March 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASoC: SSM2602: drop rate and sample bits constraints
by Mike Frysinger 30 Mar '11

30 Mar '11

From: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/ssm2602.c | 53 -------------------------------------------- 1 files changed, 0 insertions(+), 53 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 2727bef..70a33414 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -49,8 +49,6 @@ struct ssm2602_priv { unsigned int sysclk; enum snd_soc_control_type control_type; void *control_data; - struct snd_pcm_substream *master_substream; - struct snd_pcm_substream *slave_substream; }; /* @@ -278,15 +276,9 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c = codec->control_data; u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3; int i = get_coeff(ssm2602->sysclk, params_rate(params)); - if (substream == ssm2602->slave_substream) { - dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n"); - return 0; - } - /*no match is found*/ if (i == ARRAY_SIZE(coeff_div)) return -EINVAL; @@ -316,44 +308,6 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, return 0; } -static int ssm2602_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; - struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c = codec->control_data; - struct snd_pcm_runtime *master_runtime; - - /* The DAI has shared clocks so if we already have a playback or - * capture going then constrain this substream to match it. - * TODO: the ssm2602 allows pairs of non-matching PB/REC rates - */ - if (ssm2602->master_substream) { - master_runtime = ssm2602->master_substream->runtime; - dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n", - master_runtime->sample_bits, - master_runtime->rate); - - if (master_runtime->rate != 0) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - master_runtime->rate, - master_runtime->rate); - - if (master_runtime->sample_bits != 0) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - master_runtime->sample_bits, - master_runtime->sample_bits); - - ssm2602->slave_substream = substream; - } else - ssm2602->master_substream = substream; - - return 0; -} - static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -370,16 +324,10 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; - struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); /* deactivate */ if (!codec->active) ssm2602_write(codec, SSM2602_ACTIVE, 0); - - if (ssm2602->master_substream == substream) - ssm2602->master_substream = ssm2602->slave_substream; - - ssm2602->slave_substream = NULL; } static int ssm2602_mute(struct snd_soc_dai *dai, int mute) @@ -505,7 +453,6 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_ops ssm2602_dai_ops = { - .startup = ssm2602_startup, .prepare = ssm2602_pcm_prepare, .hw_params = ssm2602_hw_params, .shutdown = ssm2602_shutdown, -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: Blackfin: add ad193x sysclk configuration
by Mike Frysinger 30 Mar '11

30 Mar '11

From: Scott Jiang <scott.jiang(a)analog.com> Signed-off-by: Scott Jiang <scott.jiang(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/bf5xx-ad193x.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index d3ccb92..3367882 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -60,8 +60,16 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int clk = 0; unsigned int channel_map[] = {0, 1, 2, 3, 4, 5, 6, 7}; int ret = 0; + + switch (params_rate(params)) { + case 48000: + clk = 12288000; + break; + } + /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); @@ -74,6 +82,12 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + /* set codec DAI slots, 8 channels, all channels are enabled */ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 0xFF, 8, 32); if (ret < 0) -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: SSM2602: fix codec name
by Mike Frysinger 30 Mar '11

30 Mar '11

The codec name should not have a "-codec" suffix since this is not part of a MFD. This was incorrectly changed during the multi-component updated. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/ssm2602.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 2727bef..5a5a6b1 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -651,7 +651,7 @@ MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id); /* corgi i2c codec control layer */ static struct i2c_driver ssm2602_i2c_driver = { .driver = { - .name = "ssm2602-codec", + .name = "ssm2602", .owner = THIS_MODULE, }, .probe = ssm2602_i2c_probe, -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: fix sorting order of codecs in kconfig
by Mike Frysinger 30 Mar '11

30 Mar '11

Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f310043..b814ed0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,8 +16,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD1836 if SPI_MASTER select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS - select SND_SOC_ADS117X select SND_SOC_AD73311 + select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: ad74111: new codec driver
by Mike Frysinger 30 Mar '11

30 Mar '11

From: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Scott Jiang <scott.jiang(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/Kconfig | 8 +++- sound/soc/codecs/Makefile | 4 ++- sound/soc/codecs/ad74111.c | 76 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ad74111.h | 63 ++++++++++++++++++++++++++++++++++++ 4 files changed, 148 insertions(+), 3 deletions(-) create mode 100644 sound/soc/codecs/ad74111.c create mode 100644 sound/soc/codecs/ad74111.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index dc6d253..ee65b7c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,8 +16,9 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD183X if SPI_MASTER select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS - select SND_SOC_ADS117X select SND_SOC_AD73311 if I2C + select SND_SOC_AD74111 + select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C @@ -126,7 +127,10 @@ config SND_SOC_AD1980 config SND_SOC_AD73311 tristate - + +config SND_SOC_AD74111 + tristate + config SND_SOC_ADS117X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 44d3278..b834755 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -4,6 +4,7 @@ snd-soc-ad183x-objs := ad183x.o snd-soc-ad193x-objs := ad193x.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o +snd-soc-ad74111-objs := ad74111.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o @@ -90,7 +91,8 @@ obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD183X) += snd-soc-ad183x.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o -obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o +obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o +obj-$(CONFIG_SND_SOC_AD74111) += snd-soc-ad74111.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o diff --git a/sound/soc/codecs/ad74111.c b/sound/soc/codecs/ad74111.c new file mode 100644 index 0000000..9d247ae --- /dev/null +++ b/sound/soc/codecs/ad74111.c @@ -0,0 +1,76 @@ +/* + * ALSA Soc AD74111 codec support + * + * Copyright 2009-2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "ad74111.h" + +static struct snd_soc_dai_driver ad74111_dai = { + .name = "ad74111-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_ad74111; + +static int ad74111_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, + &soc_codec_dev_ad74111, &ad74111_dai, 1); +} + +static int __devexit ad74111_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver ad74111_codec_driver = { + .driver = { + .name = "ad74111-codec", + .owner = THIS_MODULE, + }, + + .probe = ad74111_probe, + .remove = __devexit_p(ad74111_remove), +}; + +static int __init ad74111_init(void) +{ + return platform_driver_register(&ad74111_codec_driver); +} +module_init(ad74111_init); + +static void __exit ad74111_exit(void) +{ + platform_driver_unregister(&ad74111_codec_driver); +} +module_exit(ad74111_exit); + +MODULE_DESCRIPTION("ASoC AD74111 driver"); +MODULE_AUTHOR("Cliff Cai"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad74111.h b/sound/soc/codecs/ad74111.h new file mode 100644 index 0000000..19a8183 --- /dev/null +++ b/sound/soc/codecs/ad74111.h @@ -0,0 +1,63 @@ +/* + * definitions for AD74111 registers + * + * Copyright 2006-2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#ifndef __AD74111_H__ +#define __AD74111_H__ + +#define AD_READ 0x0000 +#define AD_WRITE 0x8000 + +/* Control register A */ +#define CTRL_REG_A (0 << 11) + +#define REGA_REFAMP (1 << 2) +#define REGA_REF (1 << 3) +#define REGA_DAC (1 << 4) +#define REGA_ADC (1 << 5) +#define REGA_ADC_INPAMP (1 << 6) + +/* Control register B */ +#define CTRL_REG_B (1 << 11) + +#define REGB_FCLKDIV(x) ((x) & 0x3) +#define REGB_SCLKDIV(x) (((x) & 0x3) << 2) +#define REGB_TCLKDIV(x) (((x) & 0x3) << 4) + +/* Control register C */ +#define CTRL_REG_C (2 << 11) + +#define REGC_ADC_HP (1 << 0) +#define REGC_DAC_DEEMPH(x) (((x) & 0x3) << 1) +#define REGC_LG_DELAY (1 << 3) +#define REGC_WORD_WIDTH(x) (((x) & 0x3) << 4) + +/* Control register D */ +#define CTRL_REG_D (3 << 11) + +#define REGD_MASTER (1 << 0) +#define REGD_FDCLK (1 << 1) +#define REGD_DSP_MODE (1 << 2) +#define REGD_MIX_MODE (1 << 3) +#define REGD_MFS (1 << 9) + +/* Control register E */ +#define CTRL_REG_E (4 << 11) + +#define REGE_DAC_MUTE (1 << 0) +#define REGE_ADC_MUTE (1 << 1) +#define REGE_ADC_GAIN(x) (((x) & 0x7) << 2) +#define REGE_ADC_PEAKEN (1 << 5) + +/* Control register F */ +#define CTRL_REG_F (5 << 11) +#define REGF_DAC_VOL(x) ((x) & 0x3F) + +/* Control register G */ +#define CTRL_REG_G (6 << 11) + +#endif -- 1.7.4.1

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[alsa-devel] [PATCH 1/4] ASoC: Fix CODEC device name for Corgi
by Mark Brown 29 Mar '11

29 Mar '11

Got typoed in the multi-component changes. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/pxa/corgi.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 784cff5..9027da4 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -310,7 +310,7 @@ static struct snd_soc_dai_link corgi_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8731-codec-0.001b", + .codec_name = "wm8731-codec.0-001b", .init = corgi_wm8731_init, .ops = &corgi_ops, }; -- 1.7.4.1

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[alsa-devel] radio-maestro broken (conflicts with snd-es1968)
by Ondrej Zary 29 Mar '11

29 Mar '11

Hello, the radio-maestro driver is badly broken. It's intended to drive the radio on MediaForte ESS Maestro-based sound cards with integrated radio (like SF64-PCE2-04). But it conflicts with snd_es1968, ALSA driver for the sound chip itself. If one driver is loaded, the other one does not work - because a driver is already registered for the PCI device (there is only one). This was probably broken by conversion of PCI probing in 2006: ttp://lkml.org/lkml/2005/12/31/93 How to fix it properly? Include radio functionality in snd-es1968 and delete radio-maestro? -- Ondrej Zary

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[alsa-devel] emu10k1, midi on audigy platinum ex
by Goncalo Pinheira 29 Mar '11

29 Mar '11

Hello, I've noticed midi in (DIN connector) on an audigy platinum ex doesn't work. There's a patch in a 2009 forum thread http://ubuntuforums.org/showthread.php?t=1354637&page=3 The author seems to have contacted with the alsa project, but the latest kernel shows no sign of the code. Shouldn't this be commited?

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[alsa-devel] [PATCH alsa-kernel] snd-hda-codec-realtek: Add quirk for Jetway NF92
by Forest Bond 29 Mar '11

29 Mar '11

The Jetway NF92 mainboard works with model=3stack-6ch-dig. Add a quirk to automatically configure the codec accordingly. Signed-off-by: Forest Bond <forest.bond(a)rapidrollout.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f1a03f2..6cc8378 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -18777,6 +18777,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", ALC663_ASUS_H13), SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E), + SND_PCI_QUIRK(0x16f3, 0xa625, "Jetway NF92", ALC662_3ST_6ch_DIG), {} }; -- 1.7.0.4

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Re: [alsa-devel] Loud pops from Intel HDA onboard sound, may damage speakers (fwd)
by Takashi Iwai 29 Mar '11

29 Mar '11

At Sun, 27 Mar 2011 03:27:28 +0100 (BST), Chris Wilson wrote: > > Hi Takashi, > > On Sat, 26 Mar 2011, Takashi Iwai wrote: > > > I attached below. This is no final version and we are working on it. But > > it'd be helpful if you can test it whether it solves the issue. > > Thanks for sending the patch :) I think it makes some progress towards > solving the issue. On suspend I now get only one pop instead of two, and > none on resume. The pop that I still get on suspend is the one that causes > my amplifier to trip out (which was the second one previously). Good to hear of improvements. I cleaned up the patch a bit. Could you try the patch below instead? I removed the second msleep() in the resume callback, supposedly it's irrelevant. If you get a noise again in the resume path, try to add msleep(150) after codec->patch_ops.init() in alc_resume(). The remaining noise in the suspend path is still unclear. We have one place that we do NOT turn down the power to D3 -- in hda_set_power_state() in hda_codec.c, D3-transition is skipped on a widget with EAPD control. It's interesting whether any widgets hit this. Otherwise, we may need to check which call actually gives the noise. You can put a printk() and a long timeout at each line, and watch the kernel message.... thanks, Takashi --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 12c6f45..29eb397 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -375,6 +375,7 @@ struct alc_spec { #ifdef CONFIG_SND_HDA_POWER_SAVE void (*power_hook)(struct hda_codec *codec); #endif + void (*shutup)(struct hda_codec *codec); /* for pin sensing */ unsigned int sense_updated: 1; @@ -1236,6 +1237,15 @@ static void set_eapd(struct hda_codec *codec, hda_nid_t nid, int on) on ? 2 : 0); } +static void alc_eapd_shutup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + set_eapd(codec, 0x14, 0); + set_eapd(codec, 0x15, 0); + msleep(200); +} + static void alc_auto_init_amp(struct hda_codec *codec, int type) { unsigned int tmp; @@ -4193,6 +4203,10 @@ static int alc_build_pcms(struct hda_codec *codec) static inline void alc_shutup(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; + + if (spec && spec->shutup) + spec->shutup(codec); snd_hda_shutup_pins(codec); } @@ -4263,6 +4277,7 @@ static int alc_suspend(struct hda_codec *codec, pm_message_t state) #ifdef SND_HDA_NEEDS_RESUME static int alc_resume(struct hda_codec *codec) { + msleep(150); /* to avoid pop noise */ codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); @@ -13018,6 +13033,7 @@ static int patch_alc262(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC262_AUTO) spec->init_hook = alc262_auto_init; + spec->shutup = alc_eapd_shutup; alc_init_jacks(codec); #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -14092,6 +14108,7 @@ static int patch_alc268(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC268_AUTO) spec->init_hook = alc268_auto_init; + spec->shutup = alc_eapd_shutup; alc_init_jacks(codec); @@ -17410,6 +17418,8 @@ static int patch_alc861vd(struct hda_codec *codec) if (board_config == ALC861VD_AUTO) spec->init_hook = alc861vd_auto_init; + spec->shutup = alc_eapd_shutup; + #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc861vd_loopbacks; @@ -19611,6 +19621,7 @@ static int patch_alc662(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC662_AUTO) spec->init_hook = alc662_auto_init; + spec->shutup = alc_eapd_shutup; alc_init_jacks(codec);

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