May 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: snd-usb: fix stream info output in /proc
by Daniel Mack 21 May '12

21 May '12

Set some substream struct members to make the proc interface code work again. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-by: Felix Homann <linuxaudio(a)showlabor.de> --- sound/usb/pcm.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 2d3a04d..24839d9 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -474,6 +474,11 @@ unlock: mutex_unlock(&subs->stream->chip->shutdown_mutex); } + if (ret == 0) { + subs->interface = fmt->iface; + subs->altset_idx = fmt->altset_idx; + } + return ret; } @@ -1123,13 +1128,16 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: subs->data_endpoint->prepare_data_urb = prepare_playback_urb; subs->data_endpoint->retire_data_urb = retire_playback_urb; + subs->running = 1; return 0; case SNDRV_PCM_TRIGGER_STOP: stop_endpoints(subs, 0, 0, 0); + subs->running = 0; return 0; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: subs->data_endpoint->prepare_data_urb = NULL; subs->data_endpoint->retire_data_urb = NULL; + subs->running = 0; return 0; } @@ -1148,15 +1156,19 @@ int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int c return err; subs->data_endpoint->retire_data_urb = retire_capture_urb; + subs->running = 1; return 0; case SNDRV_PCM_TRIGGER_STOP: stop_endpoints(subs, 0, 0, 0); + subs->running = 0; return 0; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: subs->data_endpoint->retire_data_urb = NULL; + subs->running = 0; return 0; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: subs->data_endpoint->retire_data_urb = retire_capture_urb; + subs->running = 1; return 0; } -- 1.7.5.4

2 1

[alsa-devel] snd-usb-audio: Substream status not set with new streaming logic
by Felix Homann 21 May '12

21 May '12

With the new streaming logic subs->running will never be set to 1. So /proc/... will never show the streams as running. Formerly the status was set in endpoint.c. Daniel? Regards, Felix

2 1

[alsa-devel] [PATCH] sound: sh: Fix up namespace collision in sh_dac_audio.
by Paul Mundt 21 May '12

21 May '12

The module_platform_driver() conversion ended up tripping over the driver name, leading to confusion in the macro with regards to 'driver' being redefined. rename it to something slightly more suitable to avoid namespace collisions. sound/sh/sh_dac_audio.c:444:122: error: conflicting types for 'driver_init' include/linux/device.h:773:6: note: previous declaration of 'driver_init' was here make[3]: *** [sound/sh/sh_dac_audio.o] Error 1 Signed-off-by: Paul Mundt <lethal(a)linux-sh.org> --- sound/sh/sh_dac_audio.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index b11f82b..f8b01c7 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -433,7 +433,7 @@ probe_error: /* * "driver" definition */ -static struct platform_driver driver = { +static struct platform_driver sh_dac_driver = { .probe = snd_sh_dac_probe, .remove = snd_sh_dac_remove, .driver = { @@ -441,4 +441,4 @@ static struct platform_driver driver = { }, }; -module_platform_driver(driver); +module_platform_driver(sh_dac_driver);

2 1

[alsa-devel] [PATCH 1/3] of: Add prefix parameter to of_modalias_node().
by David Daney 20 May '12

20 May '12

From: David Daney <david.daney(a)cavium.com> When generating MODALIASes, it is convenient to add things like "spi:" or "i2c:" to the front of the strings. This allows the standard modprobe to find the right driver when automatically populating bus children from the device tree structure. Add a prefix parameter, and adjust callers. For of_register_spi_devices() use the "spi:" prefix. Signed-off-by: David Daney <david.daney(a)cavium.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Timur Tabi <timur(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org --- drivers/of/base.c | 22 ++++++++++++++++------ drivers/of/of_i2c.c | 2 +- drivers/of/of_spi.c | 2 +- include/linux/of.h | 3 ++- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/soc/fsl/p1022_ds.c | 2 +- 6 files changed, 22 insertions(+), 11 deletions(-) diff --git a/drivers/of/base.c b/drivers/of/base.c index 5806449..f05a520 100644 --- a/drivers/of/base.c +++ b/drivers/of/base.c @@ -575,26 +575,36 @@ EXPORT_SYMBOL(of_find_matching_node); /** * of_modalias_node - Lookup appropriate modalias for a device node * @node: pointer to a device tree node + * @prefix: prefix to be added to the compatible property, may be NULL * @modalias: Pointer to buffer that modalias value will be copied into * @len: Length of modalias value * - * Based on the value of the compatible property, this routine will attempt - * to choose an appropriate modalias value for a particular device tree node. - * It does this by stripping the manufacturer prefix (as delimited by a ',') - * from the first entry in the compatible list property. + * Based on the value of the compatible property, this routine will + * attempt to choose an appropriate modalias value for a particular + * device tree node. It does this by stripping the manufacturer + * prefix (as delimited by a ',') from the first entry in the + * compatible list property, and appending it to the prefix. * * This routine returns 0 on success, <0 on failure. */ -int of_modalias_node(struct device_node *node, char *modalias, int len) +int of_modalias_node(struct device_node *node, const char *prefix, + char *modalias, int len) { const char *compatible, *p; int cplen; + if (len < 1) + return -EINVAL; + compatible = of_get_property(node, "compatible", &cplen); if (!compatible || strlen(compatible) > cplen) return -ENODEV; p = strchr(compatible, ','); - strlcpy(modalias, p ? p + 1 : compatible, len); + if (prefix) + strlcpy(modalias, prefix, len); + else + modalias[0] = 0; + strlcat(modalias, p ? p + 1 : compatible, len); return 0; } EXPORT_SYMBOL_GPL(of_modalias_node); diff --git a/drivers/of/of_i2c.c b/drivers/of/of_i2c.c index f37fbeb..23b05ee 100644 --- a/drivers/of/of_i2c.c +++ b/drivers/of/of_i2c.c @@ -37,7 +37,7 @@ void of_i2c_register_devices(struct i2c_adapter *adap) dev_dbg(&adap->dev, "of_i2c: register %s\n", node->full_name); - if (of_modalias_node(node, info.type, sizeof(info.type)) < 0) { + if (of_modalias_node(node, NULL, info.type, sizeof(info.type)) < 0) { dev_err(&adap->dev, "of_i2c: modalias failure on %s\n", node->full_name); continue; diff --git a/drivers/of/of_spi.c b/drivers/of/of_spi.c index 6dbc074..c329c6d 100644 --- a/drivers/of/of_spi.c +++ b/drivers/of/of_spi.c @@ -42,7 +42,7 @@ void of_register_spi_devices(struct spi_master *master) } /* Select device driver */ - if (of_modalias_node(nc, spi->modalias, + if (of_modalias_node(nc, SPI_MODULE_PREFIX, spi->modalias, sizeof(spi->modalias)) < 0) { dev_err(&master->dev, "cannot find modalias for %s\n", nc->full_name); diff --git a/include/linux/of.h b/include/linux/of.h index fa7fb1d..ee34d76 100644 --- a/include/linux/of.h +++ b/include/linux/of.h @@ -233,7 +233,8 @@ extern int of_n_addr_cells(struct device_node *np); extern int of_n_size_cells(struct device_node *np); extern const struct of_device_id *of_match_node( const struct of_device_id *matches, const struct device_node *node); -extern int of_modalias_node(struct device_node *node, char *modalias, int len); +extern int of_modalias_node(struct device_node *node, + const char *prefix, char *modalias, int len); extern struct device_node *of_parse_phandle(struct device_node *np, const char *phandle_name, int index); diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 3fea5a1..1fa0682 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -254,7 +254,7 @@ static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) char temp[DAI_NAME_SIZE]; struct i2c_client *i2c; - of_modalias_node(np, temp, DAI_NAME_SIZE); + of_modalias_node(np, NULL, temp, DAI_NAME_SIZE); iprop = of_get_property(np, "reg", NULL); if (!iprop) diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 982a1c9..3ea51ec 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -257,7 +257,7 @@ static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) char temp[DAI_NAME_SIZE]; struct i2c_client *i2c; - of_modalias_node(np, temp, DAI_NAME_SIZE); + of_modalias_node(np, NULL, temp, DAI_NAME_SIZE); iprop = of_get_property(np, "reg", NULL); if (!iprop) -- 1.7.2.3

2 2

[alsa-devel] [PATCH 0/5] ASoC: sh: fsi: adds stream transfer mode
by Kuninori Morimoto 19 May '12

19 May '12

Hi Mark, Liam These patches add stream transfer mode to FSI2 Kuninori Morimoto (5): ASoC: sh: fsi: use register field macro name on IN/OUT_DMAC ASoC: sh: fsi: add fsi_version() and removed meaningless version check ASoC: sh: fsi: use same format for IN/OUT ASoC: sh: fsi: call fsi_hw_startup/shutdown from fsi_dai_trigger() ASoC: sh: fsi: enable chip specific data transfer mode This stream transfer mode is required to use HDMI sound on latest SuperH Best regards --- Kuninori Morimoto

3 6

[alsa-devel] [PATCH] ASoC: tegra: Add machine driver for WM8753 codec
by Stephen Warren 19 May '12

19 May '12

From: Stephen Warren <swarren(a)nvidia.com> One such machine is Whistler. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Note that I haven't been able to get capture working; enabling sidetone shows that the mic is working correctly (sound from mic shows up at headphones), but arecord always captures digital 0. I don't think this is due to wm8753.c using a MICBIAS widget rather than a supply widget, since I adjusted the audio-routing table to cater for that, based on neo1973_wm8753.c's routing. --- .../bindings/sound/tegra-audio-wm8753.txt | 54 +++++ sound/soc/tegra/Kconfig | 10 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_wm8753.c | 224 ++++++++++++++++++++ 4 files changed, 290 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tegra-audio-wm8753.txt create mode 100644 sound/soc/tegra/tegra_wm8753.c diff --git a/Documentation/devicetree/bindings/sound/tegra-audio-wm8753.txt b/Documentation/devicetree/bindings/sound/tegra-audio-wm8753.txt new file mode 100644 index 0000000..c4dd39c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tegra-audio-wm8753.txt @@ -0,0 +1,54 @@ +NVIDIA Tegra audio complex + +Required properties: +- compatible : "nvidia,tegra-audio-wm8753" +- 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 WM8753's pins, and the jacks on the board: + + WM8753 pins: + + * LOUT1 + * LOUT2 + * ROUT1 + * ROUT2 + * MONO1 + * MONO2 + * OUT3 + * OUT4 + * LINE1 + * LINE2 + * RXP + * RXN + * ACIN + * ACOP + * MIC1N + * MIC1 + * MIC2N + * MIC2 + * Mic Bias + + Board connectors: + + * Headphone Jack + * Mic Jack + +- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller +- nvidia,audio-codec : The phandle of the WM8753 audio codec +Example: + +sound { + compatible = "nvidia,tegra-audio-wm8753-whistler", + "nvidia,tegra-audio-wm8753" + nvidia,model = "tegra-wm8753-harmony"; + + nvidia,audio-routing = + "Headphone Jack", "LOUT1", + "Headphone Jack", "ROUT1"; + + nvidia,i2s-controller = <&i2s1>; + nvidia,audio-codec = <&wm8753>; +}; + diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 5331c61..c1c8e95 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -48,6 +48,16 @@ config SND_SOC_TEGRA30_I2S Tegra30 I2S interface. You will also need to select the individual machine drivers to support below. +config SND_SOC_TEGRA_WM8753 + tristate "SoC Audio support for Tegra boards using a WM8753 codec" + depends on SND_SOC_TEGRA && I2C + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC + select SND_SOC_WM8753 + help + Say Y or M here if you want to add support for SoC audio on Tegra + boards using the WM8753 codec, such as Whistler. + config MACH_HAS_SND_SOC_TEGRA_WM8903 bool help diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 98704b4..391e78a 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -16,10 +16,12 @@ obj-$(CONFIG_SND_SOC_TEGRA30_AHUB) += snd-soc-tegra30-ahub.o obj-$(CONFIG_SND_SOC_TEGRA30_I2S) += snd-soc-tegra30-i2s.o # Tegra machine Support +snd-soc-tegra-wm8753-objs := tegra_wm8753.o snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-trimslice-objs := trimslice.o snd-soc-tegra-alc5632-objs := tegra_alc5632.o +obj-$(CONFIG_SND_SOC_TEGRA_WM8753) += snd-soc-tegra-wm8753.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o obj-$(CONFIG_SND_SOC_TEGRA_ALC5632) += snd-soc-tegra-alc5632.o diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c new file mode 100644 index 0000000..4e77026 --- /dev/null +++ b/sound/soc/tegra/tegra_wm8753.c @@ -0,0 +1,224 @@ +/* + * tegra_wm8753.c - Tegra machine ASoC driver for boards using WM8753 codec. + * + * Author: Stephen Warren <swarren(a)nvidia.com> + * Copyright (C) 2010-2012 - NVIDIA, Inc. + * + * Based on code copyright/by: + * + * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd. + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <asm/mach-types.h> + +#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/wm8753.h" + +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-snd-wm8753" + +struct tegra_wm8753 { + struct tegra_asoc_utils_data util_data; +}; + +static int tegra_wm8753_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_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; + struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card); + int srate, mclk; + int err; + + srate = params_rate(params); + switch (srate) { + case 11025: + case 22050: + case 44100: + case 88200: + mclk = 11289600; + break; + default: + mclk = 12288000; + break; + } + + 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, WM8753_MCLK, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static struct snd_soc_ops tegra_wm8753_ops = { + .hw_params = tegra_wm8753_hw_params, +}; + +static const struct snd_soc_dapm_widget tegra_wm8753_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static struct snd_soc_dai_link tegra_wm8753_dai = { + .name = "WM8753", + .stream_name = "WM8753 PCM", + .codec_dai_name = "wm8753-hifi", + .ops = &tegra_wm8753_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_wm8753 = { + .name = "tegra-wm8753", + .owner = THIS_MODULE, + .dai_link = &tegra_wm8753_dai, + .num_links = 1, + + .dapm_widgets = tegra_wm8753_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_wm8753_dapm_widgets), + .fully_routed = true, +}; + +static __devinit int tegra_wm8753_driver_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_tegra_wm8753; + struct tegra_wm8753 *machine; + int ret; + + machine = devm_kzalloc(&pdev->dev, sizeof(struct tegra_wm8753), + GFP_KERNEL); + if (!machine) { + dev_err(&pdev->dev, "Can't allocate tegra_wm8753 struct\n"); + ret = -ENOMEM; + goto err; + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + + 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_wm8753_dai.codec_of_node = of_parse_phandle( + pdev->dev.of_node, "nvidia,audio-codec", 0); + if (!tegra_wm8753_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_wm8753_dai.cpu_dai_of_node = of_parse_phandle( + pdev->dev.of_node, "nvidia,i2s-controller", 0); + if (!tegra_wm8753_dai.cpu_dai_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_wm8753_dai.platform_of_node = + tegra_wm8753_dai.cpu_dai_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 __devexit tegra_wm8753_driver_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_wm8753 *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_wm8753_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra-audio-wm8753", }, + {}, +}; + +static struct platform_driver tegra_wm8753_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = tegra_wm8753_of_match, + }, + .probe = tegra_wm8753_driver_probe, + .remove = __devexit_p(tegra_wm8753_driver_remove), +}; +module_platform_driver(tegra_wm8753_driver); + +MODULE_AUTHOR("Stephen Warren <swarren(a)nvidia.com>"); +MODULE_DESCRIPTION("Tegra+WM8753 machine ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_wm8753_of_match); -- 1.7.0.4

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[alsa-devel] ASoC updates for 3.5
by Mark Brown 19 May '12

19 May '12

The following changes since commit fe33d4c5e2b27a4b82aa479a58f2621eec103d1b: ASoC: don't flip master of DT-instantiated DAI links (2012-05-16 09:44:18 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.5 for you to fetch changes up to 5fb86e5d4a951ddb0474cdfd809380c8e2a8d101: ARM: mx31_3ds: Add sound support (2012-05-18 16:42:44 +0100) ---------------------------------------------------------------- ASoC: mc13783 audio support A few extra patches adding a new driver and some machines using it. ---------------------------------------------------------------- Philippe Rétornaz (7): mfd: mc13xxx: add codec platform data ASoC: Add mc13783 codec ASoC: add imx-mc13783 sound support ASoC: mc13783 codec cleanups mx31moboard: Add sound support ASoC: imx-mc13783 cleanup ARM: mx31_3ds: Add sound support arch/arm/mach-imx/Kconfig | 2 + arch/arm/mach-imx/mach-mx31_3ds.c | 22 +- arch/arm/mach-imx/mach-mx31moboard.c | 20 +- drivers/mfd/mc13xxx-core.c | 3 +- include/linux/mfd/mc13xxx.h | 11 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mc13783.c | 786 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/mc13783.h | 28 ++ sound/soc/fsl/Kconfig | 8 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-mc13783.c | 156 +++++++ 12 files changed, 1041 insertions(+), 3 deletions(-) create mode 100644 sound/soc/codecs/mc13783.c create mode 100644 sound/soc/codecs/mc13783.h create mode 100644 sound/soc/fsl/imx-mc13783.c

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[alsa-devel] [PATCH 00/11] ASoC: OMAP: HDMI: Use DSS audio interface and prepare for OMAP5
by Ricardo Neri 18 May '12

18 May '12

Hello, The ASoC HDMI codec used to be embedded in the DSS HDMI driver. In order to give the OMAP HDMI code a more logical arrangement and to remove some dependency breaks[1][2], such ASoC HDMI codec was removed[3]. Instead, the DSS HDMI audio functionality[4] is now provided through the new DSS device driver audio interface [5]. Hence, the ASoC HDMI support for OMAP needs to be changed to use this new DSS device driver audio interface. Under this new approach: * The HDMI audio functionality provided by the OMAP is now regarded as a CPU DAI rather than a codec. Hence, the CPU DAI will perform the operations that were performed previously by the codec (using the DSS dev driver audio interface). * A new ASoC HDMI OMAP codec is introduced as a dummy component. In the future, this component will examine the features supported by the sink and limit the number of channels, sample rates and formats that are exposed to the user. Also, this set of patches paves the way to the introduction of the HDMI audio functionality for OMAP5. The goal is to use the same set of ASoC drivers for OMAP4 and OMAP5 (DSS will be in charge of selecting the correct set of functions at run time). For this, several patches are submitted for: * Generalizing the build files to encompass not only OMAP4. * In the HDMI sound card driver, renaming the the functions and structures from omap4_ to omap_. This set includes the suggestions and improvements that Mark Brown kindly provided some time ago [6][7]. Please note that this set of patches will not build unless the patches from [4], [5] and [8] are present. All these patches have been accepted and should be upstream for K3.5. This implementation was validated on top of: git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux-2.6.git v3.4-rc7 and Liam Girdwood's: git://gitorious.org/omap-audio/linux-audio.git lrg/topic/3.5-dev Thanks, Ricardo [1] http://www.mail-archive.com/linux-omap@vger.kernel.org/msg67295.html [2] http://www.spinics.net/lists/linux-omap/msg66178.html [3] http://www.mail-archive.com/linux-omap@vger.kernel.org/msg67809.html [4] http://www.spinics.net/lists/linux-omap/msg69466.html [5] http://www.spinics.net/lists/linux-omap/msg69451.html [6] http://mailman.alsa-project.org/pipermail/alsa-devel/2012-February/049064.h… [7] http://mailman.alsa-project.org/pipermail/alsa-devel/2012-February/049065.h… [8] http://www.mail-archive.com/linux-omap@vger.kernel.org/msg66600.html Ricardo Neri (11): ASoC: OMAP: HDMI: Introduce codec ASoC: OMAP: HDMI: Update the platform device names ASoC: OMAP: HDMI: Change error values in HDMI CPU DAI ASoC: OMAP: HDMI: Create a structure for private data of the CPU DAI ASoC: OMAP: HDMI: Use the DSS audio interface ASoC: OMAP: HDMI: Expand configuration of hw_params ASoC: OMAP: HDMI: Improve how the display state is verified ASoC: OMAP: HDMI: Expand capabilities of the HDMI DAI ASoC: OMAP: HDMI: Make build config options more generic ASoC: OMAP: HDMI: Make sound card naming more generic ASoC: OMAP: HDMI: Rename sound card driver sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/omap-hdmi.c | 69 +++++++++++ sound/soc/omap/Kconfig | 8 +- sound/soc/omap/Makefile | 4 +- sound/soc/omap/omap-hdmi-card.c | 87 ++++++++++++++ sound/soc/omap/omap-hdmi.c | 238 +++++++++++++++++++++++++++++++++++--- sound/soc/omap/omap-hdmi.h | 4 +- sound/soc/omap/omap4-hdmi-card.c | 121 ------------------- 9 files changed, 395 insertions(+), 142 deletions(-) create mode 100644 sound/soc/codecs/omap-hdmi.c create mode 100644 sound/soc/omap/omap-hdmi-card.c delete mode 100644 sound/soc/omap/omap4-hdmi-card.c -- 1.7.5.4

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[alsa-devel] [PATCH 0/4] MC13783 sound support
by Philippe Rétornaz 18 May '12

18 May '12

Hello This patchset enables sound support on boards with mc13783 codec. The first patch adds the necessary modification to the MFD driver. The second patch adds the codec driver. It's Sascha's code ported to the current kernel with DAPM. The third patch adds a generic SoC platform driver for imx boards with mc13783 codecs. The last patch enable the sound on mx31moboard boards. I will post a similar patch for mx31pdk_3ds as soon as I figure out why it hangs early at boot with the latest kernel (unrelated to this patchset). It is based on 3.4-rc4 with asoc-3.5 and the sdma channel 0 mutex to spinlock fix. Philippe Rétornaz (4): mc13xxx: add codec platform data ASoC: Add mc13783 codec ASoC: add imx-mc13783 sound support mx31moboard: Add sound support arch/arm/mach-imx/Kconfig | 1 + arch/arm/mach-imx/mach-mx31moboard.c | 20 +- drivers/mfd/mc13xxx-core.c | 3 +- include/linux/mfd/mc13xxx.h | 11 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mc13783.c | 800 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/mc13783.h | 28 ++ sound/soc/fsl/Kconfig | 8 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-mc13783.c | 163 +++++++ 11 files changed, 1040 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/mc13783.c create mode 100644 sound/soc/codecs/mc13783.h create mode 100644 sound/soc/fsl/imx-mc13783.c

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[alsa-devel] Question on Dynamic PCM
by Sebastien LEDUC 18 May '12

18 May '12

In commit "ASoC: dapm: Add API call to query valid DAPM paths<http://git.kernel.org/?p=linux/kernel/git/broonie/sound.git;a=commit;h=ec2e…>" , the signature of the functions is_connected_input_ep / is_connected_output_ep has been modified so that the 2 functions now fill list of connected DAPM widgets. What I don't understand is why the way widgets are added is not symmetric between the two: In function is_connected_output_ep: ----------------------------------------------- list_for_each_entry(path, &widget->sinks, list_source) { ... err = dapm_list_add_widget(list, path->sink); ... } In function is_connected_input_ep: --------------------------------------------- list_for_each_entry(path, &widget->sources, list_sink) { ... err = dapm_list_add_widget(list, path->sink); } I would have expected to have for the latter: err = dapm_list_add_widget(list, path->source); regards, Sebastien Le Duc

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[alsa-devel] [patch] ALSA: hda - unlock on error in azx_interrupt()
by Dan Carpenter 18 May '12

18 May '12

There is an spin_unlock() missing on this error path. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- This was added recently in a82d51ed2 "ALSA: hda - Support VGA-switcheroo". I don't have the hardware so I haven't tested it. diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bbf953f..2b6392b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1268,8 +1268,10 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) spin_lock(&chip->reg_lock); - if (chip->disabled) + if (chip->disabled) { + spin_unlock(&chip->reg_lock); return IRQ_NONE; + } status = azx_readl(chip, INTSTS); if (status == 0) {

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Re: [alsa-devel] [PATCH 2/2] ASoC: add pxa-squ dma support
by Haojian Zhuang 18 May '12

18 May '12

On Fri, May 18, 2012 at 10:39 AM, Qiao Zhou <zhouqiao(a)marvell.com> wrote: > this patch adds ASoC pxa-squ DMA audio support. > > Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> > +/* > + * Descriptor structure for PXA's SQU engine > + * Note: this structure must always be aligned to a 16-byte boundary. > + */ > +struct pxa_squ_desc { > + __iomem u32 byte_cnt; /* byte count */ > + __iomem u32 src_addr; /* source address */ > + __iomem u32 dst_addr; /* target address */ > + __iomem u32 nxt_desc; /* next descriptor dress */ Why do you use __iomem at here? > +}; > + -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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Re: [alsa-devel] [PATCH 1/2] ARM: MMP: add pxa-squ header file
by Haojian Zhuang 18 May '12

18 May '12

On Fri, May 18, 2012 at 10:39 AM, Qiao Zhou <zhouqiao(a)marvell.com> wrote: > add pxa-squ header file to support SQU DMA. > > Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> > --- > arch/arm/mach-mmp/include/mach/pxa-squ.h | 39 ++++++++++++++++++++++++++++++ > 1 files changed, 39 insertions(+), 0 deletions(-) > create mode 100644 arch/arm/mach-mmp/include/mach/pxa-squ.h > > diff --git a/arch/arm/mach-mmp/include/mach/pxa-squ.h b/arch/arm/mach-mmp/include/mach/pxa-squ.h > new file mode 100644 > index 0000000..3ee187f > --- /dev/null > +++ b/arch/arm/mach-mmp/include/mach/pxa-squ.h > @@ -0,0 +1,39 @@ > +#ifndef __MACH_SQU_H > +#define __MACH_SQU_H > + > +#include <mach/addr-map.h> > + > +#define SQU_REGS_VIRT (AXI_VIRT_BASE + 0xA0000) > +#define SQU_REG(x) (*((__iomem u32 *)(SQU_REGS_VIRT + (x)))) > + > + /*SQU*/ > +#define SDSAR(x) SQU_REG(0x810 + ((x) << 2)) > +#define SDDAR(x) SQU_REG(0x820 + ((x) << 2)) > +#define SDNDPR(x) SQU_REG(0x830 + ((x) << 2)) > +#define SDCR(x) SQU_REG(0x840 + ((x) << 2)) > +#define SDIMR(x) SQU_REG(0x880 + ((x) << 2)) > +#define SDISR(x) SQU_REG(0x8a0 + ((x) << 2)) > +#define SDCR_SSPMOD (1 << 21) /* SSPMod */ > +#define SDCR_FETCHND (1 << 13) > +#define SDCR_CHANEN (1 << 12) /* Channel Enable */ > +#define SDCR_DST_ADDR_INC (0 << 4) > +#define SDCR_DST_ADDR_HOLD (0x2 << 4) > +#define SDCR_SRC_ADDR_INC (0 << 2) > +#define SDCR_SRC_ADDR_HOLD (0x2 << 2) > +#define SDCR_DMA_BURST_4B (0x0 << 6) > +#define SDCR_DMA_BURST_8B (0x1 << 6) > +#define SDCR_DMA_BURST_16B (0x3 << 6) > +#define SDCR_DMA_BURST_1B (0x5 << 6) > +#define SDCR_DMA_BURST_2B (0x6 << 6) > +#define SDCR_DMA_BURST_32B (0x7 << 6) > +#define SDIMR_COMP (1 << 0) > + > +struct pxa_squ_platdata { > + char *pool_name; > + int irq; > + int period_max_out; /* max period size for playback */ > + int period_max_in; /* max period size for capture */ > + int buffer_max_out; /* max buffer size for playback */ > + int buffer_max_in; /* max buffer size for capture */ > +}; > +#endif /* __MACH_SQU_H */ > -- > 1.7.4.1 > Acked-by: Haojian Zhuang <haojian.zhuang(a)gmail.com> -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] Problem with snd_soc_suspend
by Patrick Lai 17 May '12

17 May '12

Hi, The CODEC I am working on has multiple digital audio interfaces included in multiple DAI-LINks. DAILINK X has ignore_suspend flag set and DAILINK Y has ignore_suspend flag cleared. The problem I encountered is that as system attempts to enter suspend while DAILINK X is active, snd_soc_suspend() calls snd_soc_dapm_stream_event( SND_SOC_DAPM_STREAM_SUSPEND) on inactive DAILINK Y since DAILINK Y ignore flag is not set. Consequently, CODEC widgets get powered off even though they should remain powered for DAILINK X. Kernel version I am working with is 3.0. Couple questions with respect to this problem 1. Should soc-core check if CODEC DAI is active beside checking ignore_suspend flag before calling snd_soc_dapm_stream_event()? I checked latest soc-core.c from kernel 3.4 RC7. The logic is the same as 3.0 2. Even if snd_soc_dapm_stream_event() is called, DAPM should not power off widgets that are powered up for DAILINK x. It seems like DAPM has the logic to only power off AIF widget if stream name of widget matches CODEC DAI stream name in soc_dapm_stream_event(). I have not gotten time to debug this problem further. However, is it a known bug on kernel 3.0. If so, has the problem been addressed in more recent kernel? Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

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[alsa-devel] Asus Xonar DG front panel not working
by Marcos Felipe Rasia de Mello 17 May '12

17 May '12

Hi, My Asus Xonar DG doesn't play through FP Headphone connector (HD Audio). Mic input doesn't work too (both rear and front connectors). Same issue: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/919809 I think its output should behave this way: Speakers - rear connector Headphones - rear connector + headphone amp FP Headphones - front connector + headphone amp Attached alsa-info.sh output. Regards, Marcos

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[alsa-devel] Support new HDA codec continue...
by Jorge Fernandez Monteagudo 17 May '12

17 May '12

Hi all, I've to take again the HDA codec for Wolfson because I see that I can't change the master volume. I would like to know how can I do to get with amixer an entry for master o pcm volume. Now I have: ~ # amixer Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Stereo Differential',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 106 [83%] [-10.50dB] [on] Front Right: Playback 106 [83%] [-10.50dB] [on] but I only can enable or disable the audio with # amixer sset 'IEC958' on / off but no master sound control is present... with # amixer sset 'Stereo Differential' 100% or # amixer sset 'Stereo Differential' 0% there is no difference. Any hint is welcome Thanks! Jorge ________________________________________ From: Mark Brown [broonie(a)opensource.wolfsonmicro.com] Sent: Monday, March 05, 2012 1:16 PM To: Jorge Fernandez Monteagudo Cc: Takashi Iwai; alsa-devel(a)alsa-project.org Subject: Re: [alsa-devel] Support new HDA codec On Thu, Mar 01, 2012 at 10:44:23AM +0100, Jorge Fernandez Monteagudo wrote: > Attached it's the patch applied to alsa 1.0.23. A new patch_wolfson.c file > is created. I would like to note that no Wolfson support was received to > develop to this code, and this code is useful for us because we only need Hrm, that's not strictly accurate - we didn't help with the code but I know there's been some help (possibly via the distributor). > the outputs differential and SPDIF. This code must be revisited in order to > support the different inputs, the stereo normal output and the I/O. Looking at the code it seems like most of this is things that I'd really expect the generic driver to be able to cope with assuming a suitable BIOS setup, but looking at the generic driver it seems like it might need some enhancement to do so. Things like the S/PIDF output should really be able to Just Work. Hrm. Este mensaje se dirige exclusivamente a su destinatario y puede contener información privilegiada o CONFIDENCIAL. Si no es vd. el destinatario indicado, queda notificado de que la utilización, divulgación y/o copia sin autorización está prohibida en virtud de la legislación vigente. Si ha recibido este mensaje por error, le rogamos que nos lo comunique inmediatamente por esta misma vía y proceda a su destrucción. This message is intended exclusively for its addressee and may contain information that is CONFIDENTIAL and protected by professional privilege. If you are not the intended recipient you are hereby notified that any dissemination, copy or disclosure of this communication is strictly prohibited by law. If this message has been received in error, please immediately notify us via e-mail and delete it.

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Re: [alsa-devel] Adjust volume may cause audio playback corruption
by cee1 17 May '12

17 May '12

Witch a patch[1] of PA which dumps playback PCM data, I found the dumped data(with ./wavformat.py[2] <dumpped_data> to convert to wav file) is not corrupted. Since PA uses mmap to transfer PCM data, the driver's DMA buffer should also be "good". It seems related to HDA controller of SB710, I tried HDMI output, and can reproduce the problem too(http://dev.lemote.com/files/upload/software/PA-apc/HDMI-corrupted_sound…. 2012/5/12 cee1 <fykcee1(a)gmail.com>: > Hi all, > > We're using PA on our product, and sometimes audio playback may > corrupt after adjusting volume. > > We found a way to (relatively) easily reproduce the problem: > 1. Play sample.mp3 through totem in repeat mode. > 2. In gnome-volume-control, Hardware tap, repeat to shift between "Off > profile" and "Analog Stereo Duplex profile". > > When the problem happens, the corruption continues until I: > * Do one more shift between "Off profile" and "Analog Stereo Duplex profile". > * Close totem (close all PA playback clients). > * Adjust volume. > > When the problem happens, open another PA client doing audio playback, > is also corrupt. > > Some additional information: > * Output of alsa-info.sh on our product: > http://dev.lemote.com/files/upload/software/PA-apc/alsa-info-lemote > * PA daemon log: > http://dev.lemote.com/files/upload/software/PA-apc/pulseaudio.log > * The corrupted sound: > http://dev.lemote.com/files/upload/software/PA-apc/corrupted_sound.ogg > * The sample mp3: http://dev.lemote.com/files/upload/software/PA-apc/sample.mp3 > > We've tried PA 1.0-0ubuntu3, 1.1-0ubuntu15 and 2.0, the problem continues. > > The problem can also be reproduced on an AMD turion 64 x2 board with > ubuntu 10.10 and ubuntu 12.04: > * Output of alsa-info: > ubuntu 10.10(32bit) > http://dev.lemote.com/files/upload/software/PA-apc/alsa-info-ubuntu-10.10 > ubuntu 12.04(32bit) > http://dev.lemote.com/files/upload/software/PA-apc/alsa-info-ubuntu-12.04 > > Note: > * The AMD turion 64 x2 board uses the same chipset as our product. > * gnome-volume-control is removed in ubuntu 12.04, thus we compiled > it: http://dev.lemote.com/files/upload/software/PA-apc/gnome-volume-control ---- [1] http://dev.lemote.com/files/upload/software/PA-apc/debug/debug-alsa-sink.pa… [2] http://dev.lemote.com/files/upload/software/PA-apc/debug/wavformat.py -- Regards, - cee1

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[alsa-devel] [PATCH v2 0/2] ASoC: OMAP4 DMIC, McPDM device tree support
by Peter Ujfalusi 16 May '12

16 May '12

Hello, Add DT support and the binding documentation for OMAP4+ DMIC, and McPDM ASoC dai drivers. Changes since v1: - Ducumentation update: add properties for register address/size, interrupt number (as Suggested by Mark and Tony). I have not added Mark's acked-by since I think the changes are big enough from v1. Regards, Peter --- Peter Ujfalusi (2): ASoC: omap-mcpdm: Add device tree bindings ASoC: omap-dmic: Add device tree bindings .../devicetree/bindings/sound/omap-dmic.txt | 21 ++++++++++++++++++++ .../devicetree/bindings/sound/omap-mcpdm.txt | 21 ++++++++++++++++++++ sound/soc/omap/omap-dmic.c | 8 +++++++ sound/soc/omap/omap-mcpdm.c | 8 +++++++ 4 files changed, 58 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/omap-dmic.txt create mode 100644 Documentation/devicetree/bindings/sound/omap-mcpdm.txt -- 1.7.8.6

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[alsa-devel] ASoC updates for 3.5
by Mark Brown 16 May '12

16 May '12

The following changes since commit 36be50515fe2aef61533b516fa2576a2c7fe7664: Linux 3.4-rc7 (2012-05-12 18:37:47 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.5 for you to fetch changes up to fe33d4c5e2b27a4b82aa479a58f2621eec103d1b: ASoC: don't flip master of DT-instantiated DAI links (2012-05-16 09:44:18 +0100) ---------------------------------------------------------------- ASoC: Updates for 3.5 The big things here in framework terms are the first two, otherwise it's just a large number of incremental and driver specific updates: - Dynamic PCM support, allowing support for SoCs with internal routing through components with tight sequencing and formatting constraints within their internal paths or where there are multiple components connected with CPU managed DMA controllers inside the SoC. - Greatly improved support for direct digital links between off-SoC devices, providing a much simpler way of connecting things like digital basebands to CODECs. - Much more fine grained and robust locking, cleaning up some of the confusion that crept in with multi-component. - CPU support for nVidia Tegra 30 I2S and audio hub controllers and ST-Ericsson MSP I2S controolers - New CODEC drivers for Cirrus CS42L52, LAPIS Semiconductor ML26124, Texas Instruments LM49453. There's also some regmap changes needed by the Tegra I2S driver. ---------------------------------------------------------------- Ashish Chavan (5): ASoC: da7210: Add support for spi regmap ASoC: da7210: Add support for PLL and SRM ASoC: da7210: Minor update for PLL and SRM ASoC: da7210: Minor bugfix for non pll slave mode ASoC: da7210: Minor improvements and a bugfix Axel Lin (3): ASoC: alc5623: Convert to module_i2c_driver() ASoC: alc5632: Convert to module_i2c_driver() ASoC: alc5632: Convert to devm_regmap_init_i2c() Brian Austin (6): ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV ASoC: core: Initialize err for snd_soc_put_volsw_sx ASoC: Add support for CS42L52 Codec ASoC: cs42l73: Convert to module_i2c_driver() ASoC: cs42l73: Remove Chip ID's from reg_default ASoC: cs42l73: Use DAPM routes to hook AIF widgets to DAI's Dan Carpenter (1): ASoC: dapm: release lock on error paths Fabio Estevam (6): ASoC: sgtl5000: Fix warning due to the lack of REGULATOR_CHANGE_VOLTAGE ASoC: wm9712: Fix build due to missing definition of "runtime" ASoC: ac97: Fix build due to removal of 'runtime' definition ASoC: wm9705: Fix build due to removal of 'runtime' definition ASoC: soc-dapm: Use '%llx' with 'u64' type. ASoC: core: Remove unused variable 'min' H Hartley Sweeten (2): ASoC: ep93xx-ac97: use devm_* helpers to cleanup probe ASoC: ep93xx-i2s: use devm_* helpers to cleanup probe Jesper Juhl (2): ASoC: wm8994: Don't test for NULL before release_firmware() ASoC: wm8994: Delete trailing whitespace from sound/soc/codecs/wm8994.c Kristoffer KARLSSON (2): ASoC: core: Add signed multi register control ASoC: core: Add strobe control Kuninori Morimoto (5): ASoC: add generic simple-card support ASoC: sh: fsi: use simple-card instead of fsi-ak4642 ASoC: sh: fsi: use simple-card instead of fsi-hdmi ASoC: sh: fsi: use simple-card instead of fsi-da7210 ASoC: sh: fsi: select simple-card on Kconfig Kyung-Kwee Ryu (1): ASoC: wm8994: Make sure we disable FLL bypass when stopping the FLL Lars-Peter Clausen (4): ASoC: bf5xx-ssm2602: Setup sysclock in init callback ASoC: SSM2602: Add sysclk based rate constraints ASoC: SSM2602: Remove driver specific version ASoC: SSM2602: Convert to direct regmap API usage Liam Girdwood (19): ASoC: core: Add card mutex locking subclasses ASoC: dapm: Use DAPM mutex for DAPM ops instead of codec mutex ASoC: dapm: lock mixer & mux update power with DAPM mutex ASoC: core: Add platform DAI widget mapping ASoC: dapm: Add platform stream event support ASoC: core: Rename card mutex subclass to better align with usage ASoC: dapm: Rename dapm mutex subclass to better match usage ASoC: dapm: Add regulator member to struct dapm_widget ASoC: DAPM: Make sure DAPM widget IO ops hold the component mutex ASoC: dapm: Add API call to query valid DAPM paths ASoC: dapm: Fix x86_64 build warning. ASoC: dpcm: Add Dynamic PCM core operations. ASoC: dpcm: Add debugFS support for DPCM ASoC: dpcm: Add runtime dynamic route update ASoC: dpcm: Add API for DAI link substream and runtime lookup ASoC: dpcm: Add bespoke trigger() ASoC: pcm: Add pcm operation for pcm ioctl. ASoC: dpcm: Fixup debugFS for DPCM state. ASoC: core: Fix dai_link dereference. M R Swami Reddy (3): ASoC: Support TI LM49453 Audio driver ASoC: MAINTAINERS: Add maintainer for TI LM49453 Audio codec driver ASoC: lm49453: fix build warnings Maarten ter Huurne (1): ASoC: JZ4740: Replaced comma operators with semicolons. Marc Reilly (2): regmap: Add support for device with 24 data bits. regmap: Use pad_bits and reg_bits when determining register format. Mark Brown (78): ASoC: ak4641: Convert to module_i2c_driver() ASoC: ak4641: Push GPIO allocation out into the I2C probe ASoC: dapm: Only lock CODEC for I/O if not using regmap ASoC: wm5100: Convert to devm_regmap_init_i2c() ASoC: wm8994: Don't lock CODEC mutex to do DAPM sync ASoC: jack: Push locking for jacks down to the jack ASoC: core: Use driver core probe deferral ASoC: wm8962: Implement DSP2 configuration initialisation ASoC: wm8962: Add 3D enhancement support ASoC: wm8962: Add Direct-Form 1 filter support ASoC: wm8962: Add Dynamic Range Control support ASoC: wm8962: Add HD Bass and VSS coefficient configuration ASoC: wm8962: Add HPF coefficient configuration support ASoC: dapm: Remove bodges for no-widget CODECs ASoC: cs4270: Check that we can enable regulators on resume ASoC: wm8994: Don't allow reconfiguration of FLL when it provides SYSCLK ASoC: sgtl5000: Convert mic bias to a supply widget ASoC: dapm: Remove SND_SOC_DAPM_MICBIAS_E() ASoC: wm8994: Don't bother lowering clock dividers inside idle AIFs ASoC: max98095: Fix build failure ASoC: sgtl5000: Convert to module_i2c_driver() ASoC: dapm: Allow DAPM registers to be 31 bit ASoC: wm8994: Don't bother updating the jackdet mode needlessly ASoC: tlv320aic23: Remove driver-specific version number ASoC: codecs: Remove rtd->codec usage from CODEC drivers ASoC: wm8400: Use snd_soc_write() and snd_soc_read() Merge tag 'v3.4-rc2' into for-3.5 ASoC: wm1250-ev1: Convert to module_i2c_driver ASoC: wm8994: Implement FLL bypass support ASoC: core: Return -ENOTSUPP instead of -EINVAL if mute is not supported ASoC: dapm: Allow DAI widgets to be routed through ASoC: core: Flip master for CODECs in the CPU slot of a CODEC<->CODEC link ASoC: core: Bind DAIs to CODECs at registration time ASoC: core: Support transparent CODEC<->CODEC DAI links ASoC: Merge tag 'v3.4-rc3' into for-3.5 ASoC: Use dai_fmt in Speyside ASoC: wm8994: Keep AIF3 tristated when not in use ASoC: samsung: Hook up AIF2 to the CODEC on Littlemill ASoC: wm1250-ev1: Support stereo ASoC: wm1250-ev1: Support sample rate configuration ASoC: Update regmap access for WM5100 DSP control registers ASoC: wm8996: Put the microphone biases into bypass mode when idle ASoC: wm8994: Tune debounce rates for jack detect mode ASoC: wm8994: Allow rate configuration with custom mic callback ASoC: wm8994: Add trace showing wm8958_micd_set_rate() ASoC: dapm: Move CODEC<->CODEC params off stack ASoC: wm_hubs: Special case headphones for digital paths in more use cases ASoC: wm_hubs: Factor out class W management ASoC: wm_hubs: Enable class W for output mixer paths ASoC: cs42l52: Remove duplicate module exit code ASoC: wm8350: Remove check for clocks in trigger() ASoC: wm8350: Replace use of custom I/O with snd_soc_read()/write() ASoC: wm_hubs: Cache multiple DCS offsets ASoC: core: Allow DAIs to specify a base address ASoC: wm5100: Set the DAI base address in the DAI drivers ASoC: pcm: Fix DPCM for aux_devs ASoC: lowland: Convert to dai_fmt ASoC: lowland: Support digital link for WM9081 ASoC: wm9081: Hook DAC up via DAPM rather than stream ASoC: rt5631: Convert to module_i2c_driver() ASoC: rt5631: Convert to direct regmap API usage ASoC: wm8994: Fix sparse warning due to use of 0 as NULL ASoC: wm8994: Add debounce to wm8994 mic detection ASoC: wm8994: Use regmap directly for wm8994_mic_work ASoC: cs42l52: Staticise non-exported symbols ASoC: ux500: Fix dependencies ASoC: ux500: Don't make the CPU I2S driver user selectable ASoC: pcm: Staticise non-exported functions ASoC: littlemill: Use a widget to keep track of AIF2 clocking ASoC: wm5100: Convert to module_i2c_driver() ASoC: wm8731: Convert to devm_ functions Merge tag 'v3.4-rc7' into for-3.5 ASoC: lm49453: Staticise non-exported symbol lm45453_dai ASoC: lm49453: Fix author e-mail address ASoC: max98095: Single bit bitfields should be unsigned ASoC: wm8350: Don't use irq_base ASoC: tlv320aic3x: Remove unused, non-standard headset detection ASoC: wm8350: Remove direct usage of codec->control_data Ola Lilja (1): ASoC: Ux500: Add MSP I2S-driver Peter Ujfalusi (1): ASoC: twl6040: Remove HS/HF gain ramp feature Philipp Zabel (1): ASoC: pxa2xx-i2s: Add clk_prepare/clk_unprepare calls Rhyland Klein (1): ASoC: max98095: add jack detection Richard Zhao (2): ASoC: imx-sgtl5000: add of_node_put when probe fail. ASoC: fsl: add sgtl5000 clock support for imx-sgtl5000 Shawn Guo (18): ASoC: fsl: separate SSI and DMA Kconfig options ASoC: imx: merge sound/soc/imx into sound/soc/fsl ASoC: fsl: rename imx-pcm Kconfig options and filename ASoC: fsl: create fsl_utils to accommodate the common functions ASoC: fsl: remove helper fsl_asoc_get_codec_dev_name ASoC: fsl: make fsl_ssi driver compilable on ARM/IMX ASoC: fsl: let fsl_ssi work with imx pcm and machine drivers ASoC: fsl: add imx-sgtl5000 machine driver ASoC: fsl: assign dma peripheral type according to bus topology ASoC: fsl: enable ssi clock in probe function ASoC: fsl: add audio routing for imx-sgtl5000 ASoC: fsl: remove redundant Kconfig option SND_SOC_FSL_SSI ASoC: fsl: remove redundant Kconfig dependency on SND_SOC_POWERPC_DMA ASoC: mxs: add __devinit for mxs_saif_probe ASoC: fsl: fix the binding of imx-sgtl5000 ASoC: mxs: mxs-pcm does not need to be a plaform_driver ASoC: mxs: add device tree support for mxs-saif ASoC: mxs: add device tree support for mxs-sgtl5000 Stephen Warren (38): ASoC: tegra: fix comment indentation in ALC5632 machine ASoC: tegra: register 'platform' from DAIs, get rid of pdev ARM: tegra: remove tegra_pcm_device ASoC: tegra: remove unnecessary includes ASoC: tegra: fix Kconfig SND_SOC_TEGRA_ALC5632 indentation ASoC: tegra: fix some checkpatch warnings ASoC: tegra: introduce separate Kconfig variable for DAS driver ASoC: tegra: sort Makefile into common and per-SoC files ASoC: tegra: remove open-coded clk reference counting ASoC: tegra: drop Kconfig description for SND_SOC_TEGRA_DAS ASoC: tegra: make Tegra20 drivers depend on Tegra20 support ASoC: tegra: rename Tegra20-specific driver files ASoC: tegra: complete Tegra->Tegra20 renaming ASoC: tegra: sort includes, remove mach/iomap.h ASoC: tegra: use devm_ APIs in SPDIF driver ASoC: tegra: set a sensible initial clock rate ASoC: tegra: utils: add support for Tegra30 devices ASoC: tegra: add runtime PM support ASoC: tegra: utils: Don't use of_have_populated_dt() regmap: introduce explicit bus_context for bus callbacks regmap: introduce fast_io busses, and use a spinlock for them regmap: add MMIO bus support regmap: mmio: convert some error returns to BUG() regmap: mmio: remove some error checks now in the core regmap: validate regmap_raw_read/write val_len regmap: allow regmap instances to be named regmap: fix compilation when !CONFIG_DEBUG_FS regmap: implement register striding regmap: fix compile errors in regmap-irq.c due to stride changes ASoC: tegra: add tegra30-ahub driver ASoC: tegra: add tegra30-i2s driver ASoC: tegra: add Kconfig and Makefile support for Tegra30 ASoC: tegra: select REGMAP_MMIO ASoC: convert Tegra20 I2S driver to regmap ASoC: convert Tegra20 SPDIF driver to regmap ASoC: convert Tegra20 DAS driver to regmap ASoC: tegra: add device tree support for TrimSlice ASoC: don't flip master of DT-instantiated DAI links Tomoya MORINAGA (1): ASoC: Add LAPIS Semiconductor ML26124 driver guoyh (1): ASoC: pxa: allocate the SSP DMA parameters in startup .../bindings/sound/imx-audio-sgtl5000.txt | 49 + .../bindings/sound/mxs-audio-sgtl5000.txt | 17 + .../devicetree/bindings/sound/mxs-saif.txt | 36 + .../bindings/sound/nvidia,tegra30-ahub.txt | 32 + .../bindings/sound/nvidia,tegra30-i2s.txt | 15 + .../bindings/sound/tegra-audio-trimslice.txt | 14 + MAINTAINERS | 6 + arch/arm/mach-shmobile/Kconfig | 2 + arch/arm/mach-shmobile/board-ap4evb.c | 35 +- arch/arm/mach-shmobile/board-mackerel.c | 35 +- arch/arm/mach-tegra/board-dt-tegra20.c | 6 +- arch/arm/mach-tegra/board-harmony.c | 1 - arch/arm/mach-tegra/board-seaboard.c | 1 - arch/arm/mach-tegra/board-trimslice.c | 1 - arch/arm/mach-tegra/devices.c | 11 +- arch/arm/mach-tegra/devices.h | 1 - arch/arm/mach-tegra/tegra2_clocks.c | 4 +- arch/powerpc/configs/86xx/mpc8610_hpcd_defconfig | 1 + arch/powerpc/configs/mpc85xx_defconfig | 1 + arch/powerpc/configs/mpc85xx_smp_defconfig | 1 + arch/sh/boards/Kconfig | 2 + arch/sh/boards/mach-ecovec24/setup.c | 26 + arch/sh/boards/mach-se/7724/setup.c | 15 +- drivers/base/regmap/Kconfig | 3 + drivers/base/regmap/Makefile | 1 + drivers/base/regmap/internal.h | 22 +- drivers/base/regmap/regcache-lzo.c | 11 +- drivers/base/regmap/regcache-rbtree.c | 44 +- drivers/base/regmap/regcache.c | 34 +- drivers/base/regmap/regmap-debugfs.c | 18 +- drivers/base/regmap/regmap-i2c.c | 13 +- drivers/base/regmap/regmap-irq.c | 37 +- drivers/base/regmap/regmap-mmio.c | 224 +++ drivers/base/regmap/regmap-spi.c | 13 +- drivers/base/regmap/regmap.c | 169 +- include/linux/regmap.h | 28 +- include/sound/cs42l52.h | 36 + include/sound/max98095.h | 12 + include/sound/sh_fsi.h | 12 - include/sound/simple_card.h | 38 + include/sound/soc-dai.h | 4 + include/sound/soc-dapm.h | 28 +- include/sound/soc-dpcm.h | 138 ++ include/sound/soc.h | 118 +- include/trace/events/asoc.h | 80 + sound/soc/Kconfig | 5 +- sound/soc/Makefile | 3 +- sound/soc/blackfin/bf5xx-ssm2602.c | 37 +- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 8 +- sound/soc/codecs/ac97.c | 6 +- sound/soc/codecs/ad1836.c | 4 +- sound/soc/codecs/ad193x.c | 4 +- sound/soc/codecs/adau1701.c | 3 +- sound/soc/codecs/ak4104.c | 3 +- sound/soc/codecs/ak4535.c | 3 +- sound/soc/codecs/ak4641.c | 113 +- sound/soc/codecs/alc5623.c | 23 +- sound/soc/codecs/alc5632.c | 31 +- sound/soc/codecs/cs4270.c | 11 +- sound/soc/codecs/cs4271.c | 3 +- sound/soc/codecs/cs42l51.c | 9 +- sound/soc/codecs/cs42l52.c | 1295 ++++++++++++++ sound/soc/codecs/cs42l52.h | 274 +++ sound/soc/codecs/cs42l73.c | 93 +- sound/soc/codecs/da7210.c | 379 ++++- sound/soc/codecs/jz4740.c | 3 +- sound/soc/codecs/lm49453.c | 1550 +++++++++++++++++ sound/soc/codecs/lm49453.h | 380 +++++ sound/soc/codecs/max98095.c | 158 +- sound/soc/codecs/max98095.h | 22 + sound/soc/codecs/ml26124.c | 681 ++++++++ sound/soc/codecs/ml26124.h | 184 ++ sound/soc/codecs/rt5631.c | 110 +- sound/soc/codecs/sgtl5000.c | 25 +- sound/soc/codecs/ssm2602.c | 138 +- sound/soc/codecs/sta32x.c | 3 +- sound/soc/codecs/tlv320aic23.c | 13 +- sound/soc/codecs/tlv320aic26.c | 3 +- sound/soc/codecs/tlv320aic3x.c | 21 +- sound/soc/codecs/tlv320dac33.c | 35 +- sound/soc/codecs/twl4030.c | 18 +- sound/soc/codecs/twl6040.c | 450 +---- sound/soc/codecs/uda134x.c | 6 +- sound/soc/codecs/uda1380.c | 6 +- sound/soc/codecs/wl1273.c | 6 +- sound/soc/codecs/wm1250-ev1.c | 65 +- sound/soc/codecs/wm5100-tables.c | 125 +- sound/soc/codecs/wm5100.c | 47 +- sound/soc/codecs/wm5100.h | 159 ++ sound/soc/codecs/wm8350.c | 187 +-- sound/soc/codecs/wm8400.c | 135 +- sound/soc/codecs/wm8510.c | 3 +- sound/soc/codecs/wm8523.c | 3 +- sound/soc/codecs/wm8728.c | 3 +- sound/soc/codecs/wm8731.c | 37 +- sound/soc/codecs/wm8737.c | 3 +- sound/soc/codecs/wm8741.c | 3 +- sound/soc/codecs/wm8750.c | 3 +- sound/soc/codecs/wm8753.c | 6 +- sound/soc/codecs/wm8900.c | 3 +- sound/soc/codecs/wm8903.c | 3 +- sound/soc/codecs/wm8940.c | 3 +- sound/soc/codecs/wm8960.c | 3 +- sound/soc/codecs/wm8962.c | 18 +- sound/soc/codecs/wm8971.c | 3 +- sound/soc/codecs/wm8978.c | 3 +- sound/soc/codecs/wm8988.c | 3 +- sound/soc/codecs/wm8990.c | 3 +- sound/soc/codecs/wm8993.c | 86 +- sound/soc/codecs/wm8994.c | 290 ++-- sound/soc/codecs/wm8994.h | 3 + sound/soc/codecs/wm8996.c | 12 + sound/soc/codecs/wm9081.c | 5 +- sound/soc/codecs/wm9705.c | 6 +- sound/soc/codecs/wm9712.c | 10 +- sound/soc/codecs/wm_hubs.c | 220 ++- sound/soc/codecs/wm_hubs.h | 12 +- sound/soc/ep93xx/ep93xx-ac97.c | 74 +- sound/soc/ep93xx/ep93xx-i2s.c | 49 +- sound/soc/fsl/Kconfig | 121 +- sound/soc/fsl/Makefile | 29 +- sound/soc/{imx => fsl}/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_ssi.c | 167 +- sound/soc/fsl/fsl_utils.c | 91 + sound/soc/fsl/fsl_utils.h | 26 + sound/soc/{imx => fsl}/imx-audmux.c | 0 sound/soc/{imx => fsl}/imx-audmux.h | 0 .../{imx/imx-pcm-dma-mx2.c => fsl/imx-pcm-dma.c} | 3 +- sound/soc/{imx => fsl}/imx-pcm-fiq.c | 0 sound/soc/{imx => fsl}/imx-pcm.c | 0 sound/soc/{imx => fsl}/imx-pcm.h | 1 + sound/soc/fsl/imx-sgtl5000.c | 221 +++ sound/soc/{imx => fsl}/imx-ssi.c | 2 +- sound/soc/{imx => fsl}/imx-ssi.h | 0 sound/soc/fsl/mpc8610_hpcd.c | 166 +- sound/soc/{imx => fsl}/mx27vis-aic32x4.c | 0 sound/soc/fsl/p1022_ds.c | 158 +- sound/soc/{imx => fsl}/phycore-ac97.c | 0 sound/soc/{imx => fsl}/wm1133-ev1.c | 0 sound/soc/generic/Kconfig | 4 + sound/soc/generic/Makefile | 3 + sound/soc/generic/simple-card.c | 114 ++ sound/soc/imx/Kconfig | 79 - sound/soc/imx/Makefile | 22 - sound/soc/jz4740/jz4740-i2s.c | 4 +- sound/soc/mxs/mxs-pcm.c | 24 +- sound/soc/mxs/mxs-pcm.h | 3 + sound/soc/mxs/mxs-saif.c | 92 +- sound/soc/mxs/mxs-saif.h | 1 - sound/soc/mxs/mxs-sgtl5000.c | 50 +- sound/soc/pxa/pxa-ssp.c | 28 +- sound/soc/pxa/pxa2xx-i2s.c | 4 +- sound/soc/samsung/littlemill.c | 102 +- sound/soc/samsung/lowland.c | 75 +- sound/soc/samsung/speyside.c | 33 +- sound/soc/sh/Kconfig | 24 +- sound/soc/sh/Makefile | 6 - sound/soc/sh/fsi-ak4642.c | 108 -- sound/soc/sh/fsi-da7210.c | 81 - sound/soc/sh/fsi-hdmi.c | 118 -- sound/soc/soc-core.c | 690 +++++--- sound/soc/soc-dapm.c | 562 +++++-- sound/soc/soc-jack.c | 5 +- sound/soc/soc-pcm.c | 1772 +++++++++++++++++++- sound/soc/tegra/Kconfig | 58 +- sound/soc/tegra/Makefile | 18 +- sound/soc/tegra/tegra20_das.c | 233 +++ sound/soc/tegra/tegra20_das.h | 134 ++ sound/soc/tegra/tegra20_i2s.c | 494 ++++++ sound/soc/tegra/tegra20_i2s.h | 164 ++ sound/soc/tegra/tegra20_spdif.c | 404 +++++ sound/soc/tegra/tegra20_spdif.h | 471 ++++++ sound/soc/tegra/tegra30_ahub.c | 631 +++++++ sound/soc/tegra/tegra30_ahub.h | 483 ++++++ sound/soc/tegra/tegra30_i2s.c | 536 ++++++ sound/soc/tegra/tegra30_i2s.h | 242 +++ sound/soc/tegra/tegra_alc5632.c | 48 +- sound/soc/tegra/tegra_asoc_utils.c | 37 +- sound/soc/tegra/tegra_asoc_utils.h | 9 +- sound/soc/tegra/tegra_das.c | 261 --- sound/soc/tegra/tegra_das.h | 135 -- sound/soc/tegra/tegra_i2s.c | 459 ----- sound/soc/tegra/tegra_i2s.h | 166 -- sound/soc/tegra/tegra_pcm.c | 28 +- sound/soc/tegra/tegra_pcm.h | 5 +- sound/soc/tegra/tegra_spdif.c | 364 ---- sound/soc/tegra/tegra_spdif.h | 473 ------ sound/soc/tegra/tegra_wm8903.c | 29 +- sound/soc/tegra/trimslice.c | 41 +- sound/soc/ux500/Kconfig | 14 + sound/soc/ux500/Makefile | 4 + sound/soc/ux500/ux500_msp_dai.c | 843 ++++++++++ sound/soc/ux500/ux500_msp_dai.h | 79 + sound/soc/ux500/ux500_msp_i2s.c | 742 ++++++++ sound/soc/ux500/ux500_msp_i2s.h | 553 ++++++ 196 files changed, 17192 insertions(+), 5071 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt create mode 100644 Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt create mode 100644 Documentation/devicetree/bindings/sound/mxs-saif.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/tegra-audio-trimslice.txt create mode 100644 drivers/base/regmap/regmap-mmio.c create mode 100644 include/sound/cs42l52.h create mode 100644 include/sound/simple_card.h create mode 100644 include/sound/soc-dpcm.h create mode 100644 sound/soc/codecs/cs42l52.c create mode 100644 sound/soc/codecs/cs42l52.h create mode 100644 sound/soc/codecs/lm49453.c create mode 100644 sound/soc/codecs/lm49453.h create mode 100644 sound/soc/codecs/ml26124.c create mode 100644 sound/soc/codecs/ml26124.h rename sound/soc/{imx => fsl}/eukrea-tlv320.c (99%) create mode 100644 sound/soc/fsl/fsl_utils.c create mode 100644 sound/soc/fsl/fsl_utils.h rename sound/soc/{imx => fsl}/imx-audmux.c (100%) rename sound/soc/{imx => fsl}/imx-audmux.h (100%) rename sound/soc/{imx/imx-pcm-dma-mx2.c => fsl/imx-pcm-dma.c} (97%) rename sound/soc/{imx => fsl}/imx-pcm-fiq.c (100%) rename sound/soc/{imx => fsl}/imx-pcm.c (100%) rename sound/soc/{imx => fsl}/imx-pcm.h (93%) create mode 100644 sound/soc/fsl/imx-sgtl5000.c rename sound/soc/{imx => fsl}/imx-ssi.c (99%) rename sound/soc/{imx => fsl}/imx-ssi.h (100%) rename sound/soc/{imx => fsl}/mx27vis-aic32x4.c (100%) rename sound/soc/{imx => fsl}/phycore-ac97.c (100%) rename sound/soc/{imx => fsl}/wm1133-ev1.c (100%) create mode 100644 sound/soc/generic/Kconfig create mode 100644 sound/soc/generic/Makefile create mode 100644 sound/soc/generic/simple-card.c delete mode 100644 sound/soc/imx/Kconfig delete mode 100644 sound/soc/imx/Makefile delete mode 100644 sound/soc/sh/fsi-ak4642.c delete mode 100644 sound/soc/sh/fsi-da7210.c delete mode 100644 sound/soc/sh/fsi-hdmi.c create mode 100644 sound/soc/tegra/tegra20_das.c create mode 100644 sound/soc/tegra/tegra20_das.h create mode 100644 sound/soc/tegra/tegra20_i2s.c create mode 100644 sound/soc/tegra/tegra20_i2s.h create mode 100644 sound/soc/tegra/tegra20_spdif.c create mode 100644 sound/soc/tegra/tegra20_spdif.h create mode 100644 sound/soc/tegra/tegra30_ahub.c create mode 100644 sound/soc/tegra/tegra30_ahub.h create mode 100644 sound/soc/tegra/tegra30_i2s.c create mode 100644 sound/soc/tegra/tegra30_i2s.h delete mode 100644 sound/soc/tegra/tegra_das.c delete mode 100644 sound/soc/tegra/tegra_das.h delete mode 100644 sound/soc/tegra/tegra_i2s.c delete mode 100644 sound/soc/tegra/tegra_i2s.h delete mode 100644 sound/soc/tegra/tegra_spdif.c delete mode 100644 sound/soc/tegra/tegra_spdif.h create mode 100644 sound/soc/ux500/Kconfig create mode 100644 sound/soc/ux500/Makefile create mode 100644 sound/soc/ux500/ux500_msp_dai.c create mode 100644 sound/soc/ux500/ux500_msp_dai.h create mode 100644 sound/soc/ux500/ux500_msp_i2s.c create mode 100644 sound/soc/ux500/ux500_msp_i2s.h

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[alsa-devel] [PATCH] ASoC: don't flip master of DT-instantiated DAI links
by Stephen Warren 16 May '12

16 May '12

From: Stephen Warren <swarren(a)nvidia.com> Commit 4924082 "ASoC: core: Flip master for CODECs in the CPU slot of a CODEC<->CODEC link" added code that was conditional on there being no PCM/DMA driver for the link. However, it failed to cover the case where the link was instantiated from device tree, and hence was specified by DT node rather than name. This prevents the following error on Toshiba AC100: aplay: pcm_write:1603: write error: Input/output error Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- I tested this fix on AC100 (ALC5632), Harmony (WM8903), and TrimSlice (TLV320AIC23). I'm not sure why the issue only affected AC100. Perhaps the other codecs support being the I2S master but the ALC5632 driver doesn't? --- sound/soc/soc-core.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2783d2b..b37ee80 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1612,7 +1612,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } /* If this is a regular CPU link there will be a platform */ - if (dai_fmt && dai_link->platform_name) { + if (dai_fmt && + (dai_link->platform_name || dai_link->platform_of_node)) { ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai, dai_fmt); if (ret != 0 && ret != -ENOTSUPP) -- 1.7.0.4

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Re: [alsa-devel] [PATCH]: sound/usb/mixer.c: Volume quirks compare empty string kctl->id.name.
by Takashi Iwai 16 May '12

16 May '12

At Wed, 16 May 2012 10:03:42 +0200, Christian Melki wrote: > > Takashi, > > I tried it on 3.3.2 on my embedded PPC machine (snd_usb_mixer_vol_tlv -> mixer_vol_tlv). Works fine for me. Thanks, it was already queued for 3.5 kernel. Takashi > > Regards, > Christian > > > > At Mon, 7 May 2012 09:51:29 +0200, > > Christian Melki wrote: > > > > > > > > Hi, > > > > > > > > > > Sorry to mail you directly. But I have tried to mail > > several times > > > > > to the mailing list but I don't get through. Everything > > checks out > > > > > fine, no html etc, other mailing lists work fine, but > > not alsa devel. > > > > > Anyway, could you please have a look at this? > > > > > > > > Could you resend the patch but just add me, Clemens and > > Daniel to Cc, so that it reaches to main developers for > > usb-audio? Even if we take your patch as is, it'd be needed > > to be applicable via git-am. > > > > > > > > > > > > thanks, > > > > > > > > Takashi > > > > > > Hi. > > > > > > I noticied that volume quirks in usb-audio did not get applied. > > > This is because kctl->id.name gets built properly _after_ > > calling the quirk adjustments. > > > My solution is a bit ugly, but check_mapped_dB needs the > > adjustment before getting called. > > > I split the call over the switch. If someone feels like > > writing something cleaner, please feel free. > > > This works for me atleast. > > > > It's not ugly, but it'd be better to split the code to handle > > the name string, call get_min_max_*(), then the rest for FU. > > > > Could you check whether the patch below works? > > > > > > thanks, > > > > Takashi > > > > --- > > From: Takashi Iwai <tiwai(a)suse.de> > > Subject: [PATCH] ALSA: usb-audio - Call get_min_max_*() after > > determining the name string > > > > get_min_max_with_quirks() must be called after the control id > > name string is determined, but the current code changes the > > id name string after calling the function. > > > > Reported-by: Christian Melki <christian.melki(a)ericsson.com> > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > --- > > sound/usb/mixer.c | 26 +++++++++++++------------- > > 1 file changed, 13 insertions(+), 13 deletions(-) > > > > diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index > > 3d70245..4f40ba8 100644 > > --- a/sound/usb/mixer.c > > +++ b/sound/usb/mixer.c > > @@ -1141,9 +1141,6 @@ static void build_feature_ctl(struct > > mixer_build *state, void *raw_desc, > > len = snd_usb_copy_string_desc(state, nameid, > > kctl->id.name, sizeof(kctl->id.name)); > > > > - /* get min/max values */ > > - get_min_max_with_quirks(cval, 0, kctl); > > - > > switch (control) { > > case UAC_FU_MUTE: > > case UAC_FU_VOLUME: > > @@ -1175,17 +1172,7 @@ static void build_feature_ctl(struct > > mixer_build *state, void *raw_desc, > > } > > append_ctl_name(kctl, control == UAC_FU_MUTE ? > > " Switch" : " Volume"); > > - if (control == UAC_FU_VOLUME) { > > - check_mapped_dB(map, cval); > > - if (cval->dBmin < cval->dBmax || > > !cval->initialized) { > > - kctl->tlv.c = snd_usb_mixer_vol_tlv; > > - kctl->vd[0].access |= > > - SNDRV_CTL_ELEM_ACCESS_TLV_READ | > > - > > SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; > > - } > > - } > > break; > > - > > default: > > if (! len) > > strlcpy(kctl->id.name, > > audio_feature_info[control-1].name, > > @@ -1193,6 +1180,19 @@ static void build_feature_ctl(struct > > mixer_build *state, void *raw_desc, > > break; > > } > > > > + /* get min/max values */ > > + get_min_max_with_quirks(cval, 0, kctl); > > + > > + if (control == UAC_FU_VOLUME) { > > + check_mapped_dB(map, cval); > > + if (cval->dBmin < cval->dBmax || !cval->initialized) { > > + kctl->tlv.c = snd_usb_mixer_vol_tlv; > > + kctl->vd[0].access |= > > + SNDRV_CTL_ELEM_ACCESS_TLV_READ | > > + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; > > + } > > + } > > + > > range = (cval->max - cval->min) / cval->res; > > /* Are there devices with volume range more than 255? I > > use a bit more > > * to be sure. 384 is a resolution magic number found > > on Logitech > > -- > > 1.7.9.2 > > > >

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[alsa-devel] [PATCH] ASoC: dmic codec: Support for DT
by Peter Ujfalusi 15 May '12

15 May '12

Add bindings to be able to probe the dmic codec driver via device tree. The device tree entry of a machine driver can use phandle to provide the device node for it's dai link structure. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- .../devicetree/bindings/sound/dmic-codec.txt | 18 ++++++++++++++++++ sound/soc/codecs/dmic.c | 6 ++++++ 2 files changed, 24 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/dmic-codec.txt diff --git a/Documentation/devicetree/bindings/sound/dmic-codec.txt b/Documentation/devicetree/bindings/sound/dmic-codec.txt new file mode 100644 index 0000000..f23f2e3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/dmic-codec.txt @@ -0,0 +1,18 @@ +Generic platform independent codec driver to represent digital microphones + +The dmic codec driver basically a dummy driver which exports the needed hooks +to be used with digital microphone interfaces. + +Required properties: +- compatible : Must be "generic,dmic-codec"; + +Example: + +dmic_codec: dmic_codec@0 { /* generic DMIC codec */ + compatible = "generic,dmic-codec"; +}; + +sound { /* ASoC machine dirver */ + ... + dai-link,codec = <&dmic_codec>; +}; diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index 3e929f0..90e92f5 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -78,12 +78,18 @@ static int __devexit dmic_dev_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id dmic_of_match[] = { + {.compatible = "generic,dmic-codec", }, + { }, +}; +MODULE_DEVICE_TABLE(of, dmic_of_match); MODULE_ALIAS("platform:dmic-codec"); static struct platform_driver dmic_driver = { .driver = { .name = "dmic-codec", .owner = THIS_MODULE, + .of_match_table = dmic_of_match, }, .probe = dmic_dev_probe, .remove = __devexit_p(dmic_dev_remove), -- 1.7.8.6

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[alsa-devel] Undocumented Sound Blaster 16 extended mixer registers
by Ondrej Zary 15 May '12

15 May '12

Hello, while playing with various SB16 types, found that the newer types have some extended mixer registers that aren't documented anywhere. Windows 9x driver detects and uses them so e.g. Auxiliary input appears when using CT2511-based card, MIDI mute becomes available on some cards, etc. Chips tested: CT1746B+CT1745A: SB16 SCSI CT1747+CT1745A: SB16 IDE CT2504: Vibra 16S CT2505: Vibra 16C (PnP) CT2511: Vibra 16XV (PnP) CT8920: AWE64 (the last one with integrated mixer) (PnP) Registers found: ---------------- 0x48: Output switches 2 all supported bits are set after reset (0x1f on CT2505 and CT8920, 0x7f on CT2511) bit 0: PC Speaker bit 1: PCM bit 2: PCM bit 3: MIDI bit 4: MIDI bit 5: AUX (CT2511 only, zero otherwise) bit 6: AUX (CT2511 only, zero otherwise) bit 7: unused (always zero) 0x49: Master switches all supported bits are set after reset (0x0f on CT2511, 0x0c on CT8920) bit 0: capture bit 1: capture bit 2: playback bit 3: playback bits 4..7: unused (always zero) 0x4a: Input switches 2 zero after reset bit 0: PCM bit 1: PCM bit 2: AUX (CT2511 only, zero otherwise) bit 3: AUX (CT2511 only, zero otherwise) bits 4..7: unused (always zero) 0x4b: ?? 0x63 after reset (on CT2511 and CT8920) bits 0..5 writable CT2511 also provides two additional volume controls: 0x50,0x51: Master Capture Volume (bits 7..3) reads as 0xE0 after reset when present 0x52,0x53: AUX Volume (bits 7..3) reads as 0xC0 after reset when present There are no extended registers on the older cards (CT1746B+CT1745A, CT1747+CT1745A and CT2504). CT2505 implements only 0x48. AUX is an additional input, used at least on SF16-FMD2 card to connect integrated FM tuner. I'm going to prepare a patch that allows sb16 driver to detect these additional controls and enable them if found. However, I still haven't found how Windows driver detects Bass/Treble control presence. All chips have the bass/treble registers (0x44-0x47) present and writable but they have no effect on all Vibra chips except CT2502 (which I don't have). There must be some other register which says that bass/treble controls are not available. -- Ondrej Zary

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[alsa-devel] [PATCH 0/1] Enable sound on mx31_3ds
by Philippe Rétornaz 15 May '12

15 May '12

Hi Here is the sound support for mx31_3ds. It successfully outputs sound on the loudspeaker. It needs the patchset: "MC13783 sound support". Philippe Rétornaz (1): mx31_3ds: Add sound support arch/arm/mach-imx/Kconfig | 1 + arch/arm/mach-imx/mach-mx31_3ds.c | 22 +++++++++++++++++++++- 2 files changed, 22 insertions(+), 1 deletions(-)

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[alsa-devel] Last minute ASoC fixes for 3.4
by Mark Brown 15 May '12

15 May '12

The following changes since commit 06412088ce98f745405b8f65cfc51ddd6b842bbf: ASoC: s3c2412-i2s: Fix dai registration (2012-04-30 23:45:25 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.4 for you to fetch changes up to c7f5f2389377b66028bc129890aa653deafe8d39: ASoC: wm8994: Fix AIF2ADC power down (2012-05-15 18:29:30 +0100) I'll send the big 3.5 update either later tonight or tomorrow. ---------------------------------------------------------------- ASoC: Last minute fixes Some last minute fixes for ASoC. Small, focused changes to specific drivers. ---------------------------------------------------------------- Brian Austin (1): ASoC: cs42l73: Sync digital mixer kcontrols to allow for 0dB Guennadi Liakhovetski (1): ASoC: sh: fix migor.c compilation Mark Brown (1): ASoC: wm8994: Fix AIF2ADC power down sound/soc/codecs/cs42l73.c | 16 ++++++++-------- sound/soc/codecs/wm8994.c | 2 +- sound/soc/sh/migor.c | 2 +- 3 files changed, 10 insertions(+), 10 deletions(-)

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