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

[alsa-devel] [PATCH] ASoC: alc5623: Fix Kconfig dependency
by Takashi Iwai 26 May '14

26 May '14

Add "depends on I2C" to shut up the build errors from randconfig. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c59943a19f94..3f7bbe61326d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -274,6 +274,7 @@ config SND_SOC_AK5386 config SND_SOC_ALC5623 tristate "Realtek ALC5623 CODEC" + depends on I2C config SND_SOC_ALC5632 tristate -- 1.9.3

2 1

[alsa-devel] [PATCH] ASoC: omap-pcm: Move omap-pcm under include/sound
by Jyri Sarha 26 May '14

26 May '14

Make including the omap-pcm.h outside sound/soc/omap more convenient. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- It would be nice to get this applied at ASAP for OMAP HDMI audio rework patches depend on this and it would help a lot if this would already be there when applying to OMAPDSS side patches. include/sound/omap-pcm.h | 30 ++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-mcasp.c | 2 +- sound/soc/omap/omap-dmic.c | 2 +- sound/soc/omap/omap-hdmi.c | 2 +- sound/soc/omap/omap-mcbsp.c | 2 +- sound/soc/omap/omap-mcpdm.c | 2 +- sound/soc/omap/omap-pcm.h | 30 ------------------------------ 7 files changed, 35 insertions(+), 35 deletions(-) create mode 100644 include/sound/omap-pcm.h delete mode 100644 sound/soc/omap/omap-pcm.h diff --git a/include/sound/omap-pcm.h b/include/sound/omap-pcm.h new file mode 100644 index 0000000..c1d2f31 --- /dev/null +++ b/include/sound/omap-pcm.h @@ -0,0 +1,30 @@ +/* + * omap-pcm.h - OMAP PCM driver + * + * Copyright (C) 2014 Texas Instruments, Inc. + * + * Author: Peter Ujfalusi <peter.ujfalusi(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * 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 + * General Public License for more details. + */ + +#ifndef __OMAP_PCM_H__ +#define __OMAP_PCM_H__ + +#if IS_ENABLED(CONFIG_SND_OMAP_SOC) +int omap_pcm_platform_register(struct device *dev); +#else +static inline int omap_pcm_platform_register(struct device *dev) +{ + return 0; +} +#endif /* CONFIG_SND_OMAP_SOC */ + +#endif /* __OMAP_PCM_H__ */ diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 14058dc..9afb146 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -33,10 +33,10 @@ #include <sound/initval.h> #include <sound/soc.h> #include <sound/dmaengine_pcm.h> +#include <sound/omap-pcm.h> #include "davinci-pcm.h" #include "davinci-mcasp.h" -#include "../omap/omap-pcm.h" #define MCASP_MAX_AFIFO_DEPTH 64 diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 53da041..6925d71 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -40,9 +40,9 @@ #include <sound/initval.h> #include <sound/soc.h> #include <sound/dmaengine_pcm.h> +#include <sound/omap-pcm.h> #include "omap-dmic.h" -#include "omap-pcm.h" struct omap_dmic { struct device *dev; diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c index 537a1ec..eb9c392 100644 --- a/sound/soc/omap/omap-hdmi.c +++ b/sound/soc/omap/omap-hdmi.c @@ -34,9 +34,9 @@ #include <sound/asoundef.h> #include <sound/dmaengine_pcm.h> #include <video/omapdss.h> +#include <sound/omap-pcm.h> #include "omap-hdmi.h" -#include "omap-pcm.h" #define DRV_NAME "omap-hdmi-audio-dai" diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 71d2266..efe2cd6 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -34,11 +34,11 @@ #include <sound/initval.h> #include <sound/soc.h> #include <sound/dmaengine_pcm.h> +#include <sound/omap-pcm.h> #include <linux/platform_data/asoc-ti-mcbsp.h> #include "mcbsp.h" #include "omap-mcbsp.h" -#include "omap-pcm.h" #define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index d8ebb52..f0e2ebe 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -40,9 +40,9 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/dmaengine_pcm.h> +#include <sound/omap-pcm.h> #include "omap-mcpdm.h" -#include "omap-pcm.h" struct mcpdm_link_config { u32 link_mask; /* channel mask for the direction */ diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h deleted file mode 100644 index c1d2f31..0000000 --- a/sound/soc/omap/omap-pcm.h +++ /dev/null @@ -1,30 +0,0 @@ -/* - * omap-pcm.h - OMAP PCM driver - * - * Copyright (C) 2014 Texas Instruments, Inc. - * - * Author: Peter Ujfalusi <peter.ujfalusi(a)ti.com> - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * 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 - * General Public License for more details. - */ - -#ifndef __OMAP_PCM_H__ -#define __OMAP_PCM_H__ - -#if IS_ENABLED(CONFIG_SND_OMAP_SOC) -int omap_pcm_platform_register(struct device *dev); -#else -static inline int omap_pcm_platform_register(struct device *dev) -{ - return 0; -} -#endif /* CONFIG_SND_OMAP_SOC */ - -#endif /* __OMAP_PCM_H__ */ -- 1.7.9.5

3 2

[alsa-devel] [PATCH 1/2] ASoC: jack: Basic GPIO descriptor conversion
by Jarkko Nikula 26 May '14

26 May '14

This patch does basic GPIO descriptor conversion to soc-jack. Even the GPIOs are still passed and requested using legacy GPIO numbers the driver internals are converted to use GPIO descriptor API. Motivation for this is to prepare soc-jack so that it will allow registering jack GPIO pins using both GPIO descriptors and legacy GPIO numbers. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- include/sound/soc.h | 1 + sound/soc/soc-jack.c | 19 +++++++++++-------- 2 files changed, 12 insertions(+), 8 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index dcdcc95efd1d..e9ec645911d9 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -607,6 +607,7 @@ struct snd_soc_jack_gpio { struct snd_soc_jack *jack; struct delayed_work work; + struct gpio_desc *desc; void *data; int (*jack_status_check)(void *data); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index b903f822d1b2..7203842fea66 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -14,6 +14,7 @@ #include <sound/jack.h> #include <sound/soc.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/interrupt.h> #include <linux/workqueue.h> #include <linux/delay.h> @@ -240,7 +241,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) int enable; int report; - enable = gpio_get_value_cansleep(gpio->gpio); + enable = gpiod_get_value_cansleep(gpio->desc); if (gpio->invert) enable = !enable; @@ -314,14 +315,16 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, if (ret) goto undo; - ret = gpio_direction_input(gpios[i].gpio); + gpios[i].desc = gpio_to_desc(gpios[i].gpio); + + ret = gpiod_direction_input(gpios[i].desc); if (ret) goto err; INIT_DELAYED_WORK(&gpios[i].work, gpio_work); gpios[i].jack = jack; - ret = request_any_context_irq(gpio_to_irq(gpios[i].gpio), + ret = request_any_context_irq(gpiod_to_irq(gpios[i].desc), gpio_handler, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, @@ -331,7 +334,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, goto err; if (gpios[i].wake) { - ret = irq_set_irq_wake(gpio_to_irq(gpios[i].gpio), 1); + ret = irq_set_irq_wake(gpiod_to_irq(gpios[i].desc), 1); if (ret != 0) dev_err(jack->codec->dev, "ASoC: " "Failed to mark GPIO %d as wake source: %d\n", @@ -339,7 +342,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, } /* Expose GPIO value over sysfs for diagnostic purposes */ - gpio_export(gpios[i].gpio, false); + gpiod_export(gpios[i].desc, false); /* Update initial jack status */ schedule_delayed_work(&gpios[i].work, @@ -372,10 +375,10 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, int i; for (i = 0; i < count; i++) { - gpio_unexport(gpios[i].gpio); - free_irq(gpio_to_irq(gpios[i].gpio), &gpios[i]); + gpiod_unexport(gpios[i].desc); + free_irq(gpiod_to_irq(gpios[i].desc), &gpios[i]); cancel_delayed_work_sync(&gpios[i].work); - gpio_free(gpios[i].gpio); + gpiod_put(gpios[i].desc); gpios[i].jack = NULL; } } -- 2.0.0.rc2

2 3

[alsa-devel] [PATCH] ASoC: tegra: free jack GPIOs before the sound card is freed
by Stephen Warren 26 May '14

26 May '14

From: Stephen Warren <swarren(a)nvidia.com> snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to generate an initial jack status report. If sound card initialization fails, that work item needs to be cancelled, so it doesn't run after the card has been freed. Specifically, freeing the card calls snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which is called from the work queue item. snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine drivers do call this function in the platform driver remove() callback. However, this happens after the sound card is freed, at least when the card is freed due to errors late during snd_soc_instantiate_card(). This leaves a window where the work item can execute after the card is freed. In next-20140522, sound card initialization does fail for unrelated reasons, and hits the problem described above. To solve this, fix the Tegra ASoC machine drivers to clean up the Jack GPIOs during the snd_soc_card's .remove() callback, which is executed before the overall card object is freed. also, gGuard the cleanup call based on whether we actually setup up the GPIOs in the first place. Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove function to match where the GPIOs get set up. However, there is no such callback. This change fixes all Tegra machine drivers. By code inspection, I believe some non-Tegra machine drivers have the same issue. I'll send a patch for that separately, once this is reviewed. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Should this be Cc: stable? --- sound/soc/tegra/tegra_alc5632.c | 16 +++++++++++++--- sound/soc/tegra/tegra_max98090.c | 16 +++++++++++++--- sound/soc/tegra/tegra_rt5640.c | 16 +++++++++++++--- sound/soc/tegra/tegra_wm8903.c | 11 ++++++----- 4 files changed, 45 insertions(+), 14 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index c61ea3a1030f..02734bd4f09b 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -125,6 +125,18 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int tegra_alc5632_card_remove(struct snd_soc_card *card) +{ + struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(machine->gpio_hp_det)) { + snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack, 1, + &tegra_alc5632_hp_jack_gpio); + } + + return 0; +} + static struct snd_soc_dai_link tegra_alc5632_dai = { .name = "ALC5632", .stream_name = "ALC5632 PCM", @@ -139,6 +151,7 @@ static struct snd_soc_dai_link tegra_alc5632_dai = { static struct snd_soc_card snd_soc_tegra_alc5632 = { .name = "tegra-alc5632", .owner = THIS_MODULE, + .remove = tegra_alc5632_card_remove, .dai_link = &tegra_alc5632_dai, .num_links = 1, .controls = tegra_alc5632_controls, @@ -223,9 +236,6 @@ static int tegra_alc5632_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(card); - snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack, 1, - &tegra_alc5632_hp_jack_gpio); - snd_soc_unregister_card(card); tegra_asoc_utils_fini(&machine->util_data); diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index 0283cfb7c031..ce73e1f62c4b 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -145,6 +145,18 @@ static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int tegra_max98090_card_remove(struct snd_soc_card *card) +{ + struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(machine->gpio_hp_det)) { + snd_soc_jack_free_gpios(&tegra_max98090_hp_jack, 1, + &tegra_max98090_hp_jack_gpio); + } + + return 0; +} + static struct snd_soc_dai_link tegra_max98090_dai = { .name = "max98090", .stream_name = "max98090 PCM", @@ -158,6 +170,7 @@ static struct snd_soc_dai_link tegra_max98090_dai = { static struct snd_soc_card snd_soc_tegra_max98090 = { .name = "tegra-max98090", .owner = THIS_MODULE, + .remove = tegra_max98090_card_remove, .dai_link = &tegra_max98090_dai, .num_links = 1, .controls = tegra_max98090_controls, @@ -241,9 +254,6 @@ static int tegra_max98090_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card); - snd_soc_jack_free_gpios(&tegra_max98090_hp_jack, 1, - &tegra_max98090_hp_jack_gpio); - snd_soc_unregister_card(card); tegra_asoc_utils_fini(&machine->util_data); diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index 4511c5a875ec..4feb16a99e02 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -128,6 +128,18 @@ static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int tegra_rt5640_card_remove(struct snd_soc_card *card) +{ + struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(machine->gpio_hp_det)) { + snd_soc_jack_free_gpios(&tegra_rt5640_hp_jack, 1, + &tegra_rt5640_hp_jack_gpio); + } + + return 0; +} + static struct snd_soc_dai_link tegra_rt5640_dai = { .name = "RT5640", .stream_name = "RT5640 PCM", @@ -141,6 +153,7 @@ static struct snd_soc_dai_link tegra_rt5640_dai = { static struct snd_soc_card snd_soc_tegra_rt5640 = { .name = "tegra-rt5640", .owner = THIS_MODULE, + .remove = tegra_rt5640_card_remove, .dai_link = &tegra_rt5640_dai, .num_links = 1, .controls = tegra_rt5640_controls, @@ -224,9 +237,6 @@ static int tegra_rt5640_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card); - snd_soc_jack_free_gpios(&tegra_rt5640_hp_jack, 1, - &tegra_rt5640_hp_jack_gpio); - snd_soc_unregister_card(card); tegra_asoc_utils_fini(&machine->util_data); diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 4ac73730d79a..0939661df60b 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -206,6 +206,12 @@ static int tegra_wm8903_remove(struct snd_soc_card *card) struct snd_soc_pcm_runtime *rtd = &(card->rtd[0]); struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; + struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(machine->gpio_hp_det)) { + snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack, 1, + &tegra_wm8903_hp_jack_gpio); + } wm8903_mic_detect(codec, NULL, 0, 0); @@ -228,9 +234,7 @@ static struct snd_soc_card snd_soc_tegra_wm8903 = { .owner = THIS_MODULE, .dai_link = &tegra_wm8903_dai, .num_links = 1, - .remove = tegra_wm8903_remove, - .controls = tegra_wm8903_controls, .num_controls = ARRAY_SIZE(tegra_wm8903_controls), .dapm_widgets = tegra_wm8903_dapm_widgets, @@ -368,9 +372,6 @@ static int tegra_wm8903_driver_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); - snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack, 1, - &tegra_wm8903_hp_jack_gpio); - snd_soc_unregister_card(card); tegra_asoc_utils_fini(&machine->util_data); -- 1.8.1.5

2 1

[alsa-devel] [PATCH] ASoC: Intel: avoid format string leak to thread name
by Kees Cook 26 May '14

26 May '14

This makes sure a format string can never get processed into the worker thread name from the device name. Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/soc/intel/sst-baytrail-ipc.c | 2 +- sound/soc/intel/sst-haswell-ipc.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index 0d31dbbf4806..1b25bf168beb 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -809,7 +809,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) /* start the IPC message thread */ init_kthread_worker(&byt->kworker); byt->tx_thread = kthread_run(kthread_worker_fn, - &byt->kworker, + &byt->kworker, "%s", dev_name(byt->dev)); if (IS_ERR(byt->tx_thread)) { err = PTR_ERR(byt->tx_thread); diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index e7996b39a484..a8fd60c67341 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1735,7 +1735,7 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) /* start the IPC message thread */ init_kthread_worker(&hsw->kworker); hsw->tx_thread = kthread_run(kthread_worker_fn, - &hsw->kworker, + &hsw->kworker, "%s", dev_name(hsw->dev)); if (IS_ERR(hsw->tx_thread)) { ret = PTR_ERR(hsw->tx_thread); -- 1.7.9.5 -- Kees Cook Chrome OS Security

3 2

[alsa-devel] [PATCH v2 0/9] DT instantiation of sound on T5325
by Andrew Lunn 26 May '14

26 May '14

This patchset uses DT to instantiate sound on the T5325. It adds a DT binding to the codec, then removes the platform device for the codec and adds a DT node. simple-card is extended to support mlck_fs. The platform device for the audio device is then replaced with a DT node. Lastly, two _defconfig's are updated to build the needed modules. Andrew Lunn (9): ASoC: alc5623: Add device tree binding ARM: Kirkwood: Remove platform driver for codec ARM: Kirkwood: Add node for audio codec ARM: Kirkwood: DT: Add missing #sound-dai-cells property ASoC: simple-card: Support setting mclk via a fixed factor ARM: Kirkwood: t5325: Remove platform device to instantiate audio ARM: Kirkwood: t5325: Use simple card to instantiate audio ARM: mvebu_v5_defconfig: Enable sound modules needed for t5325 ARM: multi_v5_defconfig: Enable sound modules needed for t5325 .../devicetree/bindings/sound/alc5623.txt | 25 +++++++++++++ .../devicetree/bindings/sound/simple-card.txt | 2 ++ arch/arm/boot/dts/kirkwood-t5325.dts | 33 +++++++++++++++++ arch/arm/boot/dts/kirkwood.dtsi | 1 + arch/arm/configs/multi_v5_defconfig | 3 +- arch/arm/configs/mvebu_v5_defconfig | 3 +- arch/arm/mach-mvebu/Kconfig | 7 ---- arch/arm/mach-mvebu/Makefile | 1 - arch/arm/mach-mvebu/board-t5325.c | 41 ---------------------- arch/arm/mach-mvebu/board.h | 6 ---- arch/arm/mach-mvebu/kirkwood.c | 3 -- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/alc5623.c | 20 +++++++++++ sound/soc/generic/simple-card.c | 28 +++++++++++++++ 14 files changed, 114 insertions(+), 61 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/alc5623.txt delete mode 100644 arch/arm/mach-mvebu/board-t5325.c -- 2.0.0.rc0

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[alsa-devel] [PATCH] ASoC: max98090: Add NI/MI values for user pclk 19.2 MHz
by Sylwester Nawrocki 26 May '14

26 May '14

From: Chen Zhen <zhen1.chen(a)samsung.com> This patch adds the clock divisor and multiplier NI, MI values for audio sampling frequencies 44100 and 48000 Hz and PCLK 19.2 MHz. This is useful for the Odroid X2/U2 boards when the codec works in master mode and its MCLK clock is fed from the I2S CDCLK output. Signed-off-by: Chen Zhen <zhen1.chen(a)samsung.com> [s.nawrocki(a)samsung.com: edited the commit description] Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/codecs/max98090.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index f7b0b37..ea44471 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1544,19 +1544,19 @@ static const int lrclk_rates[] = { }; static const int user_pclk_rates[] = { - 13000000, 13000000 + 13000000, 13000000, 19200000, 19200000, }; static const int user_lrclk_rates[] = { - 44100, 48000 + 44100, 48000, 44100, 48000, }; static const unsigned long long ni_value[] = { - 3528, 768 + 3528, 768, 441, 8 }; static const unsigned long long mi_value[] = { - 8125, 1625 + 8125, 1625, 1500, 25 }; static void max98090_configure_bclk(struct snd_soc_codec *codec) -- 1.7.9.5

2 1

[alsa-devel] [PATCH v4] ASoC: sgtl5000: Fix the cache handling
by Fabio Estevam 26 May '14

26 May '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Since commit e5d80e82e32e (ASoC: sgtl5000: Convert to use regmap directly) a kernel oops is observed after a suspend/resume sequence. According to Mark Brown: "Looking at the code what I'd expect to happen here is that set_bias_level() manages the cache enable, turning on cache only mode just before it turns the regulators off and restoring normal mode just after enabling them, then calling _restore_regs() after that. The resume call should just be a call to set_bias_level() then.." Do as suggested and also complete the cache array with the missing registers so that sgtl5000_restore_regs() can properly cache the all the registers it needs. Reported-by: Shawn Guo <shawn.guo(a)freescale.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v3: - Call sgtl5000_restore_regs from set_bias_level and let set_bias_level handle the cache handling. Changes since v2: - Use regmap_update_bits sound/soc/codecs/sgtl5000.c | 134 ++++++++++++++++++++++++-------------------- sound/soc/codecs/sgtl5000.h | 2 + 2 files changed, 76 insertions(+), 60 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 9626ee0..57bda9b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -36,18 +36,32 @@ /* default value of sgtl5000 registers */ static const struct reg_default sgtl5000_reg_defaults[] = { + { SGTL5000_CHIP_DIG_POWER, 0x0000 }, { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, { SGTL5000_CHIP_SSS_CTRL, 0x0010 }, + { SGTL5000_CHIP_ADCDAC_CTRL, 0x020c }, { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, + { SGTL5000_CHIP_ANA_ADC_CTRL, 0x0000 }, { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, { SGTL5000_CHIP_ANA_CTRL, 0x0111 }, + { SGTL5000_CHIP_LINREG_CTRL, 0x0000 }, + { SGTL5000_CHIP_REF_CTRL, 0x0000 }, + { SGTL5000_CHIP_MIC_CTRL, 0x0000 }, + { SGTL5000_CHIP_LINE_OUT_CTRL, 0x0000 }, { SGTL5000_CHIP_LINE_OUT_VOL, 0x0404 }, { SGTL5000_CHIP_ANA_POWER, 0x7060 }, { SGTL5000_CHIP_PLL_CTRL, 0x5000 }, + { SGTL5000_CHIP_CLK_TOP_CTRL, 0x0000 }, + { SGTL5000_CHIP_ANA_STATUS, 0x0000 }, + { SGTL5000_CHIP_SHORT_CTRL, 0x0000 }, + { SGTL5000_CHIP_ANA_TEST2, 0x0000 }, + { SGTL5000_DAP_CTRL, 0x0000 }, + { SGTL5000_DAP_PEQ, 0x0000 }, { SGTL5000_DAP_BASS_ENHANCE, 0x0040 }, { SGTL5000_DAP_BASS_ENHANCE_CTRL, 0x051f }, + { SGTL5000_DAP_AUDIO_EQ, 0x0000 }, { SGTL5000_DAP_SURROUND, 0x0040 }, { SGTL5000_DAP_EQ_BASS_BAND0, 0x002f }, { SGTL5000_DAP_EQ_BASS_BAND1, 0x002f }, @@ -55,6 +69,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND3, 0x002f }, { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, + { SGTL5000_DAP_MIX_CHAN, 0x0000 }, { SGTL5000_DAP_AVC_CTRL, 0x0510 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, @@ -897,6 +912,64 @@ static int ldo_regulator_remove(struct snd_soc_codec *codec) #endif /* + * restore all sgtl5000 registers, + * since a big hole between dap and regular registers, + * we will restore them respectively. + */ +static int sgtl5000_restore_regs(struct snd_soc_codec *codec) +{ + u16 reg; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + /* restore regular registers */ + for (reg = 0; reg <= SGTL5000_CHIP_SHORT_CTRL; reg += 2) { + + /* These regs should restore in particular order */ + if (reg == SGTL5000_CHIP_ANA_POWER || + reg == SGTL5000_CHIP_CLK_CTRL || + reg == SGTL5000_CHIP_LINREG_CTRL || + reg == SGTL5000_CHIP_LINE_OUT_CTRL || + reg == SGTL5000_CHIP_REF_CTRL) + continue; + + regmap_update_bits(sgtl5000->regmap, reg, reg, 0); + } + + /* restore dap registers */ + for (reg = SGTL5000_DAP_REG_OFFSET; reg < SGTL5000_MAX_REG_OFFSET; reg += 2) + regmap_update_bits(sgtl5000->regmap, reg, reg, 0); + + /* + * restore these regs according to the power setting sequence in + * sgtl5000_set_power_regs() and clock setting sequence in + * sgtl5000_set_clock(). + * + * The order of restore is: + * 1. SGTL5000_CHIP_CLK_CTRL MCLK_FREQ bits (1:0) should be restore after + * SGTL5000_CHIP_ANA_POWER PLL bits set + * 2. SGTL5000_CHIP_LINREG_CTRL should be set before + * SGTL5000_CHIP_ANA_POWER LINREG_D restored + * 3. SGTL5000_CHIP_REF_CTRL controls Analog Ground Voltage, + * prefer to resotre it after SGTL5000_CHIP_ANA_POWER restored + */ + regmap_update_bits(sgtl5000->regmap, SGTL5000_CHIP_LINREG_CTRL, + SGTL5000_CHIP_LINREG_CTRL, 0); + + regmap_update_bits(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, + SGTL5000_PLL_POWERUP_BIT, 0); + + regmap_update_bits(sgtl5000->regmap, SGTL5000_CHIP_CLK_CTRL, + SGTL5000_MCLK_FREQ_PLL, 0); + + regmap_update_bits(sgtl5000->regmap, SGTL5000_CHIP_REF_CTRL, + SGTL5000_CHIP_REF_CTRL, 0); + + regmap_update_bits(sgtl5000->regmap, SGTL5000_CHIP_LINE_OUT_CTRL, + SGTL5000_CHIP_LINE_OUT_CTRL, 0); + return 0; +} + +/* * set dac bias * common state changes: * startup: @@ -926,6 +999,7 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, udelay(10); regcache_cache_only(sgtl5000->regmap, false); + sgtl5000_restore_regs(codec); ret = regcache_sync(sgtl5000->regmap); if (ret != 0) { @@ -1068,71 +1142,11 @@ static int sgtl5000_suspend(struct snd_soc_codec *codec) return 0; } -/* - * restore all sgtl5000 registers, - * since a big hole between dap and regular registers, - * we will restore them respectively. - */ -static int sgtl5000_restore_regs(struct snd_soc_codec *codec) -{ - u16 *cache = codec->reg_cache; - u16 reg; - - /* restore regular registers */ - for (reg = 0; reg <= SGTL5000_CHIP_SHORT_CTRL; reg += 2) { - - /* These regs should restore in particular order */ - if (reg == SGTL5000_CHIP_ANA_POWER || - reg == SGTL5000_CHIP_CLK_CTRL || - reg == SGTL5000_CHIP_LINREG_CTRL || - reg == SGTL5000_CHIP_LINE_OUT_CTRL || - reg == SGTL5000_CHIP_REF_CTRL) - continue; - - snd_soc_write(codec, reg, cache[reg]); - } - - /* restore dap registers */ - for (reg = SGTL5000_DAP_REG_OFFSET; reg < SGTL5000_MAX_REG_OFFSET; reg += 2) - snd_soc_write(codec, reg, cache[reg]); - - /* - * restore these regs according to the power setting sequence in - * sgtl5000_set_power_regs() and clock setting sequence in - * sgtl5000_set_clock(). - * - * The order of restore is: - * 1. SGTL5000_CHIP_CLK_CTRL MCLK_FREQ bits (1:0) should be restore after - * SGTL5000_CHIP_ANA_POWER PLL bits set - * 2. SGTL5000_CHIP_LINREG_CTRL should be set before - * SGTL5000_CHIP_ANA_POWER LINREG_D restored - * 3. SGTL5000_CHIP_REF_CTRL controls Analog Ground Voltage, - * prefer to resotre it after SGTL5000_CHIP_ANA_POWER restored - */ - snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, - cache[SGTL5000_CHIP_LINREG_CTRL]); - - snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, - cache[SGTL5000_CHIP_ANA_POWER]); - - snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, - cache[SGTL5000_CHIP_CLK_CTRL]); - - snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, - cache[SGTL5000_CHIP_REF_CTRL]); - - snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - cache[SGTL5000_CHIP_LINE_OUT_CTRL]); - return 0; -} - static int sgtl5000_resume(struct snd_soc_codec *codec) { /* Bring the codec back up to standby to enable regulators */ sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* Restore registers by cached in memory */ - sgtl5000_restore_regs(codec); return 0; } #else diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 2f8c889..72b68c0 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -341,6 +341,8 @@ #define SGTL5000_ADC_POWERUP 0x0002 #define SGTL5000_LINE_OUT_POWERUP 0x0001 +#define SGTL5000_PLL_POWERUP_BIT 10 + /* * SGTL5000_CHIP_PLL_CTRL */ -- 1.8.3.2

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[alsa-devel] [PATCH] ASoC: fsl_ssi: Add suspend/resume support
by Fabio Estevam 26 May '14

26 May '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Doing a suspend/resume sequence while playing an audio track in the backgroung causes broken audio right after resume: root@freescale /$ aplay clarinet.wav & root@freescale /home$ Playing WAVE 'clarinet.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono root@freescale /home$ echo mem > /sys/power/state PM: Syncing filesystems ... done. Freezing user space processes ... (elapsed 0.002 seconds) done. Freezing remaining freezable tasks ... (elapsed 0.002 seconds) done. Suspending console(s) (use no_console_suspend to debug) PM: suspend of devices complete after 37.082 msecs PM: suspend devices took 0.040 seconds PM: late suspend of devices complete after 4.234 msecs PM: noirq suspend of devices complete after 4.618 msecs Disabling non-boot CPUs ... PM: noirq resume of devices complete after 4.013 msecs PM: early resume of devices complete after 4.000 msecs PM: resume of devices complete after 68.907 msecs PM: resume devices took 0.070 seconds Restarting tasks ... Suspended. Trying resume. Failed. Restarting stream. Done. Suspended. Trying resume. Failed. Restarting stream. Done. Suspended. Trying resume. Failed. Restarting stream. Done. Suspended. Trying resume. Failed. Restarting stream. Done. Suspended. Trying resume. Failed. Restarting stream. Done. Suspended. Trying resume. Failed. Restarting stream. Done. Suspended. Trying resume. Failed. Restarting stream. Done. .... Add SNDRV_PCM_TRIGGER_RESUME/SUSPEND cases so that we can gracefully handle system suspend/resume. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f233d91..090f449 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -874,6 +874,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fsl_ssi_tx_config(ssi_private, true); @@ -882,6 +883,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, break; case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fsl_ssi_tx_config(ssi_private, false); -- 1.8.3.2

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[alsa-devel] Recovering from DAI xrun condition
by Josef Ahmad 26 May '14

26 May '14

Hi, I see the ALSA framework is capable of detecting and recovering from ring buffer xrun conditions. What I'm not sure about is whether it is possible to recover from a DAI xrun error as well. Let's take the example of an I2S interface operating in tx mode. If the tx fifo overruns, one or more words will be discarded by the interface. If an odd number of words gets discarded, this may result in the left and right word being swapped. This is actually a quite common weakness in I2S controllers. Is there a mechanism to notify the ALSA subsystem that a DAI xrun condition has occurred? Regards Josef

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