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[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 05 May '09

05 May '09

The following changes since commit 6574612fbb34c63117581e68f2231ddce027e41e: Takashi Iwai (1): ASoC: Remove BROKEN from mpc5200 kconfig are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Jinyoung Park (1): ASoC: Fix errors in WM8990 sound/soc/codecs/wm8990.c | 40 ++++++++++++++++++++-------------------- 1 files changed, 20 insertions(+), 20 deletions(-)

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[alsa-devel] [PATCH 0/1] ASoC: TWL4030: VIBRA output support
by Peter Ujfalusi 05 May '09

05 May '09

Hello, The VIBRA output can be driven with audio data or with local vibrator driver, the following patch adds VIBRA output support (for driving the VIBRA output with audio data) for the TWL4030 codec. --- Peter Ujfalusi (1): ASoC: TWL4030: Add VIBRA output sound/soc/codecs/twl4030.c | 61 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 61 insertions(+), 0 deletions(-)

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[alsa-devel] [ASoC PATCH 1/1] IMote 2 ASoC Support
by Jonathan Cameron 04 May '09

04 May '09

From: Jonathan Cameron <jic23(a)cam.ac.uk> IMote2 ASoC Support This patch adds the ASoC side of the board support for the Crossbow IMB400 daughter board. Thanks to Crossbow for considerable assistance. Signed-off-by: Jonathan Cameron <jic23(a)cam.ac.uk> --- Relatively straight forward configuration with the PXA acting as the clock master. sound/soc/pxa/Kconfig | 9 +++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/imote2.c | 131 +++++++++++++++++++++++++++++++++++++++++++++++ diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index ad8a10f..eb75a1c 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -134,3 +134,12 @@ config SND_PXA2XX_SOC_MIOA701 help Say Y if you want to add support for SoC audio on the MIO A701. + +config SND_PXA2XX_SOC_IMOTE2 + tristate "SoC Audio support for IMote 2" + depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 + select SND_PXA2XX_SOC_I2S + select SND_SOC_WM8940 + help + Say Y if you want to add support for SoC audio on the + IMote 2. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 4b90c3c..6e096b4 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -22,6 +22,7 @@ snd-soc-palm27x-objs := palm27x.o snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o +snd-soc-imote2-objs := imote2.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -35,3 +36,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o +obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c new file mode 100644 index 0000000..8476151 --- /dev/null +++ b/sound/soc/pxa/imote2.c @@ -0,0 +1,131 @@ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/pxa27x.h> +#include <mach/hardware.h> + +#include <mach/audio.h> + +#include "../codecs/wm8940.h" +#include "pxa2xx-i2s.h" +#include "pxa2xx-pcm.h" + +static int imote2_asoc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int clk = 0; + int ret; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + clk = 11289600; + break; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* CPU should be clock master */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* Disable the PLL */ + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, 0); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set the I2S system clock as input (unused) */ + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, clk, + SND_SOC_CLOCK_OUT); + + return ret; +} + +static struct snd_soc_ops imote2_asoc_ops = { + .hw_params = imote2_asoc_hw_params, +}; + +static struct snd_soc_dai_link imote2_dai = { + .name = "WM8940", + .stream_name = "WM8940", + .cpu_dai = &pxa_i2s_dai, + .codec_dai = &wm8940_dai, + .ops = &imote2_asoc_ops, +}; + +static struct snd_soc_card snd_soc_imote2 = { + .name = "Imote2", + .platform = &pxa2xx_soc_platform, + .dai_link = &imote2_dai, + .num_links = 1, +}; + +static struct snd_soc_device imote2_snd_devdata = { + .card = &snd_soc_imote2, + .codec_dev = &soc_codec_dev_wm8940, +}; + +static struct platform_device *imote2_snd_device; + +static int __init imote2_asoc_init(void) +{ + int ret; + + if (!machine_is_intelmote2()) + return -ENODEV; + imote2_snd_device = platform_device_alloc("soc-audio", -1); + if (!imote2_snd_device) + return -ENOMEM; + + platform_set_drvdata(imote2_snd_device, &imote2_snd_devdata); + imote2_snd_devdata.dev = &imote2_snd_device->dev; + ret = platform_device_add(imote2_snd_device); + if (ret) + platform_device_put(imote2_snd_device); + + return ret; +} +module_init(imote2_asoc_init); + +static void __exit imote2_asoc_exit(void) +{ + platform_device_unregister(imote2_snd_device); +} +module_exit(imote2_asoc_exit); + +MODULE_AUTHOR("Jonathan Cameron"); +MODULE_DESCRIPTION("ALSA SoC Imote 2"); +MODULE_LICENSE("GPL");

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[alsa-devel] [PATCH 1/2] ASoC: TWL4030: Add VDL analog bypass
by Lopez Cruz, Misael 04 May '09

04 May '09

This patch adds voice downlink analog bypass switch. It follows the same approach as in other analog bypass switches. DAC switch is moved from 'DAC Voice' to 'Analog Voice Playback Mixer', that will also allow voice DAC to be powered in digital voice loopback (sidetone). Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 37 +++++++++++++++++++++++++++++++------ 1 files changed, 31 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index efa1a80..efb371f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -468,6 +468,10 @@ static const struct snd_kcontrol_new twl4030_dapm_abypassr2_control = static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control = SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL2_APGA_CTL, 2, 1, 0); +/* Analog bypass for Voice */ +static const struct snd_kcontrol_new twl4030_dapm_abypassv_control = + SOC_DAPM_SINGLE("Switch", TWL4030_REG_VDL_APGA_CTL, 2, 1, 0); + /* Digital bypass gain, 0 mutes the bypass */ static const unsigned int twl4030_dapm_dbypass_tlv[] = { TLV_DB_RANGE_HEAD(2), @@ -585,7 +589,7 @@ static int bypass_event(struct snd_soc_dapm_widget *w, struct soc_mixer_control *m = (struct soc_mixer_control *)w->kcontrols->private_value; struct twl4030_priv *twl4030 = w->codec->private_data; - unsigned char reg; + unsigned char reg, misc; reg = twl4030_read_reg_cache(w->codec, m->reg); @@ -597,14 +601,28 @@ static int bypass_event(struct snd_soc_dapm_widget *w, else twl4030->bypass_state &= ~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL)); + } else if (m->reg == TWL4030_REG_VDL_APGA_CTL) { + /* Analog voice bypass */ + if (reg & (1 << m->shift)) + twl4030->bypass_state |= (1 << 4); + else + twl4030->bypass_state &= ~(1 << 4); } else { /* Digital bypass */ if (reg & (0x7 << m->shift)) - twl4030->bypass_state |= (1 << (m->shift ? 5 : 4)); + twl4030->bypass_state |= (1 << (m->shift ? 6 : 5)); else - twl4030->bypass_state &= ~(1 << (m->shift ? 5 : 4)); + twl4030->bypass_state &= ~(1 << (m->shift ? 6 : 5)); } + /* Enable master analog loopback mode if any analog switch is enabled*/ + misc = twl4030_read_reg_cache(w->codec, TWL4030_REG_MISC_SET_1); + if (twl4030->bypass_state & 0x1F) + misc |= TWL4030_FMLOOP_EN; + else + misc &= ~TWL4030_FMLOOP_EN; + twl4030_write(w->codec, TWL4030_REG_MISC_SET_1, misc); + if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) { if (twl4030->bypass_state) twl4030_codec_mute(w->codec, 0); @@ -935,7 +953,7 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback", - TWL4030_REG_AVDAC_CTL, 4, 0), + SND_SOC_NOPM, 0, 0), /* Analog PGAs */ SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL, @@ -962,6 +980,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_SWITCH_E("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0, &twl4030_dapm_abypassl2_control, bypass_event, SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH_E("Voice Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassv_control, + bypass_event, SND_SOC_DAPM_POST_REG), /* Digital bypasses */ SND_SOC_DAPM_SWITCH_E("Left Digital Loopback", SND_SOC_NOPM, 0, 0, @@ -979,6 +1000,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { 2, 0, NULL, 0), SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer", TWL4030_REG_AVDAC_CTL, + 4, 0, NULL, 0), /* Output MIXER controls */ /* Earpiece */ @@ -1067,13 +1090,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Analog R1 Playback Mixer", NULL, "DAC Right1"}, {"Analog L2 Playback Mixer", NULL, "DAC Left2"}, {"Analog R2 Playback Mixer", NULL, "DAC Right2"}, + {"Analog Voice Playback Mixer", NULL, "DAC Voice"}, {"ARXL1_APGA", NULL, "Analog L1 Playback Mixer"}, {"ARXR1_APGA", NULL, "Analog R1 Playback Mixer"}, {"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"}, {"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"}, - - {"VDL_APGA", NULL, "DAC Voice"}, + {"VDL_APGA", NULL, "Analog Voice Playback Mixer"}, /* Internal playback routings */ /* Earpiece */ @@ -1169,11 +1192,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left1 Analog Loopback", "Switch", "Analog Left Capture Route"}, {"Right2 Analog Loopback", "Switch", "Analog Right Capture Route"}, {"Left2 Analog Loopback", "Switch", "Analog Left Capture Route"}, + {"Voice Analog Loopback", "Switch", "Analog Left Capture Route"}, {"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"}, {"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"}, {"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"}, {"Analog L2 Playback Mixer", NULL, "Left2 Analog Loopback"}, + {"Analog Voice Playback Mixer", NULL, "Voice Analog Loopback"}, /* Digital bypass routes */ {"Right Digital Loopback", "Volume", "TX1 Capture Route"}, -- 1.5.4.3

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Re: [alsa-devel] realtek ALC272 support
by Andres Salomon 04 May '09

04 May '09

On Tue, 28 Apr 2009 15:41:08 -0400 Andres Salomon <dilinger(a)queued.net> wrote: > On Mon, 27 Apr 2009 18:08:21 +0200 > Takashi Iwai <tiwai(a)suse.de> wrote: > [...] > > > > > > > > At Fri, 24 Apr 2009 15:24:15 -0400, > > > > > > > > Andres Salomon wrote: > > > > > > > > > > > > > > > > > > Hi Kailang, > > > > > > > > > > > > > > > > > > I noticed that your name was on the ALC272 support > > > > > > > > > patch for ALSA intel-hda. This patch basically sets > > > > > > > > > ALC272 to use (mostly) the same code as ALC662. I > > > > > > > > > have two machines that have ALC272, and both of them > > > > > > > > > need verb tables in order to function. I'm wondering > > > > > > > > > if ALC662 should really be used.. > > > > > > > > > > > > > > > > > > Here's one - > > > > > > > > > > > > > > > > > > http://kernel.ubuntu.com/git?p=ubuntu/ubuntu-hardy-lum.git;a=commitdiff;h=7… > > > > > > > > > > > > > > > > > > This isn't the final version of the patch (there are > > > > > > > > > further commits I made in order to support headphone > > > > > > > > > mic stuff), but it gives you an idea of the codec > > > > > > > > > values. The other is: > > > > > > > > > > > > > > > > > > http://kernel.ubuntu.com/git?p=ubuntu/ubuntu-hardy-lum.git;a=commitdiff;h=7… > > > > > > > > > > > > > > > > > > All of these leave me wonder if there's a specific > > > > > > > > > patch_alc272 function that could be written to rid > > > > > > > > > ourselves of these specific quirks. Are there > > > > > > > > > machines with ALC272 that are functional with the > > > > > > > > > current ALSA code? > > > > > > > > [...] > > > > Could you try sound-unstable tree a bit later again? > > I found a bug in my patch, and fixed and updated GIT tree now. > > At least, the headphone plugging should work now. > > > > The mic auto-detection still doesn't work with model=auto, though. > > So, I'm going to take your patch anyway later. But I just wanted > > to be sure that the current tree could work somehow better... > > > > Hi, > > I just updated and tried the current tree; still no sound/headphone > output. :( Ok, I believe I've made some progress on this. The problem appears to be related to the autoconfig handling of the line out nids. The current code ends up with something like the following: ALSA sound/pci/hda/hda_codec.c:3833: autoconfig: line_outs=1 (0x17/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3837: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3841: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3842: mono: mono_out=0x0 ALSA sound/pci/hda/hda_codec.c:3853: inputs: mic=0x18, fmic=0x19, line=0x0, fline=0x0, cd=0x0, aux=0x0 However, NID 0x17 is actually a speaker_out. The code that checks for line_outs in snd_hda_parse_pin_def_config() unsets the speaker_out and uses that NID for a line_out. For whatever reason, this breaks things (no sound output, no headphone out). According to the ALC272 data sheet (the block diagram on page 5), there are actually 2 line_out NIDs; 0x14 and 0x15. The default pin configs for these two NIDs (0x411111f0 for both) have the Port Connectivity field set to AC_JACK_PORT_NONE, so the auto-config skips those NIDs. If I provide an override to a) set Port Connectivity to AC_JACK_PORT_COMPLEX, and b) set Default Device to AC_JACK_LINE_OUT (from AC_JACK_SPEAKER), I get usable audio (speakers work, headphone works, headphone plugging mutes the speaker properly, and even the mic works). The patch which I've used is below. Clearly, this isn't a very nice solution, so the question is: is there a way to provide an override/quirk for pincfgs (I see one for HWDEP, but that's configuration-specific) within the current hda_codec framework for this specific codec/laptop? Alternatively, I can try and figure out what's wrong with the code in the case where line_out_pins[0] is set to the speaker_out_pins[0] (ie, NID 0x17 in this case). I can check the other ALC272 machine tomorrow (the DELL_ZM1) to see if the line_out hack enables it to work without the verb table. diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b91f6ed..5aa7a0c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3644,6 +3644,15 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, continue; def_conf = snd_hda_codec_get_pincfg(codec, nid); + snd_printd("nid %x: pincfg def_conf=%x\n", nid, def_conf); + if (codec->vendor_id == 0x10ec0272 && + (nid == 0x14 || nid == 0x15)) { + def_conf &= ~(0x3 << AC_DEFCFG_PORT_CONN_SHIFT); + def_conf &= ~(0xf << AC_DEFCFG_DEVICE_SHIFT); + snd_printd("nid %x: overriding def_conf, now %x\n", nid, + def_conf); + } + if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) continue; loc = get_defcfg_location(def_conf);

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[alsa-devel] Adding MMAP support to alsa driver
by Gabriel Johnson 04 May '09

04 May '09

What structures need to be filled out in the hardware driver to support MMAP calls?

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[alsa-devel] Too big adjustment messages on hda-intel with ALC883
by Ozan Çağlayan 04 May '09

04 May '09

When I launch pulseaudio, a bunch of the following line is dropped to dmesg: [ 1472.847085] ALSA ../../alsa-kernel/pci/hda/hda_intel.c:1102: Too big adjustment 32 [ 1472.998865] hda-intel: IRQ timing workaround is activated for card #1. Suggest a bigger bdl_pos_adj. alsa-info output is at: http://www.alsa-project.org/db/?f=fd7037731d908e4287e1d457de6a1483062b2130 I'm using 2.6.30_rc3 with: - 20090421 snapshot of alsa-driver, - 20090410 snapshot of alsa-lib. Pulseaudio version is 0.9.15, timer-based scheduling is enabled. Regards, Ozan Caglayan http://www.pardus.org.tr/eng

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[alsa-devel] [PATCH] sc6000: enable joystick port
by Krzysztof Helt 04 May '09

04 May '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Add module parameter to enable or disable joystick port (gameport) on the SC6600 and later cards. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The second version of the patch with fixed the bug spotted Takashi. Documentation/sound/alsa/ALSA-Configuration.txt | 4 +++- sound/isa/sc6000.c | 9 +++++++-- 2 files changed, 10 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 012858d..11a80a9 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1543,13 +1543,15 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module snd-sc6000 ----------------- - Module for Gallant SC-6000 soundcard. + Module for Gallant SC-6000 soundcard and later models: SC-6600 + and SC-7000. port - Port # (0x220 or 0x240) mss_port - MSS Port # (0x530 or 0xe80) irq - IRQ # (5,7,9,10,11) mpu_irq - MPU-401 IRQ # (5,7,9,10) ,0 - no MPU-401 irq dma - DMA # (1,3,0) + joystick - Enable gameport - 0 = disable (default), 1 = enable This module supports multiple cards. diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index c803b2e..9a8bbf6 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -56,6 +56,7 @@ static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x300, 0x310, 0x320, 0x330 */ static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 0 */ static int dma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0, 1, 3 */ +static bool joystick[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = false }; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for sc-6000 based soundcard."); @@ -75,6 +76,8 @@ module_param_array(mpu_irq, int, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for sc-6000 driver."); module_param_array(dma, int, NULL, 0444); MODULE_PARM_DESC(dma, "DMA # for sc-6000 driver."); +module_param_array(joystick, bool, NULL, 0444); +MODULE_PARM_DESC(joystick, "Enable gameport."); /* * Commands of SC6000's DSP (SBPRO+special). @@ -363,7 +366,7 @@ static int __devinit sc6000_init_mss(char __iomem *vport, int config, static void __devinit sc6000_hw_cfg_encode(char __iomem *vport, int *cfg, long xport, long xmpu, - long xmss_port) + long xmss_port, int joystick) { cfg[0] = 0; cfg[1] = 0; @@ -376,6 +379,8 @@ static void __devinit sc6000_hw_cfg_encode(char __iomem *vport, int *cfg, if (xmss_port == 0xe80) cfg[0] |= 0x10; cfg[0] |= 0x40; /* always set */ + if (!joystick) + cfg[0] |= 0x02; cfg[1] |= 0x80; /* enable WSS system */ cfg[1] &= ~0x40; /* disable IDE */ snd_printd("hw cfg %x, %x\n", cfg[0], cfg[1]); @@ -427,7 +432,7 @@ static int __devinit sc6000_init_board(char __iomem *vport, if (!old) { int cfg[2]; sc6000_hw_cfg_encode(vport, &cfg[0], port[dev], mpu_port[dev], - mss_port[dev]); + mss_port[dev], joystick[dev]); if (sc6000_hw_cfg_write(vport, cfg) < 0) { snd_printk(KERN_ERR "sc6000_hw_cfg_write: failed!\n"); return -EIO; -- 1.6.0.3 ---------------------------------------------------------------------- Wygraj 3000 zl, wycieczke lub lot balonem! Sprawdz >> http://link.interia.pl/f2154

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[alsa-devel] [[ALSA] Patch for HP dv4-1222nr laptop 1/1] Addition for HP dv4-1222nr laptop.
by renidragsemaj＠yahoo.com 04 May '09

04 May '09

From: James Gardiner <renidragsemaj(a)yahoo.com> Signed-off-by: James Gardiner <renidragsemaj(a)yahoo.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 917bc5d..27bc568 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -100,6 +100,7 @@ enum { STAC_HP_M4, STAC_HP_DV5, STAC_HP_HDX, + STAC_HP_DV4_1222NR, STAC_92HD71BXX_MODELS }; @@ -1836,6 +1837,7 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_HP_M4] = NULL, [STAC_HP_DV5] = NULL, [STAC_HP_HDX] = NULL, + [STAC_HP_DV4_1222NR] = NULL, }; static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { @@ -1847,6 +1849,7 @@ static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_HP_M4] = "hp-m4", [STAC_HP_DV5] = "hp-dv5", [STAC_HP_HDX] = "hp-hdx", + [STAC_HP_DV4_1222NR] = "hp-dv4-1222nr", }; static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { @@ -1855,6 +1858,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD71BXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_92HD71BXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb, + "HP dv4-1222nr", STAC_HP_DV4_1222NR), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080, "HP", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0, @@ -4520,27 +4525,38 @@ static int stac92xx_resume(struct hda_codec *codec) return 0; } - /* - * using power check for controlling mute led of HP HDX notebooks + * using power check for controlling mute led of HP notebooks * check for mute state only on Speakers (nid = 0x10) * * For this feature CONFIG_SND_HDA_POWER_SAVE is needed, otherwise * the LED is NOT working properly ! + * + * Changed name to reflect that it now works for any designated + * model, not just HP HDX. */ #ifdef CONFIG_SND_HDA_POWER_SAVE -static int stac92xx_hp_hdx_check_power_status(struct hda_codec *codec, +static int stac92xx_hp_check_power_status(struct hda_codec *codec, hda_nid_t nid) { struct sigmatel_spec *spec = codec->spec; + unsigned int gpio_bit = 0; /* gets rid of compiler warning */ + + switch (spec->board_config) { + case STAC_HP_DV4_1222NR: + gpio_bit = 0x01; + break; + case STAC_HP_HDX: + gpio_bit = 0x08; + } if (nid == 0x10) { if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & HDA_AMP_MUTE) - spec->gpio_data &= ~0x08; /* orange */ + spec->gpio_data &= ~gpio_bit; /* orange */ else - spec->gpio_data |= 0x08; /* white */ + spec->gpio_data |= gpio_bit; /* white */ stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, @@ -5219,6 +5235,22 @@ again: spec->num_smuxes = 0; spec->num_dmuxes = 1; break; + case STAC_HP_DV4_1222NR: + spec->num_dmics = 1; + /* I don't know if it needs 1 or 2 smuxes - will wait for + * bug reports to fix if needed + */ + spec->num_smuxes = 1; + spec->num_dmuxes = 1; +#ifdef CONFIG_SND_HDA_POWER_SAVE + /* This controls MUTE LED */ + spec->gpio_mask |= 0x01; + spec->gpio_dir |= 0x01; + spec->gpio_data |= 0x01; + codec->patch_ops.check_power_status = + stac92xx_hp_check_power_status; +#endif + /* fallthrough */ case STAC_HP_DV5: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); @@ -5239,7 +5271,7 @@ again: /* register check_power_status callback. */ codec->patch_ops.check_power_status = - stac92xx_hp_hdx_check_power_status; + stac92xx_hp_check_power_status; #endif break; }; -- 1.5.6.3

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[alsa-devel] [PATCH 2/2] ASoC: TWL4030: Add voice digital loopback: sidetone
by Lopez Cruz, Misael 04 May '09

04 May '09

This patch add voice digital loopback (sidetone) to the twl4030 driver. It mixes voice uplink attenuated (by sidetone gain) with voice downlink when the codec is working in option2 (voice/audio mode). Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 27 +++++++++++++++++++++++++-- 1 files changed, 25 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index efb371f..23bae74 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -491,6 +491,18 @@ static const struct snd_kcontrol_new twl4030_dapm_dbypassr_control = TWL4030_REG_ATX2ARXPGA, 0, 7, 0, twl4030_dapm_dbypass_tlv); +/* + * Voice Sidetone GAIN volume control: + * from -51 to -10 dB in 1 dB steps (mute instead of -51 dB) + */ +static DECLARE_TLV_DB_SCALE(twl4030_dapm_dbypassv_tlv, -5100, 100, 1); + +/* Digital bypass voice: sidetone (VUL -> VDL)*/ +static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control = + SOC_DAPM_SINGLE_TLV("Volume", + TWL4030_REG_VSTPGA, 0, 0x29, 0, + twl4030_dapm_dbypassv_tlv); + static int micpath_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -607,12 +619,18 @@ static int bypass_event(struct snd_soc_dapm_widget *w, twl4030->bypass_state |= (1 << 4); else twl4030->bypass_state &= ~(1 << 4); + } else if (m->reg == TWL4030_REG_VSTPGA) { + /* Voice digital bypass */ + if (reg) + twl4030->bypass_state |= (1 << 5); + else + twl4030->bypass_state &= ~(1 << 5); } else { /* Digital bypass */ if (reg & (0x7 << m->shift)) - twl4030->bypass_state |= (1 << (m->shift ? 6 : 5)); + twl4030->bypass_state |= (1 << (m->shift ? 7 : 6)); else - twl4030->bypass_state &= ~(1 << (m->shift ? 6 : 5)); + twl4030->bypass_state &= ~(1 << (m->shift ? 7 : 6)); } /* Enable master analog loopback mode if any analog switch is enabled*/ @@ -991,6 +1009,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_SWITCH_E("Right Digital Loopback", SND_SOC_NOPM, 0, 0, &twl4030_dapm_dbypassr_control, bypass_event, SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH_E("Voice Digital Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_dbypassv_control, bypass_event, + SND_SOC_DAPM_POST_REG), SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL, 0, 0, NULL, 0), @@ -1203,9 +1224,11 @@ static const struct snd_soc_dapm_route intercon[] = { /* Digital bypass routes */ {"Right Digital Loopback", "Volume", "TX1 Capture Route"}, {"Left Digital Loopback", "Volume", "TX1 Capture Route"}, + {"Voice Digital Loopback", "Volume", "TX2 Capture Route"}, {"Analog R2 Playback Mixer", NULL, "Right Digital Loopback"}, {"Analog L2 Playback Mixer", NULL, "Left Digital Loopback"}, + {"Analog Voice Playback Mixer", NULL, "Voice Digital Loopback"}, }; -- 1.5.4.3

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