December 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] endianness problems in fireworks/bebob_maudio
by Clemens Ladisch 07 Jan '15

07 Jan '15

Hi, sparse complains: sound/firewire/fireworks/fireworks_transaction.c:127:18: warning: restricted __be32 degrades to integer t = (struct snd_efw_transaction *)data; length = min_t(size_t, t->length * sizeof(t->length), length); 't->length' is still a big-endian value. This means that the driver ends up always using 'length'. sound/firewire/bebob/bebob_maudio.c:100:17: warning: incorrect type in initializer (different base types) sound/firewire/bebob/bebob_maudio.c:100:17: expected restricted __be32 sound/firewire/bebob/bebob_maudio.c:100:17: got int __be32 cues[3] = { MAUDIO_BOOTLOADER_CUE1, MAUDIO_BOOTLOADER_CUE2, MAUDIO_BOOTLOADER_CUE3 }; rcode = fw_run_transaction(device->card, TCODE_WRITE_BLOCK_REQUEST, device->node_id, device->generation, device->max_speed, BEBOB_ADDR_REG_REQ, cues, sizeof(cues)); The three MAUDIO_BOOTLOADER_CUEx values will end up as a different byte sequence on big-endian machines. The simplest way to have these twelve bytes unchanged on the bus is to have a twelve-byte array in the driver. Regards, Clemens

3 5

[alsa-devel] Intel HDA / ca0132: Multichannel + S/PDIF support
by Stefan Oltmanns 06 Jan '15

06 Jan '15

Hi, I own a Gigabyte G1.Sniper M3 Z77 mainboard with ca0132 onboard audio. Currently only stereo out and headphone out (at the back, always same signal) work by default. I was able to fix S/PDIF out by changing the headphone pin in function ca0132_config to something below 0x0d, like: spec->out_pins[1] = 0x0a; No idea why this fixes S/PDIF, but no other effects noticed. No idea if it´s going to break something on other devices. Original value is 0x10. In this function also 3 DACs are listed. The board has a total of 5 stereo outputs: -Frontpanel (HDA pin connector on mainboard) 4x 3,5mm output at backpanel: -Stereo out -Headphone out -Rear out -Center/LFE out I tried to increase the number of channels and create additional outputs and connected to the other DACs. The maximum I´m able to is to get different signals out of stereo out and front headphone (not useful, just some weird mixes) or same signal from all 3 outputs. Rear and c/lfe out always no signal. During initialization snd_hda_parse_pin_def_config is called and two outputs are found. I tried to override that and change spec->autocfg after that function is called, no success. Also no matter what I change there, a little later I get this kernel messages: kernel: [ 2.642972] input: HDA Intel PCH Line Out as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 kernel: [ 2.643142] input: HDA Intel PCH Front Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input7 kernel: [ 2.644932] input: HDA Intel PCH Line as /devices/pci0000:00/0000:00:1b.0/sound/card0/input6 kernel: [ 2.645050] input: HDA Intel PCH Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input5 I could not find the code where this messages are generated, where is that? Any ideas what I could/should try to get multichannel support working? Is there any kind of documentation on the ca0132 that could be helpful? Best regards Stefan

3 8

[alsa-devel] [PATCH] ASoC: tegra: Add platform driver for rt5677 audio codec
by Anatol Pomozov 06 Jan '15

06 Jan '15

The driver supports NVIDIA Tegra Ryu board Sponsored: Google ChromeOS Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- .../bindings/sound/nvidia,tegra-audio-rt5677.txt | 67 ++++ sound/soc/tegra/Kconfig | 10 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_rt5677.c | 351 +++++++++++++++++++++ 4 files changed, 430 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt create mode 100644 sound/soc/tegra/tegra_rt5677.c diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt new file mode 100644 index 0000000..a4589cd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt @@ -0,0 +1,67 @@ +NVIDIA Tegra audio complex, with RT5677 CODEC + +Required properties: +- compatible : "nvidia,tegra-audio-rt5677" +- clocks : Must contain an entry for each entry in clock-names. + See ../clocks/clock-bindings.txt for details. +- clock-names : Must include the following entries: + - pll_a + - pll_a_out0 + - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) +- nvidia,model : The user-visible name of this sound complex. +- nvidia,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 for sources and + sinks are the RT5677's pins (as documented in its binding), and the jacks + on the board: + + * Headphone + * Speaker + * Headset Mic + * Internal Mic 1 + * Internal Mic 2 + +- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's + connected to the CODEC. +- nvidia,audio-codec : The phandle of the RT5677 audio codec. This binding + assumes that AIF1 on the CODEC is connected to Tegra. + +Optional properties: +- nvidia,hp-det-gpios : The GPIO that detects headphones are plugged in +- nvidia,hp-en-gpios : The GPIO that enables headphone amplifier +- nvidia,mic-present-gpios: The GPIO that mic jack is plugged in +- nvidia,dmic-clk-en-gpios : The GPIO that gates DMIC clock signal + +Example: + +sound { + compatible = "nvidia,tegra-audio-rt5677-ryu", + "nvidia,tegra-audio-rt5677"; + nvidia,model = "NVIDIA Tegra Ryu"; + + nvidia,audio-routing = + "Headphone", "LOUT2", + "Headphone", "LOUT1", + "Headset Mic", "MICBIAS1", + "IN1P", "Headset Mic", + "IN1N", "Headset Mic", + "DMIC L1", "Internal Mic 1", + "DMIC R1", "Internal Mic 1", + "DMIC L2", "Internal Mic 2", + "DMIC R2", "Internal Mic 2", + "Speaker", "PDM1L", + "Speaker", "PDM1R"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&rt5677>; + + nvidia,hp-det-gpios = <&gpio TEGRA_GPIO(R, 7) GPIO_ACTIVE_HIGH>; + nvidia,mic-present-gpios = <&gpio TEGRA_GPIO(O, 5) GPIO_ACTIVE_LOW>; + nvidia,hp-en-gpios = <&rt5677 1 GPIO_ACTIVE_HIGH>; + nvidia,dmic-clk-en-gpios = <&rt5677 2 GPIO_ACTIVE_HIGH>; + + clocks = <&tegra_car TEGRA124_CLK_PLL_A>, + <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA124_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; +}; diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 31198cf7..a6768f8 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -128,3 +128,13 @@ config SND_SOC_TEGRA_MAX98090 help Say Y or M here if you want to add support for SoC audio on Tegra boards using the MAX98090 codec, such as Venice2. + +config SND_SOC_TEGRA_RT5677 + tristate "SoC Audio support for Tegra boards using a RT5677 codec" + depends on SND_SOC_TEGRA && I2C && GPIOLIB + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC + select SND_SOC_RT5677 + help + Say Y or M here if you want to add support for SoC audio on Tegra + boards using the RT5677 codec, such as Ryu. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 5ae588c..9171655 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -19,6 +19,7 @@ obj-$(CONFIG_SND_SOC_TEGRA30_I2S) += snd-soc-tegra30-i2s.o # Tegra machine Support snd-soc-tegra-rt5640-objs := tegra_rt5640.o +snd-soc-tegra-rt5677-objs := tegra_rt5677.o snd-soc-tegra-wm8753-objs := tegra_wm8753.o snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-wm9712-objs := tegra_wm9712.o @@ -27,6 +28,7 @@ snd-soc-tegra-alc5632-objs := tegra_alc5632.o snd-soc-tegra-max98090-objs := tegra_max98090.o obj-$(CONFIG_SND_SOC_TEGRA_RT5640) += snd-soc-tegra-rt5640.o +obj-$(CONFIG_SND_SOC_TEGRA_RT5677) += snd-soc-tegra-rt5677.o obj-$(CONFIG_SND_SOC_TEGRA_WM8753) += snd-soc-tegra-wm8753.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_WM9712) += snd-soc-tegra-wm9712.o diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c new file mode 100644 index 0000000..6e98d10 --- /dev/null +++ b/sound/soc/tegra/tegra_rt5677.c @@ -0,0 +1,351 @@ +/* +* tegra_rt5677.c - Tegra machine ASoC driver for boards using RT5677 codec. + * + * Copyright (c) 2014, The Chromium OS Authors. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + * Based on code copyright/by: + * + * Copyright (C) 2010-2012 - NVIDIA, Inc. + * Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> + * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd. + * Copyright 2007 Wolfson Microelectronics PLC. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../codecs/rt5677.h" + +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-snd-rt5677" + +struct tegra_rt5677 { + struct tegra_asoc_utils_data util_data; + int gpio_hp_det; + int gpio_hp_en; + int gpio_mic_present; + int gpio_dmic_clk_en; +}; + +static int tegra_rt5677_asoc_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_card *card = rtd->card; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + int srate, mclk, err; + + srate = params_rate(params); + mclk = 256 * srate; + + err = tegra_asoc_utils_set_rate(&machine->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, RT5677_SCLK_S_MCLK, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static int tegra_rt5677_event_hp(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + if (!gpio_is_valid(machine->gpio_hp_en)) + return 0; + + gpio_set_value_cansleep(machine->gpio_hp_en, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + +static struct snd_soc_ops tegra_rt5677_ops = { + .hw_params = tegra_rt5677_asoc_hw_params, +}; + +static struct snd_soc_jack tegra_rt5677_hp_jack; + +static struct snd_soc_jack_pin tegra_rt5677_hp_jack_pins = { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, +}; +static struct snd_soc_jack_gpio tegra_rt5677_hp_jack_gpio = { + .name = "Headphone detection", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, +}; + +static struct snd_soc_jack tegra_rt5677_mic_jack; + +static struct snd_soc_jack_pin tegra_rt5677_mic_jack_pins = { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, +}; + +static struct snd_soc_jack_gpio tegra_rt5677_mic_jack_gpio = { + .name = "Headset Mic detection", + .report = SND_JACK_MICROPHONE, + .debounce_time = 150, + .invert = 1 +}; + +static const struct snd_soc_dapm_widget tegra_rt5677_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_HP("Headphone", tegra_rt5677_event_hp), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Internal Mic 1", NULL), + SND_SOC_DAPM_MIC("Internal Mic 2", NULL), +}; + +static const struct snd_kcontrol_new tegra_rt5677_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Internal Mic 1"), + SOC_DAPM_PIN_SWITCH("Internal Mic 2"), +}; + +static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(rtd->card); + + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, + &tegra_rt5677_hp_jack); + snd_soc_jack_add_pins(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_pins); + + if (gpio_is_valid(machine->gpio_hp_det)) { + tegra_rt5677_hp_jack_gpio.gpio = machine->gpio_hp_det; + snd_soc_jack_add_gpios(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_gpio); + } + + + snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_rt5677_mic_jack); + snd_soc_jack_add_pins(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_pins); + + if (gpio_is_valid(machine->gpio_mic_present)) { + tegra_rt5677_mic_jack_gpio.gpio = machine->gpio_mic_present; + snd_soc_jack_add_gpios(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_gpio); + } + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + + return 0; +} + +static int tegra_rt5677_card_remove(struct snd_soc_card *card) +{ + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(machine->gpio_hp_det)) { + snd_soc_jack_free_gpios(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_gpio); + } + + if (gpio_is_valid(machine->gpio_mic_present)) { + snd_soc_jack_free_gpios(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_gpio); + } + + return 0; +} + +static struct snd_soc_dai_link tegra_rt5677_dai = { + .name = "RT5677", + .stream_name = "RT5677 PCM", + .codec_dai_name = "rt5677-aif1", + .init = tegra_rt5677_asoc_init, + .ops = &tegra_rt5677_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_tegra_rt5677 = { + .name = "tegra-rt5677", + .owner = THIS_MODULE, + .remove = tegra_rt5677_card_remove, + .dai_link = &tegra_rt5677_dai, + .num_links = 1, + .controls = tegra_rt5677_controls, + .num_controls = ARRAY_SIZE(tegra_rt5677_controls), + .dapm_widgets = tegra_rt5677_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_rt5677_dapm_widgets), + .fully_routed = true, +}; + +static int tegra_rt5677_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct snd_soc_card *card = &snd_soc_tegra_rt5677; + struct tegra_rt5677 *machine; + int ret; + +#if defined(CONFIG_MODULES) && defined(MODULE) + request_module("snd_soc_tegra30_i2s"); +#endif + + machine = devm_kzalloc(&pdev->dev, + sizeof(struct tegra_rt5677), GFP_KERNEL); + if (!machine) + return -ENOMEM; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + + machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + if (machine->gpio_hp_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + machine->gpio_mic_present = of_get_named_gpio(np, + "nvidia,mic-present-gpios", 0); + if (machine->gpio_mic_present == -EPROBE_DEFER) + return -EPROBE_DEFER; + + machine->gpio_hp_en = of_get_named_gpio(np, "nvidia,hp-en-gpios", 0); + if (machine->gpio_hp_en == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (gpio_is_valid(machine->gpio_hp_en)) { + ret = devm_gpio_request_one(&pdev->dev, machine->gpio_hp_en, + GPIOF_OUT_INIT_LOW, "hp_en"); + if (ret) { + dev_err(card->dev, "cannot get hp_en gpio\n"); + return ret; + } + } + + machine->gpio_dmic_clk_en = of_get_named_gpio(np, + "nvidia,dmic-clk-en-gpios", 0); + if (machine->gpio_dmic_clk_en == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (gpio_is_valid(machine->gpio_dmic_clk_en)) { + ret = devm_gpio_request_one(&pdev->dev, + machine->gpio_dmic_clk_en, + GPIOF_OUT_INIT_HIGH, "dmic_clk_en"); + if (ret) { + dev_err(card->dev, "cannot get dmic_clk_en gpio\n"); + return ret; + } + } + + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing"); + if (ret) + goto err; + + tegra_rt5677_dai.codec_of_node = of_parse_phandle(np, + "nvidia,audio-codec", 0); + if (!tegra_rt5677_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_rt5677_dai.cpu_of_node = of_parse_phandle(np, + "nvidia,i2s-controller", 0); + if (!tegra_rt5677_dai.cpu_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + tegra_rt5677_dai.platform_of_node = tegra_rt5677_dai.cpu_of_node; + + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); + if (ret) + goto err; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto err_fini_utils; + } + + return 0; + +err_fini_utils: + tegra_asoc_utils_fini(&machine->util_data); +err: + return ret; +} + +static int tegra_rt5677_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&machine->util_data); + + return 0; +} + +static const struct of_device_id tegra_rt5677_of_match[] = { + { .compatible = "nvidia,tegra-audio-rt5677", }, + {}, +}; + +static struct platform_driver tegra_rt5677_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = tegra_rt5677_of_match, + }, + .probe = tegra_rt5677_probe, + .remove = tegra_rt5677_remove, +}; +module_platform_driver(tegra_rt5677_driver); + +MODULE_AUTHOR("Anatol Pomozov <anatol(a)google.com>"); +MODULE_DESCRIPTION("Tegra+RT5677 machine ASoC driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_rt5677_of_match); -- 2.2.0.rc0.207.ga3a616c

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[alsa-devel] [PATCH 1/4] ASoC: wm8960: Let wm8960 codec driver manage its own MCLK
by Zidan Wang 06 Jan '15

06 Jan '15

When we want to use wm8960 codec, we should enable its MCLK in machine driver. It's reasonable for wm8960 codec driver to manage its own MCLK. When current bias_level is SND_SOC_BIAS_ON, it is preparing for a transition away from ON. In this case, disable the codec mclk. When current bias_level is not SND_SOC_BIAS_ON, it preparing for a transition to ON. In this case, enable the codec mclk. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- sound/soc/codecs/wm8960.c | 40 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 39 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 031a1ae..1a5f47b 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> #include <linux/i2c.h> #include <linux/slab.h> #include <sound/core.h> @@ -117,6 +118,7 @@ static bool wm8960_volatile(struct device *dev, unsigned int reg) } struct wm8960_priv { + struct clk *mclk; struct regmap *regmap; int (*set_bias_level)(struct snd_soc_codec *, enum snd_soc_bias_level level); @@ -618,6 +620,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int ret; switch (level) { case SND_SOC_BIAS_ON: @@ -626,6 +629,20 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: /* Set VMID to 2x50k */ snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); + + if (!IS_ERR(wm8960->mclk)) { + if (codec->dapm.bias_level == SND_SOC_BIAS_ON) + clk_disable_unprepare(wm8960->mclk); + else { + ret = clk_prepare_enable(wm8960->mclk); + if (ret) { + dev_err(codec->dev, + "Failed to enable MCLK: %d\n", + ret); + return ret; + } + } + } break; case SND_SOC_BIAS_STANDBY: @@ -661,6 +678,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, /* Disable VMID and VREF, let them discharge */ snd_soc_write(codec, WM8960_POWER1, 0); + msleep(600); break; } @@ -674,7 +692,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - int reg; + int reg, ret; switch (level) { case SND_SOC_BIAS_ON: @@ -715,9 +733,22 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, WM8960_VREF, WM8960_VREF); msleep(100); + + if (!IS_ERR(wm8960->mclk)) { + ret = clk_prepare_enable(wm8960->mclk); + if (ret) { + dev_err(codec->dev, + "Failed to enable MCLK: %d\n", + ret); + return ret; + } + } break; case SND_SOC_BIAS_ON: + if (!IS_ERR(wm8960->mclk)) + clk_disable_unprepare(wm8960->mclk); + /* Enable anti-pop mode */ snd_soc_update_bits(codec, WM8960_APOP1, WM8960_POBCTRL | WM8960_SOFT_ST | @@ -728,6 +759,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, /* Disable VMID and VREF */ snd_soc_update_bits(codec, WM8960_POWER1, WM8960_VREF | WM8960_VMID_MASK, 0); + break; case SND_SOC_BIAS_OFF: @@ -1002,6 +1034,12 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (wm8960 == NULL) return -ENOMEM; + wm8960->mclk = devm_clk_get(&i2c->dev, "codec_mclk"); + if (IS_ERR(wm8960->mclk)) { + if (PTR_ERR(wm8960->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + wm8960->regmap = devm_regmap_init_i2c(i2c, &wm8960_regmap); if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); -- 1.9.1

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[alsa-devel] [PATCH v3] ALSA: sound/atmel/ac97c.c: Add device tree support
by Alexander Stein 06 Jan '15

06 Jan '15

This adds device tree support for the AC97 controller. It uses the soc-ac97link bindings, but actually only ac97-reset is used. Signed-off-by: Alexander Stein <alexanders83(a)web.de> Acked-By Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- Changes in v3: * Renamed the compatible string to atmel,at91sam9263-ac97c * Added 'Acked-By Alexandre Belloni <alexandre.belloni(a)free-electrons.com>' Changes in v2: * It now uses the soc-ac97link bindings (partly) instead of defining its own. This should ease the transition to ASoC once a new driver has been written. sound/atmel/ac97c.c | 45 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 42 insertions(+), 3 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 4f6b14d..a0df1b1 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -22,6 +22,9 @@ #include <linux/gpio.h> #include <linux/types.h> #include <linux/io.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/initval.h> @@ -902,6 +905,40 @@ static void atmel_ac97c_reset(struct atmel_ac97c *chip) } } +#ifdef CONFIG_OF +static const struct of_device_id atmel_ac97c_dt_ids[] = { + { .compatible = "atmel,at91sam9263-ac97c", }, + { } +}; +MODULE_DEVICE_TABLE(of, atmel_ac97c_dt_ids); + +static struct ac97c_platform_data *atmel_ac97c_probe_dt(struct device *dev) +{ + struct ac97c_platform_data *pdata; + struct device_node *node = dev->of_node; + const struct of_device_id *match; + + if (!node) { + dev_err(dev, "Device does not have associated DT data\n"); + return ERR_PTR(-EINVAL); + } + + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return ERR_PTR(-ENOMEM); + + pdata->reset_pin = of_get_named_gpio(dev->of_node, "ac97-gpios", 2); + + return pdata; +} +#else +static struct ac97c_platform_data *atmel_ac97c_probe_dt(struct device *dev) +{ + dev_err(dev, "no platform data defined\n"); + return ERR_PTR(-ENXIO); +} +#endif + static int atmel_ac97c_probe(struct platform_device *pdev) { struct snd_card *card; @@ -922,10 +959,11 @@ static int atmel_ac97c_probe(struct platform_device *pdev) return -ENXIO; } - pdata = pdev->dev.platform_data; + pdata = dev_get_platdata(&pdev->dev); if (!pdata) { - dev_dbg(&pdev->dev, "no platform data\n"); - return -ENXIO; + pdata = atmel_ac97c_probe_dt(&pdev->dev); + if (IS_ERR(pdata)) + return PTR_ERR(pdata); } irq = platform_get_irq(pdev, 0); @@ -1204,6 +1242,7 @@ static struct platform_driver atmel_ac97c_driver = { .driver = { .name = "atmel_ac97c", .pm = ATMEL_AC97C_PM_OPS, + .of_match_table = of_match_ptr(atmel_ac97c_dt_ids), }, }; module_platform_driver(atmel_ac97c_driver); -- 2.2.1

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[alsa-devel] [PATCH v2 00/10] audio timestamping evolutions
by Pierre-Louis Bossart 05 Jan '15

05 Jan '15

This series of patches was inspired by recent threads on the alsa mailing list, as well issues detected with existing and upcoming hardware: 1. there was a request from the PulseAudio community to get more information from drivers to make rewinds safer. While the conclusion was that it's nearly impossible for a driver developer to provide this information, there are however ways to assess the granularity of the hw_ptr updates using timestamping capabilities, and indirectly understand how safe rewinds might be. 2. There was also a request to add a start_at capability based either on system hr timers or a wall clock, which requires a means to expose both types of information to applications. Rather than adding new sets of timestamps, it is suggested the new start_at functionality relies on the new definition provides by these patches 3. For new hardware, there is a neeed to let low-level drivers handle timestamping instead of having the ALSA core do it. Similarly there is a need to let the low-level driver update the initial estimate for the trigger timestamp if there are delays to start a stream (eg. with USB) These patches try to provide an answer to these multiple needs by building on the work done two years ago to expose wall clock information to applications. The evolution is to let application select which audio timestamp they are interested in, track the delay and drifts between recurring measurements and get, when possible, an idea of the accuracy of the underlying hardware. A backwards compatible mode is provided in case userspace doesn't provide any timestamp selection (results based on HDAudio wallclock for playback, hw_ptr in all other cases). The first 4 patches are corrections for misses in the way the system and trigger timestamps are handled, and the last 6 provide the additional audio timestamping selection. A second batch is planned to enable hardware capabilities in a low-level drivers. A corresponding set of patches is available for alsa-lib. V2 changes: trigger_tstamp: trigger_tstamp_latched, pending redefined as bool trigger_tstamp_latched reset in snd_pcm_pre_start() audio_ts_config, report: keep separate structure but use different bitfields for in and out. use u32 instead of __u32, add comments that these structures are internal COMPAT backwards compatible mode, uses WALL_CLOCK/LINK for HDAudio playback and DEFAULT (hw_ptr) everywhere else INFO bits: reclaimed 32-bits from hw_params, renamed as info_ext moved all timestamp info to info_ext snd_pcm_status: read only 32-bit audio_tstamp_data, ignore all other fields Pierre-Louis Bossart (10): ALSA: core: don't override timestamp unconditionally ALSA: core: allow for trigger_tstamp snapshot in .trigger ALSA: hda: read trigger_timestamp immediately after starting DMA ALSA: usb: update trigger timestamp on first non-zero URB submitted ALSA: core: add info_ext field in hw_params and pcm_hardware ALSA: core: selection of audio_tstamp type and accuracy reports ALSA: core: pass audio tstamp config from userspace ALSA: core: pass audio tstamp config from userspace in compat mode ALSA: core: replace .wall_clock by .get_time_info ALSA: hda: replace .wallclock by .get_time_info Documentation/sound/alsa/timestamping.txt | 176 ++++++++++++++++++++++++++++++ include/sound/pcm.h | 71 +++++++++++- include/uapi/sound/asound.h | 31 +++++- sound/core/pcm_compat.c | 15 ++- sound/core/pcm_lib.c | 81 +++++++++----- sound/core/pcm_native.c | 41 ++++++- sound/pci/hda/hda_controller.c | 44 ++++++-- sound/usb/pcm.c | 9 ++ 8 files changed, 417 insertions(+), 51 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt -- 1.9.1

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[alsa-devel] [PATCH] ASoC: adi: Add missing return statement.
by Andrew Jackson 05 Jan '15

05 Jan '15

From: Andrew Jackson <Andrew.Jackson(a)arm.com> The probe routine was disabling the clock even if the system was configured successfully. Add a return statement to leave clocks enabled. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- Spotted while reviewing clock preparation sound/soc/adi/axi-i2s.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index 7752860..4c23381 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -240,6 +240,8 @@ static int axi_i2s_probe(struct platform_device *pdev) if (ret) goto err_clk_disable; + return 0; + err_clk_disable: clk_disable_unprepare(i2s->clk); return ret; -- 1.7.1

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[alsa-devel] [patch] ALSA: asihpi: fix an information leak in asihpi_hpi_ioctl()
by Dan Carpenter 05 Jan '15

05 Jan '15

So far as I can see "hr->h.size" is set to "res_max_size" which is a user controlled value between 12 and USHRT_MAX. If it's larger than sizeof(*hr), then that leads to an information leak. I am not very familiar with this code, my other question here is that on lines before we set "hr->h.size = sizeof(hr->h)". It think this is a bug. I also think this particular code is never executed and I added a comment to that effect. But we do it in earlier in the function as well: copy_to_user(puhr, hr, sizeof(hr->h)); It doesn't make sense to me. Anyway, I think my patch is safe and it seems to fix a real information leak. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 6aa677e..f88109a 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -283,6 +283,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) goto out; } + /* FIXME: isn't this a no-op? */ if (hr->h.size > res_max_size) { HPI_DEBUG_LOG(ERROR, "response too big %d %d\n", hr->h.size, res_max_size); @@ -290,6 +291,9 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) hr->h.specific_error = hr->h.size; hr->h.size = sizeof(hr->h); } + /* prevent an information leak */ + if (hr->h.size > sizeof(*hr)) + hr->h.size = sizeof(*hr); uncopied_bytes = copy_to_user(puhr, hr, hr->h.size); if (uncopied_bytes) {

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[alsa-devel] DSD over HDMI: audio data packetization
by Xuebing Wang 05 Jan '15

05 Jan '15

Hi Community, I am trying to implement DSD (one bit audio) over HDMI, for Intel integrated graphics. I am referencing HDMI specification 1.3a. Regarding audio data packetization, I am referencing spec: 5.3.4 Audio Sample Packet 7.6 Audio Data Packetization Q1) Would you please point to me where is the source code to handle L-PCM audio sample packetization (to implement 5.3.4 Audio Sample Packet)? Q2) I admit that I did not carefully read HD-Audio specification. Is this possible that this L-PCM audio data packetization is done by *hardware* (e.g HDMI related widgets/pins)? Q3) Did we implement "7.2 Audio Sample Clock Capture and Regeneration"? Thanks. Xuebing Wang

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[alsa-devel] [PATCH - ALSA: snd-virtuoso 1/1] Xonar DSX support
by sgiurgiu11＠gmail.com 04 Jan '15

04 Jan '15

From: Sergiu Giurgiu <sgiurgiu11(a)gmail.com> This patch adds support for ASUS - Xonar DSX sound cards. Tested on openSUSE 12.2 with kernel: Linux 3.4.6-2.10-desktop #1 SMP PREEMPT Thu Jul 26 09:36:26 UTC 2012 (641c197) x86_64 x86_64 x86_64 GNU/Linux Works: - play sounds - adjust volume on master channel. - mute . Since Xonar DS uses the same chip, everything that works for DS should work for DSX as well. Thank you, Sergiu Giurgiu. Signed-off-by: Sergiu Giurgiu <sgiurgiu11(a)gmail.com> diff --git a/mirror/sound/pci/oxygen/virtuoso.c b/mirror/sound/pci/oxygen/virtuoso.c index 3d71423..8104eab 100644 --- a/mirror/sound/pci/oxygen/virtuoso.c +++ b/mirror/sound/pci/oxygen/virtuoso.c @@ -52,6 +52,7 @@ static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, { OXYGEN_PCI_SUBID(0x1043, 0x835e) }, { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8522) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; diff --git a/mirror/sound/pci/oxygen/xonar_wm87x6.c b/mirror/sound/pci/oxygen/xonar_wm87x6.c index 63cff90..b555b62 100644 --- a/mirror/sound/pci/oxygen/xonar_wm87x6.c +++ b/mirror/sound/pci/oxygen/xonar_wm87x6.c @@ -1255,7 +1255,6 @@ static void dump_wm87x6_registers(struct oxygen *chip, } static const struct oxygen_model model_xonar_ds = { - .shortname = "Xonar DS", .longname = "Asus Virtuoso 66", .chip = "AV200", .init = xonar_ds_init, @@ -1327,6 +1326,11 @@ int __devinit get_xonar_wm87x6_model(struct oxygen *chip, switch (id->subdevice) { case 0x838e: chip->model = model_xonar_ds; + chip->model.shortname = "Xonar DS"; + break; + case 0x8522: + chip->model = model_xonar_ds; + chip->model.shortname = "Xonar DSX"; break; case 0x835e: chip->model = model_xonar_hdav_slim; -- 1.7.10.4

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