + regcache_sync(wm8962->regmap); return 0; -- 1.7.10.4

4 4

[alsa-devel] [PATCH] ASoC: Add Kconfig and Makefile to support SPEAr audio driver
by Rajeev Kumar 10 Jun '13

10 Jun '13

This patch adds Kconfig and Makefile to support SPEAr Audio driver. Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar(a)st.com> --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/spear/Kconfig | 39 +++++++++++++++++++++++++++++++++++++++ sound/soc/spear/Makefile | 7 +++++++ 4 files changed, 48 insertions(+), 0 deletions(-) create mode 100644 sound/soc/spear/Kconfig create mode 100644 sound/soc/spear/Makefile diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 5da8ca7..c6d98e2 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -47,6 +47,7 @@ source "sound/soc/pxa/Kconfig" source "sound/soc/samsung/Kconfig" source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" +source "sound/soc/spear/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/ux500/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 99f32f7..b86bf5a 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -25,6 +25,7 @@ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += samsung/ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ +obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += ux500/ diff --git a/sound/soc/spear/Kconfig b/sound/soc/spear/Kconfig new file mode 100644 index 0000000..3b7cdad --- /dev/null +++ b/sound/soc/spear/Kconfig @@ -0,0 +1,39 @@ +config SND_SPEAR_EVM + tristate "SoC Audio support for SPEAr EVM" + select SND_DESIGNWARE_I2S + select SND_SOC_STA529 + select SND_SPEAR_SOC + help + Say Y if you want to add support for SoC audio on SPEAr + platform + +config SND_SPEAR1340_EVM + tristate "SoC Audio support for SPEAr1340 EVM" + select SND_DESIGNWARE_I2S + select SND_SOC_STA529 + select SND_SPEAR_SPDIF_OUT + select SND_SPEAR_SPDIF_IN + select SND_SOC_SPDIF + select SND_SPEAR_SOC + help + Say Y if you want to add support for SoC audio on SPEAr1340 + platform + +config SND_SPEAR_SOC + tristate "SoC Audio for the ST chip" + depends on SND_DESIGNWARE_I2S || SND_SPEAR_SPDIF_OUT || \ + SND_SPEAR_SPDIF_IN + help + Say Y or M if you want to add support for any of the audio + controllers (I2S/SPDIF). You will also need to select + the audio interface codecs to support below. + +config SND_SPEAR_SPDIF_OUT + tristate "SPEAr SPDIF Out Device Driver" + help + Say Y or M if you want to add support for SPDIF OUT driver. + +config SND_SPEAR_SPDIF_IN + tristate "SPEAr SPDIF IN Device Driver" + help + Say Y or M if you want to add support for SPDIF IN driver. diff --git a/sound/soc/spear/Makefile b/sound/soc/spear/Makefile new file mode 100644 index 0000000..b365126 --- /dev/null +++ b/sound/soc/spear/Makefile @@ -0,0 +1,7 @@ +# SPEAR Platform Support +obj-$(CONFIG_SND_SPEAR_SOC) += spear_pcm.o +obj-$(CONFIG_SND_SPEAR_SPDIF_IN) += spdif_in.o +obj-$(CONFIG_SND_SPEAR_SPDIF_OUT) += spdif_out.o + +# SPEAR Machine Support +obj-$(CONFIG_SND_SPEAR_EVM) += spear_evb.o -- 1.6.0.2

2 1

Re: [alsa-devel] [PATCH 12/15] ASoC: spear: Convert to use devm_ioremap_resource
by Sachin Kamat 10 Jun '13

10 Jun '13

On 10 June 2013 17:05, Tushar Behera <tushar.behera(a)linaro.org> wrote: > Commit 75096579c3ac ("lib: devres: Introduce devm_ioremap_resource()") > introduced devm_ioremap_resource() and deprecated the use of > devm_request_and_ioremap(). This patch does a little more than just conversion. May that change could be mentioned here. > > Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> > CC: alsa-devel(a)alsa-project.org > CC: Liam Girdwood <lgirdwood(a)gmail.com> > CC: Mark Brown <broonie(a)kernel.org> > --- > sound/soc/spear/spdif_out.c | 20 +++++--------------- > 1 file changed, 5 insertions(+), 15 deletions(-) > > diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c > index 4bde512..896ee7f 100644 > --- a/sound/soc/spear/spdif_out.c > +++ b/sound/soc/spear/spdif_out.c > @@ -282,27 +282,17 @@ static int spdif_out_probe(struct platform_device *pdev) > struct resource *res; > int ret; > > - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); > - if (!res) > - return -EINVAL; > - > - if (!devm_request_mem_region(&pdev->dev, res->start, > - resource_size(res), pdev->name)) { > - dev_warn(&pdev->dev, "Failed to get memory resourse\n"); > - return -ENOENT; > - } > - > host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); > if (!host) { > dev_warn(&pdev->dev, "kzalloc fail\n"); > return -ENOMEM; > } > > - host->io_base = devm_request_and_ioremap(&pdev->dev, res->start, > - resource_size(res)); > - if (!host->io_base) { > - dev_warn(&pdev->dev, "ioremap failed\n"); > - return -ENOMEM; > + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); > + host->io_base = devm_ioremap_resource(&pdev->dev, res); > + if (IS_ERR(host->io_base)) { > + dev_warn(&pdev->dev, "devm_ioremap_resource failed\n"); This print is not needed as devm_ioremap_resource prints its own messages. > + return PTR_ERR(host->io_base); > } > > host->clk = devm_clk_get(&pdev->dev, NULL); > -- > 1.7.9.5 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ -- With warm regards, Sachin

1 0

[alsa-devel] Missing Kconfig and Makefile for SPEAr platform
by Rajeev kumar 10 Jun '13

10 Jun '13

Hello Mark, The Makefile and Kconfig is missing for SPEAr platform. Please find below the link of circulated patch for the same. http://mailman.alsa-project.org/pipermail/alsa-devel/2012-June/052872.html Please update. Thanks and Best Regards Rajeev

2 1

Re: [alsa-devel] [PATCH] ASoC: dmaengine: add runtime status checking in dmaengine_pcm_dma_complete
by Mark Brown 10 Jun '13

10 Jun '13

On Sat, Jun 08, 2013 at 04:34:02PM +0800, Qiao Zhou wrote: > This call back only takes substream as parameter. In case that a > wrong/fake interrupt of other drivers triggers this call back, like > a careless/bad design, then it panics in audio driver. Can we add > such checking and give a warning log rather than to let the system > die directly. Could you consider it? Why not fix the problem in the paths that do the teardown instead?

1 0

[alsa-devel] Codec driver gets probed prior to the machine driver
by Fabio Estevam 10 Jun '13

10 Jun '13

Hi, When running linux-next 20130607 on a mx6qsabrelite I see that the audio drivers fail to probe. Added some printk's to better illustrate the problem: mmcblk0: mmc0:b5a1 SD04G 3.69 GiB **** reading the chip ID sgtl5000: probe of 0-000a failed with error -5 ****** enabling the codec clock imx-sgtl5000 sound.13: ASoC: CODEC (null) not registered imx-sgtl5000 sound.13: snd_soc_register_card failed (-517) platform sound.13: Driver imx-sgtl5000 requests probe defer The problem here is that the codec driver (sgtl5000) is being probed prior to the machine driver (imx-sgtl5000). The reading of the chip ID fails because at this point in time there is no codec clock, which is turned on by imx-sgtl5000. Running 3.10-rc4, the order of the events is as expected: imx-sgtl5000 is called first and sgtl5000 is called later. In this case the probe works as usual. Tried doing a bisect, but this was not succesful due to several bisect breakages (build issues and non-booting kernels) on its way. Would anyone have some suggestion as to why the codec driver is getting probed first? Thanks, Fabio Estevam

3 3

[alsa-devel] Device enumeration
by Robert Billing 10 Jun '13

10 Jun '13

I am currently developing a package to play theatrical sound effects, which has two play channels that I want to be able to attach to two different ALSA devices. To allow the users, who are theatre directors and are *not* technical to do this I am trying to provide a drop-down list of play devices. I can get the raw devices by parsing either the output from aplay -L, or from /proc/asound. The problem comes with devices defined in .asoundrc. I have this in .asoundrc on the development system: pcm.speakers { type pulse device alsa_output.pci-0000_02_06.0.analog-stereo } ctl.speakers { type pulse device alsa_output.pci-0000_02_06.0.analog-stereo } pcm.cans { type pulse device alsa_output.pci-0000_00_1f.5.analog-stereo } ctl.cans { type pulse device alsa_output.pci-0000_00_1f.5.analog-stereo } This works and it is possible to play to the devices "cans" or "speakers" to get output on headphones and speakers respectively, but these names don't appear in the output from aplay. What I need is some clean way of getting a list of the aliases. This package will have to be capable of being installed on any Linux system without needing to edit configuration files manually. However if some tool or some part of the installation process has edited a configuration file I want the aliases to appear in my list of devices. What is the right way to do this? -- Robert Billing, Phone/fax +44 (0) 1344-772849 bob(a)tnglwood.demon.co.uk Tanglewood, 64, Pinehill Road, Crowthorne, Berks, RG45 7JR

1 0

[alsa-devel] [PATCH] ALSA: hda - Headset mic support for two more machines
by David Henningsson 10 Jun '13

10 Jun '13

They need these quirks to have headset mic support. BugLink: https://bugs.launchpad.net/bugs/1189363 Tested-by: Shawn Wang <shawn.wang(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 02e22b4..cbd443d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3483,6 +3483,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05ca, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05de, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05e0, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05ea, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05eb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), @@ -3494,6 +3495,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), -- 1.7.9.5

1 0

[alsa-devel] [PATCH 0/8] [RFC] new driver for Echo Audio's Fireworks based devices
by o-takashi＠sakamocchi.jp 08 Jun '13

08 Jun '13

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> This driver module is originally developed by Clemens Ladish 2010. But it includes duplicate codes which current snd-firewire-lib has and lack of automatical sampling rate/the number of channels arrangement. http://git.alsa-project.org/?p=alsa-kprivate.git;a=shortlog;h=refs/heads/fi… This driver module is based on snd-firewire-lib with my previous patches. [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 [PATCH 0/2] snd-firewire-lib: add MIDI stream support http://mailman.alsa-project.org/pipermail/alsa-devel/2013-June/062610.html Currently this driver module support just PCM/MIDI kernel streaming via ALSA interfaces. So there are some issues about this driver. 1.whether adding each PCM devices for analog and digital interface 2.where the codes for device control like volume, routing and etc are 3.card definition files for ALSA applications like PulseAudio I hope this driver module promote developers to discuss about how to implement Firewire devices which has many channels and options. Takashi Sakamoto (8): add main file of driver module add device specific command add control interfaces add handling AMDTP stream add MIDI interface add PCM interface add proc interface modify Makefile and Kconfig to build this module sound/firewire/Kconfig | 16 + sound/firewire/Makefile | 2 + sound/firewire/fireworks/Makefile | 2 + sound/firewire/fireworks/fireworks.c | 488 +++++++++++++++++++++++ sound/firewire/fireworks/fireworks.h | 241 ++++++++++++ sound/firewire/fireworks/fireworks_command.c | 535 +++++++++++++++++++++++++ sound/firewire/fireworks/fireworks_control.c | 537 ++++++++++++++++++++++++++ sound/firewire/fireworks/fireworks_midi.c | 236 +++++++++++ sound/firewire/fireworks/fireworks_pcm.c | 530 +++++++++++++++++++++++++ sound/firewire/fireworks/fireworks_proc.c | 183 +++++++++ sound/firewire/fireworks/fireworks_stream.c | 107 +++++ 11 files changed, 2877 insertions(+) create mode 100644 sound/firewire/fireworks/Makefile create mode 100644 sound/firewire/fireworks/fireworks.c create mode 100644 sound/firewire/fireworks/fireworks.h create mode 100644 sound/firewire/fireworks/fireworks_command.c create mode 100644 sound/firewire/fireworks/fireworks_control.c create mode 100644 sound/firewire/fireworks/fireworks_midi.c create mode 100644 sound/firewire/fireworks/fireworks_pcm.c create mode 100644 sound/firewire/fireworks/fireworks_proc.c create mode 100644 sound/firewire/fireworks/fireworks_stream.c -- 1.7.10.4

3 18

[alsa-devel] [PATCH] ASoC: sgtl5000: Make device cache only when powered off
by Mark Brown 08 Jun '13

08 Jun '13

When the regulators have been disabled mark the device as cache only so that we don't try to interact with the hardware. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/sgtl5000.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1c3b20f..85a32e6 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -911,10 +911,25 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, if (ret) return ret; udelay(10); + + regcache_cache_only(sgtl5000->regmap, false); + + ret = regcache_sync(sgtl5000->regmap); + if (ret != 0) { + dev_err(codec->dev, + "Failed to restore cache: %d\n", ret); + + regcache_cache_only(sgtl5000->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + + return ret; + } } break; case SND_SOC_BIAS_OFF: + regcache_cache_only(sgtl5000->regmap, true); regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); break; -- 1.7.10.4

3 4

[alsa-devel] [PATCH] ASoC: codecs: tlv320aic3x: Remove deadlock from snd_soc_dapm_put_volsw_aic3x()
by Andreas Irestal 07 Jun '13

07 Jun '13

From: Andreas Irestål <Andreas.Irestal(a)axis.com> When calling snd_soc_dapm_sync(), it eventually tries to lock the same mutex already locked in snd_soc_dapm_put_volsw_aic3x() and a deadlock occurs. By moving the mutex unlock to just before snd_soc_dapm_sync(), this deadlock is prevented. This problem was introduced in Linux 3.5 Signed-off-by: Andreas Irestål <Andreas.Irestal(a)axis.com> --- sound/soc/codecs/tlv320aic3x.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 65d09d6..1514bf8 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -187,14 +187,14 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, break; } - - if (found) - snd_soc_dapm_sync(widget->dapm); } - ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); - mutex_unlock(&widget->codec->mutex); + + if (found) + snd_soc_dapm_sync(widget->dapm); + + ret = snd_soc_update_bits_locked(widget->codec, reg, val_mask, val); return ret; }

2 1

[alsa-devel] [PATCH] ASoC: WM8962: Set CLKREG_OVD if GP5_FN is not its reset value
by Nicolin Chen 07 Jun '13

07 Jun '13

According to WM8962 datasheet, the GP5_FN bit of R516 (GPIO 5) depends on the CLKREG_OVD bit of R8 (Clocking2): "Note that GPIO5 functions are only supported when CLKREG_OVD=1. When CLKREG_OVD=0, the contents of Register R516 must not be changed from the default value." So set CLKREG_OVD bit before regcache_sync() updates the GP5_FN bit to a non-reset value. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/codecs/wm8962.c | 19 +++++++++++++++++++ 1 files changed, 19 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 26219ea..9c8fd11 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3739,6 +3739,23 @@ static int wm8962_i2c_remove(struct i2c_client *client) } #ifdef CONFIG_PM_RUNTIME +static void wm8962_check_gp5_fn(struct snd_soc_codec *codec) +{ + int changed; + + /* + * If GP5_FN is not going to be the reset value 0x0, + * need to set CLKREG_OVD bit of Register CLOCKING2 + * before regcache_sync() updates GP5_FN bit to a + * non-reset value. The reset value should be 0x0. + */ + changed = snd_soc_test_bits(codec, WM8962_GPIO_5, + WM8962_GP5_FN_MASK, 0x0); + if (changed) + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_CLKREG_OVD_MASK, WM8962_CLKREG_OVD); +} + static int wm8962_runtime_resume(struct device *dev) { struct wm8962_priv *wm8962 = dev_get_drvdata(dev); @@ -3756,6 +3773,8 @@ static int wm8962_runtime_resume(struct device *dev) wm8962_reset(wm8962); + wm8962_check_gp5_fn(wm8962->codec); + regcache_sync(wm8962->regmap); return 0; -- 1.7.1

3 5

[alsa-devel] [PATCH] ALSA: get rid of CONFIG_SND_VERBOSE_PRINTK
by Alan Stern 07 Jun '13

07 Jun '13

According to Takashi Iwai, CONFIG_SND_VERBOSE_PRINTK is "useless". All it does is add filenames and line numbers to certain log messages. This patch gets rid of it. Signed-off-by: Alan Stern <stern(a)rowland.harvard.edu> CC: Joe Perches <joe(a)perches.com> --- Updating all the snd_printk() and snd_printd() calls will be a huge job. Getting rid of the verbose logging is a lot easier, so that's where I'm starting. Alan Stern [as1688] Documentation/DocBook/writing-an-alsa-driver.tmpl | 7 +--- arch/arm/configs/eseries_pxa_defconfig | 1 arch/arm/configs/omap2plus_defconfig | 1 arch/arm/configs/s3c2410_defconfig | 1 arch/arm/configs/trizeps4_defconfig | 1 arch/ia64/configs/generic_defconfig | 1 arch/ia64/configs/gensparse_defconfig | 1 arch/mips/configs/pnx8335_stb225_defconfig | 1 arch/powerpc/configs/ppc6xx_defconfig | 1 arch/sh/configs/edosk7760_defconfig | 1 arch/sh/configs/sh7785lcr_32bit_defconfig | 1 include/sound/core.h | 4 +- sound/core/Kconfig | 9 ------ sound/core/misc.c | 33 ---------------------- sound/pci/hda/hda_intel.c | 4 -- 15 files changed, 5 insertions(+), 62 deletions(-) Index: usb-3.10/Documentation/DocBook/writing-an-alsa-driver.tmpl =================================================================== --- usb-3.10.orig/Documentation/DocBook/writing-an-alsa-driver.tmpl +++ usb-3.10/Documentation/DocBook/writing-an-alsa-driver.tmpl @@ -6087,11 +6087,8 @@ struct _snd_pcm_runtime { <title><function>snd_printk()</function> and friends</title> <para> ALSA provides a verbose version of the - <function>printk()</function> function. If a kernel config - <constant>CONFIG_SND_VERBOSE_PRINTK</constant> is set, this - function prints the given message together with the file name - and the line of the caller. The <constant>KERN_XXX</constant> - prefix is processed as + <function>printk()</function> function. + The <constant>KERN_XXX</constant> prefix is processed as well as the original <function>printk()</function> does, so it's recommended to add this prefix, e.g. Index: usb-3.10/arch/arm/configs/eseries_pxa_defconfig =================================================================== --- usb-3.10.orig/arch/arm/configs/eseries_pxa_defconfig +++ usb-3.10/arch/arm/configs/eseries_pxa_defconfig @@ -90,7 +90,6 @@ CONFIG_SND=m CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m CONFIG_SND_DYNAMIC_MINORS=y -CONFIG_SND_VERBOSE_PRINTK=y # CONFIG_SND_PCMCIA is not set CONFIG_SND_SOC=m CONFIG_SND_PXA2XX_SOC=m Index: usb-3.10/arch/arm/configs/omap2plus_defconfig =================================================================== --- usb-3.10.orig/arch/arm/configs/omap2plus_defconfig +++ usb-3.10/arch/arm/configs/omap2plus_defconfig @@ -192,7 +192,6 @@ CONFIG_SOUND=m CONFIG_SND=m CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_SND_USB_AUDIO=m CONFIG_SND_SOC=m Index: usb-3.10/arch/arm/configs/s3c2410_defconfig =================================================================== --- usb-3.10.orig/arch/arm/configs/s3c2410_defconfig +++ usb-3.10/arch/arm/configs/s3c2410_defconfig @@ -315,7 +315,6 @@ CONFIG_SND_SEQUENCER=m CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m CONFIG_SND_SEQUENCER_OSS=y -CONFIG_SND_VERBOSE_PRINTK=y # CONFIG_SND_DRIVERS is not set # CONFIG_SND_ARM is not set # CONFIG_SND_SPI is not set Index: usb-3.10/arch/arm/configs/trizeps4_defconfig =================================================================== --- usb-3.10.orig/arch/arm/configs/trizeps4_defconfig +++ usb-3.10/arch/arm/configs/trizeps4_defconfig @@ -159,7 +159,6 @@ CONFIG_SND=y CONFIG_SND_SEQUENCER=y CONFIG_SND_MIXER_OSS=y CONFIG_SND_PCM_OSS=y -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_SND_PXA2XX_AC97=y CONFIG_SND_USB_AUDIO=m Index: usb-3.10/arch/ia64/configs/generic_defconfig =================================================================== --- usb-3.10.orig/arch/ia64/configs/generic_defconfig +++ usb-3.10/arch/ia64/configs/generic_defconfig @@ -127,7 +127,6 @@ CONFIG_SND_SEQ_DUMMY=m CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m CONFIG_SND_SEQUENCER_OSS=y -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DUMMY=m CONFIG_SND_VIRMIDI=m CONFIG_SND_MTPAV=m Index: usb-3.10/arch/ia64/configs/gensparse_defconfig =================================================================== --- usb-3.10.orig/arch/ia64/configs/gensparse_defconfig +++ usb-3.10/arch/ia64/configs/gensparse_defconfig @@ -115,7 +115,6 @@ CONFIG_SND_SEQ_DUMMY=m CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m CONFIG_SND_SEQUENCER_OSS=y -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DUMMY=m CONFIG_SND_VIRMIDI=m CONFIG_SND_MTPAV=m Index: usb-3.10/arch/mips/configs/pnx8335_stb225_defconfig =================================================================== --- usb-3.10.orig/arch/mips/configs/pnx8335_stb225_defconfig +++ usb-3.10/arch/mips/configs/pnx8335_stb225_defconfig @@ -73,7 +73,6 @@ CONFIG_SND_SEQUENCER=m CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m CONFIG_SND_SEQUENCER_OSS=y -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_EXT2_FS=m # CONFIG_DNOTIFY is not set Index: usb-3.10/arch/powerpc/configs/ppc6xx_defconfig =================================================================== --- usb-3.10.orig/arch/powerpc/configs/ppc6xx_defconfig +++ usb-3.10/arch/powerpc/configs/ppc6xx_defconfig @@ -870,7 +870,6 @@ CONFIG_SND_PCM_OSS=m CONFIG_SND_SEQUENCER_OSS=y CONFIG_SND_DYNAMIC_MINORS=y # CONFIG_SND_SUPPORT_OLD_API is not set -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_SND_DEBUG_VERBOSE=y CONFIG_SND_PCM_XRUN_DEBUG=y Index: usb-3.10/arch/sh/configs/edosk7760_defconfig =================================================================== --- usb-3.10.orig/arch/sh/configs/edosk7760_defconfig +++ usb-3.10/arch/sh/configs/edosk7760_defconfig @@ -90,7 +90,6 @@ CONFIG_SOUND=y CONFIG_SND=y # CONFIG_SND_SUPPORT_OLD_API is not set # CONFIG_SND_VERBOSE_PROCFS is not set -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_SOC=y # CONFIG_HID_SUPPORT is not set # CONFIG_USB_SUPPORT is not set Index: usb-3.10/arch/sh/configs/sh7785lcr_32bit_defconfig =================================================================== --- usb-3.10.orig/arch/sh/configs/sh7785lcr_32bit_defconfig +++ usb-3.10/arch/sh/configs/sh7785lcr_32bit_defconfig @@ -105,7 +105,6 @@ CONFIG_SND_HRTIMER=y CONFIG_SND_DYNAMIC_MINORS=y # CONFIG_SND_SUPPORT_OLD_API is not set # CONFIG_SND_VERBOSE_PROCFS is not set -CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y CONFIG_SND_DEBUG_VERBOSE=y # CONFIG_SND_DRIVERS is not set Index: usb-3.10/include/sound/core.h =================================================================== --- usb-3.10.orig/include/sound/core.h +++ usb-3.10/include/sound/core.h @@ -335,7 +335,7 @@ enum { SND_PR_VERBOSE, }; -#if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK) +#if defined(CONFIG_SND_DEBUG) __printf(4, 5) void __snd_printk(unsigned int level, const char *file, int line, const char *format, ...); @@ -349,7 +349,7 @@ void __snd_printk(unsigned int level, co * @fmt: format string * * Works like printk() but prints the file and the line of the caller - * when configured with CONFIG_SND_VERBOSE_PRINTK. + * when configured with CONFIG_SND_DEBUG. */ #define snd_printk(fmt, args...) \ __snd_printk(0, __FILE__, __LINE__, fmt, ##args) Index: usb-3.10/sound/core/Kconfig =================================================================== --- usb-3.10.orig/sound/core/Kconfig +++ usb-3.10/sound/core/Kconfig @@ -173,15 +173,6 @@ config SND_VERBOSE_PROCFS useful information to developers when a problem occurs). On the other side, it makes the ALSA subsystem larger. -config SND_VERBOSE_PRINTK - bool "Verbose printk" - help - Say Y here to enable verbose log messages. These messages - will help to identify source file and position containing - printed messages. - - You don't need this unless you're debugging ALSA. - config SND_DEBUG bool "Debug" help Index: usb-3.10/sound/core/misc.c =================================================================== --- usb-3.10.orig/sound/core/misc.c +++ usb-3.10/sound/core/misc.c @@ -51,27 +51,11 @@ void release_and_free_resource(struct re EXPORT_SYMBOL(release_and_free_resource); -#ifdef CONFIG_SND_VERBOSE_PRINTK -/* strip the leading path if the given path is absolute */ -static const char *sanity_file_name(const char *path) -{ - if (*path == '/') - return strrchr(path, '/') + 1; - else - return path; -} -#endif - -#if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK) +#if defined(CONFIG_SND_DEBUG) void __snd_printk(unsigned int level, const char *path, int line, const char *format, ...) { va_list args; -#ifdef CONFIG_SND_VERBOSE_PRINTK - int kern_level; - struct va_format vaf; - char verbose_fmt[] = KERN_DEFAULT "ALSA %s:%d %pV"; -#endif #ifdef CONFIG_SND_DEBUG if (debug < level) @@ -79,22 +63,7 @@ void __snd_printk(unsigned int level, co #endif va_start(args, format); -#ifdef CONFIG_SND_VERBOSE_PRINTK - vaf.fmt = format; - vaf.va = &args; - - kern_level = printk_get_level(format); - if (kern_level) { - const char *end_of_header = printk_skip_level(format); - memcpy(verbose_fmt, format, end_of_header - format); - vaf.fmt = end_of_header; - } else if (level) - memcpy(verbose_fmt, KERN_DEBUG, sizeof(KERN_DEBUG) - 1); - printk(verbose_fmt, sanity_file_name(path), line, &vaf); - -#else vprintk(format, args); -#endif va_end(args); } EXPORT_SYMBOL_GPL(__snd_printk); Index: usb-3.10/sound/pci/hda/hda_intel.c =================================================================== --- usb-3.10.orig/sound/pci/hda/hda_intel.c +++ usb-3.10/sound/pci/hda/hda_intel.c @@ -189,11 +189,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}, "{ULI, M5461}}"); MODULE_DESCRIPTION("Intel HDA driver"); -#ifdef CONFIG_SND_VERBOSE_PRINTK -#define SFX /* nop */ -#else #define SFX "hda-intel " -#endif #if defined(CONFIG_PM) && defined(CONFIG_VGA_SWITCHEROO) #ifdef CONFIG_SND_HDA_CODEC_HDMI

4 37

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

07 Jun '13

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

1 0

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

07 Jun '13

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

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

07 Jun '13

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

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

07 Jun '13

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

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