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Re: [alsa-devel] [PATCH 1/4] ASoC: add a WM8978 codec driver
by Guennadi Liakhovetski 22 Jan '10

22 Jan '10

On Tue, 19 Jan 2010, Mark Brown wrote: > On Tue, Jan 19, 2010 at 09:08:57AM +0100, Guennadi Liakhovetski wrote: > > This all looks pretty good - there's a few issues below but they're > largely stylistic rather than anything fundamental. > > > + 0x0000, 0x0000, 0x0000, 0x01ff, /* 0x08...0x0b */ /* 0x0b contains the VU bit */ > > + 0x01ff, 0x0000, 0x0100, 0x01ff, /* 0x0c...0x0f */ /* 0x0c and 0x0f contain the VU bit */ > > + 0x01ff, 0x0000, 0x012c, 0x002c, /* 0x10...0x13 */ /* 0x10 contains the VU bit */ > > What do these comments refer to - do you mean to say that these are not > the actual chip register defaults? The normal way to deal with those is > to have a write (or other update of the cache defaults). This avoids > potential confusion later on if the chip updates the register defaults > or when reviewing against the datasheet. No, I meant, that there's a "Value Update" bit in that register, usually bit 8. So, that bit, of course, is not in default, but it was suggested to me on IRC to set it in cache to force automatic value update. > > +#define ARRAY_SINGLE(xreg, xshift, xtexts) SOC_ENUM_SINGLE(xreg, xshift, \ > > + ARRAY_SIZE(xtexts), xtexts) > > This looks generic - please namespace it and add it to the header file, > other drivers can benefit from it. SOC_ARRAY_SINGLE()? > > +static const struct soc_enum wm8978_enum[] = { > > + /* adc */ > > + ARRAY_SINGLE(WM8978_COMPANDING_CONTROL, 1, wm8978_companding), > > Please define individual variables for this - indexing into the array > gets hard to follow, it's far too easy to have an off by one errors > trying to match things up. > > > +#define MIXER_ARRAY(n, r, s, i, m) SND_SOC_DAPM_MIXER(n, r, s, i, \ > > + m, ARRAY_SIZE(m)) > > > > +/* Mixer #3: Boost (Input) mixer */ > > +static const struct snd_kcontrol_new wm8978_left_boost_mixer[] = { > > +SOC_DAPM_SINGLE("Left PGA Mute", WM8978_LEFT_INP_PGA_CONTROL, 6, 1, 0), > > +}; > > +static const struct snd_kcontrol_new wm8978_right_boost_mixer[] = { > > +SOC_DAPM_SINGLE("Right PGA Mute", WM8978_RIGHT_INP_PGA_CONTROL, 6, 1, 0), > > +}; > > These should have "Switch" at the end of the name. Also, the names are > going be concatenated with the mixer names so you'll end up with things > "Left Boost Mixer Left PGA Switch" - the individual controls should > probably be just "Switch" so you end up with "Left Boost Mixer Switch". > > > +/* Mic Input boost vol */ > > +static const struct snd_kcontrol_new wm8978_mic_boost_controls[] = { > > +SOC_DAPM_SINGLE("Left Mic Volume", WM8978_LEFT_ADC_BOOST_CONTROL, 4, 7, 0), > > +SOC_DAPM_SINGLE("Right Mic Volume", WM8978_RIGHT_ADC_BOOST_CONTROL, 4, 7, 0), > > +}; > > This doesn't seem to be referenced anywhere? > > > +#define MIXER_ARRAY(n, r, s, i, m) SND_SOC_DAPM_MIXER(n, r, s, i, \ > > + m, ARRAY_SIZE(m)) > > Again, this should be done somewhere generic. Probably by going through > and changing the SND_SOC_DAPM_MIXER definition. Well, having discussed this privately and having looked at the heades, I'm not sure, this is a good idea. First, there are SND_SOC_DAPM_PGA() SND_SOC_DAPM_MIXER() SND_SOC_DAPM_MIXER_NAMED_CTL() SND_SOC_DAPM_PGA_E() SND_SOC_DAPM_MIXER_E() SND_SOC_DAPM_MIXER_NAMED_CTL_E() - 6 macros that follow the pattern .kcontrols = wcontrols, .num_kcontrols = wncontrols, and lots of macros like SOC_ENUM_DOUBLE() SOC_ENUM_SINGLE() SOC_ENUM_SINGLE_EXT() ... doing .max = xmax. BTW, many of the users even open-code array sizes instead of using ARRAY_SIZE() like static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", "Off"}; static const char *spk_function[] = {"On", "Off"}; static const struct soc_enum tosa_enum[] = { SOC_ENUM_SINGLE_EXT(5, jack_function), SOC_ENUM_SINGLE_EXT(2, spk_function), }; Changing only one of them would make the headers and the API look inconsistent. Changing all of them... I really don't think we want to endeavor this now, at least not me;) Besides, doing this locally is very convenient, and is available to everyone - there's no space science involved with this. One can be even as evil as doing +#define _A(n, r, s, i, m) SND_SOC_DAPM_MIXER(n, r, s, i, \ + m, ARRAY_SIZE(m)) ... +#undef _A +#define _A(xreg, xshift, xtexts) SOC_ENUM_SINGLE(xreg, xshift, \ + ARRAY_SIZE(xtexts), xtexts) ... +#undef _A I think, it's best to leave these things as they are. Otherwise we can give this to kernel janitors as a small exercise;) > > + /* Output PLL to GPIO1 */ > > + snd_soc_write(codec, WM8978_GPIO_CONTROL, > > + snd_soc_read(codec, WM8978_GPIO_CONTROL) | 0x4); > > This should be being done unconditionally. Put a switch in via the > set_dai_clkdiv() call. Ok, I moved it to wm8978_set_dai_clkdiv() under "case WM8978_OPCLKDIV:" > > + /* Mic bias */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, > > + (snd_soc_read(codec, 1) & ~4) | 0x10); > > + > > + /* Out-1 enabled, left/right input channel enabled */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0x1bf); > > + > > + /* Out-2 disabled, right/left output channel enabled, dac enabled */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_3, 0x10f); > > These should all be being managed via DAPM. Yes, I'm having quite a bit of trouble with DAPM, but I'm getting through slowly. Will need some more time for a next version. > > > +static int wm8978_set_bias_level(struct snd_soc_codec *codec, > > + enum snd_soc_bias_level level) > > +{ > > + u16 power1 = snd_soc_read(codec, WM8978_POWER_MANAGEMENT_1) & ~3; > > This bitmask maintains everything except the two LSB... > > > + switch (level) { > > + case SND_SOC_BIAS_ON: > > + case SND_SOC_BIAS_PREPARE: > > + power1 |= 1; /* VMID 75k */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, power1); > > + break; > > + case SND_SOC_BIAS_STANDBY: > > + power1 |= 0xC; > > ...but this is also managing other bits. Right, bits 7-6 are either kept or set, nothing wrong with that. > > + if (codec->bias_level == SND_SOC_BIAS_OFF) { > > + /* Initial cap charge at VMID 5k */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, > > + power1 | 0x3); > > + mdelay(100); > > + } > > > +/* Also supports 12kHz */ > > +#define WM8978_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ > > + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | \ > > + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) > > SNDRV_PCM_RATE_8000_48000 > > > +static int wm8978_suspend(struct platform_device *pdev, pm_message_t state) > > +{ > > + struct snd_soc_device *socdev = platform_get_drvdata(pdev); > > + struct snd_soc_codec *codec = socdev->card->codec; > > + > > + /* we only need to suspend if we are a valid card */ > > + if (!codec->card) > > + return 0; > > Don't need to check for the card any more, the core will stop you > getting called without a card. > > > + /* Sync reg_cache with the hardware */ > > + for (i = 0; i < ARRAY_SIZE(wm8978_reg); i++) { > > + if (i == WM8978_RESET) > > + continue; > > You can also skip the write if the register has the default value (this > will speed up resume slightly). > > > + data = cpu_to_be16((i << 9) | (cache[i] & 0x1ff)); > > + codec->hw_write(codec->control_data, (char *)&data, 2); > > + } > > Just use snd_soc_write()? I'll take care about the rest. Thanks Guennadi --- Guennadi Liakhovetski, Ph.D. Freelance Open-Source Software Developer http://www.open-technology.de/

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[alsa-devel] [PATCH - hda 1/1] ALSA: hda - Add PCI quirk for 17" MacBook Pro 5, 2 model (July 2009) to use existing mb5 model
by murray.alex＠gmail.com 22 Jan '10

22 Jan '10

From: Alex Murray <murray.alex(a)gmail.com> Signed-off-by: Alex Murray <murray.alex(a)gmail.com> --- pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/pci/hda/patch_realtek.c b/pci/hda/patch_realtek.c index 7e0ea37..d044db9 100644 --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -9223,6 +9223,7 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { */ SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5), + SND_PCI_QUIRK(0x10ec, 0x0885, "MacbookPro 5,2", ALC885_MB5), {} /* terminator */ }; -- 1.6.3.3

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[alsa-devel] Update information: HD-Audio-models list
by Mark Brown 22 Jan '10

22 Jan '10

Sir/Madam, I have an update for the HD-Audio-Models.txt file. Under ALC262, the toshiba-s06 model setting is applicable to the Sony Vaio VGN-CS60B laptop, and may also be applicable to similar models. It also needs configuring in alsamixer to enable the internal microphone afterwards. Best regards, Mark Brown.

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Re: [alsa-devel] Reforming s3c2443 ac97 driver
by SeungHyun Choi 22 Jan '10

22 Jan '10

3 2

[alsa-devel] [PATCH] ASoC: add helper macros to declare struct soc_enum instances
by Guennadi Liakhovetski 21 Jan '10

21 Jan '10

Several shortcuts for popular uses of some of SOC_ENUM_* and SOC_VALUE_ENUM_* macros. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- diff --git a/include/sound/soc.h b/include/sound/soc.h index 08909cc..9537144 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -169,6 +169,23 @@ .private_value = (unsigned long)&xenum } /* + * Simplified versions of above macros, declaring a struct and calculating + * ARRAY_SIZE internally + */ +#define SOC_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xtexts) \ + struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \ + ARRAY_SIZE(xtexts), xtexts) +#define SOC_ENUM_SINGLE_DECL(name, xreg, xshift, xtexts) \ + SOC_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xtexts) +#define SOC_ENUM_SINGLE_EXT_DECL(name, xtexts) \ + struct soc_enum name = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(xtexts), xtexts) +#define SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xmask, xtexts, xvalues) \ + struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \ + ARRAY_SIZE(xtexts), xtexts, xvalues) +#define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \ + SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) + +/* * Bias levels * * @ON: Bias is fully on for audio playback and capture operations.

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[alsa-devel] [PATCH 2/5] ASoC: DaVinci: Voice Codec Interface
by miguel.aguilar＠ridgerun.com 21 Jan '10

21 Jan '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This patch adds the support for the interface needed by the DaVinci Voice Codec CQ93VC. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/davinci/Kconfig | 3 + sound/soc/davinci/Makefile | 2 + sound/soc/davinci/davinci-vcif.c | 272 ++++++++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-vcif.h | 28 ++++ 4 files changed, 305 insertions(+), 0 deletions(-) create mode 100644 sound/soc/davinci/davinci-vcif.c create mode 100644 sound/soc/davinci/davinci-vcif.h diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 047ee39..47e7cce 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -12,6 +12,9 @@ config SND_DAVINCI_SOC_I2S config SND_DAVINCI_SOC_MCASP tristate +config SND_DAVINCI_SOC_VCIF + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a6939d7..a93679d 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -2,10 +2,12 @@ snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o +snd-soc-davinci-vcif-objs:= davinci-vcif.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o +obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c new file mode 100644 index 0000000..03f3feb --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.c @@ -0,0 +1,272 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2010 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/mfd/davinci_voicecodec.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-i2s.h" +#include "davinci-vcif.h" + +#define MOD_REG_BIT(val, mask, set) do { \ + if (set) { \ + val |= mask; \ + } else { \ + val &= ~mask; \ + } \ +} while (0) + +struct davinci_vcif_dev { + struct davinci_vc *davinci_vc; + struct davinci_pcm_dma_params dma_params[2]; +}; + +static void davinci_vcif_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *davinci_vcif_dev = + rtd->dai->cpu_dai->private_data; + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + u32 w; + + /* Start the sample generator and enable transmitter/receiver */ + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); + else + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); +} + +static void davinci_vcif_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *davinci_vcif_dev = + rtd->dai->cpu_dai->private_data; + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + u32 w; + + /* Reset transmitter/receiver and sample rate/frame sync generators */ + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); + else + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); +} + +static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct davinci_vcif_dev *davinci_vcif_dev = dai->private_data; + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + struct davinci_pcm_dma_params *dma_params = + &davinci_vcif_dev->dma_params[substream->stream]; + u32 w; + + /* Restart the codec before setup */ + davinci_vcif_stop(substream); + davinci_vcif_start(substream); + + /* General line settings */ + writel(DAVINCI_VC_CTRL_MASK, davinci_vc->base + DAVINCI_VC_CTRL); + + writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTCLR); + + writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTEN); + + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + + /* Determine xfer data type */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + dma_params->data_type = 0; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_BITS_8 | + DAVINCI_VC_CTRL_WD_UNSIGNED, 1); + break; + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_WD_BITS_8, 1); + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_UNSIGNED, 0); + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_BITS_8 | + DAVINCI_VC_CTRL_WD_UNSIGNED, 0); + break; + default: + printk(KERN_WARNING "davinci-vcif: unsupported PCM format"); + return -EINVAL; + } + + dma_params->acnt = dma_params->data_type; + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); + + return 0; +} + +static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_vcif_start(substream); + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_vcif_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +#define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000 + +static struct snd_soc_dai_ops davinci_vcif_dai_ops = { + .trigger = davinci_vcif_trigger, + .hw_params = davinci_vcif_hw_params, +}; + +struct snd_soc_dai davinci_vcif_dai = { + .name = "davinci-vcif", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &davinci_vcif_dai_ops, + +}; +EXPORT_SYMBOL_GPL(davinci_vcif_dai); + +static int davinci_vcif_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct davinci_vcif_dev *davinci_vcif_dev; + int ret; + + davinci_vcif_dev = kzalloc(sizeof(struct davinci_vcif_dev), GFP_KERNEL); + if (!davinci_vc) { + dev_dbg(&pdev->dev, + "could not allocate memory for private data\n"); + return -ENOMEM; + } + + /* DMA tx params */ + davinci_vcif_dev->davinci_vc = davinci_vc; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = + davinci_vc->davinci_vcif.dma_tx_channel; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = + davinci_vc->davinci_vcif.dma_tx_addr; + + /* DMA rx params */ + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = + davinci_vc->davinci_vcif.dma_rx_channel; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = + davinci_vc->davinci_vcif.dma_rx_addr; + + davinci_vcif_dai.dev = &pdev->dev; + davinci_vcif_dai.dma_data = davinci_vcif_dev->dma_params; + davinci_vcif_dai.private_data = davinci_vcif_dev; + + ret = snd_soc_register_dai(&davinci_vcif_dai); + if (ret != 0) { + dev_err(&pdev->dev, "could not register dai\n"); + goto fail; + } + + return 0; + +fail: + kfree(davinci_vcif_dev); + + return ret; +} + +static int davinci_vcif_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&davinci_vcif_dai); + + return 0; +} + +static struct platform_driver davinci_vcif_driver = { + .probe = davinci_vcif_probe, + .remove = davinci_vcif_remove, + .driver = { + .name = "davinci_vcif", + .owner = THIS_MODULE, + }, +}; + +static int __init davinci_vcif_init(void) +{ + return platform_driver_probe(&davinci_vcif_driver, davinci_vcif_probe); +} +module_init(davinci_vcif_init); + +static void __exit davinci_vcif_exit(void) +{ + platform_driver_unregister(&davinci_vcif_driver); +} +module_exit(davinci_vcif_exit); + +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC Voice Codec Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-vcif.h b/sound/soc/davinci/davinci-vcif.h new file mode 100644 index 0000000..571c994 --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.h @@ -0,0 +1,28 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2010 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _DAVINCI_VCIF_H +#define _DAVINCI_VCIF_H + +extern struct snd_soc_dai davinci_vcif_dai; + +#endif -- 1.6.0.4

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[alsa-devel] Problem with PPH plugin
by Colin Guthrie 21 Jan '10

21 Jan '10

Hi, Just wanted to highlight a bug over on Mozilla relating to a crash in the PPH plugin. https://bugzilla.mozilla.org/show_bug.cgi?id=540460 It was initially though to be in PulseAudio, but on closer inspection this doesn't involve PA at all and in fact seems to relate to the PPH plugin from what I can gather. The backtrace is here: https://bug540460.bugzilla.mozilla.org/attachment.cgi?id=422264 The user did submit a bug report to the bugtracker (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4878) but given previous experience there (see other mail!) I figured I'd post a message here to highlight it seeing as the reporter has already done a lot to try and work out what is going on and posting a message on his behalf seemed like a good idea :) Cheers All the best. -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/]

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[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 21 Jan '10

21 Jan '10

The following changes since commit 9135f6db09aa9d186f05d41e9c516ec8e3550d78: Mark Brown (1): Merge branch 'mxc-audio' into for-2.6.34 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Guennadi Liakhovetski (1): ASoC: fix compile breakage - add a missing header include Mark Brown (3): ASoC: Remove console DAPM debug code ASoC: Support turning off bias when the CODEC is idle ASoC: Use BIAS_OFF when idle for wm_hubs devices Peter Ujfalusi (3): ASoC: tlv320dac33: Correct the prefill number of samples ASoC: tlv320dac33: BCLK divider fix ASoC: tlv320dac33: Burst mode BCLK divider configuration include/sound/soc-dai.h | 2 + include/sound/soc.h | 2 + include/sound/tlv320dac33-plat.h | 1 + sound/soc/codecs/tlv320dac33.c | 24 +++++++-- sound/soc/codecs/wm_hubs.c | 6 ++ sound/soc/soc-dapm.c | 105 ++++++++++---------------------------- 6 files changed, 56 insertions(+), 84 deletions(-)

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[alsa-devel] [PATCH] ASoC: Support turning off bias when the CODEC is idle
by Mark Brown 21 Jan '10

21 Jan '10

Currently ASoC always maintains the bias of the CODEC while the system is active. With older mobile CODECs this is required since the outputs are referenced to a non-zero voltage and enabling or disabling this voltage without audible pops or clicks in the output takes too long to do when starting or stopping audio. As a result of features such as ground referenced outputs and class D speaker drivers current generation devices are able to power on and off much more quickly without these system level issues so provide a new flag idle_bias_off in snd_soc_codec which will cause the core to turn off the CODEC bias. The distinction between STANDBY and OFF is still maintained. This is partly for consistency but also allows for potential future extensions such as per-machine overrides or deferring the bias removal. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- include/sound/soc.h | 2 ++ sound/soc/soc-dapm.c | 25 ++++++++++++++++++++++++- 2 files changed, 26 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 08909cc..a8768ea 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -405,6 +405,8 @@ struct snd_soc_codec { short reg_cache_size; short reg_cache_step; + unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ + /* dapm */ u32 pop_time; struct list_head dapm_widgets; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d8e9374..6c33510 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1012,13 +1012,28 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) sys_power = 0; break; case SND_SOC_DAPM_STREAM_NOP: - sys_power = codec->bias_level != SND_SOC_BIAS_STANDBY; + switch (codec->bias_level) { + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + sys_power = 0; + break; + default: + sys_power = 1; + break; + } break; default: break; } } + if (sys_power && codec->bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_STANDBY); + if (ret != 0) + pr_err("Failed to turn on bias: %d\n", ret); + } + /* If we're changing to all on or all off then prepare */ if ((sys_power && codec->bias_level == SND_SOC_BIAS_STANDBY) || (!sys_power && codec->bias_level == SND_SOC_BIAS_ON)) { @@ -1042,6 +1057,14 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) pr_err("Failed to apply standby bias: %d\n", ret); } + /* If we're in standby and can support bias off then do that */ + if (codec->bias_level == SND_SOC_BIAS_STANDBY && + codec->idle_bias_off) { + ret = snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_OFF); + if (ret != 0) + pr_err("Failed to turn off bias: %d\n", ret); + } + /* If we just powered up then move to active bias */ if (codec->bias_level == SND_SOC_BIAS_PREPARE && sys_power) { ret = snd_soc_dapm_set_bias_level(socdev, -- 1.6.6

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[alsa-devel] Reforming s3c2443 ac97 driver
by JASWINDERSINGH BRAR 21 Jan '10

21 Jan '10

Hi, Right now the S3C2443's AC97_ver2.0 controller driver isn't flexible enough to handle newer SoCs. In order to prepare ground for supporting 64xx and S5Pxxxx the extant driver needs to be reformed, and that is what i am planning to do next. I plan to do the following:- 1) User platform_driver structure to detect and manage AC97 controller platform devices. 2) Remove hardcoded parameters(DMA, IO address, IRQ etc) and pass via platform resources. 3) Initialize platform specific stuff by callback pointers to platform code(cfg_gpio) Objections, suggestions welcome. ------------------------------- Jaswinder Singh Brar Senior Engineer Linux OS Team Application Processor Development System LSI Division SAMSUNG Electronics Co., Ltd. --------------------------------

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