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

[alsa-devel] [PATCH 1/3] ASoC: Convert WM8962 MICBIAS to a supply widget
by Mark Brown 23 Sep '11

23 Sep '11

A supply widget is generally clearer than a MICBIAS widget and a mic bias is just a type of supply so use a supply widget for the MICBIAS. This also avoids confusion with the routing when connected to multiple inputs. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 2 +- sound/soc/samsung/speyside_wm8962.c | 10 +++++----- 2 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8679f06..59b2ad6 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2669,7 +2669,7 @@ SND_SOC_DAPM_INPUT("IN4R"), SND_SOC_DAPM_INPUT("Beep"), SND_SOC_DAPM_INPUT("DMICDAT"), -SND_SOC_DAPM_MICBIAS("MICBIAS", WM8962_PWR_MGMT_1, 1, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS", WM8962_PWR_MGMT_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Class G", WM8962_CHARGE_PUMP_B, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("SYSCLK", WM8962_CLOCKING2, 5, 0, sysclk_event, diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 98b28ea..35539eb 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -152,12 +152,12 @@ static struct snd_soc_dapm_route audio_paths[] = { { "Main Speaker", NULL, "SPKOUTL" }, { "Main Speaker", NULL, "SPKOUTR" }, - { "MICBIAS", NULL, "Headset Mic" }, - { "IN4L", NULL, "MICBIAS" }, - { "IN4R", NULL, "MICBIAS" }, + { "Headset Mic", NULL, "MICBIAS" }, + { "IN4L", NULL, "Headset Mic" }, + { "IN4R", NULL, "Headset Mic" }, - { "MICBIAS", NULL, "DMIC" }, - { "DMICDAT", NULL, "MICBIAS" }, + { "DMIC", NULL, "MICBIAS" }, + { "DMICDAT", NULL, "DMIC" }, }; static struct snd_soc_jack speyside_wm8962_headset; -- 1.7.6.3

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[alsa-devel] [PATCH] ASoC: Rename WM8962 DMIC widget to DMIC_ENA
by Mark Brown 23 Sep '11

23 Sep '11

Matches the register name and avoids confusion with board widgets. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 431b976..59b2ad6 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2690,7 +2690,7 @@ SND_SOC_DAPM_MIXER("MIXINL", WM8962_PWR_MGMT_1, 5, 0, SND_SOC_DAPM_MIXER("MIXINR", WM8962_PWR_MGMT_1, 4, 0, mixinr, ARRAY_SIZE(mixinr)), -SND_SOC_DAPM_AIF_IN("DMIC", NULL, 0, WM8962_PWR_MGMT_1, 10, 0), +SND_SOC_DAPM_AIF_IN("DMIC_ENA", NULL, 0, WM8962_PWR_MGMT_1, 10, 0), SND_SOC_DAPM_ADC("ADCL", "Capture", WM8962_PWR_MGMT_1, 3, 0), SND_SOC_DAPM_ADC("ADCR", "Capture", WM8962_PWR_MGMT_1, 2, 0), @@ -2769,18 +2769,18 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "MICBIAS", NULL, "SYSCLK" }, - { "DMIC", NULL, "DMICDAT" }, + { "DMIC_ENA", NULL, "DMICDAT" }, { "ADCL", NULL, "SYSCLK" }, { "ADCL", NULL, "TOCLK" }, { "ADCL", NULL, "MIXINL" }, - { "ADCL", NULL, "DMIC" }, + { "ADCL", NULL, "DMIC_ENA" }, { "ADCL", NULL, "DSP2" }, { "ADCR", NULL, "SYSCLK" }, { "ADCR", NULL, "TOCLK" }, { "ADCR", NULL, "MIXINR" }, - { "ADCR", NULL, "DMIC" }, + { "ADCR", NULL, "DMIC_ENA" }, { "ADCR", NULL, "DSP2" }, { "STL", "Left", "ADCL" }, -- 1.7.6.3

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[alsa-devel] [PATCH] ASoC: Support a wider range of sample rates on Speyside WM8962
by Mark Brown 23 Sep '11

23 Sep '11

As we've only got one audio interface and it is symmetric we can just set SYSCLK based on the sample rate requested by the application layer. Provide a default so bypass paths work before audio playback. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/speyside_wm8962.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 3820a6b..98b28ea 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -16,6 +16,8 @@ #include "../codecs/wm8962.h" +static int sample_rate = 44100; + static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) @@ -31,13 +33,13 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, 32768, - 44100 * 512); + sample_rate * 512); if (ret < 0) pr_err("Failed to start FLL: %d\n", ret); ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, - 44100 * 512, + sample_rate * 512, SND_SOC_CLOCK_IN); if (ret < 0) { pr_err("Failed to set SYSCLK: %d\n", ret); @@ -109,6 +111,8 @@ static int speyside_wm8962_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + sample_rate = params_rate(params); + return 0; } -- 1.7.6.3

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[alsa-devel] [PATCH] sound/usb/card.c: clear chip->probing on error exit
by Thomas Pfaff 23 Sep '11

23 Sep '11

The Terratec Aureon 5.1 USB sound card support is broken since kernel 2.6.39. 2.6.39 introduced power management support for USB sound cards that added a probing flag in struct snd_usb_audio. During the probe of the card it gives following error message : usb 7-2: new full speed USB device number 2 using uhci_hcd cannot find UAC_HEADER snd-usb-audio: probe of 7-2:1.3 failed with error -5 input: USB Audio as /devices/pci0000:00/0000:00:1d.1/usb7/7-2/7-2:1.3/input/input6 generic-usb 0003:0CCD:0028.0001: input: USB HID v1.00 Device [USB Audio] on usb-0000:00:1d.1-2/input3 I can not comment about that "cannot find UAC_HEADER" error, but until 2.6.38 the card worked anyway. With 2.6.39 chip->probing remains 1 on error exit, and any later ioctl stops in snd_usb_autoresume with -ENODEV. Signed-off-by: Thomas Pfaff <tpfaff(a)gmx.net> --- diff -urp a/sound/usb/card.c b/sound/usb/card.c --- a/sound/usb/card.c 2011-09-22 08:42:29.138247328 +0200 +++ b/sound/usb/card.c 2011-09-22 09:02:09.382079980 +0200 @@ -532,6 +532,7 @@ snd_usb_audio_probe(struct usb_device *d __error: if (chip && !chip->num_interfaces) snd_card_free(chip->card); + chip->probing = 0; mutex_unlock(&register_mutex); __err_val: return NULL;

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[alsa-devel] ASoC updates for 3.2
by Mark Brown 23 Sep '11

23 Sep '11

The following changes since commit f648de832dbf6d1947ce5a7c0ed24a3a71d8545b: Merge branch 'for-3.1' into for-3.2 (2011-09-20 12:59:35 +0100) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 Axel Lin (5): ASoC: Staticise bf5xx_pcm_ac97_new() ASoC: Staticise bf5xx_pcm_i2s_new() ASoC: Staticise jz4740_pcm_new() ASoC: Staticise nuc900_dma_getposition() ASoC: Add Kconfig and Makefile entries for rt5631 codec Bas Vermeulen (2): ASoC: 88pm860x-codec - Allow independent use of both I2S playback and capture ASoC: 88pm860x-codec - reset the codec correctly Dong Aisheng (1): ASoC: soc-core: symmetry checking for each DAIs separately Mark Brown (10): ASoC: Add WM5100 driver ASoC: Trace and collect statistics for DAPM graph walking ASoC: Factor out per-widget DAPM power checks ASoC: Ensure we generate a driver name Merge branch 'for-3.1' into for-3.2 ASoC: Include delay.h in 88pm860x Merge branch 'peter/topic/for-mark/mcpdm_for-3.2' of git://gitorious.org/omap-audio/linux-audio into for-3.2 ASoC: Dynamically manage DBVDD2 and DBVDD3 on WM5100 ASoC: Add another DAPM stat for neighbour checks ASoC: Add missed BCLK rate to WM5100 driver Misael Lopez Cruz (1): ASoC: omap-mcpdm: Replace legacy driver Peter Ujfalusi (16): ASoC: tpa6130a2: Model support cleanup ASoC: tpa6130a2: Remove model_id from platform data OMAP: McPDM: Convert McPDM device to omap_device OMAP4: hwmod: enable mcpdm hwmod device. ASoC: omap-mcpdm: Fix threshold and dma configuration ASoC: OMAP4: McPDM: Convert to hwmod/omap_device ASoC: twl6040: Lower the power on gain values at startup ASoC: twl6040: Fix comments for register names ASoC: twl6040: Remove strings "NULL" from DAPM route ASoC: twl6040: Introduce SW only shadow register ASoC: twl6040: Earphone path correction ASoC: twl6040: Use consistent names for Handsfree path ASoC: twl6040: Use consistent names for Headset path ASoC: twl6040: Support for AUX L/R output ASoC: twl6040/sdp4430: Change legacy DAI name ASoC/MFD: twl6040: Combine bit definitions for Headset control registers johnnyhsu(a)realtek.com (1): ASoC: Add driver for rt5631 arch/arm/mach-omap2/board-rx51-peripherals.c | 1 - arch/arm/mach-omap2/devices.c | 33 + arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 +- arch/arm/plat-omap/devices.c | 36 - include/linux/mfd/twl6040.h | 13 +- include/sound/soc-dai.h | 3 + include/sound/soc-dapm.h | 6 + include/sound/soc.h | 3 +- include/sound/tpa6130a2-plat.h | 6 - include/sound/wm5100.h | 59 + include/trace/events/asoc.h | 25 + sound/soc/blackfin/bf5xx-ac97-pcm.c | 2 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 2 +- sound/soc/codecs/88pm860x-codec.c | 14 +- sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/rt5631.c | 1789 +++++++++ sound/soc/codecs/rt5631.h | 701 ++++ sound/soc/codecs/tpa6130a2.c | 12 +- sound/soc/codecs/twl6040.c | 223 +- sound/soc/codecs/wm5100-tables.c | 1530 ++++++++ sound/soc/codecs/wm5100.c | 2641 +++++++++++++ sound/soc/codecs/wm5100.h | 5146 ++++++++++++++++++++++++++ sound/soc/jz4740/jz4740-pcm.c | 2 +- sound/soc/nuc900/nuc900-pcm.c | 2 +- sound/soc/omap/Makefile | 2 +- sound/soc/omap/mcpdm.c | 470 --- sound/soc/omap/mcpdm.h | 153 - sound/soc/omap/omap-mcpdm.c | 442 ++- sound/soc/omap/omap-mcpdm.h | 95 + sound/soc/omap/sdp4430.c | 6 +- sound/soc/soc-core.c | 18 +- sound/soc/soc-dapm.c | 133 +- sound/soc/soc-pcm.c | 40 +- 34 files changed, 12671 insertions(+), 951 deletions(-) create mode 100644 include/sound/wm5100.h create mode 100644 sound/soc/codecs/rt5631.c create mode 100644 sound/soc/codecs/rt5631.h create mode 100644 sound/soc/codecs/wm5100-tables.c create mode 100644 sound/soc/codecs/wm5100.c create mode 100644 sound/soc/codecs/wm5100.h delete mode 100644 sound/soc/omap/mcpdm.c delete mode 100644 sound/soc/omap/mcpdm.h create mode 100644 sound/soc/omap/omap-mcpdm.h

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[alsa-devel] [PATCH 1/1] HDA - Add Independent Headphone for all models of ad1988/ad1989
by Raymond Yau 23 Sep '11

23 Sep '11

- Add "AD198x Headphone" playback device for independent headphone playback while playing 7.1 surround using rear panel audio jacks. - Remove "6stack-dig-fp" model since "Headphone Playback Volume" control using DAC0 instead of DAC1 (HDA_FRONT) was already added to all models. - Add "Independent HP" switch to enable/disable this playback device. When the switch is OFF, headphone use "copy front" mode to get the front channel as the green jack. When the switch is ON, you can play stereo sound through "AD198x Headphone" device to headphone while playing 7.1 surround sound through "AD198x Analog" device. The switch cannot be changed when either "AD198x Headphone" or "AD198X Analog" is open.

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Re: [alsa-devel] [PATCH] alsa: 6fire: don't use custom hex_to_bin()
by Takashi Iwai 23 Sep '11

23 Sep '11

At Fri, 23 Sep 2011 14:32:11 +0300, Andy Shevchenko wrote: > > Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: alsa-devel(a)alsa-project.org Applied now. Thanks. Takashi > --- > sound/usb/6fire/firmware.c | 25 ++++++++++++------------- > 1 files changed, 12 insertions(+), 13 deletions(-) > > diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c > index 3ebbdec..3b5f517 100644 > --- a/sound/usb/6fire/firmware.c > +++ b/sound/usb/6fire/firmware.c > @@ -17,6 +17,7 @@ > #include <linux/firmware.h> > #include <linux/module.h> > #include <linux/bitrev.h> > +#include <linux/kernel.h> > > #include "firmware.h" > #include "chip.h" > @@ -60,21 +61,19 @@ struct ihex_record { > unsigned int txt_offset; /* current position in txt_data */ > }; > > -static u8 usb6fire_fw_ihex_nibble(const u8 n) > -{ > - if (n >= '0' && n <= '9') > - return n - '0'; > - else if (n >= 'A' && n <= 'F') > - return n - ('A' - 10); > - else if (n >= 'a' && n <= 'f') > - return n - ('a' - 10); > - return 0; > -} > - > static u8 usb6fire_fw_ihex_hex(const u8 *data, u8 *crc) > { > - u8 val = (usb6fire_fw_ihex_nibble(data[0]) << 4) | > - usb6fire_fw_ihex_nibble(data[1]); > + u8 val = 0; > + int hval; > + > + hval = hex_to_bin(data[0]); > + if (hval >= 0) > + val |= (hval << 4); > + > + hval = hex_to_bin(data[1]); > + if (hval >= 0) > + val |= hval; > + > *crc += val; > return val; > } > -- > 1.7.6.3 >

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[alsa-devel] [PATCH] ASoC: twl6040: No need to read the INTID register
by Peter Ujfalusi 23 Sep '11

23 Sep '11

Since our irq handler has been called, it is granted, that the reason was either PLUGINT, or UNPLUGINT. The INTID register has been checked in the MFD part of twl6040 driver (twl6040-irq.c). We have no reason to read from chip again here. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/twl6040.c | 9 ++------- 1 files changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index f924827..d5c4bb4 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -747,15 +747,10 @@ static void twl6040_accessory_work(struct work_struct *work) static irqreturn_t twl6040_audio_handler(int irq, void *data) { struct snd_soc_codec *codec = data; - struct twl6040 *twl6040 = codec->control_data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - u8 intid; - - intid = twl6040_reg_read(twl6040, TWL6040_REG_INTID); - if ((intid & TWL6040_PLUGINT) || (intid & TWL6040_UNPLUGINT)) - queue_delayed_work(priv->workqueue, &priv->delayed_work, - msecs_to_jiffies(200)); + queue_delayed_work(priv->workqueue, &priv->delayed_work, + msecs_to_jiffies(200)); return IRQ_HANDLED; } -- 1.7.6.1

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Re: [alsa-devel] [PATCH 1/2] ASoC: Staticize rt5631_dai
by Mark Brown 23 Sep '11

23 Sep '11

On Fri, Sep 23, 2011 at 04:22:07PM +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied both, thanks.

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Re: [alsa-devel] [PATCH] ASoC: Add missed free_irq in wm5100_remove and wm5100_probe error path
by Mark Brown 23 Sep '11

23 Sep '11

On Fri, Sep 23, 2011 at 01:23:10PM +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied, thanks.

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