October 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASoC: rt5645: Correct the naming and setting of ADC Boost Volume Control
by Oder Chiou 01 Oct '15

01 Oct '15

Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 6 +++--- sound/soc/codecs/rt5645.h | 16 +++++++++------- 2 files changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 3eb184c..080cc1c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -519,11 +519,11 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = { RT5645_L_VOL_SFT + 1, RT5645_R_VOL_SFT + 1, 63, 0, adc_vol_tlv), /* ADC Boost Volume Control */ - SOC_DOUBLE_TLV("STO1 ADC Boost Gain", RT5645_ADC_BST_VOL1, + SOC_DOUBLE_TLV("ADC Boost Capture Volume", RT5645_ADC_BST_VOL1, RT5645_STO1_ADC_L_BST_SFT, RT5645_STO1_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), - SOC_DOUBLE_TLV("STO2 ADC Boost Gain", RT5645_ADC_BST_VOL1, - RT5645_STO2_ADC_L_BST_SFT, RT5645_STO2_ADC_R_BST_SFT, 3, 0, + SOC_DOUBLE_TLV("Mono ADC Boost Capture Volume", RT5645_ADC_BST_VOL2, + RT5645_MONO_ADC_L_BST_SFT, RT5645_MONO_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), /* I2S2 function select */ diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index f45861c..61bc8ab 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -39,8 +39,8 @@ #define RT5645_STO1_ADC_DIG_VOL 0x1c #define RT5645_MONO_ADC_DIG_VOL 0x1d #define RT5645_ADC_BST_VOL1 0x1e -/* Mixer - D-D */ #define RT5645_ADC_BST_VOL2 0x20 +/* Mixer - D-D */ #define RT5645_STO1_ADC_MIXER 0x27 #define RT5645_MONO_ADC_MIXER 0x28 #define RT5645_AD_DA_MIXER 0x29 @@ -315,12 +315,14 @@ #define RT5645_STO1_ADC_R_BST_SFT 12 #define RT5645_STO1_ADC_COMP_MASK (0x3 << 10) #define RT5645_STO1_ADC_COMP_SFT 10 -#define RT5645_STO2_ADC_L_BST_MASK (0x3 << 8) -#define RT5645_STO2_ADC_L_BST_SFT 8 -#define RT5645_STO2_ADC_R_BST_MASK (0x3 << 6) -#define RT5645_STO2_ADC_R_BST_SFT 6 -#define RT5645_STO2_ADC_COMP_MASK (0x3 << 4) -#define RT5645_STO2_ADC_COMP_SFT 4 + +/* ADC Boost Volume Control (0x20) */ +#define RT5645_MONO_ADC_L_BST_MASK (0x3 << 14) +#define RT5645_MONO_ADC_L_BST_SFT 14 +#define RT5645_MONO_ADC_R_BST_MASK (0x3 << 12) +#define RT5645_MONO_ADC_R_BST_SFT 12 +#define RT5645_MONO_ADC_COMP_MASK (0x3 << 10) +#define RT5645_MONO_ADC_COMP_SFT 10 /* Stereo2 ADC Mixer Control (0x26) */ #define RT5645_STO2_ADC_SRC_MASK (0x1 << 15) -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] Prevent writing reserved registers on tlv320aic3104 CODECs
by Rick Mann 01 Oct '15

01 Oct '15

The current code writes a set of registers that are reserved on the tlc320aic3104. The change skips those registers for that IC. Signed-off-by: Rick Mann <rmann(a)latencyzero.com> --- sound/soc/codecs/tlv320aic3x.c | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index f1c9fffcd3a6..8d893e94e78d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1531,14 +1531,17 @@ static int aic3x_init(struct snd_soc_codec *codec) snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL); snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL); - /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ - snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL); - /* Line2 Line Out default volume, disconnect from Output Mixer */ - snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); + /* On tlv320aic3104, these registers are reserved and must not be written */ + if (aic3x->model != AIC3X_MODEL_3104) { + /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL); + /* Line2 Line Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); + } switch (aic3x->model) { case AIC3X_MODEL_3X: -- 1.9.1

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[alsa-devel] Driver for audio power amplifier ICs?
by Rick Mann 01 Oct '15

01 Oct '15

I've been working on this Beaglebone Black custom audio cape. It has a tlv320aic3104 codec (that I'm still trying to get working properly), and connected to its line outs is a tpa3140d2 analog 2x10W power amplifier. This chip has very little in the way of CPU interface, but it does have /shutdown and /fault lines. I'm wondering how ALSA becomes aware of that chip, in order to enable it when it enables certain routes in the tlv320aic3104, and to inform application software if the /fault line asserts. I don't mind writing a driver, if necessary, I just don't know where such a thing would fit. I also wonder what the device tree entries for the amp might look like. Thanks! -- Rick Mann rmann(a)latencyzero.com

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[alsa-devel] aic31xx_set_bias_level not called
by Matthias Weißer 01 Oct '15

01 Oct '15

Hi I currently try to port an existing alsa driver to a new hardware. The only difference between these to board is the used codec. Where the original board (iMX6SX SDB from Freescale) uses a wm8962 our custom board uses a tlv320aic3110. So I took the file sound\soc\fsl\imx-wm8962.c and changed it according to the new codec. From what I understand the main things where just renaming. I then setup my device tree according to the new environment and the kernel seemed to setup stuff correctly. But when I try to play some sound using speaker-test I see some errors and don't hear anything. Tracking stuff down using dev_dbg outputs I see that aic31xx_set_bias_level is never called with a value > 1 which causes that the clocks of the codec are never enabled. Maybe someone can give me some hints on what can be wrong. Here are some things which might be useful: ~ # dmesg | grep -i -A 2 ALSA [ 5.058905] ALSA device list: [ 5.061885] #0: tlv320aic31xx-audio ~ # dmesg | grep tlv [ 1.598495] tlv320aic31xx-codec 2-0018: ## aic31xx_i2c_probe: tlv320aic3110 codec_type = 1 [ 1.613239] tlv320aic31xx-codec 2-0018: ## aic31xx_codec_probe [ 1.621895] tlv320aic31xx-codec 2-0018: ## aic31xx_set_dai_fmt: fmt = 0x1001 [ 1.630390] imx-tlv320aic31xx sound: tlv320aic31xx-hifi <-> 202c000.ssi mapping ok [ 1.638145] tlv320aic31xx-codec 2-0018: ## aic31xx_set_dai_sysclk: clk_id = 0, freq = 24000000, dir = 0 [ 1.663566] tlv320aic31xx-codec 2-0018: ## aic31xx_set_bias_level: 0 -> 1 [ 5.061885] #0: tlv320aic31xx-audio [ 36.860404] tlv320aic31xx-codec 2-0018: ## aic31xx_hw_params: width 16 rate 48000 [ 36.886777] tlv320aic31xx-codec 2-0018: pll 8.1920/2 dosr 128 n 8 m 2 aosr 128 n 8 m 2 bclk_n 16 # aplay -L null Discard all samples (playback) or generate zero samples (capture) default:CARD=tlv320aic31xxau tlv320aic31xx-audio, Default Audio Device sysdefault:CARD=tlv320aic31xxau tlv320aic31xx-audio, Default Audio Device # speaker-test -Ddefault speaker-test 1.0.29 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 128 to 32768 Period size range from 64 to 16384 Using max buffer size 32768 Periods = 4 was set period_size = 8192 was set buffer_size = 32768 0 - Front Left Write error: -5,Input/output error xrun_recovery failed: -5,Input/output error Transfer failed: Input/output error # uname -a Linux imx6sx-g343-devel 4.1.4-dirty #49 Thu Oct 1 10:39:44 CEST 2015 armv7l GNU/Linux /* * Copyright 2013 Freescale Semiconductor, Inc. * * Based on imx-wm8962.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 */ #define DEBUG #include <linux/module.h> #include <linux/of_platform.h> #include <linux/i2c.h> #include <linux/slab.h> #include <linux/clk.h> #include <sound/soc.h> #include <sound/pcm_params.h> #include <sound/soc-dapm.h> #include <linux/pinctrl/consumer.h> #include "../codecs/tlv320aic31xx.h" #include "imx-audmux.h" #define DAI_NAME_SIZE 32 struct imx_tlv320aic31xx_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 { struct platform_device *pdev; }; static struct imx_priv card_priv; static const struct snd_soc_dapm_widget imx_tlv320aic31xx_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_tlv320aic31xx_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_tlv320aic31xx_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; int ret = 0; ret = snd_soc_dai_set_sysclk(codec_dai, 0, data->clk_frequency, SND_SOC_CLOCK_IN); if (ret < 0) dev_err(dev, "failed to set sysclk in %s\n", __func__); return ret; } static int imx_tlv320aic31xx_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_tlv320aic31xx_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; } 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->dev.driver) { dev_err(&pdev->dev, "failed to find codec platform device\n"); ret = -EINVAL; goto fail; } 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 = "tlv320aic31xx-hifi"; 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.owner = THIS_MODULE; data->card.dai_link = &data->dai; data->card.dapm_widgets = imx_tlv320aic31xx_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_tlv320aic31xx_dapm_widgets); data->card.late_probe = imx_tlv320aic31xx_late_probe; platform_set_drvdata(pdev, &data->card); snd_soc_card_set_drvdata(&data->card, data); ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto clk_fail; } of_node_put(ssi_np); of_node_put(codec_np); return 0; clk_fail: clk_disable_unprepare(data->codec_clk); fail: of_node_put(ssi_np); of_node_put(codec_np); return ret; } static int imx_tlv320aic31xx_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); struct imx_tlv320aic31xx_data *data = snd_soc_card_get_drvdata(card); if (!IS_ERR(data->codec_clk)) clk_disable_unprepare(data->codec_clk); return 0; } static const struct of_device_id imx_tlv320aic31xx_dt_ids[] = { { .compatible = "fsl,imx-audio-tlv320aic31xx", }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, imx_tlv320aic31xx_dt_ids); static struct platform_driver imx_tlv320aic31xx_driver = { .driver = { .name = "imx-tlv320aic31xx", .pm = &snd_soc_pm_ops, .of_match_table = imx_tlv320aic31xx_dt_ids, }, .probe = imx_tlv320aic31xx_probe, .remove = imx_tlv320aic31xx_remove, }; module_platform_driver(imx_tlv320aic31xx_driver); MODULE_AUTHOR("Freescale Semiconductor, Inc."); MODULE_DESCRIPTION("Freescale i.MX tlv320aic31xx ASoC machine driver"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:imx-tlv320aic31xx");

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