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

[alsa-devel] Creative ca0132 on Gigabyte Z170X Gaming 7 motherboard: no sound
by Scott Doty 04 Mar '16

04 Mar '16

Hello, Didn't see a response on alsa-user, so thought I'd better roll up my sleeves and get to work. :) The ca0132 codec seems to do everything on my Gigabyte Z170X Gaming 7 motherboard but make sound. I can see the mixer settings, change them, even see the DSP settings -- but no sound out, either with HP or SPDIF. http://www.alsa-project.org/db/?f=6373811de293125e4e8c345c0c2d4a736c1fc2b8 I do see this in the startup messages: [ 9.314893] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 9.321116] snd_hda_intel 0000:00:1f.3: failed to add i915 component master (-19) I've tried all the fixes I could find reference to, as well as using iecset, but no joy. I also tried editing the driver to disable the "vga_switcheroo" flag, but that didn't work either. (Which is too bad, I thought that would be a quick fix that I could report. :/ ) If someone could point me in the right direction, I think I can investigate this and get it working. Any assistance greatly appreciated. Thanks, -Scott

2 2

[alsa-devel] [PATCH] Provide sequencer sound card number / PID via alsa-lib
by Martin Koegler 03 Mar '16

03 Mar '16

From: Martin Koegler <martin.koegler(a)chello.at> rawmidi devices expose the card number via IOCTLs, which allows to find the corresponding device in sysfs. The sequencer provides no identifing data. Chromium works around this issue by scanning rawmidi as well as sequencer devices and matching them by using assumtions, how the kernel register sequencer devices. This changes adds support for exposing the card number for kernel clients as well as the PID for user client. It supports kernels with and without the required support. Signed-off-by: Martin Koegler <martin.koegler(a)chello.at> --- include/seq.h | 2 ++ include/sound/asequencer.h | 6 ++++-- src/seq/seq.c | 26 ++++++++++++++++++++++++++ src/seq/seq_hw.c | 12 +++++++++++- 4 files changed, 43 insertions(+), 3 deletions(-) diff --git a/include/seq.h b/include/seq.h index 9576822..d05940e 100644 --- a/include/seq.h +++ b/include/seq.h @@ -143,6 +143,8 @@ snd_seq_client_type_t snd_seq_client_info_get_type(const snd_seq_client_info_t * const char *snd_seq_client_info_get_name(snd_seq_client_info_t *info); int snd_seq_client_info_get_broadcast_filter(const snd_seq_client_info_t *info); int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info); +int snd_seq_client_info_get_card(const snd_seq_client_info_t *info); +int snd_seq_client_info_get_pid(const snd_seq_client_info_t *info); const unsigned char *snd_seq_client_info_get_event_filter(const snd_seq_client_info_t *info); int snd_seq_client_info_get_num_ports(const snd_seq_client_info_t *info); int snd_seq_client_info_get_event_lost(const snd_seq_client_info_t *info); diff --git a/include/sound/asequencer.h b/include/sound/asequencer.h index 09c8a00..3ac70fd 100644 --- a/include/sound/asequencer.h +++ b/include/sound/asequencer.h @@ -24,7 +24,7 @@ /** version of the sequencer */ -#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1) +#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 2) /** * definition of sequencer event types @@ -356,7 +356,9 @@ struct snd_seq_client_info { unsigned char event_filter[32]; /* event filter bitmap */ int num_ports; /* RO: number of ports */ int event_lost; /* number of lost events */ - char reserved[64]; /* for future use */ + int card; /* RO: card number[kernel] */ + int pid; /* RO: pid[user] */ + char reserved[56]; /* for future use */ }; diff --git a/src/seq/seq.c b/src/seq/seq.c index 620ca3f..4405e68 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -1522,6 +1522,32 @@ int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info) } /** + * \brief Get the sound card number. + * \param info client_info container + * \return card number or -1 if value is not available. + * + * Only available for SND_SEQ_KERNEL_CLIENT clients. + */ +int snd_seq_client_info_get_card(const snd_seq_client_info_t *info) +{ + assert(info); + return info->card; +} + +/** + * \brief Get the owning PID. + * \param info client_info container + * \return pid or -1 if value is not available. + * + * Only available for SND_SEQ_USER_CLIENT clients. + */ +int snd_seq_client_info_get_pid(const snd_seq_client_info_t *info) +{ + assert(info); + return info->pid; +} + +/** * \brief (DEPRECATED) Get the event filter bitmap of a client_info container * \param info client_info container * \return NULL if no event filter, or pointer to event filter bitmap diff --git a/src/seq/seq_hw.c b/src/seq/seq_hw.c index d033367..578ef12 100644 --- a/src/seq/seq_hw.c +++ b/src/seq/seq_hw.c @@ -32,10 +32,11 @@ const char *_snd_module_seq_hw = ""; #ifndef DOC_HIDDEN #define SNDRV_FILE_SEQ ALSA_DEVICE_DIRECTORY "seq" #define SNDRV_FILE_ALOADSEQ ALOAD_DEVICE_DIRECTORY "aloadSEQ" -#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 1) +#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 2) typedef struct { int fd; + int version; } snd_seq_hw_t; #endif /* DOC_HIDDEN */ @@ -100,6 +101,10 @@ static int snd_seq_hw_get_client_info(snd_seq_t *seq, snd_seq_client_info_t * in /*SYSERR("SNDRV_SEQ_IOCTL_GET_CLIENT_INFO failed");*/ return -errno; } + if (hw->version < SNDRV_PROTOCOL_VERSION(1, 0, 2)) { + info->card = -1; + info->pid = -1; + } return 0; } @@ -368,6 +373,10 @@ static int snd_seq_hw_query_next_client(snd_seq_t *seq, snd_seq_client_info_t *i /*SYSERR("SNDRV_SEQ_IOCTL_QUERY_NEXT_CLIENT failed");*/ return -errno; } + if (hw->version < SNDRV_PROTOCOL_VERSION(1, 0, 2)) { + info->card = -1; + info->pid = -1; + } return 0; } @@ -480,6 +489,7 @@ int snd_seq_hw_open(snd_seq_t **handle, const char *name, int streams, int mode) return -ENOMEM; } hw->fd = fd; + hw->version = ver; if (streams & SND_SEQ_OPEN_OUTPUT) { seq->obuf = (char *) malloc(seq->obufsize = SND_SEQ_OBUF_SIZE); if (!seq->obuf) { -- 2.1.4

1 0

[alsa-devel] [PATCH v3] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Zidan Wang 03 Mar '16

03 Mar '16

This is the initial imx-wm8958 device-tree-only machine driver working with fsl_sai driver. This sound card has three dai link, HIFI, VOICE and BT dai. HIFI dai link will support codec master and slave mode. VOICE an BT dai link have dummy cpu dai, and just support codec master mode. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v2->v3: 1. move codec mclk enable to codec driver 2. remove rate constraints 3. remove GPIO configurations, which can be configured in device tree .../devicetree/bindings/sound/imx-audio-wm8958.txt | 49 +++ arch/arm/configs/imx_v6_v7_defconfig | 1 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8958.c | 327 +++++++++++++++++++++ 5 files changed, 393 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..7a2b8a9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt @@ -0,0 +1,49 @@ +Freescale i.MX audio complex with WM8958 codec + +Required properties: + + - compatible : "fsl,imx-audio-wm8958" + + - model : The user-visible name of this sound complex + + - cpu-dai : The phandle of an CPU DAI controller + + - 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 + + - fsl,hifi-dai-master : If present, hifi dai works as master, and will + provide bit clock and frame clock. Otherwise, + hifi dai works as slave. + +Example: + +sound { + compatible = "fsl,imx6ul-evk-wm8958", + "fsl,imx-audio-wm8958"; + model = "wm8958-audio"; + cpu-dai = <&sai1>; + 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"; + + fsl,hifi-dai-master; +}; diff --git a/arch/arm/configs/imx_v6_v7_defconfig b/arch/arm/configs/imx_v6_v7_defconfig index 978c5de..7ac5856 100644 --- a/arch/arm/configs/imx_v6_v7_defconfig +++ b/arch/arm/configs/imx_v6_v7_defconfig @@ -251,6 +251,7 @@ CONFIG_SND_SOC_FSL_ASRC=y CONFIG_SND_IMX_SOC=y CONFIG_SND_SOC_PHYCORE_AC97=y CONFIG_SND_SOC_EUKREA_TLV320=y +CONFIG_SND_SOC_IMX_WM8958=y CONFIG_SND_SOC_IMX_WM8962=y CONFIG_SND_SOC_IMX_SGTL5000=y CONFIG_SND_SOC_IMX_SPDIF=y 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..01e7044 --- /dev/null +++ b/sound/soc/fsl/imx-wm8958.c @@ -0,0 +1,327 @@ +/* + * 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 "../fsl/fsl_sai.h" +#include "../codecs/wm8994.h" + +#define DAI_NAME_SIZE 32 + +#define DAI_LINK_NUM (3) +#define HIFI_DAI (0) +#define VOICE_DAI (1) +#define BT_DAI (2) + +#define WM8958_MCLK_MAX (2) + +#define WM8994_FLL(id) (id == HIFI_DAI ? WM8994_FLL1 : WM8994_FLL2) +#define WM8994_SYSCLK_FLL(id) (id == HIFI_DAI ? WM8994_SYSCLK_FLL1 : WM8994_SYSCLK_FLL2) +#define WM8994_FLL_SRC_MCLK(id) (id == HIFI_DAI ? WM8994_FLL_SRC_MCLK1 : WM8994_FLL_SRC_MCLK2) + +struct imx_wm8958_data { + struct snd_soc_dai_link *dai_link; + struct snd_soc_card card; + struct clk *mclk[WM8958_MCLK_MAX]; + u32 mclk_freq[WM8958_MCLK_MAX]; + bool is_hifi_dai_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 == HIFI_DAI) + mclk_id = WM8994_FLL_SRC_MCLK2; + else + mclk_id = WM8994_FLL_SRC_MCLK1; + + if (id == HIFI_DAI) { + if (!data->is_hifi_dai_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_hifi_dai_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 == HIFI_DAI && !data->is_hifi_dai_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_hifi_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_ops imx_voice_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_dai_link imx_wm8958_dai_link[] = { + [HIFI_DAI] = { + .name = "HiFi", + .stream_name = "HiFi", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif1", + .ops = &imx_hifi_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF, + }, + [VOICE_DAI] = { + .name = "Voice", + .stream_name = "Voice", + .cpu_dai_name = "snd-soc-dummy-dai", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif2", + .platform_name = "snd-soc-dummy", + .ignore_pmdown_time = 1, + .ops = &imx_voice_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + }, + [BT_DAI] = { + .name = "Bluetooth", + .stream_name = "Bluetooth", + .cpu_dai_name = "snd-soc-dummy-dai", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif3", + .platform_name = "snd-soc-dummy", + .ignore_pmdown_time = 1, + }, +}; + +static int imx_wm8958_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np, *codec_np; + struct device_node *np = pdev->dev.of_node; + struct platform_device *cpu_pdev; + struct i2c_client *codec_dev; + struct imx_wm8958_data *data; + char tmp[8]; + int ret, i; + + cpu_np = of_parse_phandle(np, "cpu-dai", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "cpu dai phandle missing or invalid\n"); + return -EINVAL; + } + + 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; + } + + cpu_pdev = of_find_device_by_node(cpu_np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find cpu dai 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) + return -ENOMEM; + + data->dai_link = imx_wm8958_dai_link; + + /* + * AIF1 support codec master and slave mode + * AIF2 and AIF3 just support codec master mode + */ + if (of_property_read_bool(pdev->dev.of_node, "fsl,hifi-dai-master")) { + data->is_hifi_dai_master = true; + data->dai_link[HIFI_DAI].dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else { + data->is_hifi_dai_master = false; + data->dai_link[HIFI_DAI].dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } + + for (i = 0; i < WM8958_MCLK_MAX; i++) { + sprintf(tmp, "MCLK%d", i + 1); + data->mclk[i] = devm_clk_get(&codec_dev->dev, tmp); + if (!IS_ERR(data->mclk[i])) + data->mclk_freq[i] = clk_get_rate(data->mclk[i]); + } + + if (!data->mclk_freq[0] && !data->mclk_freq[1]) { + dev_err(&pdev->dev, "failed to get mclk clock\n"); + ret = -EINVAL; + goto fail; + } + + data->dai_link[HIFI_DAI].cpu_dai_name = dev_name(&cpu_pdev->dev); + data->dai_link[HIFI_DAI].platform_of_node = cpu_np; + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + + data->card.num_links = DAI_LINK_NUM; + 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); + 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 v2] ASoC: cht_bsw_rt5645: Fix writing to string literal
by Carlo Caione 03 Mar '16

03 Mar '16

From: Carlo Caione <carlo(a)endlessm.com> We cannot use strcpy() to write to a const char * location. This is causing a 'BUG: unable to handle kernel paging request' error at boot when using the cht-bsw-rt5645 driver. With this patch we also fix a wrong indexing in the driver where the codec_name of the wrong dai_link is being overwritten. Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 2d3afdd..a7b96a9 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -367,8 +367,12 @@ static int snd_cht_mc_probe(struct platform_device *pdev) } card->dev = &pdev->dev; sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id); + /* set correct codec name */ - strcpy((char *)card->dai_link[2].codec_name, codec_name); + for (i = 0; i < ARRAY_SIZE(cht_dailink); i++) + if (!strcmp(card->dai_link[i].codec_name, "i2c-10EC5645:00")) + card->dai_link[i].codec_name = kstrdup(codec_name, GFP_KERNEL); + snd_soc_card_set_drvdata(card, drv); ret_val = devm_snd_soc_register_card(&pdev->dev, card); if (ret_val) { -- 2.5.0

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Re: [alsa-devel] Compute Stick baytrail audio problem
by Giacomo Comes 03 Mar '16

03 Mar '16

On Tue, Feb 23, 2016 at 12:32:23PM +0900, Toyo Abe wrote: > > Hi, > > I'd struggled a similar underrun problem on my baytrail device. The dmesg showed > the following warnings. > > [15356.757938] had: Driver detected 2 missed buffer done interrupt(s)!!!! > [15356.761594] had: Driver detected 1 missed buffer done interrupt(s)!!!! > [15356.774775] had: Unable to clear UNDERRUN bits > [15356.791312] had: Driver detected 1 missed buffer done interrupt(s)!!!! > [15356.808148] had: Unable to clear UNDERRUN bits > [15356.824692] had: Driver detected 2 missed buffer done interrupt(s)!!!! > > After some investigation, I found some problems on the driver and fixed a few of > them. I pushed the fix to my github. I hope it might help you. > https://raw.githubusercontent.com/toyoabe/baytrailaudio/fix-underrun/hdmi_a… > > Best Regards, > -toyo Hi, that was it!. I added your changes and now the audio works. I'm happy. I got finally every subsystem of the Compute Stick working. I have a minor question tought. Alsamixer says: This sound device does not have any controls. Is there a way to change the volume without enabling pulseaudio? Just for documentation, here is the small change that Toyo did to make the underrun disappear: diff -Nraub a/drivers/gpu/drm/i915/i915_irq.c b/drivers/gpu/drm/i915/i915_irq.c --- a/drivers/gpu/drm/i915/i915_irq.c 2016-02-23 08:03:42.406016576 -0400 +++ b/drivers/gpu/drm/i915/i915_irq.c 2016-02-23 08:03:58.714016576 -0400 @@ -2717,11 +2717,14 @@ int pipe = 1; spin_lock_irqsave(&dev_priv->irq_lock, irqflags); + i915_enable_lpe_pipestat(dev_priv, pipe); + imr = I915_READ(VLV_IMR); /* Audio is on Stream A */ imr &= ~I915_LPE_PIPE_A_INTERRUPT; I915_WRITE(VLV_IMR, imr); - i915_enable_lpe_pipestat(dev_priv, pipe); + I915_WRITE(VLV_IER, ~imr); + POSTING_READ(VLV_IER); spin_unlock_irqrestore(&dev_priv->irq_lock, irqflags); return 0; @@ -2769,7 +2772,10 @@ spin_lock_irqsave(&dev_priv->irq_lock, irqflags); imr = I915_READ(VLV_IMR); imr |= I915_LPE_PIPE_A_INTERRUPT; + I915_WRITE(VLV_IER, ~imr); I915_WRITE(VLV_IMR, imr); + POSTING_READ(VLV_IMR); + i915_disable_lpe_pipestat(dev_priv, pipe); spin_unlock_irqrestore(&dev_priv->irq_lock, irqflags); diff -Nraub a/sound/hdmi_audio/intel_mid_hdmi_audio.c b/sound/hdmi_audio/intel_mid_hdmi_audio.c --- a/sound/hdmi_audio/intel_mid_hdmi_audio.c 2016-02-23 08:03:42.414016576 -0400 +++ b/sound/hdmi_audio/intel_mid_hdmi_audio.c 2016-02-23 08:03:58.722016576 -0400 @@ -1128,7 +1128,6 @@ pr_debug("HDMI status =0x%x\n", hdmi_status); if (hdmi_status & AUD_CONFIG_MASK_UNDERRUN) { i++; - hdmi_status &= ~AUD_CONFIG_MASK_UNDERRUN; had_write_register(AUD_HDMI_STATUS_v2, hdmi_status); } else break; Thanks again. Giacomo > > Pierre-Louis Bossart wrote: > > On 02/19/2016 02:39 PM, Giacomo Comes wrote: > >> Hi, > >> I have recently bought a Intel Compute Stick and I am having trouble with= > > the audio. > >> Intel provides a custom Ubuntu release that support audio, but I want to = > > run the > >> linux distribution of my choice on it (openSUSE 13.2 for the moment). > >> > >> In order to have HDMI audio support it is necessary to compile a kernel w= > > ith this patch > >> (https://raw.githubusercontent.com/01org/baytrailaudio/master/hdmi_audio_= > > 20150319.patch) > >> provided by Intel for kernel 3.16.7 (https://01.org/ubuntu-hdmi) > >> > >> I compiled the openSUSE kernel using the default openSUSE .config plus th= > > e following: > >> CONFIG_SUPPORT_HDMI=3Dy > >> CONFIG_SND_SOC=3Dm > >> CONFIG_SND_SOC_INTEL_SST=3Dm > >> CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH=3Dm > >> > >> I booted the new kernel and aplay -l gives this output: > >> **** List of PLAYBACK Hardware Devices **** > >> card 0: IntelHDMI [IntelHDMI], device 0: IntelHDMI [IntelHDMI] > >> Subdevices: 0/1 > >> Subdevice #0: subdevice #0 > >> > >> but when I run: > >> aplay /usr/share/sounds/alsa/test.wav > >> the audio plays for .2 seconds and then nothing more. > >> Sometime it will play later a little bit more and stop again. > >> At some point these messages appears: > >> underrun!!! (at least 29325.946 ms long) > >> underrun!!! (at least 1535.889 ms long) > >> underrun!!! (at least 3196.213 ms long) > >> > >> > >> I have looked around on google for a hint on how to fix this issue but I = > > found nothing. > >> Any idea where to look for a solution? > >> > >> Attached there are the dmesg and lsmod output for ubuntu > >> with the working audio and openSUSE 13.2 plus patch > >> without the working audio. > >> I can provide any other information you need. > > > > There are enough messages that make me believe something is very wrong > > from the start > > > > 19.664486] had: ******** HAD DRIVER loading.. Ver: 0.01.003 > > [ 19.664549] hdmi_audio_probe dma_mask: 0 > > [ 19.664937] card->dev is NULL!!!!! Should not be this case > > [ 19.668176] [drm] mid_hdmi_audio_register: Scheduling HDMI audio work qu= > > eue > > [ 19.670745] [drm:i915_had_wq] *ERROR* Checking for HDMI connection at bo= > > ot > > > > Some of us at Intel are trying to clean this up but it'll take time. > > > > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >

3 4

[alsa-devel] feeding back pkgsrc changs
by Thomas Klausner 03 Mar '16

03 Mar '16

Hi! I've just updated the pkgsrc packages for alsa-lib, alsa-utils and alsa-plugins to 1.1.0 and noticed that there are quite a number of local patches we have. I'd like to get them integrated. When I visited the alsa-projects web page, I noticed that the link to the bug tracker is dead. What is the recommended method to provide the patches? Thanks, Thomas

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[alsa-devel] [PATCH 1/5] Introduce bswap.h for portable definitions of byte swap macros.
by Thomas Klausner 03 Mar '16

03 Mar '16

From: Thomas Klausner <wiz(a)NetBSD.org> Signed-off-by: Thomas Klausner <wiz(a)NetBSD.org> --- include/Makefile.am | 2 +- include/bswap.h | 39 +++++++++++++++++++++++++++++++++++++++ src/pcm/pcm_adpcm.c | 2 +- src/pcm/pcm_alaw.c | 2 +- src/pcm/pcm_copy.c | 2 +- src/pcm/pcm_dmix_generic.c | 2 +- src/pcm/pcm_file.c | 2 +- src/pcm/pcm_iec958.c | 2 +- src/pcm/pcm_lfloat.c | 2 +- src/pcm/pcm_linear.c | 2 +- src/pcm/pcm_meter.c | 2 +- src/pcm/pcm_misc.c | 2 +- src/pcm/pcm_mulaw.c | 2 +- src/pcm/pcm_null.c | 2 +- src/pcm/pcm_rate.c | 2 +- src/pcm/pcm_rate_linear.c | 2 +- src/pcm/pcm_route.c | 2 +- src/pcm/pcm_softvol.c | 2 +- 18 files changed, 56 insertions(+), 17 deletions(-) create mode 100644 include/bswap.h diff --git a/include/Makefile.am b/include/Makefile.am index 8c83c11..31a3f74 100644 --- a/include/Makefile.am +++ b/include/Makefile.am @@ -59,7 +59,7 @@ alsainclude_HEADERS += alisp.h endif noinst_HEADERS = alsa sys.h search.h list.h aserver.h local.h alsa-symbols.h \ - asoundlib-head.h asoundlib-tail.h type_compat.h + asoundlib-head.h asoundlib-tail.h bswap.h type_compat.h DISTCLEANFILES = stamp-vh version.h alsa asoundlib.h diff --git a/include/bswap.h b/include/bswap.h new file mode 100644 index 0000000..68adce3 --- /dev/null +++ b/include/bswap.h @@ -0,0 +1,39 @@ +/* + * ALSA lib - compatibility header for providing byte swapping macros + * Copyright (c) 2016 by Thomas Klausner <wiz(a)NetBSD.org> + * + * + * This library is free software; you can redistribute it and/or modify + * it under the terms of the GNU Lesser General Public License as + * published by the Free Software Foundation; either version 2.1 of + * the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#ifndef __BSWAP_H +#define __BSWAP_H + +#if defined(__FreeBSD__) || defined(__DragonFly__) || defined(__NetBSD__) +#include <sys/endian.h> +#define bswap_16 bswap16 +#define bswap_32 bswap32 +#define bswap_64 bswap64 +#elif defined (__sun) +#include <sys/byteorder.h> +#define bswap_16 BSWAP_16 +#define bswap_32 BSWAP_32 +#define bswap_64 BSWAP_64 +#else +#include <byteswap.h> +#endif + +#endif diff --git a/src/pcm/pcm_adpcm.c b/src/pcm/pcm_adpcm.c index 1a83c5a..1c88c83 100644 --- a/src/pcm/pcm_adpcm.c +++ b/src/pcm/pcm_adpcm.c @@ -56,7 +56,7 @@ IMA compatibility project proceedings, Vol 2, Issue 2, May 1992. come across a good description of XA yet. */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_alaw.c b/src/pcm/pcm_alaw.c index db759e3..fa58441 100644 --- a/src/pcm/pcm_alaw.c +++ b/src/pcm/pcm_alaw.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_copy.c b/src/pcm/pcm_copy.c index 66d3a47..0dbd5bd 100644 --- a/src/pcm/pcm_copy.c +++ b/src/pcm/pcm_copy.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_dmix_generic.c b/src/pcm/pcm_dmix_generic.c index 9e9d3c3..40c0874 100644 --- a/src/pcm/pcm_dmix_generic.c +++ b/src/pcm/pcm_dmix_generic.c @@ -125,7 +125,7 @@ static void mix_select_callbacks(snd_pcm_direct_t *dmix) (1ULL << SND_PCM_FORMAT_S24_LE) | (1ULL << SND_PCM_FORMAT_S24_3LE) | \ (1ULL << SND_PCM_FORMAT_U8)) -#include <byteswap.h> +#include "bswap.h" static void generic_mix_areas_16_native(unsigned int size, volatile signed short *dst, diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 9ab6964..92eb072 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include <ctype.h> #include <string.h> #include "pcm_local.h" diff --git a/src/pcm/pcm_iec958.c b/src/pcm/pcm_iec958.c index 38c4ce7..86ac9cf 100644 --- a/src/pcm/pcm_iec958.c +++ b/src/pcm/pcm_iec958.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_lfloat.c b/src/pcm/pcm_lfloat.c index 2f3e578..22201f8 100644 --- a/src/pcm/pcm_lfloat.c +++ b/src/pcm/pcm_lfloat.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_linear.c b/src/pcm/pcm_linear.c index 9a92abd..50df779 100644 --- a/src/pcm/pcm_linear.c +++ b/src/pcm/pcm_linear.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_meter.c b/src/pcm/pcm_meter.c index 034f582..1b0ccb4 100644 --- a/src/pcm/pcm_meter.c +++ b/src/pcm/pcm_meter.c @@ -27,7 +27,7 @@ */ -#include <byteswap.h> +#include "bswap.h" #include <time.h> #include <pthread.h> #include <dlfcn.h> diff --git a/src/pcm/pcm_misc.c b/src/pcm/pcm_misc.c index 5870f82..7d2b05d 100644 --- a/src/pcm/pcm_misc.c +++ b/src/pcm/pcm_misc.c @@ -23,7 +23,7 @@ #include <stdlib.h> #include <unistd.h> #include <string.h> -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" diff --git a/src/pcm/pcm_mulaw.c b/src/pcm/pcm_mulaw.c index 011b2a5..15dbdc1 100644 --- a/src/pcm/pcm_mulaw.c +++ b/src/pcm/pcm_mulaw.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_null.c b/src/pcm/pcm_null.c index 5e63caa..685618e 100644 --- a/src/pcm/pcm_null.c +++ b/src/pcm/pcm_null.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include <limits.h> #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index c76db25..41bddac 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -28,7 +28,7 @@ * */ #include <inttypes.h> -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" #include "pcm_rate.h" diff --git a/src/pcm/pcm_rate_linear.c b/src/pcm/pcm_rate_linear.c index 7481b38..70399e0 100644 --- a/src/pcm/pcm_rate_linear.c +++ b/src/pcm/pcm_rate_linear.c @@ -21,7 +21,7 @@ */ #include <inttypes.h> -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" #include "pcm_rate.h" diff --git a/src/pcm/pcm_route.c b/src/pcm/pcm_route.c index 646517d..3611603 100644 --- a/src/pcm/pcm_route.c +++ b/src/pcm/pcm_route.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include <math.h> #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c index c6cfd88..802aa4b 100644 --- a/src/pcm/pcm_softvol.c +++ b/src/pcm/pcm_softvol.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include <math.h> #include "pcm_local.h" #include "pcm_plugin.h" -- 2.7.1

1 4

[alsa-devel] [PATCH v2 0/7] ALSA: hda: Move chmap helpers to core
by Subhransu S. Prusty 03 Mar '16

03 Mar '16

The HDMI channel map helper APIs and control registration can be reused by ASoC skylake HDMI drivers. A common chmap object is created, channel map control callback handlers, helper APIs and data structures are moved to the core. Driver object store an instance of this chmap object to register chmap controls and program channel allocation for HDMI infoframe. Driver registers ops for specific operations. changes in v2: - Splitted patches for easier review - Access most of chmap helpers through ops, thus reducing number of exported APIs for chmap support Subhransu S. Prusty (7): ALSA: hda - Create common chmap object ALSA: hda - Register chmap obj as priv data instead of codec ALSA: hda - Use hdmi name space for CEA spk alloc structure ALSA: hda - Add hdmi chmap verb programming ops to chmap object ALSA: hda - Use ops instead of directly accessing chmap helpers ALSA: hda - Move chmap support helpers/ops to core ALSA: hda - Use snd_hdmi namespace prefix for printing ch alloc api include/sound/hdmi_chmap.h | 75 ++++ sound/hda/Makefile | 2 +- sound/hda/hdmi_chmap.c | 791 ++++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_eld.c | 31 +- sound/pci/hda/patch_hdmi.c | 830 +++++---------------------------------------- 5 files changed, 957 insertions(+), 772 deletions(-) create mode 100644 include/sound/hdmi_chmap.h create mode 100644 sound/hda/hdmi_chmap.c -- 1.9.1

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[alsa-devel] [PATCH - hda 1/1] ALSA: hda - Fix mic issues on Acer Aspire E1-472
by Simon South 03 Mar '16

03 Mar '16

This patch applies the microphone-related fix created for the Acer Aspire E1-572 to the E1-472 as well, as it uses the same Realtek ALC282 CODEC and demonstrates the same issues. This patch allows an external, headset microphone to be used and limits the gain on the (quite noisy) internal microphone. Signed-off-by: Simon South <simon(a)simonsouth.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c1527fc..5c672f8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5224,6 +5224,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x080d, "Acer Aspire V5-122P", ALC269_FIXUP_ASPIRE_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), + SND_PCI_QUIRK(0x1025, 0x0762, "Acer Aspire E1-472", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), -- 2.5.0

2 1

[alsa-devel] [PATCH 0/5] ASoC: pxa: Fix enum ctl accesses
by Takashi Iwai 02 Mar '16

02 Mar '16

Hi, here is a patchset to correct the wrong type accesses in pxa drivers for enum ctls. I didn't put Cc to stable for them, supposing these are only for 32bit systems. If not, feel free to put Cc to stable. thanks, Takashi === Takashi Iwai (5): ASoC: pxa: corgi: Fix enum ctl accesses in a wrong type ASoC: pxa: magician: Fix enum ctl accesses in a wrong type ASoC: pxa: poodle: Fix enum ctl accesses in a wrong type ASoC: pxa: spitz: Fix enum ctl accesses in a wrong type ASoC: pxa: tosa: Fix enum ctl accesses in a wrong type sound/soc/pxa/corgi.c | 12 ++++++------ sound/soc/pxa/magician.c | 6 +++--- sound/soc/pxa/poodle.c | 12 ++++++------ sound/soc/pxa/spitz.c | 12 ++++++------ sound/soc/pxa/tosa.c | 12 ++++++------ 5 files changed, 27 insertions(+), 27 deletions(-) -- 2.7.2

3 11