March 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v5] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Zidan Wang 15 Mar '16

15 Mar '16

This is an initial imx-wm8958 device-tree-only machine driver. The sound card driver support three dai link, HIFI, VOICE and BT. We can configure the cpu dai from device tree, if present it will create corresponding sound card subdevice. So at least one cpu dai phandle should be set from device tree. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v4->v5: support set dai link name and stream name from device tree. .../devicetree/bindings/sound/imx-audio-wm8958.txt | 69 +++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8958.c | 328 +++++++++++++++++++++ 4 files changed, 413 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt create mode 100644 sound/soc/fsl/imx-wm8958.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt new file mode 100644 index 0000000..7978c62 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt @@ -0,0 +1,69 @@ +Freescale i.MX audio complex with WM8958 codec + +Required properties: + + - compatible : "fsl,imx-audio-wm8958" + + - model : The user-visible name of this sound complex + + - audio-codec : The phandle of the WM8958 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, WM8958 pins, and the jacks on the board: + + Power supplies: + * MICBIAS1 + * MICBIAS2 + + Board connectors: + * Headphone Jack + * Ext Spk + +Optional properties: + + - cpu-dai1 : The phandle of CPU DAI1 controller, which is for + aif1 of wm8958. At least one cpu dai is required. + + - cpu-dai2 : The phandle of CPU DAI2 controller, which is for + aif2 of wm8958. At least one cpu dai is required. + + - cpu-dai3 : The phandle of CPU DAI3 controller, which is for + aif3 of wm8958. At least one cpu dai is required. + + - dai-link1-name : If present, set the dai link1 name and stream name + to the specific strings. Otherwise, set the dai + link1 name and stream name to "AIF1-DAI". + + - dai-link2-name : If present, set the dai link2 name and stream name + to the specific strings. Otherwise, set the dai + link2 name and stream name to "AIF2-DAI". + + - dai-link3-name : If present, set the dai link3 name and stream name + to the specific strings. Otherwise, set the dai + link3 name and stream name to "AIF3-DAI". + + - fsl,cpu-dai1-master : If present, cpu dai1 works as master, and will + provide bit clock and frame clock. Otherwise, + cpu dai1 works as slave. + +Example: + +sound { + compatible = "fsl,imx6ul-ddr3-arm2-wm8958", + "fsl,imx-audio-wm8958"; + model = "wm8958-audio"; + cpu-dai1 = <&sai1>; + dai-link1-name = "Hifi"; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUT1L", + "Headphone Jack", "HPOUT1R", + "Ext Spk", "SPKOUTLP", + "Ext Spk", "SPKOUTLN", + "Ext Spk", "SPKOUTRP", + "Ext Spk", "SPKOUTRN", + "IN1LN", "MICBIAS2"; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 35aabf9..f6e5d96 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -229,6 +229,20 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8958 + tristate "SoC Audio support for i.MX boards with wm8958" + depends on OF && I2C + select MFD_WM8994 + select SND_SOC_WM8994 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SAI + select SND_SOC_FSL_UTILS + select SND_KCTL_JACK + help + SoC Audio support for i.MX boards with WM8958 + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8958 codec. + config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C && INPUT diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index d28dc25..2a781c0 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -54,6 +54,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8958-objs := imx-wm8958.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8958) += snd-soc-imx-wm8958.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8958.c b/sound/soc/fsl/imx-wm8958.c new file mode 100644 index 0000000..650b0c8 --- /dev/null +++ b/sound/soc/fsl/imx-wm8958.c @@ -0,0 +1,328 @@ +/* + * Copyright (C) 2015-2016 Freescale Semiconductor, Inc. + * + * 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_platform.h> +#include <linux/i2c.h> +#include <linux/of_gpio.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/control.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" + +#define DAI_NAME_SIZE 32 + +#define DAI_LINK_NUM (3) +#define AIF1_DAI (0) +#define AIF2_DAI (1) +#define AIF3_DAI (2) + +#define WM8958_MCLK_MAX (2) + +#define WM8994_FLL(id) (id == AIF1_DAI ? WM8994_FLL1 : WM8994_FLL2) +#define WM8994_SYSCLK_FLL(id) (id == AIF1_DAI ? WM8994_SYSCLK_FLL1 : WM8994_SYSCLK_FLL2) +#define WM8994_FLL_SRC_MCLK(id) (id == AIF1_DAI ? WM8994_FLL_SRC_MCLK1 : WM8994_FLL_SRC_MCLK2) + +struct imx_wm8958_data { + struct snd_soc_dai_link dai_link[DAI_LINK_NUM]; + struct snd_soc_card card; + unsigned long mclk_freq[WM8958_MCLK_MAX]; + bool is_cpu_dai1_master; + bool is_stream_in_use[DAI_LINK_NUM][2]; +}; + +static const struct snd_soc_dapm_widget imx_wm8958_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static int imx_wm8958_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_card *card = rtd->card; + struct device *dev = card->dev; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool hifi_dai_sysclk_dir = SND_SOC_CLOCK_OUT; + u32 mclk_id, id = codec_dai->id - 1; + u32 pll_out; + int ret; + + data->is_stream_in_use[id][tx] = true; + + if (data->mclk_freq[id]) + mclk_id = WM8994_FLL_SRC_MCLK(id); + else if (id == AIF1_DAI) + mclk_id = WM8994_FLL_SRC_MCLK2; + else + mclk_id = WM8994_FLL_SRC_MCLK1; + + if (id == AIF1_DAI) { + if (data->is_cpu_dai1_master) + hifi_dai_sysclk_dir = SND_SOC_CLOCK_IN; + + ret = snd_soc_dai_set_sysclk(cpu_dai, + 0, 0, !hifi_dai_sysclk_dir); + + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + + if (data->is_cpu_dai1_master) { + ret = snd_soc_dai_set_sysclk(codec_dai, mclk_id, + data->mclk_freq[mclk_id - 1], + hifi_dai_sysclk_dir); + if (ret) { + dev_err(dev, + "failed to set codec sysclk: %d\n", + ret); + return ret; + } + + return 0; + } + } + + if (params_width(params) == 24) + pll_out = params_rate(params) * 384; + else + pll_out = params_rate(params) * 256; + + ret = snd_soc_dai_set_pll(codec_dai, + WM8994_FLL(id), + mclk_id, + data->mclk_freq[mclk_id - 1], + pll_out); + if (ret) { + dev_err(dev, "failed to set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8994_SYSCLK_FLL(id), + pll_out, + SND_SOC_CLOCK_OUT); + if (ret) { + dev_err(dev, "failed to set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_wm8958_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = rtd->card; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int id = codec_dai->id - 1; + + data->is_stream_in_use[id][tx] = false; + + if (id == AIF1_DAI && data->is_cpu_dai1_master) + return 0; + + if (!data->is_stream_in_use[id][!tx]) { + /* + * We should connect AIFxCLK source to FLL after enable FLL, + * and disconnect AIF1CLK source to FLL before disable FLL, + * otherwise FLL worked abnormal. + */ + snd_soc_dai_set_sysclk(codec_dai, WM8994_FLL_SRC_MCLK(id), + data->mclk_freq[id], SND_SOC_CLOCK_OUT); + + /* Disable FLL1 after all stream finished. */ + snd_soc_dai_set_pll(codec_dai, WM8994_FLL(id), 0, 0, 0); + } + + return 0; +} + +static struct snd_soc_ops imx_wm8958_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_dai_link imx_wm8958_dai_link[] = { + [AIF1_DAI] = { + .name = "AIF1-DAI", + .stream_name = "AIF1-DAI", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif1", + .ops = &imx_wm8958_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF, + }, + [AIF2_DAI] = { + .name = "AIF2-DAI", + .stream_name = "AIF2-DAI", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif2", + .ops = &imx_wm8958_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + }, + [AIF3_DAI] = { + .name = "AIF3-DAI", + .stream_name = "AIF3-DAI", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif3", + }, +}; + +static int imx_wm8958_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np[DAI_LINK_NUM], *codec_np; + struct device_node *np = pdev->dev.of_node; + struct platform_device *cpu_pdev[DAI_LINK_NUM]; + struct i2c_client *codec_dev; + struct imx_wm8958_data *data; + struct clk *mclk; + char tmp[8], *name; + int num_links = 0; + int ret, i; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + for (i = 0; i < DAI_LINK_NUM; i++) { + sprintf(tmp, "cpu-dai%d", i + 1); + cpu_np[i] = of_parse_phandle(np, tmp, 0); + if (!cpu_np[i]) + continue; + + cpu_pdev[i] = of_find_device_by_node(cpu_np[i]); + if (!cpu_pdev[i]) { + dev_err(&pdev->dev, + "failed to get cpu dai%d platform device\n", i); + ret = -EINVAL; + goto fail; + } + data->dai_link[num_links] = imx_wm8958_dai_link[i]; + data->dai_link[num_links].cpu_dai_name = dev_name(&cpu_pdev[i]->dev); + data->dai_link[num_links].platform_of_node = cpu_np[i]; + + sprintf(tmp, "dai-link%d-name", i + 1); + if(!of_property_read_string(np, tmp, (const char **)&name)) { + data->dai_link[num_links].name = name; + data->dai_link[num_links].stream_name = name; + } + num_links++; + } + + if (!num_links) { + dev_err(&pdev->dev, "cpu dai phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\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; + } + + /* cpu dai1 link support cpu dai master and slave mode */ + if (of_property_read_bool(np, "fsl,cpu-dai1-master")) { + data->is_cpu_dai1_master = true; + data->dai_link[AIF1_DAI].dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } else { + data->is_cpu_dai1_master = false; + data->dai_link[AIF1_DAI].dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } + + for (i = 0; i < WM8958_MCLK_MAX; i++) { + sprintf(tmp, "MCLK%d", i + 1); + mclk = devm_clk_get(&codec_dev->dev, tmp); + if (!IS_ERR(mclk)) + data->mclk_freq[i] = clk_get_rate(mclk); + } + + if (!data->mclk_freq[0] && !data->mclk_freq[1]) { + dev_err(&pdev->dev, "failed to get mclk clock\n"); + ret = -EINVAL; + goto fail; + } + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + + data->card.num_links = num_links; + data->card.dai_link = data->dai_link; + data->card.dapm_widgets = imx_wm8958_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8958_dapm_widgets); + data->card.owner = THIS_MODULE; + + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto fail; + + 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 fail; + } + +fail: + of_node_put(cpu_np[AIF1_DAI]); + of_node_put(cpu_np[AIF2_DAI]); + of_node_put(cpu_np[AIF3_DAI]); + of_node_put(codec_np); + + return ret; +} + +static const struct of_device_id imx_wm8958_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8958", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8958_dt_ids); + +static struct platform_driver imx_wm8958_driver = { + .driver = { + .name = "imx-wm8958", + .pm = &snd_soc_pm_ops, + .of_match_table = imx_wm8958_dt_ids, + }, + .probe = imx_wm8958_probe, +}; +module_platform_driver(imx_wm8958_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8958 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8958"); -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda - Fix unconditional GPIO toggle via automute
by Takashi Iwai 15 Mar '16

15 Mar '16

Cirrus HD-audio driver may adjust GPIO pins for EAPD dynamically depending on the jack plug state. This works fine for the auto-mute mode where the speaker gets muted upon the HP jack plug. OTOH, when the auto-mute mode is off, this turns off the EAPD unexpectedly depending on the jack state, which results in the silent speaker output. This patch fixes the silent speaker output issue by setting GPIO bits constantly when the auto-mute mode is off. Reported-and-tested-by: moosotc(a)gmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_cirrus.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index c1c855a6c0af..a47e8ae0eb30 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -174,8 +174,12 @@ static void cs_automute(struct hda_codec *codec) snd_hda_gen_update_outputs(codec); if (spec->gpio_eapd_hp || spec->gpio_eapd_speaker) { - spec->gpio_data = spec->gen.hp_jack_present ? - spec->gpio_eapd_hp : spec->gpio_eapd_speaker; + if (spec->gen.automute_speaker) + spec->gpio_data = spec->gen.hp_jack_present ? + spec->gpio_eapd_hp : spec->gpio_eapd_speaker; + else + spec->gpio_data = + spec->gpio_eapd_hp | spec->gpio_eapd_speaker; snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, spec->gpio_data); } -- 2.7.3

1 0

[alsa-devel] No audio output via built-in speaker on a MacMini while headphone jack is plugged in
by moosotc＠gmail.com 15 Mar '16

15 Mar '16

`amixer set Speaker unmute' makes no difference physically unplugging the headphone jack from the socket does, i.e. `amixer set Speaker (un)mute' does the right thing. -- mailto:moosotc@gmail.com

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[alsa-devel] [PATCH] [media] sound/usb: fix NULL dereference in usb_audio_probe()
by Nicolai Stange 15 Mar '16

15 Mar '16

With commit aebb2b89bff0 ("[media] sound/usb: Use Media Controller API to share media resources") an access to quirk->media_device without checking for quirk != NULL has been introduced in usb_audio_probe(). With a Plantronics USB headset (device ID 0x047f:0xc010) attached, this results in the following splat at boot time: BUG: unable to handle kernel NULL pointer dereference at 0000000000000014 IP: [<ffffffffa089aa6c>] usb_audio_probe+0x2cc/0x9a0 [snd_usb_audio] Oops: 0000 [#1] SMP [...] CPU: 2 PID: 696 Comm: systemd-udevd Not tainted 4.5.0-next-20160315 #13 Hardware name: Dell Inc. Latitude E6540/0725FP, BIOS A10 06/26/2014 task: ffff88021c88d7c0 ti: ffff88003d5b0000 task.ti: ffff88003d5b0000 RIP: 0010:[<ffffffffa089aa6c>] [<ffffffffa089aa6c>] usb_audio_probe+0x2cc/0x9a0 [snd_usb_audio] [...] Call Trace: [<ffffffff815a8e16>] usb_probe_interface+0x136/0x2d0 [<ffffffff81509edc>] driver_probe_device+0x22c/0x440 [<ffffffff8150a1c1>] __driver_attach+0xd1/0xf0 [<ffffffff8150a0f0>] ? driver_probe_device+0x440/0x440 [<ffffffff815077ec>] bus_for_each_dev+0x6c/0xc0 [<ffffffff815095ce>] driver_attach+0x1e/0x20 [<ffffffff81509013>] bus_add_driver+0x1c3/0x280 [<ffffffff8150ab10>] driver_register+0x60/0xe0 [<ffffffff815a7711>] usb_register_driver+0x81/0x140 [<ffffffffa08c7000>] ? 0xffffffffa08c7000 [<ffffffffa08c701e>] usb_audio_driver_init+0x1e/0x1000 [snd_usb_audio] [<ffffffff81002123>] do_one_initcall+0xb3/0x1f0 [<ffffffff811fb091>] ? __vunmap+0x81/0xd0 [<ffffffff8121b8d2>] ? kmem_cache_alloc_trace+0x182/0x1d0 [<ffffffff811b0267>] ? do_init_module+0x27/0x1d8 [<ffffffff811b029f>] do_init_module+0x5f/0x1d8 [<ffffffff8112ce35>] load_module+0x1fe5/0x27a0 [<ffffffff81129870>] ? __symbol_put+0x60/0x60 [<ffffffff81241690>] ? vfs_read+0x110/0x130 [<ffffffff8112d866>] SYSC_finit_module+0xe6/0x120 [<ffffffff8112d8be>] SyS_finit_module+0xe/0x10 [<ffffffff81003d94>] do_syscall_64+0x64/0x110 [<ffffffff817c0b61>] entry_SYSCALL64_slow_path+0x25/0x25 After encountering this, the system-udevd process seems to be blocked until it is killed when hitting its timeout of 3min. In analogy to the other accesses to members of quirk in usb_audio_probe(), check for quirk != NULL before accessing its ->media_device. Fixes: aebb2b89bff0 ("[media] sound/usb: Use Media Controller API to share media resources") Signed-off-by: Nicolai Stange <nicstange(a)gmail.com> --- Applicable to linux-next-20160315. sound/usb/card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 63244bb..479621e 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -612,7 +612,7 @@ static int usb_audio_probe(struct usb_interface *intf, if (err < 0) goto __error; - if (quirk->media_device) { + if (quirk && quirk->media_device) { /* don't want to fail when media_snd_device_create() fails */ media_snd_device_create(chip, intf); } -- 2.7.2

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Re: [alsa-devel] ASoC: TLV320AIC3x: Adding additional functionality for 3106 with [Patch] for discuss
by Peter Ujfalusi 15 Mar '16

15 Mar '16

On 03/14/2016 04:57 PM, Timur Karaldin wrote: > Hi, Peter! > What's about placing capture controls in the Playback section of Alsamixer > view? Yeah, it is annoying. They can be fixed by: "Line1L Switch" -> "Line1L Capture Switch" and so on. On the other hand we had these unfortunate control names I think since the beginning and it is kind of an ABI for existing products. I don't have issue changing them, but existing setups relying on the control names might break... > Does resetting of some registers during start/stop of recording > connect with it? Not sure what you mean. -- Péter > > 14.03.2016 17:47, Peter Ujfalusi пишет: >> Hi Timur, >> >> On 03/14/16 13:06, Timur Karaldin wrote: >>> Hi, colleagues! >>> >>> I'm newbie with ALSA devel. I need to get extra functionality from codec >>> through ALSA mixer. >>> Task: 1. making possibility to set-up gain for each input (XXX_2_LADCCTRL and >>> XXX_2_RADCCTRL registers) >> This part is a bit tricky as in the registers: >> value 0..8 is gain from 0dB to -12dB in steps of -1.5dB and value 0xf means >> basically mute/disconnect. >> >>> 2. making possibility to set-up debounce for button and headset detection ( >>> HEADSET_DETECT_CTRL_A register) >>> 3. making possibility to enable high-power AC-coupled output >>> (HEADSET_DETECT_CTRL_A register) >>> 4. making possibility to enable stereo pseudo differential output >>> (HEADSET_DETECT_CTRL_B register) >>> 5. enable headset detection (HEADSET_DETECT_CTRL_A) >>> 6. getting headset detect status (HEADSET_DETECT_CTRL_A) >>> >>> Well, my patch works in generally, but I don't like it very much, because: >>> 1. All controls for input amplifiers' gain are shown in Alsamixer-Playback >>> section. But not only mine, capture controls from base codec's version are >>> shown in playback section too, for example (Right or Left) PGA Mixer Line1(L >>> or R)/2(L or R)/MIC3(L or R), which is definitely capture settings. Should I >>> worried about it? because, as I understand, it could not be setted correctly >>> during changing playback/capture mode for power-saving reason. >>> 2. "Right PGA Mixer Line1R" control (and similar) using the same bits in >>> LINE1R_2_RADCCTRL register as my "Line1R RADC Gain" control (and similar) and >>> put there b"1111" when user switch disconnect it from ADC. So my control is >>> not shown when user disconnect it from ADC. >>> 3. When I start/stop arecord some of my options were reset (debounces, headset >>> enable, high-power AC-coupled, pseudo differential output). When I set up it >>> in shell and then start arecord, it's reset again. Is it the reason because of >>> displaying controls in playback mixer section? >>> >>> I will be very pleased for any advise, thank you! >>> >>> >>> here is my patch for tlv320aic3x.c >>> --- >>> sound/soc/codecs/tlv320aic3x.c | 55 >>> +++++++++++++++++++++++++++++++++++++++--- >>> 1 file changed, 52 insertions(+), 3 deletions(-) >>> >>> diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c >>> index d7349bc..27d031e 100644 >>> --- a/sound/soc/codecs/tlv320aic3x.c >>> +++ b/sound/soc/codecs/tlv320aic3x.c >>> @@ -122,11 +122,24 @@ static const struct reg_default aic3x_reg[] = { >>> { 108, 0x00 }, { 109, 0x00 }, >>> }; >>> >>> +static bool aic3106_volatile(struct device *dev, unsigned int reg) >>> +{ >>> + switch(reg) >>> + { >>> + case AIC3X_HEADSET_DETECT_CTRL_A: >>> + case AIC3X_HEADSET_DETECT_CTRL_B: >>> + return true; >>> + default: >>> + return false; >>> + } >>> +} >>> + >>> static const struct regmap_config aic3x_regmap = { >>> .reg_bits = 8, >>> .val_bits = 8, >>> >>> .max_register = DAC_ICC_ADJ, >>> + .volatile_reg = aic3106_volatile, >>> .reg_defaults = aic3x_reg, >>> .num_reg_defaults = ARRAY_SIZE(aic3x_reg), >>> .cache_type = REGCACHE_RBTREE, >>> @@ -196,7 +209,7 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, >>> struct snd_kcontrol *kcontrol, int event) >>> { >>> struct snd_soc_codec *codec = w->codec; >>> - struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); >>> + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); >> You have couple of these nop changes in the patch... >> >>> switch (event) { >>> case SND_SOC_DAPM_POST_PMU: >>> @@ -270,6 +283,26 @@ static const struct soc_enum aic3x_agc_decay_enum[] = { >>> SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), >>> }; >>> >>> +static const char *aic3x_adc_gain[] = { "0dB", "-1.5dB", "-3dB", "-4.5dB", >>> "-6dB", "-7.5dB", "-9dB", "-10.5dB", "-12dB" }; >>> +static const struct soc_enum aic3x_adc_gain_enum[] = { >>> + SOC_ENUM_SINGLE(MIC3LR_2_LADC_CTRL, 4, 9, aic3x_adc_gain), >>> + SOC_ENUM_SINGLE(MIC3LR_2_RADC_CTRL, 0, 9, aic3x_adc_gain), >>> + SOC_ENUM_SINGLE(LINE1L_2_LADC_CTRL, 3, 9, aic3x_adc_gain), >>> + SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 3, 9, aic3x_adc_gain), >>> + SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 3, 9, aic3x_adc_gain), >>> + SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 3, 9, aic3x_adc_gain), >>> +}; >> So the issue is that we have the DAPM switches controlling exactly the same >> registers. I believe if you would set the gain in a way that bit0 is 1, then >> DAPM will think that the path is disconnected. >> Also if you would set the gain and then mute and unmute the path you would >> have lost the gain you wanted to have.. >> >> The only way I can think of implementing these mixers is to have two sets of >> custom callbacks. one set is to set/get the gain and the other is to set/get >> the mute/disconnect on these. >> When the path is disconnected you should not write the gain change to the >> chip, but cache it and if the path is unmuted, you write the cached gain. When >> you mute the path you should take the set gain first, cache it, then >> disconnect the path. >> >> For these gains you should have DECLARE_TLV_DB_SCALE() and use >> SOC_SINGLE_TLV(). Make sure that the control name matches with the >> corresponding DAPM widget's name so ALSA can match them correctly. >> >>> +static const char *aic3x_headset_debounce[] = { "16ms", "32ms", "64ms", >>> "128ms", "256ms", "512ms" }; >>> +static const struct soc_enum aic3x_headset_debounce_enum[] = { >>> + SOC_ENUM_SINGLE(AIC3X_HEADSET_DETECT_CTRL_A, 2, 6, >>> aic3x_headset_debounce), >>> +}; >>> +static const char *aic3x_button_debounce[] = { "0ms", "8ms", "16ms", >>> "32ms" }; >>> +static const struct soc_enum aic3x_button_debounce_enum[] = { >>> + SOC_ENUM_SINGLE(AIC3X_HEADSET_DETECT_CTRL_A, 0, 4, >>> aic3x_button_debounce), >>> +}; >>> + >>> + >>> + >>> /* >>> * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps >>> */ >>> @@ -399,6 +432,19 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] >>> = { >>> SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), >>> >>> SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), >>> + /* Additional controls */ >>> + SOC_ENUM("Mic3L LADC Gain", aic3x_adc_gain_enum[0]), >>> + SOC_ENUM("Mic3R RADC Gain", aic3x_adc_gain_enum[1]), >>> + SOC_ENUM("Line1L LADC Gain", aic3x_adc_gain_enum[2]), >>> + SOC_ENUM("Line1R RADC Gain", aic3x_adc_gain_enum[3]), >>> + SOC_ENUM("Line2L LADC Gain", aic3x_adc_gain_enum[4]), >>> + SOC_ENUM("Line2R RADC Gain", aic3x_adc_gain_enum[5]), >> Please avoid using arrays as the driver no longer use these anymore. >> >>> + SOC_ENUM("Headset Debounce", aic3x_headset_debounce_enum), >>> + SOC_ENUM("Button Debounce", aic3x_button_debounce_enum), >>> + SOC_SINGLE("Headset Detected", AIC3X_HEADSET_DETECT_CTRL_A, >>> AIC3X_HEADSET_DETECT_SHIFT, AIC3X_HEADSET_DETECT_MASK, 0), >>> + SOC_SINGLE("Headset Detect Enable", AIC3X_HEADSET_DETECT_CTRL_A, 7, 1, >>> 0), >>> + SOC_SINGLE("High power output Ac-coupled", AIC3X_HEADSET_DETECT_CTRL_B, >>> 7, 1, 0), >>> + SOC_SINGLE("Stereo pseudodifferential output", >>> AIC3X_HEADSET_DETECT_CTRL_B, 3, 1, 0), >>> }; >>> >>> static const struct snd_kcontrol_new aic3x_mono_controls[] = { >>> @@ -614,7 +660,7 @@ static const struct snd_soc_dapm_widget >>> aic3x_dapm_widgets[] = { >>> SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 32", >>> AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0), >>> >>> - /* Mic Bias */ >>> + /* Mic Bias */ >> nop change >> >>> SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0, >>> mic_bias_event, >>> SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), >>> @@ -692,7 +738,7 @@ static const struct snd_soc_dapm_route intercon[] = { >>> {"Left Line2L Mux", "single-ended", "LINE2L"}, >>> {"Left Line2L Mux", "differential", "LINE2L"}, >>> >>> - {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"}, >>> + {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"}, >> nop change >> >>> {"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"}, >>> {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"}, >>> {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, >>> @@ -814,6 +860,7 @@ static const struct snd_soc_dapm_route intercon[] = { >>> {"Right HPCOM Mux", "external feedback", "Right HPCOM Mixer"}, >>> {"Right HP Com", NULL, "Right HPCOM Mux"}, >>> {"HPRCOM", NULL, "Right HP Com"}, >>> + >> why? >> >>> }; >>> >>> static const struct snd_soc_dapm_route intercon_mono[] = { >>> @@ -918,7 +965,9 @@ static int aic3x_hw_params(struct snd_pcm_substream >>> *substream, >>> data = (LDAC2LCH | RDAC2RCH); >>> data |= (fsref == 44100) ? FSREF_44100 : FSREF_48000; >>> if (params_rate(params) >= 64000) >>> + { >>> data |= DUAL_RATE_MODE; >>> + } >> why? >> >>> snd_soc_write(codec, AIC3X_CODEC_DATAPATH_REG, data); >>> >>> /* codec sample rate select */ >>> -- >>> and tlv320aic3x.h >>> --- >>> sound/soc/codecs/tlv320aic3x.h | 6 ++++++ >>> 1 file changed, 6 insertions(+) >>> >>> diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h >>> index e521ac3..8ff42d2 100644 >>> --- a/sound/soc/codecs/tlv320aic3x.h >>> +++ b/sound/soc/codecs/tlv320aic3x.h >>> @@ -271,6 +271,12 @@ enum { >>> AIC3X_BUTTON_DEBOUNCE_32MS = 3 >>> }; >>> >>> +typedef struct { >>> + struct platform_device* pdev; >>> + struct proc_dir_entry *proc_value; >>> + struct snd_soc_codec *codec; >>> +} aic3106_detect_t; >>> + >>> #define AIC3X_HEADSET_DETECT_ENABLED 0x80 >>> #define AIC3X_HEADSET_DETECT_SHIFT 5 >>> #define AIC3X_HEADSET_DETECT_MASK 3 >>> -- >> >> When submitting please separate the patches by functionality. It is easier to >> review the changes. >> > >

2 2

[alsa-devel] [PATCH v3] ASoC: cs35l32: avoid uninitialized variable access
by Arnd Bergmann 15 Mar '16

15 Mar '16

gcc warns about the possibilty of accessing a property read from devicetree in cs35l32_i2c_probe() when it has not been initialized because CONFIG_OF is disabled: sound/soc/codecs/cs35l32.c: In function 'cs35l32_i2c_probe': sound/soc/codecs/cs35l32.c:278:2: warning: 'val' may be used uninitialized in this function [-Wmaybe-uninitialized] The code is actually correct because it checks the dev->of_node variable first and we know this is NULL here when CONFIG_OF is disabled, but Russell King noticed that it's broken when we probe the device using DT, and the properties are absent. The code already has some checking for incorrect values, and I keep that checking unchanged here, but add an additional check for an error returned by the property accessor functions that now gets handled the same way as incorrect data in the properties. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- v3: Restore a line that was accidentally removed, as pointed out by Brian Austin. v2: fix bug in case of CONFIG_OF and missing properties v1: only address warning for the !CONFIG_OF case sound/soc/codecs/cs35l32.c | 17 +++++++++++++---- 1 file changed, 13 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 44c30fe3e315..287d13740be4 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -274,7 +274,9 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, if (of_property_read_u32(np, "cirrus,sdout-share", &val) >= 0) pdata->sdout_share = val; - of_property_read_u32(np, "cirrus,boost-manager", &val); + if (of_property_read_u32(np, "cirrus,boost-manager", &val)) + val = -1u; + switch (val) { case CS35L32_BOOST_MGR_AUTO: case CS35L32_BOOST_MGR_AUTO_AUDIO: @@ -282,13 +284,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_BOOST_MGR_FIXED: pdata->boost_mng = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,boost-manager DT value %d\n", val); pdata->boost_mng = CS35L32_BOOST_MGR_BYPASS; } - of_property_read_u32(np, "cirrus,sdout-datacfg", &val); + if (of_property_read_u32(np, "cirrus,sdout-datacfg", &val)) + val = -1u; switch (val) { case CS35L32_DATA_CFG_LR_VP: case CS35L32_DATA_CFG_LR_STAT: @@ -296,13 +300,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_DATA_CFG_LR_VPSTAT: pdata->sdout_datacfg = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,sdout-datacfg DT value %d\n", val); pdata->sdout_datacfg = CS35L32_DATA_CFG_LR; } - of_property_read_u32(np, "cirrus,battery-threshold", &val); + if (of_property_read_u32(np, "cirrus,battery-threshold", &val)) + val = -1u; switch (val) { case CS35L32_BATT_THRESH_3_1V: case CS35L32_BATT_THRESH_3_2V: @@ -310,13 +316,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_BATT_THRESH_3_4V: pdata->batt_thresh = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,battery-threshold DT value %d\n", val); pdata->batt_thresh = CS35L32_BATT_THRESH_3_3V; } - of_property_read_u32(np, "cirrus,battery-recovery", &val); + if (of_property_read_u32(np, "cirrus,battery-recovery", &val)) + val = -1u; switch (val) { case CS35L32_BATT_RECOV_3_1V: case CS35L32_BATT_RECOV_3_2V: @@ -326,6 +334,7 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_BATT_RECOV_3_6V: pdata->batt_recov = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,battery-recovery DT value %d\n", val); -- 2.7.0

3 2

[alsa-devel] [PATCH] ASoC: nau8825: improve FLL function for better performance
by John Hsu 15 Mar '16

15 Mar '16

In FLL calculation, increase VCO/DCO frequency for better performance. Besides, have different register configuration according to fraction or not when apply FLL parameters. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 47 +++++++++++++++++++++++++++++++++------------- sound/soc/codecs/nau8825.h | 13 ++++++++----- 2 files changed, 42 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 1c87299..9023dbd 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -31,7 +31,7 @@ #include "nau8825.h" #define NAU_FREF_MAX 13500000 -#define NAU_FVCO_MAX 100000000 +#define NAU_FVCO_MAX 124000000 #define NAU_FVCO_MIN 90000000 struct nau8825_fll { @@ -957,8 +957,8 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, struct nau8825_fll *fll_param) { - u64 fvco; - unsigned int fref, i; + u64 fvco, fvco_max; + unsigned int fref, i, fvco_sel; /* Ensure the reference clock frequency (FREF) is <= 13.5MHz by dividing * freq_in by 1, 2, 4, or 8 using FLL pre-scalar. @@ -983,18 +983,23 @@ static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, fll_param->ratio = fll_ratio[i].val; /* Calculate the frequency of DCO (FDCO) given freq_out = 256 * Fs. - * FDCO must be within the 90MHz - 100MHz or the FFL cannot be + * FDCO must be within the 90MHz - 124MHz or the FFL cannot be * guaranteed across the full range of operation. * FDCO = freq_out * 2 * mclk_src_scaling */ + fvco_max = 0; + fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { fvco = 256 * fs * 2 * mclk_src_scaling[i].param; - if (NAU_FVCO_MIN < fvco && fvco < NAU_FVCO_MAX) - break; + if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && + fvco_max < fvco) { + fvco_max = fvco; + fvco_sel = i; + } } - if (i == ARRAY_SIZE(mclk_src_scaling)) + if (ARRAY_SIZE(mclk_src_scaling) == fvco_sel) return -EINVAL; - fll_param->mclk_src = mclk_src_scaling[i].val; + fll_param->mclk_src = mclk_src_scaling[fvco_sel].val; /* Calculate the FLL 10-bit integer input and the FLL 16-bit fractional * input based on FDCO, FREF and FLL ratio. @@ -1009,7 +1014,8 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, struct nau8825_fll *fll_param) { regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, - NAU8825_CLK_MCLK_SRC_MASK, fll_param->mclk_src); + NAU8825_CLK_SRC_MASK | NAU8825_CLK_MCLK_SRC_MASK, + NAU8825_CLK_SRC_MCLK | fll_param->mclk_src); regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL1, NAU8825_FLL_RATIO_MASK, fll_param->ratio); /* FLL 16-bit fractional input */ @@ -1022,10 +1028,25 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div); /* select divided VCO input */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, - NAU8825_FLL_FILTER_SW_MASK, 0x0000); - /* FLL sigma delta modulator enable */ - regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, - NAU8825_SDM_EN_MASK, NAU8825_SDM_EN); + NAU8825_FLL_CLK_SW_MASK, NAU8825_FLL_CLK_SW_REF); + /* Disable free-running mode */ + regmap_update_bits(nau8825->regmap, + NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); + if (fll_param->fll_frac) { + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FTR_SW_MASK, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FTR_SW_FILTER); + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN, NAU8825_SDM_EN); + } else { + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FTR_SW_MASK, NAU8825_FLL_FTR_SW_ACCU); + regmap_update_bits(nau8825->regmap, + NAU8825_REG_FLL6, NAU8825_SDM_EN, 0); + } } /* freq_out must be 256*Fs in order to achieve the best performance */ diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 8ceb5f3..01bd386 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -118,15 +118,18 @@ #define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) /* FLL5 (0x08) */ -#define NAU8825_FLL_FILTER_SW_MASK (0x1 << 14) +#define NAU8825_FLL_PDB_DAC_EN (0x1 << 15) +#define NAU8825_FLL_LOOP_FTR_EN (0x1 << 14) +#define NAU8825_FLL_CLK_SW_MASK (0x1 << 13) +#define NAU8825_FLL_CLK_SW_N2 (0x1 << 13) +#define NAU8825_FLL_CLK_SW_REF (0x0 << 13) +#define NAU8825_FLL_FTR_SW_MASK (0x1 << 12) +#define NAU8825_FLL_FTR_SW_ACCU (0x1 << 12) +#define NAU8825_FLL_FTR_SW_FILTER (0x0 << 12) /* FLL6 (0x9) */ -#define NAU8825_DCO_EN_MASK (0x1 << 15) #define NAU8825_DCO_EN (0x1 << 15) -#define NAU8825_DCO_DIS (0x0 << 15) -#define NAU8825_SDM_EN_MASK (0x1 << 14) #define NAU8825_SDM_EN (0x1 << 14) -#define NAU8825_SDM_DIS (0x0 << 14) /* HSD_CTRL (0xc) */ #define NAU8825_HSD_AUTO_MODE (1 << 6) -- 2.6.4

2 1

[alsa-devel] [PATCH 0/3] *** new features for alsabat ***
by han.lu＠intel.com 15 Mar '16

15 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> Add 3 features for alsabat: 1. add default device name for playback and capture 2. add standalone mode, to bypass data analysis 3. add bash test script to cover basic alsabat features Lu, Han (3): alsabat: add default device name for playback and capture alsabat: add standalone mode alsabat: add bash test script bat/Makefile.am | 13 +++--- bat/alsa.c | 12 ------ bat/alsabat-test.sh | 112 ++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/alsabat.1 | 9 +++++ bat/bat.c | 18 ++++++++- bat/common.h | 3 ++ configure.ac | 5 ++- 7 files changed, 153 insertions(+), 19 deletions(-) create mode 100755 bat/alsabat-test.sh -- 2.5.0

2 4

[alsa-devel] [patch] ALSA: mixart: silence unitialized variable warnings
by Dan Carpenter 15 Mar '16

15 Mar '16

We print can print the uninitialized memory on error. Which is an info leak, I suppose but it's basically harmless. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/mixart/mixart_mixer.c b/sound/pci/mixart/mixart_mixer.c index 24a1955..58fd79eb 100644 --- a/sound/pci/mixart/mixart_mixer.c +++ b/sound/pci/mixart/mixart_mixer.c @@ -726,7 +726,7 @@ int mixart_update_playback_stream_level(struct snd_mixart* chip, int is_aes, int int volume[2]; struct mixart_msg request; struct mixart_set_out_stream_level_req set_level; - u32 status; + u32 status = 0; struct mixart_pipe *pipe; memset(&set_level, 0, sizeof(set_level)); @@ -778,7 +778,7 @@ int mixart_update_capture_stream_level(struct snd_mixart* chip, int is_aes) struct mixart_pipe *pipe; struct mixart_msg request; struct mixart_set_in_audio_level_req set_level; - u32 status; + u32 status = 0; if(is_aes) { idx = 1;

2 1

[alsa-devel] Max9768 driver in kernel 4.1
by Micka 15 Mar '16

15 Mar '16

Hi, I'm trying to use the max9768, I have defined it correctly in the dts file : max9768: max9768@49 { compatible = "max,max9768"; reg = <0x49>; }; I can see that in the driver, i go in the function max9768_i2c_probe <http://lxr.free-electrons.com/ident?v=4.1;i=max9768_i2c_probe> without problem, and I don't see errors when executing this line : err <http://lxr.free-electrons.com/ident?v=4.1;i=err> = snd_soc_register_codec <http://lxr.free-electrons.com/ident?v=4.1;i=snd_soc_register_codec>(&client <http://lxr.free-electrons.com/ident?v=4.1;i=client>->dev <http://lxr.free-electrons.com/ident?v=4.1;i=dev>, &max9768_codec_driver <http://lxr.free-electrons.com/ident?v=4.1;i=max9768_codec_driver>, NULL <http://lxr.free-electrons.com/ident?v=4.1;i=NULL>, 0); I have put some log : [ 9.213797] ASoC: dai register 2-0049 #0 [ 9.213806] ASoC: Registered codec 'max9768.2-0049' my problem is that i don't know how to configure the volume ? Any idea ? where is it ? I don't see where is it in the amixer controls list : root@beaglebone:~# amixer controls numid=1,iface=MIXER,name='PCM Playback Volume' numid=15,iface=MIXER,name='Line DAC Playback Volume' numid=14,iface=MIXER,name='Line PGA Bypass Volume' numid=20,iface=MIXER,name='Line Playback Switch' numid=33,iface=MIXER,name='ADC HPF Cut-off' numid=23,iface=MIXER,name='AGC Switch' numid=30,iface=MIXER,name='De-emphasis Switch' numid=17,iface=MIXER,name='HP DAC Playback Volume' numid=16,iface=MIXER,name='HP PGA Bypass Volume' numid=21,iface=MIXER,name='HP Playback Switch' numid=19,iface=MIXER,name='HPCOM DAC Playback Volume' numid=18,iface=MIXER,name='HPCOM PGA Bypass Volume' numid=22,iface=MIXER,name='HPCOM Playback Switch' numid=26,iface=MIXER,name='Left AGC Attack time' numid=28,iface=MIXER,name='Left AGC Decay time' numid=24,iface=MIXER,name='Left AGC Target level' numid=75,iface=MIXER,name='Left DAC Mux' numid=49,iface=MIXER,name='Left HP Mixer DACL1 Switch' numid=7,iface=MIXER,name='Left HP Mixer DACR1 Playback Volume' numid=51,iface=MIXER,name='Left HP Mixer DACR1 Switch' numid=48,iface=MIXER,name='Left HP Mixer PGAL Bypass Switch' numid=50,iface=MIXER,name='Left HP Mixer PGAR Bypass Switch' numid=6,iface=MIXER,name='Left HP Mixer PGAR Bypass Volume' numid=41,iface=MIXER,name='Left HPCOM Mixer DACL1 Switch' numid=11,iface=MIXER,name='Left HPCOM Mixer DACR1 Playback Volume' numid=43,iface=MIXER,name='Left HPCOM Mixer DACR1 Switch' numid=40,iface=MIXER,name='Left HPCOM Mixer PGAL Bypass Switch' numid=42,iface=MIXER,name='Left HPCOM Mixer PGAR Bypass Switch' numid=10,iface=MIXER,name='Left HPCOM Mixer PGAR Bypass Volume' numid=74,iface=MIXER,name='Left HPCOM Mux' numid=57,iface=MIXER,name='Left Line Mixer DACL1 Switch' numid=3,iface=MIXER,name='Left Line Mixer DACR1 Playback Volume' numid=59,iface=MIXER,name='Left Line Mixer DACR1 Switch' numid=56,iface=MIXER,name='Left Line Mixer PGAL Bypass Switch' numid=58,iface=MIXER,name='Left Line Mixer PGAR Bypass Switch' numid=2,iface=MIXER,name='Left Line Mixer PGAR Bypass Volume' numid=71,iface=MIXER,name='Left Line1L Mux' numid=70,iface=MIXER,name='Left Line1R Mux' numid=64,iface=MIXER,name='Left PGA Mixer Line1L Switch' numid=65,iface=MIXER,name='Left PGA Mixer Line1R Switch' numid=66,iface=MIXER,name='Left PGA Mixer Mic2L Switch' numid=67,iface=MIXER,name='Left PGA Mixer Mic2R Switch' numid=34,iface=MIXER,name='Output Driver Power-On time' numid=35,iface=MIXER,name='Output Driver Ramp-up step' numid=32,iface=MIXER,name='PGA Capture Switch' numid=31,iface=MIXER,name='PGA Capture Volume' numid=27,iface=MIXER,name='Right AGC Attack time' numid=29,iface=MIXER,name='Right AGC Decay time' numid=25,iface=MIXER,name='Right AGC Target level' numid=73,iface=MIXER,name='Right DAC Mux' numid=9,iface=MIXER,name='Right HP Mixer DACL1 Playback Volume' numid=45,iface=MIXER,name='Right HP Mixer DACL1 Switch' numid=47,iface=MIXER,name='Right HP Mixer DACR1 Switch' numid=44,iface=MIXER,name='Right HP Mixer PGAL Bypass Switch' numid=8,iface=MIXER,name='Right HP Mixer PGAL Bypass Volume' numid=46,iface=MIXER,name='Right HP Mixer PGAR Bypass Switch' numid=13,iface=MIXER,name='Right HPCOM Mixer DACL1 Playback Volume' numid=37,iface=MIXER,name='Right HPCOM Mixer DACL1 Switch' numid=39,iface=MIXER,name='Right HPCOM Mixer DACR1 Switch' numid=36,iface=MIXER,name='Right HPCOM Mixer PGAL Bypass Switch' numid=12,iface=MIXER,name='Right HPCOM Mixer PGAL Bypass Volume' numid=38,iface=MIXER,name='Right HPCOM Mixer PGAR Bypass Switch' numid=72,iface=MIXER,name='Right HPCOM Mux' numid=5,iface=MIXER,name='Right Line Mixer DACL1 Playback Volume' numid=53,iface=MIXER,name='Right Line Mixer DACL1 Switch' numid=55,iface=MIXER,name='Right Line Mixer DACR1 Switch' numid=52,iface=MIXER,name='Right Line Mixer PGAL Bypass Switch' numid=4,iface=MIXER,name='Right Line Mixer PGAL Bypass Volume' numid=54,iface=MIXER,name='Right Line Mixer PGAR Bypass Switch' numid=69,iface=MIXER,name='Right Line1L Mux' numid=68,iface=MIXER,name='Right Line1R Mux' numid=61,iface=MIXER,name='Right PGA Mixer Line1L Switch' numid=60,iface=MIXER,name='Right PGA Mixer Line1R Switch' numid=62,iface=MIXER,name='Right PGA Mixer Mic2L Switch' numid=63,iface=MIXER,name='Right PGA Mixer Mic2R Switch' Thank you for your work and help !

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