September 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] sscape: convert to firmware loader framework
by Krzysztof Helt 28 Sep '09

28 Sep '09

The conversion solves the problem that firmware size was set to 64KB while non PnP cards have 128KB firmware files. An additional post firmware initialization code has been moved from the OSS driver. --- Now, the MIDI is audible but not correct. It is not played timely. The sscape_ctl alsa-tool is not needed any more. Documentation/sound/alsa/ALSA-Configuration.txt | 8 +- include/sound/sscape_ioctl.h | 21 -- sound/isa/Kconfig | 8 +- sound/isa/sscape.c | 305 +++++++---------------- 4 files changed, 101 insertions(+), 241 deletions(-) delete mode 100644 include/sound/sscape_ioctl.h diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 1c8eb45..cf98525 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1631,7 +1631,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module snd-sscape ----------------- - Module for ENSONIQ SoundScape PnP cards. + Module for ENSONIQ SoundScape cards. port - Port # (PnP setup) wss_port - WSS Port # (PnP setup) @@ -1640,9 +1640,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. dma - DMA # (PnP setup) dma2 - 2nd DMA # (PnP setup, -1 to disable) - This module supports multiple cards. ISA PnP must be enabled. - You need sscape_ctl tool in alsa-tools package for loading - the microcode. + This module supports multiple cards. + + The driver requires the firmware loader support on kernel. Module snd-sun-amd7930 (on sparc only) -------------------------------------- diff --git a/include/sound/sscape_ioctl.h b/include/sound/sscape_ioctl.h deleted file mode 100644 index 0d88859..0000000 --- a/include/sound/sscape_ioctl.h +++ /dev/null @@ -1,21 +0,0 @@ -#ifndef SSCAPE_IOCTL_H -#define SSCAPE_IOCTL_H - - -struct sscape_bootblock -{ - unsigned char code[256]; - unsigned version; -}; - -#define SSCAPE_MICROCODE_SIZE 65536 - -struct sscape_microcode -{ - unsigned char __user *code; -}; - -#define SND_SSCAPE_LOAD_BOOTB _IOWR('P', 100, struct sscape_bootblock) -#define SND_SSCAPE_LOAD_MCODE _IOW ('P', 101, struct sscape_microcode) - -#endif diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index b90fc16..02fe81c 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -372,9 +372,9 @@ config SND_SGALAXY config SND_SSCAPE tristate "Ensoniq SoundScape driver" - select SND_HWDEP select SND_MPU401_UART select SND_WSS_LIB + select FW_LOADER help Say Y here to include support for Ensoniq SoundScape and Ensoniq OEM soundcards. @@ -382,7 +382,11 @@ config SND_SSCAPE The PCM audio is supported on SoundScape Classic, Elite, PnP and VIVO cards. The supported OEM cards are SPEA Media FX and Reveal SC-600. - The MIDI support is very experimental. + The MIDI support is very experimental and requires binary + firmware files called "scope.cod" and "sndscape.co?" where the + ? is digit 0, 1, 2, 3 or 4. The firmware files can be found + in DOS or Windows driver packages. One has to put the firmware + files into the /lib/firmware directory. To compile this driver as a module, choose M here: the module will be called snd-sscape. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index b11c35f..08389a2 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -1,5 +1,5 @@ /* - * Low-level ALSA driver for the ENSONIQ SoundScape PnP + * Low-level ALSA driver for the ENSONIQ SoundScape * Copyright (c) by Chris Rankin * * This driver was written in part using information obtained from @@ -25,12 +25,12 @@ #include <linux/err.h> #include <linux/isa.h> #include <linux/delay.h> +#include <linux/firmware.h> #include <linux/pnp.h> #include <linux/spinlock.h> #include <linux/moduleparam.h> #include <asm/dma.h> #include <sound/core.h> -#include <sound/hwdep.h> #include <sound/wss.h> #include <sound/mpu401.h> #include <sound/initval.h> @@ -39,7 +39,7 @@ MODULE_AUTHOR("Chris Rankin"); -MODULE_DESCRIPTION("ENSONIQ SoundScape PnP driver"); +MODULE_DESCRIPTION("ENSONIQ SoundScape driver"); MODULE_LICENSE("GPL"); static int index[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IDX; @@ -142,14 +142,12 @@ struct soundscape { struct resource *wss_res; struct snd_wss *chip; struct snd_mpu401 *mpu; - struct snd_hwdep *hw; /* * The MIDI device won't work until we've loaded * its firmware via a hardware-dependent device IOCTL */ spinlock_t fwlock; - int hw_in_use; unsigned long midi_usage; unsigned char midi_vol; }; @@ -167,12 +165,6 @@ static inline struct soundscape *get_mpu401_soundscape(struct snd_mpu401 * mpu) return (struct soundscape *) (mpu->private_data); } -static inline struct soundscape *get_hwdep_soundscape(struct snd_hwdep * hw) -{ - return (struct soundscape *) (hw->private_data); -} - - /* * Allocates some kernel memory that we can use for DMA. * I think this means that the memory has to map to @@ -438,14 +430,14 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) * Upload a byte-stream into the SoundScape using DMA channel A. */ static int upload_dma_data(struct soundscape *s, - const unsigned char __user *data, + const unsigned char *data, size_t size) { unsigned long flags; struct snd_dma_buffer dma; int ret; - if (!get_dmabuf(&dma, PAGE_ALIGN(size))) + if (!get_dmabuf(&dma, PAGE_ALIGN(32 * 1024))) return -ENOMEM; spin_lock_irqsave(&s->lock, flags); @@ -468,35 +460,17 @@ static int upload_dma_data(struct soundscape *s, sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x80); /* - * Upload the user's data (firmware?) to the SoundScape + * Upload the firmware to the SoundScape * board through the DMA channel ... */ while (size != 0) { unsigned long len; - /* - * Apparently, copying to/from userspace can sleep. - * We are therefore forbidden from holding any - * spinlocks while we copy ... - */ - spin_unlock_irqrestore(&s->lock, flags); - - /* - * Remember that the data that we want to DMA - * comes from USERSPACE. We have already verified - * the userspace pointer ... - */ len = min(size, dma.bytes); - len -= __copy_from_user(dma.area, data, len); + memcpy(dma.area, data, len); data += len; size -= len; - /* - * Grab that spinlock again, now that we've - * finished copying! - */ - spin_lock_irqsave(&s->lock, flags); - snd_dma_program(s->chip->dma1, dma.addr, len, DMA_MODE_WRITE); sscape_start_dma_unsafe(s->io_base, GA_DMAA_REG); if (!sscape_wait_dma_unsafe(s->io_base, GA_DMAA_REG, 5000)) { @@ -512,6 +486,7 @@ static int upload_dma_data(struct soundscape *s, } /* while */ set_host_mode_unsafe(s->io_base); + outb(0x0, s->io_base); /* * Boot the board ... (I think) @@ -537,7 +512,7 @@ _release_dma: /* * NOTE!!! We are NOT holding any spinlocks at this point !!! */ - sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_ODIE ? 0x70 : 0x40)); + sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_OPUS ? 0x40 : 0x70)); free_dmabuf(&dma); return ret; @@ -547,162 +522,70 @@ _release_dma: * Upload the bootblock(?) into the SoundScape. The only * purpose of this block of code seems to be to tell * us which version of the microcode we should be using. - * - * NOTE: The boot-block data resides in USER-SPACE!!! - * However, we have already verified its memory - * addresses by the time we get here. */ -static int sscape_upload_bootblock(struct soundscape *sscape, struct sscape_bootblock __user *bb) +static int sscape_upload_bootblock(struct snd_card *card) { + struct soundscape *sscape = get_card_soundscape(card); unsigned long flags; + const struct firmware *init_fw = NULL; int data = 0; int ret; - ret = upload_dma_data(sscape, bb->code, sizeof(bb->code)); - - spin_lock_irqsave(&sscape->lock, flags); - if (ret == 0) { - data = host_read_ctrl_unsafe(sscape->io_base, 100); - } - set_midi_mode_unsafe(sscape->io_base); - spin_unlock_irqrestore(&sscape->lock, flags); - - if (ret == 0) { - if (data < 0) { - snd_printk(KERN_ERR "sscape: timeout reading firmware version\n"); - ret = -EAGAIN; - } - else if (__copy_to_user(&bb->version, &data, sizeof(bb->version))) { - ret = -EFAULT; - } + ret = request_firmware(&init_fw, "scope.cod", card->dev); + if (ret < 0) { + snd_printk(KERN_ERR "Error loading scope.cod"); + return ret; } + ret = upload_dma_data(sscape, init_fw->data, init_fw->size); - return ret; -} + release_firmware(init_fw); -/* - * Upload the microcode into the SoundScape. The - * microcode is 64K of data, and if we try to copy - * it into a local variable then we will SMASH THE - * KERNEL'S STACK! We therefore leave it in USER - * SPACE, and save ourselves from copying it at all. - */ -static int sscape_upload_microcode(struct soundscape *sscape, - const struct sscape_microcode __user *mc) -{ - unsigned long flags; - char __user *code; - int err; - - /* - * We are going to have to copy this data into a special - * DMA-able buffer before we can upload it. We shall therefore - * just check that the data pointer is valid for now. - * - * NOTE: This buffer is 64K long! That's WAY too big to - * copy into a stack-temporary anyway. - */ - if ( get_user(code, &mc->code) || - !access_ok(VERIFY_READ, code, SSCAPE_MICROCODE_SIZE) ) - return -EFAULT; + spin_lock_irqsave(&sscape->lock, flags); + if (ret == 0) + data = host_read_ctrl_unsafe(sscape->io_base, 100); - if ((err = upload_dma_data(sscape, code, SSCAPE_MICROCODE_SIZE)) == 0) { - snd_printk(KERN_INFO "sscape: MIDI firmware loaded\n"); - } + if (data & 0x10) + sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2f); - spin_lock_irqsave(&sscape->lock, flags); - set_midi_mode_unsafe(sscape->io_base); spin_unlock_irqrestore(&sscape->lock, flags); - initialise_mpu401(sscape->mpu); + data &= 0xf; + if (ret == 0 && data > 7) { + snd_printk(KERN_ERR + "sscape: timeout reading firmware version\n"); + ret = -EAGAIN; + } - return err; + return (ret == 0) ? data : ret; } /* - * Hardware-specific device functions, to implement special - * IOCTLs for the SoundScape card. This is how we upload - * the microcode into the card, for example, and so we - * must ensure that no two processes can open this device - * simultaneously, and that we can't open it at all if - * someone is using the MIDI device. + * Upload the microcode into the SoundScape. */ -static int sscape_hw_open(struct snd_hwdep * hw, struct file *file) +static int sscape_upload_microcode(struct snd_card *card, int version) { - register struct soundscape *sscape = get_hwdep_soundscape(hw); - unsigned long flags; + struct soundscape *sscape = get_card_soundscape(card); + const struct firmware *init_fw = NULL; + char name[14] = "sndscape.co0"; int err; - spin_lock_irqsave(&sscape->fwlock, flags); + name[11] = '0' + version; - if ((sscape->midi_usage != 0) || sscape->hw_in_use) { - err = -EBUSY; - } else { - sscape->hw_in_use = 1; - err = 0; + err = request_firmware(&init_fw, name, card->dev); + if (err < 0) { + snd_printk(KERN_ERR "Error loading sndscape.co0"); + return err; } + err = upload_dma_data(sscape, init_fw->data, init_fw->size); + if (err == 0) + snd_printk(KERN_INFO "MIDI firmware loaded %d KBs\n", + init_fw->size >> 10); - spin_unlock_irqrestore(&sscape->fwlock, flags); - return err; -} - -static int sscape_hw_release(struct snd_hwdep * hw, struct file *file) -{ - register struct soundscape *sscape = get_hwdep_soundscape(hw); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - sscape->hw_in_use = 0; - spin_unlock_irqrestore(&sscape->fwlock, flags); - return 0; -} - -static int sscape_hw_ioctl(struct snd_hwdep * hw, struct file *file, - unsigned int cmd, unsigned long arg) -{ - struct soundscape *sscape = get_hwdep_soundscape(hw); - int err = -EBUSY; - - switch (cmd) { - case SND_SSCAPE_LOAD_BOOTB: - { - register struct sscape_bootblock __user *bb = (struct sscape_bootblock __user *) arg; - - /* - * We are going to have to copy this data into a special - * DMA-able buffer before we can upload it. We shall therefore - * just check that the data pointer is valid for now ... - */ - if ( !access_ok(VERIFY_READ, bb->code, sizeof(bb->code)) ) - return -EFAULT; - - /* - * Now check that we can write the firmware version number too... - */ - if ( !access_ok(VERIFY_WRITE, &bb->version, sizeof(bb->version)) ) - return -EFAULT; - - err = sscape_upload_bootblock(sscape, bb); - } - break; - - case SND_SSCAPE_LOAD_MCODE: - { - register const struct sscape_microcode __user *mc = (const struct sscape_microcode __user *) arg; - - err = sscape_upload_microcode(sscape, mc); - } - break; - - default: - err = -EINVAL; - break; - } /* switch */ + release_firmware(init_fw); return err; } - /* * Mixer control for the SoundScape's MIDI device. */ @@ -920,7 +803,7 @@ static int mpu401_open(struct snd_mpu401 * mpu) spin_lock_irqsave(&sscape->fwlock, flags); - if (sscape->hw_in_use || (sscape->midi_usage == ULONG_MAX)) { + if (sscape->midi_usage == ULONG_MAX) { err = -EBUSY; } else { ++(sscape->midi_usage); @@ -1053,13 +936,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, } } - strcpy(card->driver, "SoundScape"); - strcpy(card->shortname, pcm->name); - snprintf(card->longname, sizeof(card->longname), - "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %d\n", - pcm->name, chip->port, chip->irq, - chip->dma1, chip->dma2); - sscape->chip = chip; } @@ -1162,29 +1038,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) return -ENXIO; } - if (sscape->type != SSCAPE_VIVO) { - /* - * Now create the hardware-specific device so that we can - * load the microcode into the on-board processor. - * We cannot use the MPU-401 MIDI system until this firmware - * has been loaded into the card. - */ - err = snd_hwdep_new(card, "MC68EC000", 0, &(sscape->hw)); - if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " - "firmware device\n"); - goto _release_dma; - } - strlcpy(sscape->hw->name, "SoundScape M68K", - sizeof(sscape->hw->name)); - sscape->hw->name[sizeof(sscape->hw->name) - 1] = '\0'; - sscape->hw->iface = SNDRV_HWDEP_IFACE_SSCAPE; - sscape->hw->ops.open = sscape_hw_open; - sscape->hw->ops.release = sscape_hw_release; - sscape->hw->ops.ioctl = sscape_hw_ioctl; - sscape->hw->private_data = sscape; - } - /* * Tell the on-board devices where their resources are (I think - * I can't be sure without a datasheet ... So many magic values!) @@ -1222,28 +1075,51 @@ static int __devinit create_sscape(int dev, struct snd_card *card) wss_port[dev], irq[dev]); goto _release_dma; } + strcpy(card->driver, "SoundScape"); + strcpy(card->shortname, name); + snprintf(card->longname, sizeof(card->longname), + "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %d\n", + name, sscape->chip->port, sscape->chip->irq, + sscape->chip->dma1, sscape->chip->dma2); + #define MIDI_DEVNUM 0 if (sscape->type != SSCAPE_VIVO) { - err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); - if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " - "MPU-401 device at 0x%lx\n", - port[dev]); - goto _release_dma; - } + err = sscape_upload_bootblock(card); + if (err >= 0) + err = sscape_upload_microcode(card, err); + + if (err == 0) { + err = create_mpu401(card, MIDI_DEVNUM, port[dev], + mpu_irq[dev]); + if (err < 0) { + printk(KERN_ERR "sscape: Failed to create " + "MPU-401 device at 0x%lx\n", + port[dev]); + goto _release_dma; + } - /* - * Enable the master IRQ ... - */ - sscape_write(sscape, GA_INTENA_REG, 0x80); - - /* - * Initialize mixer - */ - sscape->midi_vol = 0; - host_write_ctrl_unsafe(sscape->io_base, CMD_SET_MIDI_VOL, 100); - host_write_ctrl_unsafe(sscape->io_base, 0, 100); - host_write_ctrl_unsafe(sscape->io_base, CMD_XXX_MIDI_VOL, 100); + /* + * Initialize mixer + */ + spin_lock_irqsave(&sscape->lock, flags); + sscape->midi_vol = 0; + host_write_ctrl_unsafe(sscape->io_base, + CMD_SET_MIDI_VOL, 100); + host_write_ctrl_unsafe(sscape->io_base, + sscape->midi_vol, 100); + host_write_ctrl_unsafe(sscape->io_base, + CMD_XXX_MIDI_VOL, 100); + host_write_ctrl_unsafe(sscape->io_base, + sscape->midi_vol, 100); + host_write_ctrl_unsafe(sscape->io_base, + CMD_SET_EXTMIDI, 100); + host_write_ctrl_unsafe(sscape->io_base, + 0, 100); + host_write_ctrl_unsafe(sscape->io_base, 80, 1); + + set_midi_mode_unsafe(sscape->io_base); + spin_unlock_irqrestore(&sscape->lock, flags); + } } /* @@ -1301,11 +1177,12 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) sscape->type = SSCAPE; dma[dev] &= 0x03; + snd_card_set_dev(card, pdev); + ret = create_sscape(dev, card); if (ret < 0) goto _release_card; - snd_card_set_dev(card, pdev); if ((ret = snd_card_register(card)) < 0) { printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; @@ -1426,12 +1303,12 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, wss_port[idx] = pnp_port_start(dev, 1); dma2[idx] = pnp_dma(dev, 1); } + snd_card_set_dev(card, &pcard->card->dev); ret = create_sscape(idx, card); if (ret < 0) goto _release_card; - snd_card_set_dev(card, &pcard->card->dev); if ((ret = snd_card_register(card)) < 0) { printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; -- 1.6.0.3 ---------------------------------------------------------------------- Co godzine losujemy 15 000 z�! http://link.interia.pl/f2376

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[alsa-devel] [PATCHv2 0/7] TWL6030 audio codec initial support
by Lopez Cruz, Misael 28 Sep '09

28 Sep '09

Following patch series adds initial support for TWL6030 codec driver. TWL6030 codec depends on TWL6030_CORE, which is not accepted yet. The main intention then is to get comments, I'll repost the patches of this series when TWL6030_CORE support gets accepted and rework patches as needed. Changes from v1: - Renamed dai format for PDM as SND_SOC_DAIFMT_PDM - CODEC driver registered as a platform driver - AUDPWRON gpio line handled in the CODEC driver - Handle scenario when no AUDPWRON gpio line is provided - Corrected names of some widgets - Avoid multiple calls to power functions in set_bias_level Thanks, -Misa --- Misael Lopez Cruz (7): OMAP4: PMIC: Add support for twl6030 codec ASoC: Add PDM DAI format definition ASoC: TWL6030: Add twl6030 codec driver ASoC: TWL6030: Manual power-up/down sequences ASoC: TWL6030: Add support for low-power mode ASoC: TWL6030: Enable audio interrupt ASoC: TWL6030: Power-up seq completion through audio interrupt drivers/mfd/twl-core.c | 15 + include/linux/i2c/twl.h | 11 + include/sound/soc-dai.h | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl6030.c | 1179 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6030.h | 137 +++++ 7 files changed, 1349 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/twl6030.c create mode 100644 sound/soc/codecs/twl6030.h

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Re: [alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 28 Sep '09

28 Sep '09

Hi Peter, Thanks for the help. I don't believe it's possible for me to try the master/slave switch. Please take a look at http://www.efn.org/~rick/pub/wm8731.jpg Note that DACLRC and ADCLRC are connected. This requires me to run as slave, right? Our bit clock line is buffered so I don't think that will be a problem. Rick > > Yes, I had tried 8 before and just tried it now. Same result. It > > really seems like there is a DMA issue. Can you think of anything > > else to try before I dive into the DMA code? > > > Ok, that wasn't the reason. > > Have you tried to reverse roles of codec and omap? I.e. by using > the codec as a master and omap as a slave with SND_SOC_DAIFMT_CBM_CFM. > > I don't think there is any HW reason as you have seen the bit-clock and > frame sync signals toggling but I remember one case few years back > where codec had slight too high capacitive load on the bit-clock line. > That was then causing unstable frame sync as it was internally derived > directly from the bit-clock pin without any buffer. Bit-clock signal > looked just fine on oscilloscope but wasn't well enough for internal > circuits of the cpu. > > So if assuming similar problem here, I could imagine that if the FSX is > used similar way directly inside the omap and if signal is not well > enough, the McBSP and DMA may not be running. > > This migh be a bit far reason to look for but at least easy to try by > switching the roles as then the codec is driving the both bit-clock and > fs. > >

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[alsa-devel] [PATCHv2 1/7] OMAP4: PMIC: Add support for twl6030 codec
by Lopez Cruz, Misael 28 Sep '09

28 Sep '09

In order to have TWL6030 CODEC driver as a platform driver, codec data should be passed through twl_platform_data structure. For twl6030 audio codec, the following data is required: - audpwron_gpio: gpio line used to power-up/down the codec. A low-to-high transition powers codec up. Setting audpwron_gpio to a negative value means that codec will use manual power sequence instead of automatic sequence - naudint_irq: irq line for audio interrupt. twl6030 drives NAUDINT line to low when an interrupt (codec ready, plug insertion/removal, etc) is detected Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- drivers/mfd/twl-core.c | 15 +++++++++++++++ include/linux/i2c/twl.h | 11 +++++++++++ 2 files changed, 26 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/twl-core.c b/drivers/mfd/twl-core.c index af4cf47..e9d1ff1 100644 --- a/drivers/mfd/twl-core.c +++ b/drivers/mfd/twl-core.c @@ -108,6 +108,12 @@ #define twl_has_mmc() false #endif +#if defined(CONFIG_SND_SOC_TWL6030) +#define twl_has_codec() true +#else +#define twl_has_codec() false +#endif + #if defined(CONFIG_TWL4030_USB) || defined(CONFIG_TWL4030_USB_MODULE) #define twl_has_usb() true #else @@ -190,6 +196,7 @@ #define BCI_SUB_CHIP_ID SUB_CHIP_ID1 #define GPIO_SUB_CHIP_ID 0 /* NOT SUPPORTED IN TWL6030 */ #define KEYPAD_SUB_CHIP_ID 0 /* ADDED FOR COMPILATION ONLY */ +#define CODEC_SUB_CHIP_ID SUB_CHIP_ID3 /* subchip/slave 0 0x48 - POWER */ #define TWL6030_BASEADD_RTC 0x0000 @@ -632,6 +639,14 @@ add_children(struct twl_platform_data *pdata, unsigned long features) if (IS_ERR(child)) return PTR_ERR(child); } + + if (twl_has_codec() && pdata->codec) { + child = add_child(CODEC_SUB_CHIP_ID, "twl6030_codec", + pdata->codec, sizeof(*pdata->codec), false, + 0, 0); + if (IS_ERR(child)) + return PTR_ERR(child); + } #endif if (twl_has_usb() && pdata->usb) { diff --git a/include/linux/i2c/twl.h b/include/linux/i2c/twl.h index b687a8b..d8a8823 100644 --- a/include/linux/i2c/twl.h +++ b/include/linux/i2c/twl.h @@ -472,6 +472,16 @@ struct twl_usb_data { enum twl_usb_mode usb_mode; }; +struct snd_soc_codec; + +struct twl_codec_data { + struct snd_soc_codec *codec; + + /* twl6030 */ + int audpwron_gpio; /* audio power-on gpio */ + int naudint_irq; /* audio interrupt */ +}; + struct twl_platform_data { unsigned irq_base, irq_end; struct twl_bci_platform_data *bci; @@ -479,6 +489,7 @@ struct twl_platform_data { struct twl_madc_platform_data *madc; struct twl_keypad_data *keypad; struct twl_usb_data *usb; + struct twl_codec_data *codec; /* LDO regulators common to TWL4030/TWL6030 */ struct regulator_init_data *vdac; -- 1.5.4.3

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[alsa-devel] snd-hda-intel and MSI
by Elimar Riesebieter 28 Sep '09

28 Sep '09

Hi folks, are there any contradictions to enable-msi=1 by default? There are users with chips like an 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) Subsystem: Hewlett-Packard Company Device 30f7 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin B routed to IRQ 33 Region 0: Memory at dc500000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel which only works with MSI. And it is somewhat hard to find the appropriate Documantation in HD-Audio.txt. 2nd it will be very hard for Distributers to maintain modul options by chip ID's. Thanks for cooperation. Elimar (one of Debian's alsa maintainers) -- Do you smell something burning or ist it me?

5 15

[alsa-devel] [PATCHv2 2/7] ASoC: Add PDM DAI format definition
by Lopez Cruz, Misael 28 Sep '09

28 Sep '09

Add DAI format definition for PDM interfaces. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- include/sound/soc-dai.h | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 352d7ee..9239cb0 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -30,6 +30,7 @@ struct snd_pcm_substream; #define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM LRC */ #define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM LRC */ #define SND_SOC_DAIFMT_AC97 5 /* AC97 */ +#define SND_SOC_DAIFMT_PDM 6 /* Pulse density modulation */ /* left and right justified also known as MSB and LSB respectively */ #define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J -- 1.5.4.3

2 1

[alsa-devel] [PATCHv2 7/7] ASoC: TWL6030: Power-up seq completion through audio interrupt
by Lopez Cruz, Misael 28 Sep '09

28 Sep '09

When the codec is powered-up through external AUDPWRON line it starts its power-up sequence. The completion of the sequence is signaled through the audio interrupt, and then codec is operational. CODEC driver starts a wait_for_completion just after AUDPWRON line transitions from low to high. It's signaled as complete when servicing READYINT interrupt. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 46 ++++++++++++++++++++++++++++--------------- 1 files changed, 30 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index 5cf2099..f8bd8ee 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -48,6 +48,7 @@ struct twl6030_data { int codec_powered; unsigned int sysclk; struct work_struct audint_work; + struct completion ready_completion; }; /* @@ -58,7 +59,7 @@ static const u8 twl6030_reg[TWL6030_CACHEREGNUM] = { 0x4B, /* TWL6030_ASICID (ro) 0x01 */ 0x00, /* TWL6030_ASICREV (ro) 0x02 */ 0x00, /* TWL6030_INTID 0x03 */ - 0x41, /* TWL6030_INTMR 0x04 */ + 0x00, /* TWL6030_INTMR 0x04 */ 0x00, /* TWL6030_NCPCTRL 0x05 */ 0x00, /* TWL6030_LDOCTL 0x06 */ 0x00, /* TWL6030_HPPLLCTL 0x07 */ @@ -186,6 +187,23 @@ static inline void twl6030_write_reg_cache(struct snd_soc_codec *codec, } /* + * read from twl6030 hardware register + */ +static int twl6030_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 value; + + if (reg > TWL6030_CACHEREGNUM) + return -EIO; + + twl_i2c_read_u8(TWL6030_MODULE_AUDIO, &value, reg); + twl6030_write_reg_cache(codec, reg, value); + + return value; +} + +/* * write to the twl6030 register space */ static int twl6030_write(struct snd_soc_codec *codec, @@ -372,7 +390,8 @@ void twl6030_naudint_work(struct work_struct *work) dev_alert(codec->dev, "vib drivers over current detection\n"); break; case TWL6030_READYINT: - dev_alert(codec->dev, "codec is ready\n"); + priv->codec_powered = 1; + complete(&priv->ready_completion); break; default: dev_err(codec->dev, "unknown audio interrupt %d\n", intid); @@ -617,25 +636,21 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, /* use AUDPWRON line */ gpio_set_value(audpwron_gpio, 1); - /* power-up sequence latency */ - mdelay(16); + /* wait for ready interrupt */ + wait_for_completion(&priv->ready_completion); /* sync registers updated during power-up sequence */ - twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, - 0x81); - twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, - 0x45); - twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, - 0x01); + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_read(codec, TWL6030_REG_LPPLLCTL); } else { /* use manual power-up sequence */ twl6030_power_up(codec); + priv->codec_powered = 1; } /* initialize vdd/vss registers with reg_cache */ twl6030_init_vdd_regs(codec); - - priv->codec_powered = 1; break; case SND_SOC_BIAS_OFF: if (!priv->codec_powered) @@ -649,10 +664,8 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, udelay(500); /* sync registers updated during power-down sequence */ - twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, - 0x00); - twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, - 0x00); + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x00); } else { @@ -1061,6 +1074,7 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) /* audio interrupt */ INIT_WORK(&priv->audint_work, twl6030_naudint_work); + init_completion(&priv->ready_completion); ret = request_irq(naudint_irq, twl6030_naudint_handler, -- 1.5.4.3

2 1

[alsa-devel] [PATCHv2 6/7] ASoC: TWL6030: Enable audio interrupt
by Lopez Cruz, Misael 28 Sep '09

28 Sep '09

NAUDINT interrupt line is provided by the TWL6030 codec to signal externally events like headset plug/unplug, hook, power-up sequence completion, etc. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 78 +++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/twl6030.h | 10 ++++++ 2 files changed, 87 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index 032619d..5cf2099 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -47,6 +47,7 @@ struct twl6030_data { struct snd_soc_codec codec; int codec_powered; unsigned int sysclk; + struct work_struct audint_work; }; /* @@ -329,6 +330,58 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) return 0; } +/* audio interrupt handler */ +irqreturn_t twl6030_naudint_handler(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct twl6030_data *priv = codec->private_data; + + schedule_work(&priv->audint_work); + + /* disable audint irq to let workqueue to execute */ + disable_irq_nosync(irq); + + return IRQ_HANDLED; +} + +void twl6030_naudint_work(struct work_struct *work) +{ + struct twl_codec_data *twl_codec; + struct snd_soc_codec *codec; + struct twl6030_data *priv; + u8 intid; + + priv = container_of(work, struct twl6030_data, audint_work); + codec = &priv->codec; + twl_codec = codec->control_data; + + twl_i2c_read_u8(TWL6030_MODULE_AUDIO, &intid, TWL6030_REG_INTID); + + switch (intid) { + case TWL6030_THINT: + dev_alert(codec->dev, "die temp over-limit detection\n"); + break; + case TWL6030_PLUGINT: + case TWL6030_UNPLUGINT: + case TWL6030_HOOKINT: + break; + case TWL6030_HFINT: + dev_alert(codec->dev, "hf drivers over current detection\n"); + break; + case TWL6030_VIBINT: + dev_alert(codec->dev, "vib drivers over current detection\n"); + break; + case TWL6030_READYINT: + dev_alert(codec->dev, "codec is ready\n"); + break; + default: + dev_err(codec->dev, "unknown audio interrupt %d\n", intid); + break; + } + + enable_irq(twl_codec->naudint_irq); +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -610,6 +663,7 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, priv->codec_powered = 0; break; } + codec->bias_level = level; return 0; @@ -954,8 +1008,15 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) struct twl6030_data *priv; struct snd_soc_codec *codec; int audpwron_gpio = twl_codec->audpwron_gpio; + int naudint_irq = twl_codec->naudint_irq; int ret = 0; + /* prerequisites */ + if (!naudint_irq) { + dev_err(&pdev->dev, "no audio interrupt irq supplied\n"); + return -EINVAL; + } + priv = kzalloc(sizeof(struct twl6030_data), GFP_KERNEL); if (priv == NULL) return -ENOMEM; @@ -998,6 +1059,17 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) priv->codec_powered = 0; } + /* audio interrupt */ + INIT_WORK(&priv->audint_work, twl6030_naudint_work); + + ret = request_irq(naudint_irq, + twl6030_naudint_handler, + IRQF_TRIGGER_LOW | IRQF_DISABLED, + "twl6030-codec", + codec); + if (ret) + goto gpio2_err; + /* init vio registers */ twl6030_init_vio_regs(codec); @@ -1006,7 +1078,7 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) ret = snd_soc_register_codec(codec); if (ret) - goto gpio2_err; + goto reg_err; twl6030_codec = codec; @@ -1019,6 +1091,8 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) dai_err: snd_soc_unregister_codec(codec); twl6030_codec = NULL; +reg_err: + free_irq(naudint_irq, twl_codec); gpio2_err: if (gpio_is_valid(audpwron_gpio)) gpio_free(audpwron_gpio); @@ -1036,6 +1110,8 @@ static int __devexit twl6030_codec_remove(struct platform_device *pdev) if (gpio_is_valid(twl_codec->audpwron_gpio)) gpio_free(twl_codec->audpwron_gpio); + free_irq(twl_codec->naudint_irq, twl6030_codec); + snd_soc_unregister_dai(&twl6030_dai); snd_soc_unregister_codec(twl6030_codec); diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index 15d3e1b..8a106f2 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -67,6 +67,16 @@ #define TWL6030_VIOREGNUM 18 #define TWL6030_VDDREGNUM 21 +/* INTID (0x03) fields */ + +#define TWL6030_THINT 0x01 +#define TWL6030_PLUGINT 0x02 +#define TWL6030_UNPLUGINT 0x04 +#define TWL6030_HOOKINT 0x08 +#define TWL6030_HFINT 0x10 +#define TWL6030_VIBINT 0x20 +#define TWL6030_READYINT 0x40 + /* NCPCTL (0x05) fields */ #define TWL6030_NCPENA 0x01 -- 1.5.4.3

2 1

[alsa-devel] [PATCHv2 4/7] ASoC: TWL6030: Manual power-up/down sequences
by Lopez Cruz, Misael 28 Sep '09

28 Sep '09

TWL6030 codec device can be powered-up/down through a specific register writes sequence. These sequences can be used when no gpio line is provided for AUDPWRON. When the codec is powered-up in this way, automatic power-down sequence (triggered by thermal shutdown) is not possible. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 117 +++++++++++++++++++++++++++++++++++++++----- sound/soc/codecs/twl6030.h | 16 ++++++ 2 files changed, 120 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index 92797e7..f1e333f 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -223,6 +223,88 @@ static void twl6030_init_vdd_regs(struct snd_soc_codec *codec) } } +/* twl6030 codec manual power-up sequence */ +static void twl6030_power_up(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6030_read_reg_cache(codec, TWL6030_REG_NCPCTL); + ldoctl = twl6030_read_reg_cache(codec, TWL6030_REG_LDOCTL); + lppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); + accctl = twl6030_read_reg_cache(codec, TWL6030_REG_ACCCTL); + + /* enable reference system */ + ldoctl |= TWL6030_REFENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + mdelay(10); + /* enable internal oscillator */ + ldoctl |= TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(10); + /* enable high-side ldo */ + ldoctl |= TWL6030_HSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* enable negative charge pump */ + ncpctl |= TWL6030_NCPENA | TWL6030_NCPOPEN; + twl6030_write(codec, TWL6030_REG_NCPCTL, ncpctl); + udelay(488); + /* enable low-side ldo */ + ldoctl |= TWL6030_LSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* enable low-power pll */ + lppllctl |= TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + /* reset state machine */ + accctl |= TWL6030_RESETSPLIT; + twl6030_write(codec, TWL6030_REG_ACCCTL, accctl); + mdelay(5); + accctl &= ~TWL6030_RESETSPLIT; + twl6030_write(codec, TWL6030_REG_ACCCTL, accctl); + /* disable internal oscillator */ + ldoctl &= ~TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); +} + +/* twl6030 codec manual power-down sequence */ +static void twl6030_power_down(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6030_read_reg_cache(codec, TWL6030_REG_NCPCTL); + ldoctl = twl6030_read_reg_cache(codec, TWL6030_REG_LDOCTL); + lppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); + accctl = twl6030_read_reg_cache(codec, TWL6030_REG_ACCCTL); + + /* enable internal oscillator */ + ldoctl |= TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(10); + /* disable low-power pll */ + lppllctl &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + /* disable low-side ldo */ + ldoctl &= ~TWL6030_LSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* disable negative charge pump */ + ncpctl &= ~(TWL6030_NCPENA | TWL6030_NCPOPEN); + twl6030_write(codec, TWL6030_REG_NCPCTL, ncpctl); + udelay(488); + /* disable high-side ldo */ + ldoctl &= ~TWL6030_HSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* disable internal oscillator */ + ldoctl &= ~TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + /* disable reference system */ + ldoctl &= ~TWL6030_REFENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + mdelay(10); +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -460,12 +542,18 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, /* power-up sequence latency */ mdelay(16); - } - /* sync registers updated during power-up sequence */ - twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, 0x81); - twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, 0x45); - twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x01); + /* sync registers updated during power-up sequence */ + twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, + 0x81); + twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, + 0x45); + twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, + 0x01); + } else { + /* use manual power-up sequence */ + twl6030_power_up(codec); + } /* initialize vdd/vss registers with reg_cache */ twl6030_init_vdd_regs(codec); @@ -482,12 +570,18 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, /* power-down sequence latency */ udelay(500); - } - /* sync registers updated during power-down sequence */ - twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, 0x00); - twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, 0x00); - twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x00); + /* sync registers updated during power-down sequence */ + twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, + 0x00); + twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, + 0x00); + twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, + 0x00); + } else { + /* use manual power-down sequence */ + twl6030_power_down(codec); + } priv->codec_powered = 0; break; @@ -746,9 +840,6 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) goto gpio2_err; priv->codec_powered = 0; - } else { - /* if no gpio is provided, then assume its always on */ - priv->codec_powered = 1; } /* init vio registers */ diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index d99d469..7375ae8 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -67,6 +67,18 @@ #define TWL6030_VIOREGNUM 18 #define TWL6030_VDDREGNUM 21 +/* NCPCTL (0x05) fields */ + +#define TWL6030_NCPENA 0x01 +#define TWL6030_NCPOPEN 0x40 + +/* LDOCTL (0x06) fields */ + +#define TWL6030_LSLDOENA 0x01 +#define TWL6030_HSLDOENA 0x04 +#define TWL6030_REFENA 0x40 +#define TWL6030_OSCENA 0x80 + /* HPPLLCTL (0x07) fields */ #define TWL6030_HPLLENA 0x01 @@ -88,6 +100,10 @@ #define TWL6030_LPLLFIN 0x08 #define TWL6030_HPLLSEL 0x10 +/* ACCCTL (0x2D) fields */ + +#define TWL6030_RESETSPLIT 0x04 + extern struct snd_soc_dai twl6030_dai; extern struct snd_soc_codec_device soc_codec_dev_twl6030; -- 1.5.4.3

2 1

[alsa-devel] [PATCHv2 3/7] ASoC: TWL6030: Add twl6030 codec driver
by Lopez Cruz, Misael 28 Sep '09

28 Sep '09

Initial version of TWL6030 codec driver. The TWL6030 codec uses a propietary PDM-based digital audio interface. TWL6030 codec has two power modes: low-power and high-performance, this initial version only supports high-performance mode. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl6030.c | 842 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6030.h | 94 +++++ 4 files changed, 942 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/twl6030.c create mode 100644 sound/soc/codecs/twl6030.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bbc97fd..fd9f8c1 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -25,6 +25,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TWL4030 if TWL4030_CORE + select SND_SOC_TWL6030 if TWL6030_CORE && GPIOLIB select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WM8350 if MFD_WM8350 @@ -114,6 +115,9 @@ config SND_SOC_TLV320AIC3X config SND_SOC_TWL4030 tristate +config SND_SOC_TWL6030 + tristate + config SND_SOC_UDA134X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8b75305..b70c8a1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -13,6 +13,7 @@ snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-twl4030-objs := twl4030.o +snd-soc-twl6030-objs := twl6030.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8350-objs := wm8350.o @@ -50,6 +51,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o +obj-$(CONFIG_SND_SOC_TWL6030) += snd-soc-twl6030.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c new file mode 100644 index 0000000..92797e7 --- /dev/null +++ b/sound/soc/codecs/twl6030.c @@ -0,0 +1,842 @@ +/* + * ALSA SoC TWL6030 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.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 <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/platform_device.h> +#include <linux/i2c/twl.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "twl6030.h" + +#define TWL6030_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define TWL6030_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct twl6030_data { + struct snd_soc_codec codec; + int codec_powered; +}; + +/* + * twl6030 register cache & default register settings + */ +static const u8 twl6030_reg[TWL6030_CACHEREGNUM] = { + 0x00, /* not used 0x00 */ + 0x4B, /* TWL6030_ASICID (ro) 0x01 */ + 0x00, /* TWL6030_ASICREV (ro) 0x02 */ + 0x00, /* TWL6030_INTID 0x03 */ + 0x41, /* TWL6030_INTMR 0x04 */ + 0x00, /* TWL6030_NCPCTRL 0x05 */ + 0x00, /* TWL6030_LDOCTL 0x06 */ + 0x00, /* TWL6030_HPPLLCTL 0x07 */ + 0x00, /* TWL6030_LPPLLCTL 0x08 */ + 0x00, /* TWL6030_LPPLLDIV 0x09 */ + 0x00, /* TWL6030_AMICBCTL 0x0A */ + 0x00, /* TWL6030_DMICBCTL 0x0B */ + 0x18, /* TWL6030_MICLCTL 0x0C */ + 0x18, /* TWL6030_MICRCTL 0x0D */ + 0x00, /* TWL6030_MICGAIN 0x0E */ + 0x1B, /* TWL6030_LINEGAIN 0x0F */ + 0x00, /* TWL6030_HSLCTL 0x10 */ + 0x00, /* TWL6030_HSRCTL 0x11 */ + 0x00, /* TWL6030_HSGAIN 0x12 */ + 0x06, /* TWL6030_EARCTL 0x13 */ + 0x00, /* TWL6030_HFLCTL 0x14 */ + 0x03, /* TWL6030_HFLGAIN 0x15 */ + 0x00, /* TWL6030_HFRCTL 0x16 */ + 0x03, /* TWL6030_HFRGAIN 0x17 */ + 0x00, /* TWL6030_VIBCTLL 0x18 */ + 0x00, /* TWL6030_VIBDATL 0x19 */ + 0x00, /* TWL6030_VIBCTLR 0x1A */ + 0x00, /* TWL6030_VIBDATR 0x1B */ + 0x00, /* TWL6030_HKCTL1 0x1C */ + 0x00, /* TWL6030_HKCTL2 0x1D */ + 0x00, /* TWL6030_GPOCTL 0x1E */ + 0x00, /* TWL6030_ALB 0x1F */ + 0x00, /* TWL6030_DLB 0x20 */ + 0x00, /* not used 0x21 */ + 0x00, /* not used 0x22 */ + 0x00, /* not used 0x23 */ + 0x00, /* not used 0x24 */ + 0x00, /* not used 0x25 */ + 0x00, /* not used 0x26 */ + 0x00, /* not used 0x27 */ + 0x00, /* TWL6030_TRIM1 0x28 */ + 0x00, /* TWL6030_TRIM2 0x29 */ + 0x00, /* TWL6030_TRIM3 0x2A */ + 0x00, /* TWL6030_HSOTRIM 0x2B */ + 0x00, /* TWL6030_HFOTRIM 0x2C */ + 0x09, /* TWL6030_ACCCTL 0x2D */ + 0x00, /* TWL6030_STATUS (ro) 0x2E */ +}; + +/* + * twl6030 vio/gnd registers: + * registers under vio/gnd supply can be accessed + * before the power-up sequence, after NRESPWRON goes high + */ +static const int twl6030_vio_reg[TWL6030_VIOREGNUM] = { + TWL6030_REG_ASICID, + TWL6030_REG_ASICREV, + TWL6030_REG_INTID, + TWL6030_REG_INTMR, + TWL6030_REG_NCPCTL, + TWL6030_REG_LDOCTL, + TWL6030_REG_AMICBCTL, + TWL6030_REG_DMICBCTL, + TWL6030_REG_HKCTL1, + TWL6030_REG_HKCTL2, + TWL6030_REG_GPOCTL, + TWL6030_REG_TRIM1, + TWL6030_REG_TRIM2, + TWL6030_REG_TRIM3, + TWL6030_REG_HSOTRIM, + TWL6030_REG_HFOTRIM, + TWL6030_REG_ACCCTL, + TWL6030_REG_STATUS, +}; + +/* + * twl6030 vdd/vss registers: + * registers under vdd/vss supplies can only be accessed + * after the power-up sequence + */ +static const int twl6030_vdd_reg[TWL6030_VDDREGNUM] = { + TWL6030_REG_HPPLLCTL, + TWL6030_REG_LPPLLCTL, + TWL6030_REG_LPPLLDIV, + TWL6030_REG_MICLCTL, + TWL6030_REG_MICRCTL, + TWL6030_REG_MICGAIN, + TWL6030_REG_LINEGAIN, + TWL6030_REG_HSLCTL, + TWL6030_REG_HSRCTL, + TWL6030_REG_HSGAIN, + TWL6030_REG_EARCTL, + TWL6030_REG_HFLCTL, + TWL6030_REG_HFLGAIN, + TWL6030_REG_HFRCTL, + TWL6030_REG_HFRGAIN, + TWL6030_REG_VIBCTLL, + TWL6030_REG_VIBDATL, + TWL6030_REG_VIBCTLR, + TWL6030_REG_VIBDATR, + TWL6030_REG_ALB, + TWL6030_REG_DLB, +}; + +/* + * read twl6030 register cache + */ +static inline unsigned int twl6030_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6030_CACHEREGNUM) + return -EIO; + + return cache[reg]; +} + +/* + * write twl6030 register cache + */ +static inline void twl6030_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6030_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the twl6030 register space + */ +static int twl6030_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + if ((reg < TWL6030_REG_INTID) || (reg > TWL6030_REG_ACCCTL)) + /* read-only registers */ + return -EIO; + + twl6030_write_reg_cache(codec, reg, value); + return twl_i2c_write_u8(TWL6030_MODULE_AUDIO, value, reg); +} + +static void twl6030_init_vio_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + /* allow registers to be accessed by i2c */ + twl6030_write(codec, TWL6030_REG_ACCCTL, cache[TWL6030_REG_ACCCTL]); + + /* avoid read-only registers (ASICID, ASICREV, STATUS) */ + for (i = 2; i < (TWL6030_VIOREGNUM - 1); i++) { + reg = twl6030_vio_reg[i]; + twl6030_write(codec, reg, cache[reg]); + } +} + +static void twl6030_init_vdd_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + for (i = 0; i < TWL6030_VDDREGNUM; i++) { + reg = twl6030_vdd_reg[i]; + twl6030_write(codec, reg, cache[reg]); + } +} + +/* + * MICATT volume control: + * from -6 to 0 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0); + +/* + * MICGAIN volume control: + * from 6 to 30 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0); + +/* + * HSGAIN volume control: + * from -30 to 0 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hs_tlv, -3000, 200, 0); + +/* + * HFGAIN volume control: + * from -52 to 6 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hf_tlv, -5200, 200, 0); + +/* Left analog microphone selection */ +static const char *twl6030_amicl_texts[] = + {"Headset Mic", "Main Mic", "Aux/FM Left", "Off"}; + +/* Right analog microphone selection */ +static const char *twl6030_amicr_texts[] = + {"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"}; + +static const struct soc_enum twl6030_enum[] = { + SOC_ENUM_SINGLE(TWL6030_REG_MICLCTL, 3, 3, twl6030_amicl_texts), + SOC_ENUM_SINGLE(TWL6030_REG_MICRCTL, 3, 3, twl6030_amicr_texts), +}; + +static const struct snd_kcontrol_new amicl_control = + SOC_DAPM_ENUM("Route", twl6030_enum[0]); + +static const struct snd_kcontrol_new amicr_control = + SOC_DAPM_ENUM("Route", twl6030_enum[1]); + +/* Headset DAC playback switches */ +static const struct snd_kcontrol_new hsdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSLCTL, 5, 1, 0); + +static const struct snd_kcontrol_new hsdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSRCTL, 5, 1, 0); + +/* Handsfree DAC playback switches */ +static const struct snd_kcontrol_new hfdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hfdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFRCTL, 2, 1, 0); + +/* Headset driver switches */ +static const struct snd_kcontrol_new hsl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hsr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSRCTL, 2, 1, 0); + +/* Handsfree driver switches */ +static const struct snd_kcontrol_new hfl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFLCTL, 4, 1, 0); + +static const struct snd_kcontrol_new hfr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFRCTL, 4, 1, 0); + +static const struct snd_kcontrol_new twl6030_snd_controls[] = { + /* Capture gains */ + SOC_DOUBLE_TLV("Capture Preamplifier Volume", + TWL6030_REG_MICGAIN, 6, 7, 1, 1, mic_preamp_tlv), + SOC_DOUBLE_TLV("Capture Volume", + TWL6030_REG_MICGAIN, 0, 3, 4, 0, mic_amp_tlv), + + /* Playback gains */ + SOC_DOUBLE_TLV("Headset Playback Volume", + TWL6030_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + TWL6030_REG_HFLGAIN, TWL6030_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), + +}; + +static const struct snd_soc_dapm_widget twl6030_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("MAINMIC"), + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("SUBMIC"), + SND_SOC_DAPM_INPUT("AFML"), + SND_SOC_DAPM_INPUT("AFMR"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HSOL"), + SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("HFL"), + SND_SOC_DAPM_OUTPUT("HFR"), + + /* Analog input muxes for the capture amplifiers */ + SND_SOC_DAPM_MUX("Analog Left Capture Route", + SND_SOC_NOPM, 0, 0, &amicl_control), + SND_SOC_DAPM_MUX("Analog Right Capture Route", + SND_SOC_NOPM, 0, 0, &amicr_control), + + /* Analog capture PGAs */ + SND_SOC_DAPM_PGA("MicAmpL", + TWL6030_REG_MICLCTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MicAmpR", + TWL6030_REG_MICRCTL, 0, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture", + TWL6030_REG_MICLCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right Front Capture", + TWL6030_REG_MICRCTL, 2, 0), + + /* Microphone bias */ + SND_SOC_DAPM_MICBIAS("Headset Mic Bias", + TWL6030_REG_AMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Main Mic Bias", + TWL6030_REG_AMICBCTL, 4, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic1 Bias", + TWL6030_REG_DMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic2 Bias", + TWL6030_REG_DMICBCTL, 4, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", + TWL6030_REG_HSLCTL, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", + TWL6030_REG_HSRCTL, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Left", "Handsfree Playback", + TWL6030_REG_HFLCTL, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Right", "Handsfree Playback", + TWL6030_REG_HFRCTL, 0, 0), + + /* Analog playback switches */ + SND_SOC_DAPM_SWITCH("HSDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hsdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HSDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hsdacr_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hfdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls), + + SND_SOC_DAPM_SWITCH("Headset Left Driver", + SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Headset Right Driver", + SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Handsfree Left Driver", + SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Handsfree Right Driver", + SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls), + + /* Analog playback PGAs */ + SND_SOC_DAPM_PGA("HFDAC Left PGA", + TWL6030_REG_HFLCTL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HFDAC Right PGA", + TWL6030_REG_HFRCTL, 1, 0, NULL, 0), + +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Capture path */ + {"Analog Left Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Left Capture Route", "Main Mic", "MAINMIC"}, + {"Analog Left Capture Route", "Aux/FM Left", "AFML"}, + + {"Analog Right Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Right Capture Route", "Sub Mic", "SUBMIC"}, + {"Analog Right Capture Route", "Aux/FM Right", "AFMR"}, + + {"MicAmpL", NULL, "Analog Left Capture Route"}, + {"MicAmpR", NULL, "Analog Right Capture Route"}, + + {"ADC Left", NULL, "MicAmpL"}, + {"ADC Right", NULL, "MicAmpR"}, + + /* Headset playback path */ + {"HSDAC Left Playback", "Switch", "HSDAC Left"}, + {"HSDAC Right Playback", "Switch", "HSDAC Right"}, + + {"Headset Left Driver", "Switch", "HSDAC Left Playback"}, + {"Headset Right Driver", "Switch", "HSDAC Right Playback"}, + + {"HSOL", NULL, "Headset Left Driver"}, + {"HSOR", NULL, "Headset Right Driver"}, + + /* Handsfree playback path */ + {"HFDAC Left Playback", "Switch", "HFDAC Left"}, + {"HFDAC Right Playback", "Switch", "HFDAC Right"}, + + {"HFDAC Left PGA", NULL, "HFDAC Left Playback"}, + {"HFDAC Right PGA", NULL, "HFDAC Right Playback"}, + + {"Handsfree Left Driver", "Switch", "HFDAC Left PGA"}, + {"Handsfree Right Driver", "Switch", "HFDAC Right PGA"}, + + {"HFL", NULL, "Handsfree Left Driver"}, + {"HFR", NULL, "Handsfree Right Driver"}, +}; + +static int twl6030_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, twl6030_dapm_widgets, + ARRAY_SIZE(twl6030_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int twl6030_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct twl_codec_data *twl_codec = codec->control_data; + struct twl6030_data *priv = codec->private_data; + int audpwron_gpio = twl_codec->audpwron_gpio; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + if (priv->codec_powered) + return 0; + + if (gpio_is_valid(audpwron_gpio)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron_gpio, 1); + + /* power-up sequence latency */ + mdelay(16); + } + + /* sync registers updated during power-up sequence */ + twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, 0x81); + twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, 0x45); + twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x01); + + /* initialize vdd/vss registers with reg_cache */ + twl6030_init_vdd_regs(codec); + + priv->codec_powered = 1; + break; + case SND_SOC_BIAS_OFF: + if (!priv->codec_powered) + return 0; + + if (gpio_is_valid(audpwron_gpio)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron_gpio, 0); + + /* power-down sequence latency */ + udelay(500); + } + + /* sync registers updated during power-down sequence */ + twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, 0x00); + twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, 0x00); + twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x00); + + priv->codec_powered = 0; + break; + } + codec->bias_level = level; + + return 0; +} + +static int twl6030_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + int rate, format; + + /* hardware dai (McPDM) requires bit stream of twice + * the actual rate, application processor (audio backend) + * should upsample accordingly + */ + rate = params_rate(params); + switch (rate) { + case 44100: + case 48000: + break; + default: + dev_err(codec->dev, "unknown rate %d\n", rate); + return -EINVAL; + } + + /* the sample lenght handled by codec at A-D and D-A + * conversion is 16-bits, but the dai requires 32-bits + */ + format = params_format(params); + if (format != SNDRV_PCM_FORMAT_S32_LE) { + dev_err(codec->dev, "unknown format %d\n", format); + return -EINVAL; + } + + return 0; +} + +static int twl6030_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 hppll, lppll; + + hppll = twl6030_read_reg_cache(codec, TWL6030_REG_HPPLLCTL); + hppll &= TWL6030_HPLLRST; + + switch (freq) { + case 12000000: + /* MCLK input, PLL enabled */ + hppll = TWL6030_MCLK_12000KHZ + | TWL6030_HPLLSQRBP + | TWL6030_HPLLENA; + break; + case 19200000: + /* MCLK input, PLL disabled */ + hppll = TWL6030_MCLK_19200KHZ + | TWL6030_HPLLSQRBP + | TWL6030_HPLLBP; + break; + case 26000000: + /* MCLK input, PLL enabled */ + hppll = TWL6030_MCLK_26000KHZ + | TWL6030_HPLLSQRBP + | TWL6030_HPLLENA; + break; + case 38400000: + /* clk slicer input, PLL disabled */ + hppll = TWL6030_MCLK_38400KHZ + | TWL6030_HPLLSQRENA + | TWL6030_HPLLBP; + break; + default: + dev_err(codec->dev, "unknown sysclk rate %d\n", freq); + return -EINVAL; + } + + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + + /* Disable LPPLL and select HPPLL */ + lppll = TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + return 0; +} + +static int twl6030_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + int format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + /* propietary pdm dai format */ + if (format != SND_SOC_DAIFMT_PDM) + return -EINVAL; + + return 0; +} + +static struct snd_soc_dai_ops twl6030_dai_ops = { + .hw_params = twl6030_hw_params, + .set_sysclk = twl6030_set_dai_sysclk, + .set_fmt = twl6030_set_dai_fmt, +}; + +struct snd_soc_dai twl6030_dai = { + .name = "twl6030", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TWL6030_RATES, + .formats = TWL6030_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6030_RATES, + .formats = TWL6030_FORMATS, + }, + .ops = &twl6030_dai_ops, +}; +EXPORT_SYMBOL_GPL(twl6030_dai); + +static int twl6030_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; + + twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int twl6030_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl6030_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static struct snd_soc_codec *twl6030_codec; + +static int twl6030_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + BUG_ON(!twl6030_codec); + + codec = twl6030_codec; + socdev->card->codec = codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create pcms\n"); + return ret; + } + + snd_soc_add_controls(codec, twl6030_snd_controls, + ARRAY_SIZE(twl6030_snd_controls)); + twl6030_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(&pdev->dev, "failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + return ret; +} + +static int twl6030_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_twl6030 = { + .probe = twl6030_probe, + .remove = twl6030_remove, + .suspend = twl6030_suspend, + .resume = twl6030_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl6030); + +static int __devinit twl6030_codec_probe(struct platform_device *pdev) +{ + struct twl_codec_data *twl_codec = pdev->dev.platform_data; + struct twl6030_data *priv; + struct snd_soc_codec *codec; + int audpwron_gpio = twl_codec->audpwron_gpio; + int ret = 0; + + priv = kzalloc(sizeof(struct twl6030_data), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + codec = &priv->codec; + twl_codec->codec = codec; + codec->dev = &pdev->dev; + twl6030_dai.dev = &pdev->dev; + + codec->name = "twl6030"; + codec->owner = THIS_MODULE; + codec->read = twl6030_read_reg_cache; + codec->write = twl6030_write; + codec->set_bias_level = twl6030_set_bias_level; + codec->private_data = priv; + codec->control_data = twl_codec; + codec->dai = &twl6030_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(twl6030_reg); + codec->reg_cache = kmemdup(twl6030_reg, sizeof(twl6030_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + if (gpio_is_valid(audpwron_gpio)) { + ret = gpio_request(audpwron_gpio, "audpwron"); + if (ret) + goto gpio1_err; + + ret = gpio_direction_output(audpwron_gpio, 0); + if (ret) + goto gpio2_err; + + priv->codec_powered = 0; + } else { + /* if no gpio is provided, then assume its always on */ + priv->codec_powered = 1; + } + + /* init vio registers */ + twl6030_init_vio_regs(codec); + + /* power on device */ + twl6030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ret = snd_soc_register_codec(codec); + if (ret) + goto gpio2_err; + + twl6030_codec = codec; + + ret = snd_soc_register_dai(&twl6030_dai); + if (ret) + goto dai_err; + + return 0; + +dai_err: + snd_soc_unregister_codec(codec); + twl6030_codec = NULL; +gpio2_err: + if (gpio_is_valid(audpwron_gpio)) + gpio_free(audpwron_gpio); +gpio1_err: + kfree(codec->reg_cache); +cache_err: + kfree(priv); + return ret; +} + +static int __devexit twl6030_codec_remove(struct platform_device *pdev) +{ + struct twl_codec_data *twl_codec = pdev->dev.platform_data; + + if (gpio_is_valid(twl_codec->audpwron_gpio)) + gpio_free(twl_codec->audpwron_gpio); + + snd_soc_unregister_dai(&twl6030_dai); + snd_soc_unregister_codec(twl6030_codec); + + kfree(twl6030_codec); + twl6030_codec = NULL; + + return 0; +} + +#ifdef CONFIG_PM +static int twl6030_codec_suspend(struct platform_device *pdev, + pm_message_t msg) +{ + return snd_soc_suspend_device(&pdev->dev); +} + +static int twl6030_codec_resume(struct platform_device *pdev) +{ + return snd_soc_resume_device(&pdev->dev); +} +#else +#define twl6030_codec_suspend NULL +#define twl6030_codec_resume NULL +#endif + +static struct platform_driver twl6030_codec_driver = { + .driver = { + .name = "twl6030_codec", + .owner = THIS_MODULE, + }, + .probe = twl6030_codec_probe, + .remove = __devexit_p(twl6030_codec_remove), + .suspend = twl6030_codec_suspend, + .resume = twl6030_codec_resume, +}; + +static int __init twl6030_codec_init(void) +{ + return platform_driver_register(&twl6030_codec_driver); +} +module_init(twl6030_codec_init); + +static void __exit twl6030_codec_exit(void) +{ + platform_driver_unregister(&twl6030_codec_driver); +} +module_exit(twl6030_codec_exit); + +MODULE_DESCRIPTION("ASoC TWL6030 codec driver"); +MODULE_AUTHOR("Misael Lopez Cruz"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h new file mode 100644 index 0000000..d99d469 --- /dev/null +++ b/sound/soc/codecs/twl6030.h @@ -0,0 +1,94 @@ +/* + * ALSA SoC TWL6030 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.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 + * + */ + +#ifndef __TWL6030_H__ +#define __TWL6030_H__ + +#define TWL6030_REG_ASICID 0x01 +#define TWL6030_REG_ASICREV 0x02 +#define TWL6030_REG_INTID 0x03 +#define TWL6030_REG_INTMR 0x04 +#define TWL6030_REG_NCPCTL 0x05 +#define TWL6030_REG_LDOCTL 0x06 +#define TWL6030_REG_HPPLLCTL 0x07 +#define TWL6030_REG_LPPLLCTL 0x08 +#define TWL6030_REG_LPPLLDIV 0x09 +#define TWL6030_REG_AMICBCTL 0x0A +#define TWL6030_REG_DMICBCTL 0x0B +#define TWL6030_REG_MICLCTL 0x0C +#define TWL6030_REG_MICRCTL 0x0D +#define TWL6030_REG_MICGAIN 0x0E +#define TWL6030_REG_LINEGAIN 0x0F +#define TWL6030_REG_HSLCTL 0x10 +#define TWL6030_REG_HSRCTL 0x11 +#define TWL6030_REG_HSGAIN 0x12 +#define TWL6030_REG_EARCTL 0x13 +#define TWL6030_REG_HFLCTL 0x14 +#define TWL6030_REG_HFLGAIN 0x15 +#define TWL6030_REG_HFRCTL 0x16 +#define TWL6030_REG_HFRGAIN 0x17 +#define TWL6030_REG_VIBCTLL 0x18 +#define TWL6030_REG_VIBDATL 0x19 +#define TWL6030_REG_VIBCTLR 0x1A +#define TWL6030_REG_VIBDATR 0x1B +#define TWL6030_REG_HKCTL1 0x1C +#define TWL6030_REG_HKCTL2 0x1D +#define TWL6030_REG_GPOCTL 0x1E +#define TWL6030_REG_ALB 0x1F +#define TWL6030_REG_DLB 0x20 +#define TWL6030_REG_TRIM1 0x28 +#define TWL6030_REG_TRIM2 0x29 +#define TWL6030_REG_TRIM3 0x2A +#define TWL6030_REG_HSOTRIM 0x2B +#define TWL6030_REG_HFOTRIM 0x2C +#define TWL6030_REG_ACCCTL 0x2D +#define TWL6030_REG_STATUS 0x2E + +#define TWL6030_CACHEREGNUM (TWL6030_REG_STATUS + 1) + +#define TWL6030_VIOREGNUM 18 +#define TWL6030_VDDREGNUM 21 + +/* HPPLLCTL (0x07) fields */ + +#define TWL6030_HPLLENA 0x01 +#define TWL6030_HPLLRST 0x02 +#define TWL6030_HPLLBP 0x04 +#define TWL6030_HPLLSQRENA 0x08 +#define TWL6030_HPLLSQRBP 0x10 +#define TWL6030_MCLK_12000KHZ (0 << 5) +#define TWL6030_MCLK_19200KHZ (1 << 5) +#define TWL6030_MCLK_26000KHZ (2 << 5) +#define TWL6030_MCLK_38400KHZ (3 << 5) +#define TWL6030_MCLK_MSK 0x60 + +/* LPPLLCTL (0x08) fields */ + +#define TWL6030_LPLLENA 0x01 +#define TWL6030_LPLLRST 0x02 +#define TWL6030_LPLLSEL 0x04 +#define TWL6030_LPLLFIN 0x08 +#define TWL6030_HPLLSEL 0x10 + +extern struct snd_soc_dai twl6030_dai; +extern struct snd_soc_codec_device soc_codec_dev_twl6030; + +#endif /* End of __TWL6030_H__ */ -- 1.5.4.3

2 1

[alsa-devel] [PATCH 0/3] Add quirks for Vaio VGN
by Pascal Terjan 28 Sep '09

28 Sep '09

Here are some quirks for 3 Vaio VGN model reported on launchpad to need model=vaio Pascal Terjan (3): [ALSA] Add quirk for Sony Vaio VGN-SZ650N [ALSA] Add quirk for Sony Vaio VGN-AR51M [ALSA] Add quirk for Sony Vaio VGN-FZ15G sound/pci/hda/patch_sigmatel.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-)

2 7

[alsa-devel] ctl_elems hooks with dmix plugin
by George Kuriakose 28 Sep '09

28 Sep '09

Hi I was trying to use ctl_elems hooks with dmix plugin but figured that this combination doesn't work. Is there any workaround for this. I am trying the following. There are two softvol devices s0 and s1 going through dmix plugins. I wan't to mute s0 when any app opens s1 and unmute s0 when s1 is closed. I am using the below asoundrc file which results in "No card for this PCM" . Any workarounds or any minor change in libasound which will make this work? Thanks George asoundrc: pcm.softvol0 { type softvol slave { pcm "dmix:1" } control { name "SoftControl0" card 1 } } pcm.softvol1 { type softvol slave { pcm "dmix:1" } control { name "SoftControl1" card 1 } } pcm.s0 { type plug slave.pcm "softvol0" } pcm.s1 { type plug slave.pcm "softvol1" } pcm.s1hook { type hooks # PCM with hooks slave.pcm "s1" hooks.0 { type ctl_elems hook_args [ { name "SoftControl0" preserve true lock true optional true value [ 0 ] } { name "SoftControl1" preserve true lock true optional true value [ 100 ] } ] } }

2 4

[alsa-devel] [PATCH] sscape: add supoort for SPEA Media FX/Reveal SC-600
by Krzysztof Helt 28 Sep '09

28 Sep '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Move code from the OSS sscape driver in order to support old Soundscape OEM models. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Only PCM works. sound/isa/Kconfig | 6 ++- sound/isa/sscape.c | 116 ++++++++++++++++++++++++++++++++++++--------------- 2 files changed, 86 insertions(+), 36 deletions(-) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 51a7e37..b90fc16 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -377,10 +377,12 @@ config SND_SSCAPE select SND_WSS_LIB help Say Y here to include support for Ensoniq SoundScape - soundcards. + and Ensoniq OEM soundcards. The PCM audio is supported on SoundScape Classic, Elite, PnP - and VIVO cards. The MIDI support is very experimental. + and VIVO cards. The supported OEM cards are SPEA Media FX and + Reveal SC-600. + The MIDI support is very experimental. To compile this driver as a module, choose M here: the module will be called snd-sscape. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 6618712..b11c35f 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -127,7 +127,8 @@ enum GA_REG { enum card_type { - SSCAPE, + MEDIA_FX, /* Sequoia S-1000 */ + SSCAPE, /* Sequoia S-2000 */ SSCAPE_PNP, SSCAPE_VIVO, }; @@ -784,20 +785,25 @@ static struct snd_kcontrol_new midi_mixer_ctl = { * These IRQs are encoded as bit patterns so that they can be * written to the control registers. */ -static unsigned __devinit get_irq_config(int irq) +static unsigned __devinit get_irq_config(int sscape_type, int irq) { static const int valid_irq[] = { 9, 5, 7, 10 }; + static const int old_irq[] = { 9, 7, 5, 15 }; unsigned cfg; - for (cfg = 0; cfg < ARRAY_SIZE(valid_irq); ++cfg) { - if (irq == valid_irq[cfg]) - return cfg; - } /* for */ + if (sscape_type == MEDIA_FX) { + for (cfg = 0; cfg < ARRAY_SIZE(old_irq); ++cfg) + if (irq == old_irq[cfg]) + return cfg; + } else { + for (cfg = 0; cfg < ARRAY_SIZE(valid_irq); ++cfg) + if (irq == valid_irq[cfg]) + return cfg; + } return INVALID_IRQ; } - /* * Perform certain arcane port-checks to see whether there * is a SoundScape board lurking behind the given ports. @@ -842,11 +848,39 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if (s->type != SSCAPE_VIVO && (d & 0x9f) != 0x0e) goto _done; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); + if (s->ic_type == IC_OPUS) + activate_ad1845_unsafe(s->io_base); if (s->type == SSCAPE_VIVO) wss_io += 4; + + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); + + /* wait for WSS codec */ + for (d = 0; d < 500; d++) { + if ((inb(wss_io) & 0x80) == 0) + break; + spin_unlock_irqrestore(&s->lock, flags); + msleep(1); + spin_lock_irqsave(&s->lock, flags); + } + snd_printd(KERN_INFO "init delay = %d ms\n", d); + + if ((inb(wss_io) & 0x80) != 0) + goto _done; + + if (inb(wss_io + 2) == 0xff) + goto _done; + + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d); + + if ((inb(wss_io) & 0x80) != 0) + s->type = MEDIA_FX; + + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ for (d = 0; d < 500; d++) { if ((inb(wss_io) & 0x80) == 0) @@ -954,9 +988,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, if (sscape->type == SSCAPE_VIVO) port += 4; - if (dma1 == dma2) - dma2 = -1; - err = snd_wss_create(card, port, -1, irq, dma1, dma2, WSS_HW_DETECT, WSS_HWSHARE_DMA1, &chip); if (!err) { @@ -1051,21 +1082,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) struct resource *wss_res; unsigned long flags; int err; - - /* - * Check that the user didn't pass us garbage data ... - */ - irq_cfg = get_irq_config(irq[dev]); - if (irq_cfg == INVALID_IRQ) { - snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); - return -ENXIO; - } - - mpu_irq_cfg = get_irq_config(mpu_irq[dev]); - if (mpu_irq_cfg == INVALID_IRQ) { - printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); - return -ENXIO; - } + const char *name; /* * Grab IO ports that we will need to probe so that we @@ -1109,8 +1126,41 @@ static int __devinit create_sscape(int dev, struct snd_card *card) goto _release_dma; } - printk(KERN_INFO "sscape: hardware detected at 0x%x, using IRQ %d, DMA %d\n", - sscape->io_base, irq[dev], dma[dev]); + switch (sscape->type) { + case MEDIA_FX: + name = "MediaFX/SoundFX"; + break; + case SSCAPE: + name = "Soundscape"; + break; + case SSCAPE_PNP: + name = "Soundscape PnP"; + break; + case SSCAPE_VIVO: + name = "Soundscape VIVO"; + break; + default: + name = "unknown Soundscape"; + break; + } + + printk(KERN_INFO "sscape: %s card detected at 0x%x, using IRQ %d, DMA %d\n", + name, sscape->io_base, irq[dev], dma[dev]); + + /* + * Check that the user didn't pass us garbage data ... + */ + irq_cfg = get_irq_config(sscape->type, irq[dev]); + if (irq_cfg == INVALID_IRQ) { + snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); + return -ENXIO; + } + + mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); + if (mpu_irq_cfg == INVALID_IRQ) { + printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); + return -ENXIO; + } if (sscape->type != SSCAPE_VIVO) { /* @@ -1141,8 +1191,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ spin_lock_irqsave(&sscape->lock, flags); - activate_ad1845_unsafe(sscape->io_base); - sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x00); /* disable */ sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2e); sscape_write_unsafe(sscape->io_base, GA_SMCFGB_REG, 0x00); @@ -1151,12 +1199,12 @@ static int __devinit create_sscape(int dev, struct snd_card *card) * Enable and configure the DMA channels ... */ sscape_write_unsafe(sscape->io_base, GA_DMACFG_REG, 0x50); - dma_cfg = (sscape->ic_type == IC_ODIE ? 0x70 : 0x40); + dma_cfg = (sscape->ic_type == IC_OPUS ? 0x40 : 0x70); sscape_write_unsafe(sscape->io_base, GA_DMAA_REG, dma_cfg); sscape_write_unsafe(sscape->io_base, GA_DMAB_REG, 0x20); - sscape_write_unsafe(sscape->io_base, - GA_INTCFG_REG, 0xf0 | (mpu_irq_cfg << 2) | mpu_irq_cfg); + mpu_irq_cfg |= mpu_irq_cfg << 2; + sscape_write_unsafe(sscape->io_base, GA_INTCFG_REG, 0xf0 | mpu_irq_cfg); sscape_write_unsafe(sscape->io_base, GA_CDCFG_REG, 0x09 | DMA_8BIT | (dma[dev] << 4) | (irq_cfg << 1)); -- 1.6.0.3 ---------------------------------------------------------------------- Kawalerka za 79 000 zl! Sprawdz >>> http://link.interia.pl/f2348

2 1

[alsa-devel] ASoC update for 2.6.33
by Mark Brown 28 Sep '09

28 Sep '09

The following changes since commit 2c9ee33d37a6f3808c4319f0849671fafa8a6f4c: Mark Brown (1): Merge branch 'for-2.6.32' into for-2.6.33 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Graeme Gregory (1): ASoC: pxa-ssp increase max_channels to 8 sound/soc/pxa/pxa-ssp.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-)

2 1

[alsa-devel] Patch for Zotac IONITX model detection
by gus.alsa＠bourg.net 28 Sep '09

28 Sep '09

Tested this on my Zotac IONITX motherboard (N330 model). Dmesg output: [ 16.464533] ALSA hda_codec.c:3081: hda_codec: model '3stack-6ch' is selected for config 174b:437b (Zotac IONITX) Patch: --- alsa-kernel/pci/hda/patch_realtek.c.orig 2009-09-24 14:50:45.000000000 -0700 +++ alsa-kernel/pci/hda/patch_realtek.c 2009-09-24 14:38:07.000000000 -0700 @@ -16724,6 +16724,7 @@ static struct snd_pci_quirk alc662_cfg_t ALC662_3ST_6ch_DIG), SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x174b, 0x437b, "Zotac IONITX", ALC662_3ST_6ch), {} }; Thanks, Gus ------------------------- Message sent via Atmail Open - http://atmail.org/

2 1

[alsa-devel] No sound after returning from suspend-to-ram
by Ozan Çağlayan 28 Sep '09

28 Sep '09

Hi, A user complains that after returning from suspend-to-ram, he isn't able to get any sound from his laptop until rebooting. I delayed reporting this one because I once thought that it may be related to Pulseaudio or some kind of mixer level problem but now I tried to suspect the kernel and the drivers as well. This is 2.6.30.5 with alsa-* 1.0.21a. I have dmesg and alsa-info outputs before and after s2ram: dmesg before s2ram: http://bugs.pardus.org.tr/attachment.cgi?id=4454 alsa-info before s2ram: http://bugs.pardus.org.tr/attachment.cgi?id=4455 dmesg after s2ram: http://bugs.pardus.org.tr/attachment.cgi?id=4456 alsa-info after s2ram: http://bugs.pardus.org.tr/attachment.cgi?id=4457 Note that a traceback is dumped in dmesg during resume. Maybe it will give a clue about the problem, dunno. And there's something like hda_codec: invalid CONNECT_LIST verb 12[2]:2100 in regular dmesg output, i wonder what does that mean too. Any ideas? Thanks, Ozan Caglayan

2 5

[alsa-devel] Problem with mplayer and OSS on at91sam9g20(kernel 2.6.30)
by Sedji Gaouaou 28 Sep '09

28 Sep '09

Hi, We've been doing some tests here and we noticed a problem when using mplayer that we didn't have previously(on 2.6.27). While playing audio through OSS emulation on 2.6.30 / at91sam9g20ek there seems to be issue with hardware parameters. The sound does not play and a bunch of error messages are comming on the console. Here is the test case on angstrom: -- mplayer output sample ------------------------- mplayer -quiet myfile.mp3 MPlayer 1.0rc1-4.1.2 (C) 2000-2006 MPlayer Team CPU: ARM Playing /media/sda/audio/01.mp3. [..] AUDIO: 44100 Hz, 2 ch, s16le, 192.0 kbit/13.61% (ratio: 24000->176400) Selected audio codec: [mad] afm: libmad (libMAD MPEG layer 1-2-3) [AO OSS]setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-( -- kernel output sample ------------------------- Atmel_ssc_dai: Entering atmel_ssc_hw_params() natmel_ssc_hw_params: RCMR=00010420 RFMR=001f008f TCMR=4a010404 TFMR=001f008f atmel_ssc_dai: Starting clock atmel_ssc_dai,hw_params: SSC initialized Atmel_ssc_dai: Entering atmel_ssc_hw_params() atmel_ssc_hw_params: RCMR=00010420 RFMR=001f008f TCMR=4a010404 TFMR=001f008f atmel_ssc_dai,hw_params: SSC initialized Atmel_ssc_dai: Entering atmel_ssc_hw_params() atmel_ssc_hw_params: RCMR=00010420 RFMR=001f018f TCMR=4a010404 TFMR=001f018f atmel_ssc_dai,hw_params: SSC initialized asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed asoc: machine hw_params failed [..] asoc: machine hw_params failed MPlayer interrupted by signal 2 in module: ao2_init atmel_ssc_shutdown: transmit disabled SSC_SR=0x000004cc atmel_ssc_dau: Stopping clock soc-audio soc-audio: shutdown pop wq checking: Playback status: inactive waiting: yes pop wq D1 WM8731 Playback pop wq D3 WM8731 Playback -- 2 error messages to note ------------------------- mplayer: "[AO OSS]setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(" kernel: "asoc: machine hw_params failed" => please note also that it is working correctly using the alsa output: mplayer -ao alsa myfile.mp3 Here is the log while using the alsa output (it is working in this case): root@at91sam:~$ mplayer -quiet -ao alsa /media/sda/audio/01.mp3 MPlayer 1.0rc1-4.1.2 (C) 2000-2006 MPlayer Team CPU: ARM Playing /media/sda/audio/01.mp3. Audio file file format detected. Clip info: Title: Angel Artist: Massive Attack Album: Mezzanine Year: 1998 Comment: Ripped by Beto Track: 1 Genre: Rock ========================================================================== Forced audio codec: mad Opening audio decoder: [libmad] libmad mpeg audio decoder AUDIO: 44100 Hz, 2 ch, s16le, 192.0 kbit/13.61% (ratio: 24000->176400) Selected audio codec: [mad] afm: libmad (libMAD MPEG layer 1-2-3) ========================================================================== atmel_ssc_startup: SSC_SR=0x204 asoc: WM8731 <-> atmel-ssc0 info: asoc: rate mask 0x6fe asoc: min ch 1 max ch 2 asoc: min rate 8000 max rate 96000 Atmel_ssc_dai: Entering atmel_ssc_hw_params() atmel_ssc_hw_params: RCMR=00010420 RFMR=001f018f TCMR=19010404 TFMR=001f018f atmel_ssc_dai: Starting clock atmel_ssc_dai,hw_params: SSC initialized transmit enabled SSC_SR=0x000104cf AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample) Video: no video Starting playback... [..] atmel_ssc_shutdown: transmit disabled SSC_SR=0x000104c3 atmel_ssc_dau: Stopping clock soc-audio soc-audio: shutdown Exiting... (Quit) pop wq checking: Playback status: inactive waiting: yes pop wq D1 WM8731 Playback pop wq D3 WM8731 Playback egards, Sedji

3 4

[alsa-devel] [PATCH 0/1] alsa-lib: pcm_hw: Always use delay ioctl in snd_pcm_delay()
by Kai Vehmanen 28 Sep '09

28 Sep '09

Hello all, this is a updated version of the alsa-lib patch I mentioned in the earlier thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-August/020377.html That was meant to just describe the possible problem, but now I'm more confident that this should be considered for merging. The related alsa-kernel patch is 'ALSA: Add extra delay count in PCM', which added the 'runtime->delay' property. Please review if this patch makes sense. Kai Vehmanen (1): pcm_hw: Always use delay ioctl in snd_pcm_delay() src/pcm/pcm_hw.c | 22 ---------------------- 1 files changed, 0 insertions(+), 22 deletions(-)

3 2

Re: [alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 28 Sep '09

28 Sep '09

Hi Peter, > Yes it is indicating so. I read through this thread I didn't find any > simple reason why it's not working. Your setup is similar with the > Pandora, i.e. McBSP functional clock is coming the codec and codec is > slave. Yes, I pretty much used the pandora as an example to write mine. > The McBSP is operating if the FSX and CLKX are toggling in this setup. > Is it so that you don't get any DMA interrupts or just few of them? I don't get any DMA int's: ~# cat /proc/interrupts CPU0 11: 0 INTC prcm 12: 0 INTC DMA 25: 0 INTC dispc 37: 5003 INTC gp timer 56: 3425 INTC i2c_omap 62: 0 INTC McBSP 63: 0 INTC McBSP > Your CLKGDV divisor value 256 means too low bit clock and sample rate > for both external 12.288 MHz and internal 96 MHz but still the DMA > should be running (if there is no bug with this divisor value). > Can you try divisor value 8 does it work then? For 48 kHz sample rate > with I2S you need a bit clock of 48 kHz*2*16 = 1.536 MHz and this you > get by dividing the 12.288 MHz with 8. > Yes, I had tried 8 before and just tried it now. Same result. It really seems like there is a DMA issue. Can you think of anything else to try before I dive into the DMA code? Rick

2 1

Re: [alsa-devel] Problems with Intel HDA driver
by Munzir Taha (منذر طه) 27 Sep '09

27 Sep '09

Hi Takashi, Sorry, the correct link to alsa-info report is http://www.alsa-project.org/db/?f=2e50beab11390b9687490feae899217710e81afa On Yaum al-Ithnain 24 Ramadan 1430 4:23:05 pm Takashi Iwai wrote: > At Mon, 14 Sep 2009 15:15:53 +0200, > > I wrote: > > At Mon, 14 Sep 2009 16:15:02 +0300, > > > > Ozan Çağlayan wrote: > > > Takashi Iwai wrote On 14-09-2009 15:36: > > > > Yeah, I think we can add EAPD handling as default for IDT92HD71xx > > > > codecs. I've never seen machines that can't live with GPIO0 bit up. > > > > > > enable_msi doesn't seem to affect the auto-muting problem. I played > > > with every single node in hda-analyzer but can't get auto-mute work. I > > > can manually mute the HP output and the internal speakers by > > > checking/unchecking OUT checkboxes on relevant NIDs. There's a 8xn > > > matrix for GPIO settings, I played with them as well to set unsol but > > > nope. > > > > > > I want to remember that dell-m4-1 makes automute work but breaks the > > > mixer channels: > > > > > > http://www.spinics.net/lists/alsa-devel/msg26714.html > > > > > > Err, I know that I didn't answer the last question in the thread as I'm > > > a little bit lost in the NID pool and didn't have time to investigate > > > further. > > > > Could you check whether this still valid? > > > > Basically the model option doesn't change the auto-mute behavior. > > It just changes which pin to be detected and which pin to mute. > > So, if model=dell-m4-1 makes the auto-muting working, it means > > that the BIOS setup is broken. > > I looked through alsa-info output now, and found the culprit. > NID 0x0f is set as a line-out, and currently the driver doesn't > don't do auto-muting for line-out unless a headphone is present. > Changing it to a headphone should fix. Or you can give hp_detect=1 > hint via sysfs and reconfigure. > > > Takashi -- Munzir Taha Telecommunications and Electronics Engineer Master CIW Designer, ICDL, MOUS, Linux+, LPI, SCSA Riyadh, SA

2 6

[alsa-devel] Problems with Intel HDA driver
by Andrius Štikonas 27 Sep '09

27 Sep '09

On my new laptop there were no sound with snd-hda-intel driver, so I tried to change model parameter. It seems that with model=hp-m4 my soundcard mostly works (except that main speakers still play when I plug headphones). I've attached my alsa-info.txt script. -- Andrius Štikonas

4 11

[PATCH 1/1] ALSA: hda - Analog Devices AD1984A add HP Touchsmart model
by Miguel de Barros 27 Sep '09

27 Sep '09

Reference: ALSA bug #0004614 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4614 port-A (0x11) - front hp-out port-D (0x12) - rear line out port-E (0x1c) - front mic-in port-F (0x16) - Internal speakers digital-mic (0x17) - Internal mic init verbs, mixers, jack sensing and PCI_QUIRK to support this hardware Signed-off-by: Miguel de Barros <miguel.de.barros(a)bluewin.ch> diff --git a/Documentation/HD-Audio-Models.txt b/Documentation/HD-Audio-Models.txt index abffd41..9e850ed 100644 --- a/Documentation/HD-Audio-Models.txt +++ b/Documentation/HD-Audio-Models.txt @@ -208,6 +208,7 @@ AD1884A / AD1883 / AD1984A / AD1984B laptop laptop with HP jack sensing mobile mobile devices with HP jack sensing thinkpad Lenovo Thinkpad X300 + touchsmart HP Touchsmart AD1884 ====== diff --git a/pci/hda/patch_analog.c b/pci/hda/patch_analog.c index 215e72a..2d603f6 100644 --- a/pci/hda/patch_analog.c +++ b/pci/hda/patch_analog.c @@ -4032,6 +4032,127 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec) } /* + * HP Touchsmart + * port-A (0x11) - front hp-out + * port-B (0x14) - unused + * port-C (0x15) - unused + * port-D (0x12) - rear line out + * port-E (0x1c) - front mic-in + * port-F (0x16) - Internal speakers + * digital-mic (0x17) - Internal mic + */ + +static struct hda_verb ad1984a_touchsmart_verbs[] = { + /* DACs; unmute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + /* Port-A (HP) mixer - route only from analog mixer */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-A pin */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Port-A (HP) pin - always unmuted */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Port-E (int speaker) mixer - route only from analog mixer */ + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03}, + /* Port-E pin */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + /* Port-F (int speaker) mixer - route only from analog mixer */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-F pin */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* capture sources */ + /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* unsolicited event for pin-sense */ + {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, + /* allow to touch GPIO1 (for mute control) */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */ + /* internal mic - dmic */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* set magic COEFs for dmic */ + {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, + {0x01, AC_VERB_SET_PROC_COEF, 0x08}, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), +/* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = ad1884a_mobile_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT), + { } /* end */ +}; + +/* switch to external mic if plugged */ +static void ad1984a_touchsmart_automic(struct hda_codec *codec) +{ + if (snd_hda_codec_read(codec, 0x1c, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000) { + snd_hda_codec_write(codec, 0x0c, 0, + AC_VERB_SET_CONNECT_SEL, 0x4); + } else { + snd_hda_codec_write(codec, 0x0c, 0, + AC_VERB_SET_CONNECT_SEL, 0x5); + } +} + + +/* unsolicited event for HP jack sensing */ +static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case AD1884A_HP_EVENT: + ad1884a_hp_automute(codec); + break; + case AD1884A_MIC_EVENT: + ad1984a_touchsmart_automic(codec); + break; + } +} + +/* initialize jack-sensing, too */ +static int ad1984a_touchsmart_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1884a_hp_automute(codec); + ad1984a_touchsmart_automic(codec); + return 0; +} + + +/* */ enum { @@ -4039,6 +4160,7 @@ enum { AD1884A_LAPTOP, AD1884A_MOBILE, AD1884A_THINKPAD, + AD1984A_TOUCHSMART, AD1884A_MODELS }; @@ -4047,6 +4169,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = { [AD1884A_LAPTOP] = "laptop", [AD1884A_MOBILE] = "mobile", [AD1884A_THINKPAD] = "thinkpad", + [AD1984A_TOUCHSMART] = "touchsmart", }; static struct snd_pci_quirk ad1884a_cfg_tbl[] = { @@ -4059,6 +4182,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), + SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART), {} }; @@ -4142,6 +4266,21 @@ static int patch_ad1884a(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event; codec->patch_ops.init = ad1984a_thinkpad_init; break; + case AD1984A_TOUCHSMART: + spec->mixers[0] = ad1984a_touchsmart_mixers; + spec->init_verbs[0] = ad1984a_touchsmart_verbs; + spec->multiout.dig_out_nid = 0; + codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event; + codec->patch_ops.init = ad1984a_touchsmart_init; + /* set the upper-limit for mixer amp to 0dB for avoiding the + * possible damage by overloading + */ + snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); + break; } return 0; -- 1.6.0.4 --=-J56U1/hvhPtbcxv8DR5c--

1 0

[PATCH 1/1] ALSA: hda - Analog Devices AD1984A add HP Touchsmart model modified: Documentation/HD-Audio-Models.txt modified: pci/hda/patch_analog.c
by Miguel de Barros 27 Sep '09

27 Sep '09

Signed-off-by: Miguel de Barros <miguel.de.barros(a)bluewin.ch> diff --git a/Documentation/HD-Audio-Models.txt b/Documentation/HD-Audio-Models.txt index abffd41..9e850ed 100644 --- a/Documentation/HD-Audio-Models.txt +++ b/Documentation/HD-Audio-Models.txt @@ -208,6 +208,7 @@ AD1884A / AD1883 / AD1984A / AD1984B laptop laptop with HP jack sensing mobile mobile devices with HP jack sensing thinkpad Lenovo Thinkpad X300 + touchsmart HP Touchsmart AD1884 ====== diff --git a/pci/hda/patch_analog.c b/pci/hda/patch_analog.c index 215e72a..7df8a08 100644 --- a/pci/hda/patch_analog.c +++ b/pci/hda/patch_analog.c @@ -4032,6 +4032,143 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec) } /* + * HP Touchsmart + * port-A (0x11) - front hp-out + * port-B (0x14) - unused + * port-C (0x15) - unused + * port-D (0x12) - rear line out + * port-E (0x1c) - front mic-in + * port-F (0x16) - Internal speakers + * digital-mic (0x17) - Internal mic + */ + +static struct hda_verb ad1984a_touchsmart_verbs[] = { + /* DACs; unmute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + /* Port-A (HP) mixer - route only from analog mixer */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-A pin */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Port-A (HP) pin - always unmuted */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Port-E (int speaker) mixer - route only from analog mixer */ + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03}, + /* Port-E pin */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + /* Port-F (int speaker) mixer - route only from analog mixer */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-F pin */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* capture sources */ + /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* unsolicited event for pin-sense */ + {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, + /* allow to touch GPIO1 (for mute control) */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */ + /* internal mic - dmic */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* set magic COEFs for dmic */ + {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, + {0x01, AC_VERB_SET_PROC_COEF, 0x08}, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), +/* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = ad1884a_mobile_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Source", + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_input_mux ad1984a_touchsmart_capture_source = { + .num_items = 3, + .items = { + { "Mix", 0x3 }, /* Mixer */ + { "Mic", 0x4 }, /* port-E */ + { "Internal Mic", 0x5 }, /* Digital Mic */ + }, +}; + +/* switch to external mic if plugged */ +static void ad1984a_touchsmart_automic(struct hda_codec *codec) +{ + if (snd_hda_codec_read(codec, 0x1c, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000) { + snd_hda_codec_write(codec, 0x0c, 0, + AC_VERB_SET_CONNECT_SEL, 0x4); + } else { + snd_hda_codec_write(codec, 0x0c, 0, + AC_VERB_SET_CONNECT_SEL, 0x5); + } +} + + +/* unsolicited event for HP jack sensing */ +static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case AD1884A_HP_EVENT: + ad1884a_hp_automute(codec); + break; + case AD1884A_MIC_EVENT: + ad1984a_touchsmart_automic(codec); + break; + } +} + +/* initialize jack-sensing, too */ +static int ad1984a_touchsmart_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1884a_hp_automute(codec); + ad1984a_touchsmart_automic(codec); + return 0; +} + + +/* */ enum { @@ -4039,6 +4176,7 @@ enum { AD1884A_LAPTOP, AD1884A_MOBILE, AD1884A_THINKPAD, + AD1984A_TOUCHSMART, AD1884A_MODELS }; @@ -4047,6 +4185,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = { [AD1884A_LAPTOP] = "laptop", [AD1884A_MOBILE] = "mobile", [AD1884A_THINKPAD] = "thinkpad", + [AD1984A_TOUCHSMART] = "touchsmart", }; static struct snd_pci_quirk ad1884a_cfg_tbl[] = { @@ -4059,6 +4198,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), + SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART), {} }; @@ -4142,6 +4282,22 @@ static int patch_ad1884a(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event; codec->patch_ops.init = ad1984a_thinkpad_init; break; + case AD1984A_TOUCHSMART: + spec->mixers[0] = ad1984a_touchsmart_mixers; + spec->init_verbs[0] = ad1984a_touchsmart_verbs; + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1984a_touchsmart_capture_source; + codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event; + codec->patch_ops.init = ad1984a_touchsmart_init; + /* set the upper-limit for mixer amp to 0dB for avoiding the + * possible damage by overloading + */ + snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); + break; } return 0; -- 1.6.0.4 --=-w/QltLfzo6NHdpL3XZCU--

1 0

Re: [alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 26 Sep '09

26 Sep '09

Hi Peter, I forgot to tell you that if I deliberately misconfigure the McBXP_CLKS pin so that it's an output, the FSX -> DACLRC/ADCLRC signal stops. This should indicate that the CLKOUT signal is actually making it to the CLKS pin on the OMAP. Right? Rick

2 1

Re: [alsa-devel] [Pkg-alsa-devel] Bug#527074: /etc/init.d/alsa-utils: `/etc/init.d/alsa-utils stop` triggers amixer error
by Elimar Riesebieter 26 Sep '09

26 Sep '09

* Piotr Lewandowski [090925 20:54 +0200] > Hi alsa-devel@! > > One of the maintainers of alsa-utils package in Debian advised me to > ask for your help. I've encountered following problem: > > #v+ > $ amixer -c 0 -q set Mic 0% mute > amixer: Invalid command! > #v- > > This command is invoked from /etc/init.d/alsa-utils script during > system shutdown. Do you have any ideas how to fix that? > > Here is some basic info about my hardware: > > #v+ > $ lspci -v | sed -ne '/Audio/,/^$/p' > 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High > Definition Audio Controller (rev 01) > Subsystem: Hewlett-Packard Company Device 30a2 > Flags: bus master, fast devsel, latency 0, IRQ 16 > Memory at f4580000 (64-bit, non-prefetchable) [size=16K] > Capabilities: <access denied> > Kernel driver in use: HDA Intel > > $ cat /proc/asound/version > Advanced Linux Sound Architecture Driver Version 1.0.18a. 1. It would be an option to insert: 'echo option "snd-hda-intel enable_msi=1"' >> /etc/modprobe.d/local.conf 2. It is a good idea to update to alsa-source 1.0.21-2 and build biuld your driver up from there. Elimar -- Never make anything simple and efficient when a way can be found to make it complex and wonderful ;-)

1 0

[alsa-devel] [PATCH] [PATCH] [ASOC] pxa-ssp increase max_channels to 8
by Graeme Gregory 25 Sep '09

25 Sep '09

When running in TDM mode there can be more than 2 channels used. Datasheet has figures for upto 8 channels so increase max_channels on all SSP interfaces to this figure. Signed-off-by: Graeme Gregory <dp(a)xora.org.uk> --- sound/soc/pxa/pxa-ssp.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 19c4540..e8a5629 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -767,13 +767,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -787,13 +787,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -808,13 +808,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -829,13 +829,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, -- 1.6.3.3

2 1

Re: [alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 25 Sep '09

25 Sep '09

Hi Peter, Thanks very much for the help. > I think the problem is here. > The MCBSP_CLKS pin is connected to CLK256FS pin of TWL4030 (or TPA). > Since you are not using the TWL codec, there will be no clock on the MCBSP_CLKS > pin, thus no functional clock for McBSP, thus no bits are shifted in or out. > You can try to use the internal functional clock: > > ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_FCLK, 0, > SND_SOC_CLOCK_IN); I have checked the CLKOUT (WM8731) -> McBSP_CLKS (OMAP3) pin and it has a 12288000 clock on it. I've tried using the internal clock as you show above, just to see if I get anything out on DX but I get the same thing: DX just goes low for the "aplay" duration. Any idea why I'm not getting DMA interrupts? This would indeed cause no data at DX. omap2_dma_irq_handler() never gets called but omap_init_dma() and omap_start_dma() get called. Do you think there may be any issues with my DMA code (specifically arch/arm/plat-omap/dma.c). As I mentioned, I'm at "2.6.29-rc3-omap1" Thanks, Rick

1 0

[alsa-devel] [PATCH] [ASOC] pxa-ssp increase max_channels to 8
by Graeme Gregory 25 Sep '09

25 Sep '09

When running in TDM mode there can be more than 2 channels used. Datasheet has figures for upto 8 channels so increase max_channels on all SSP interfaces to this figure. --- sound/soc/pxa/pxa-ssp.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 19c4540..e8a5629 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -767,13 +767,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -787,13 +787,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -808,13 +808,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -829,13 +829,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, -- 1.6.3.3

1 0

[alsa-devel] Mic stops working after installation (ALC861)
by remrot＠gmail.com 25 Sep '09

25 Sep '09

/* Sorry for posting this to the devel list, but the problem seems to be trickier than I first thought * * and I don't have much time left to fix it (not my machine)! Thanks... */ Hi! I've got a very strange problem with a ALC861 soundchip on a friend's Toshiba A100-507 Laptop. Sound output and input work perfectly within a live session using the Ubuntu Jaunty and Karmic (Alpha6) CDs but immediately stops working after installing the system. Currently I've installed the latest Karmic containing kernel 2.6.31 and ALSA 1.0.20. I've attached the output of the alsa-info script as well as the output of 'pacmd list' for both systems (live and installed). Right after the installation the sound stopped working and I couldn't record anything. Alsamixer showed only output devices, no capture devices at all (everything has been correct within the live session). In 'dmesg' I found the following error messages: [ 279.924019] hda-intel: azx_get_response timeout, switching to polling mode: last cmd=0x301f000a [ 280.928019] hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x301f000a After doing some research I found this guide (thx Mr. Iwai): http://www.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.html#_cod… And indeed, adding 'probe_mask=8' fixed the sound output and brought back the capture devices (remember that they were there within the live session without that option). I've also tried the 'model=toshiba' option (combined with 'probe_mask' and standalone) but that didn't help at all (no capture devices and no sound). The current situation: I've already posted to the PulseAudio mailing ist and got some help from Mr. Colin Guthrie but we haven't found a solution (everything seems OK regarding PA). I still can't record anything within the installed system. I tried audacity, alsarec and the gnome-sound-recorder without luck. When recording in audacity, the timeline cursor doesn't move, it only flickers. In gnome-recorder, recording stops right after it has started and the application hangs on exit. PA's volume meter doesn't show anything either, it is empty. I compared the configurations on both systems but I see no differences that are not related to the 'probe_mask' option... I really believe that there is a solution because everything works in a live session. Can anybody help me with this? Regards JK

1 1

[alsa-devel] WM8731 using I2S on omap3 McBSP2 issues
by Rick Bronson 25 Sep '09

25 Sep '09

Hi, I'm working on getting a WM8731 (in SLAVE) working using I2S on omap3 McBSP2 (in MASTER). Here is the hardware: http://www.efn.org/~rick/pub/wm8731.jpg Here are the PADCONF's and the pins they hook to on the WM8731: /*Audio Interface for csb740 */\ MUX_VAL(CP(McBSP2_FSX), (OFF_OUT_PD | IDIS | PTU | EN | M0)) /* McBSP2_FSX -> I2S_FRM -> LRCLK -> DACLRC and ADCLRC*/\ MUX_VAL(CP(McBSP2_CLKX), (OFF_OUT_PD | IDIS | PTU | EN | M0)) /* McBSP2_CLKX -> I2S_CLK -> I2S_BCLK -> BCLK */\ MUX_VAL(CP(McBSP2_DR), (OFF_IN_PD | IEN | PTD | DIS | M0)) /* McBSP2_DR <- I2S_RXD <- ACDDAT */\ MUX_VAL(CP(McBSP2_DX), (OFF_OUT_PD | IDIS | PTU | EN | M0)) /* McBSP2_DX -> I2S_TXD -> DACDAT */\ MUX_VAL(CP(McBSP_CLKS), (OFF_IN_PD | IEN | PTD | DIS | M0)) /* McBSP_CLKS <- I2S_MCLK <- CLKOUT */\ My kernel is "2.6.29-rc3-omap1", here's what I see at bootup: 1 WM8731 Audio Codec 0.13<6>asoc: WM8731 <-> omap-mcbsp-dai-0 mapping ok 2 ALSA device list: 3 #0: omap3csb740 (WM8731) ... 4 beagle:~# aplay test48000.wav 5 Playing WAVE 'test48000.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo 6 aplay: pcm_write:1269: write error: Input/output error I get no audio output, other than a "click" when the 8731 is initialized upon boot. When I look at the signals the McBSP2_DX line just goes low for the duration of the "aplay", it does not toggle. The other lines look fine. The file is at http://www.efn.org/~rick/pub/omap3csb740.c but roughly, here's what I'm doing: static struct snd_soc_dai_link omap3csb740_dai = { ... .cpu_dai = &omap_mcbsp_dai[0], /* use 1st one */ static int __init omap3csb740_soc_init(void) ... *(unsigned int *)omap_mcbsp_dai[0].private_data = 1; /* McBSP2 */ *(unsigned int *)omap3csb740_dai.cpu_dai->private_data = 1; /* McBSP2 debug only? */ static int omap3csb740_hw_params(struct snd_pcm_substream *substream, ... ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); ret = snd_soc_dai_set_sysclk(codec_dai, 0, 12288000, SND_SOC_CLOCK_IN); ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0, SND_SOC_CLOCK_IN); ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 256); Any help is greatly appreciated. Rick Bronson

3 7

[alsa-devel] Exposure of the input devices of Conexant CX20561 amplifier
by Matthias Bläsing 24 Sep '09

24 Sep '09

Hello, it would be nice if I could control the input channels of my lenovo laptop (Lenovo R500). These are configured for microphone input, but the amplifier can be deactivated, so that it can work with the higher line-in level. The resulting quality is not good, but at least usable. The volume control is joint, but the amplifier control are seperated. With the HDAAnalyser I could find the Nodes (0x17 (rear) and 0x18 (front)) can be used to control the amplification. The setting can be changed between 0 and 4, where 0 makes it possible to use the input for line-in, while 4 is better suited for microphone input. I'm not sure whether it would be better to expose this as a stepped control or just a switch (amplifier on/off). These are the loaded modules (from kernel 2.6.31): snd_usb_audio 74152 3 snd_usb_lib 13904 1 snd_usb_audio snd_hda_codec_conexant 17684 1 snd_seq_dummy 2748 0 snd_seq_oss 24620 0 snd_hda_intel 24064 9 snd_hda_codec 65724 2 snd_hda_codec_conexant,snd_hda_intel snd_seq_midi 6232 0 snd_hwdep 6772 3 snd_usb_audio,snd_hda_codec snd_rawmidi 19600 2 snd_usb_lib,snd_seq_midi snd_seq_midi_event 6580 2 snd_seq_oss,snd_seq_midi snd_pcm_oss 33516 0 snd_mixer_oss 13416 1 snd_pcm_oss snd_seq 44744 7 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_pcm 64872 6 snd_usb_audio,snd_hda_intel,snd_hda_codec,snd_pcm_oss snd_timer 18448 2 snd_seq,snd_pcm snd_seq_device 6424 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd_page_alloc 8556 2 snd_hda_intel,snd_pcm snd 52452 35 snd_usb_audio,snd_hda_codec_conexant,snd_seq_oss,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_rawmidi,snd_pcm_oss,snd_mixer_oss,snd_seq,snd_pcm,snd_timer,snd_seq_device soundcore 6876 1 snd I attached the /proc/asound/card0/codec#0 info to this email. If you need further information - please ask. Matthias

2 2

[alsa-devel] [PATCH 1/2] ASoC: DaVinci: Fix divide by zero error during 1st execution
by Troy Kisky 24 Sep '09

24 Sep '09

When both playback and capture stream were open davinci_i2s_hw_params was setting parameters for the wrong stream. The fix for davinci_i2s_hw_params is sufficient, but it looks like a race still happens in davici_pcm_open. This patch also makes the race smaller but the next patch provides a better fix. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- sound/soc/davinci/davinci-i2s.c | 3 ++- sound/soc/davinci/davinci-pcm.c | 12 +++++------- 2 files changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 12a6c54..d32e197 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -353,8 +353,9 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct davinci_pcm_dma_params *dma_params = dai->dma_data; struct davinci_mcbsp_dev *dev = dai->private_data; + struct davinci_pcm_dma_params *dma_params = + dev->dma_params[substream->stream]; struct snd_interval *i = NULL; int mcbsp_word_length; unsigned int rcr, xcr, srgr; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 091dacb..002808b 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -126,16 +126,9 @@ static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data) static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; struct edmacc_param p_ram; int ret; - if (!dma_data) - return -ENODEV; - - prtd->params = dma_data; - /* Request master DMA channel */ ret = edma_alloc_channel(prtd->params->channel, davinci_pcm_dma_irq, substream, @@ -244,6 +237,10 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd; int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_pcm_dma_params *params = rtd->dai->cpu_dai->dma_data; + if (!params) + return -ENODEV; snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware); /* ensure that buffer size is a multiple of period size */ @@ -257,6 +254,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) return -ENOMEM; spin_lock_init(&prtd->lock); + prtd->params = params; runtime->private_data = prtd; -- 1.5.6.3

2 5

[alsa-devel] AIC23 playback volume control not working
by Aggarwal, Anuj 24 Sep '09

24 Sep '09

Hi, I am trying to change the playback volume using amixer commands but am not able to see any effect. Register writes are happening correctly but no change in volume. Here is a short log: root@omap3evm:~# amixer cset numid=1 0 tlv320aic23_write: reg=0x2, value=0x0 data[0]=0x4, data[1]=0x0 tlv320aic23_write: reg=0x3, value=0x0 data[0]=0x6, data[1]=0x0 numid=1,iface=MIXER,name='Digital Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=127,step=0 : values=0,0 | dBscale-min=-121.00dB,step=1.00dB,mute=0 root@omap3evm:~# root@omap3evm:~# root@omap3evm:~# root@omap3evm:~# root@omap3evm:~# root@omap3evm:~# amixer cset numid=1 127 tlv320aic23_write: reg=0x2, value=0x7f data[0]=0x4, data[1]=0x7f tlv320aic23_write: reg=0x3, value=0x7f data[0]=0x6, data[1]=0x7f numid=1,iface=MIXER,name='Digital Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=127,step=0 : values=127,127 | dBscale-min=-121.00dB,step=1.00dB,mute=0 root@omap3evm:~# root@omap3evm:~# Any suggestions are appreciated. Regards, Anuj Aggarwal

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[alsa-devel] Capture not working in MMAP mode with '-v'
by Aggarwal, Anuj 24 Sep '09

24 Sep '09

Hi, When I try to capture audio in MMAP mode using the following command, it works properly: arecord -f cd -d 10 -M rec01.wav However, if I specify '-v' along with the above command, it fails to capture anything and prints log and returns immediately: Recording WAVE 'rec03.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Hardware PCM card 0 'omap3evm' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22052 period_size : 5513 period_time : 125011 tstamp_mode : NONE period_step : 1 avail_min : 5513 period_event : 0 start_threshold : 1 stop_threshold : 22052 silence_threshold: 0 silence_size : 0 boundary : 1445199872 appl_ptr : 0 hw_ptr : 0 mmap_area[0] = 0x4030e000,0,32 (16) mmap_area[1] = 0x4030e000,16,32 (16) I have not observed this problem while playing audio, only while capture this issue crops up. Any pointers? Thanks & Regards, Anuj Aggarwal

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Re: [alsa-devel] [PATCH - hda-intel conexant benq r55 laptop 1/1] [ALSA] CD-audio sound for hda-intel conexant benq laptop
by Takashi Iwai 24 Sep '09

24 Sep '09

At Tue, 22 Sep 2009 21:42:40 +0200, nowymarluk(a)wp.pl wrote: > > From: edoiks <edoiks(a)edoiks-laptop.(none)> > > After puting a cd-audio inside my laptop there was no sound out here, so I decided to install alsa-driver with debug level and setup a model=test, it didn't help, but then I look at source code and added this few lines, now cd-audio is working both when playback/recording. > > Signed-off-by: Lukasz Marcinowski <nowymarluk(a)wp.pl> > Signed-off-by: edoiks <edoiks(a)edoiks-laptop.(none)> Judging from your mail address, I stripped this edoiks line. Also, I fixed some strings for mixer elements and capture source items to follow the standard rule. E.g. "CD Audio xxx" is "CD xxx", and remove trailing spaces from the string. Now applied to sound git tree, and will be eventually merged to the upstream, too. thanks, Takashi > diff --git a/pci/hda/patch_conexant.c b/pci/hda/patch_conexant.c > index 9d899ed..9afacb6 100644 > --- a/pci/hda/patch_conexant.c > +++ b/pci/hda/patch_conexant.c > @@ -682,11 +682,13 @@ static struct hda_input_mux cxt5045_capture_source = { > }; > > static struct hda_input_mux cxt5045_capture_source_benq = { > - .num_items = 3, > + .num_items = 5, > .items = { > { "IntMic", 0x1 }, > { "ExtMic", 0x2 }, > { "LineIn", 0x3 }, > + { "CD Pin", 0x4 }, > + { "Mixer ", 0x0 }, > } > }; > > @@ -811,11 +813,19 @@ static struct snd_kcontrol_new cxt5045_mixers[] = { > }; > > static struct snd_kcontrol_new cxt5045_benq_mixers[] = { > + HDA_CODEC_VOLUME("CD Audio Capture Volume", 0x1a, 0x04, HDA_INPUT), > + HDA_CODEC_MUTE("CD Audio Capture Switch", 0x1a, 0x04, HDA_INPUT), > + HDA_CODEC_VOLUME("CD Audio Playback Volume", 0x17, 0x4, HDA_INPUT), > + HDA_CODEC_MUTE("CD Audio Playback Switch", 0x17, 0x4, HDA_INPUT), > + > HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT), > HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT), > HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT), > HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT), > > + HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT), > + HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT), > + > {} > }; > > -- > 1.6.0.4 >

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 24 Sep '09

24 Sep '09

The following changes since commit 877ae70763fa0b19ffbe25e7e119fc96f52fbbd3: Phil Vandry (1): ASoC: wm8753: fix mapping when MONOMIX is set to Stereo are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Chaithrika U S (1): ASoC: DaVinci: Correct McASP FIFO initialization Cliff Cai (1): ASoC: Blackfin: fix inverted handling of SPORT0 on PORT F/G Troy Kisky (2): ASoC: DaVinci: Fix divide by zero error during 1st execution ASoC: Davinci: Fix race with cpu_dai->dma_data sound/soc/blackfin/bf5xx-i2s.c | 8 ++-- sound/soc/blackfin/bf5xx-tdm.c | 8 ++-- sound/soc/davinci/davinci-i2s.c | 37 ++++++----------- sound/soc/davinci/davinci-mcasp.c | 80 +++++++++++++------------------------ sound/soc/davinci/davinci-mcasp.h | 7 +++- sound/soc/davinci/davinci-pcm.c | 13 +++--- sound/soc/davinci/davinci-pcm.h | 1 - 7 files changed, 60 insertions(+), 94 deletions(-)

2 1

[alsa-devel] Audio problems with AD1981A (Subsystem: 0e11:00ad)
by Simos Xenitellis 24 Sep '09

24 Sep '09

Hi All, I am helping a user with the following hardware, http://www.alsa-project.org/db/?f=7f926200ee31b5b22222336b9e25bb0e60671a90 (Desktop computer Compaq Evo D510 SFF, with onboard sound card) lspci -vvc shows: 00:1f.5 0401: 8086:24c5 (rev 01) Subsystem: 0e11:00ad Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin B routed to IRQ 17 Region 0: I/O ports at 2000 [size=256] Region 1: I/O ports at 2400 [size=64] Region 2: Memory at fc480400 (32-bit, non-prefetchable) [size=512] Region 3: Memory at fc480600 (32-bit, non-prefetchable) [size=256] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: Intel ICH Kernel modules: snd-intel8x0 What does not work: 1. Jack sense. Audio plays on onboard speakers (the computer has in-built speakers) but does not work with headset. 2. Recording. Only a 'Capture' control is shown in the recording tab. No other controls for recording are available. When trying to record with gnome-sound-recorder, the application hangs, which then kills pulseaudio. Cheers, Simos

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[alsa-devel] Is there a way to initialize mixer elements?
by Ozan Çağlayan 24 Sep '09

24 Sep '09

Hi, Sometimes specific hardwares needs special tweaks (e.g. a mixer element is muted by default which avoids getting sound from HP, or the master volume's gain is very low, etc.). Do we have some kind of quirks (kernel-space/user-space) for that situations? As a real-world example, I have a user with a VIA VT1708B codec which is not able to get sound from HP unless he mutes the 'Independent HP' control: http://www.alsa-project.org/db/?f=4e7672aee15a64f67286a951e75e4168c6cd43d4 Thanks, Ozan Caglayan

2 1

[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 23 Sep '09

23 Sep '09

The following changes since commit 9f072b7b2260a588bfb5fb43418c3358104e7c13: Mark Brown (1): Merge branch 'for-2.6.32' into for-2.6.33 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Barry Song (2): ASoC: some minor changes for AD1836 and AD1938 codec drivers ASoC: Blackfin I2S: use dai state rather than local counter Chaithrika U S (1): ASoC: DaVinci: Correct McASP FIFO initialization Cliff Cai (1): ASoC: Blackfin: fix inverted handling of SPORT0 on PORT F/G Mark Brown (9): ASoC: Update WM8711 to driver model registration method Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into upstream/wm8711 ASoC: Factor out WM8711 cache I/O ASoC: Add SPI support to WM8711 ASoC: WM8711 minor cleanups ASoC: Add TLV information to WM8711 ASoC: Convert soc-cache to use C99 style initialisers for the table Merge branch 'upstream/wm8711' into for-2.6.33 Merge branch 'for-2.6.32' into for-2.6.33 Mike Arthur (1): ASoC: Add WM8711 CODEC driver Phil Vandry (1): ASoC: wm8753: fix mapping when MONOMIX is set to Stereo Troy Kisky (2): ASoC: DaVinci: Fix divide by zero error during 1st execution ASoC: Davinci: Fix race with cpu_dai->dma_data jassi brar (1): ASoC: Support WM8580 based audio subsystem on SMDK64xx machines sound/soc/blackfin/bf5xx-i2s.c | 23 +- sound/soc/blackfin/bf5xx-tdm.c | 8 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad1836.c | 3 +- sound/soc/codecs/ad1938.c | 2 +- sound/soc/codecs/wm8711.c | 658 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8711.h | 42 +++ sound/soc/codecs/wm8753.c | 1 + sound/soc/davinci/davinci-i2s.c | 37 +-- sound/soc/davinci/davinci-mcasp.c | 80 ++--- sound/soc/davinci/davinci-mcasp.h | 7 +- sound/soc/davinci/davinci-pcm.c | 13 +- sound/soc/davinci/davinci-pcm.h | 1 - sound/soc/s3c24xx/Kconfig | 9 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smdk64xx_wm8580.c | 273 +++++++++++++++ sound/soc/soc-cache.c | 18 +- 18 files changed, 1068 insertions(+), 115 deletions(-) create mode 100644 sound/soc/codecs/wm8711.c create mode 100644 sound/soc/codecs/wm8711.h create mode 100644 sound/soc/s3c24xx/smdk64xx_wm8580.c

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[alsa-devel] [PATCH] ASoC: DaVinci: Correct McASP FIFO initialization
by Chaithrika U S 23 Sep '09

23 Sep '09

McASP write FIFO registers should be modified for playback and read FIFO registers for capture. Check the PCM mode before manipulating the FIFO registers. Currently, irrespective of playback/capture both the FIFOs are enabled or disbaled. This resulted in errors in audio loopback mode. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to ALSA GIT tree on branch topic/asoc at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; h=topic/asoc sound/soc/davinci/davinci-mcasp.c | 36 ++++++++++++++++++------------------ 1 files changed, 18 insertions(+), 18 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 7a06c0a..613fd98 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -386,17 +386,17 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (dev->txnumevt) /* enable FIFO */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + FIFO_ENABLE); mcasp_start_tx(dev); - else + } else { + if (dev->rxnumevt) /* enable FIFO */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + FIFO_ENABLE); mcasp_start_rx(dev); - - /* enable FIFO */ - if (dev->txnumevt) - mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); - - if (dev->rxnumevt) - mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } } static void mcasp_stop_rx(struct davinci_audio_dev *dev) @@ -413,17 +413,17 @@ static void mcasp_stop_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (dev->txnumevt) /* disable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + FIFO_ENABLE); mcasp_stop_tx(dev); - else + } else { + if (dev->rxnumevt) /* disable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + FIFO_ENABLE); mcasp_stop_rx(dev); - - /* disable FIFO */ - if (dev->txnumevt) - mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); - - if (dev->rxnumevt) - mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } } static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, -- 1.5.6

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Re: [alsa-devel] sound issues on an HP Pavilion DV3-1075 (solved)
by Liviu Andronic 23 Sep '09

23 Sep '09

On 9/23/09, Daniel Chen <seven.steps(a)gmail.com> wrote: > Yes, both of these options are necessary. What's your lspci -nv|grep -A1 0403? > debian-liv:/home/liviu# lspci -nv|grep -A1 0403 00:14.2 0403: 1002:4383 Subsystem: 103c:1506 -- 01:05.1 0403: 1002:960f Subsystem: 1002:960f Best Liviu

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[alsa-devel] [PATCH 1/2] ASoC: Blackfin I2S: use dai state rather than local counter
by Mike Frysinger 23 Sep '09

23 Sep '09

From: Barry Song <barry.song(a)analog.com> Since the active field of the dai already tells us the stream activity, the local counter variable is redundant and can be replaced. Signed-off-by: Barry Song <barry.song(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/bf5xx-i2s.c | 15 +-------------- 1 files changed, 1 insertions(+), 14 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 876abad..66e7333 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -49,7 +49,6 @@ struct bf5xx_i2s_port { u16 rcr1; u16 tcr2; u16 rcr2; - int counter; int configured; }; @@ -133,16 +132,6 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return ret; } -static int bf5xx_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - pr_debug("%s enter\n", __func__); - - /*this counter is used for counting how many pcm streams are opened*/ - bf5xx_i2s.counter++; - return 0; -} - static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -201,9 +190,8 @@ static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { pr_debug("%s enter\n", __func__); - bf5xx_i2s.counter--; /* No active stream, SPORT is allowed to be configured again. */ - if (!bf5xx_i2s.counter) + if (!dai->active) bf5xx_i2s.configured = 0; } @@ -288,7 +276,6 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai) SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = { - .startup = bf5xx_i2s_startup, .shutdown = bf5xx_i2s_shutdown, .hw_params = bf5xx_i2s_hw_params, .set_fmt = bf5xx_i2s_set_dai_fmt, -- 1.6.5.rc1

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[alsa-devel] sound issues on an HP Pavilion DV3-1075 (solved)
by Liviu Andronic 23 Sep '09

23 Sep '09

Dear ALSA devels I am writing on an ALSA-related issue that I managed to solve, on Debian testing with kernel 2.6.30 and ALSA 1.0.21. For background and exact symptoms please see these two threads. [1] [2] The first issue was that there was no sound. I solved it by manually configuring it as follows: debian-liv:/home/liviu# cat /etc/modprobe.d/sound ## ALSA portion alias char-major-116 snd alias snd-card-0 snd-hda-intel alias snd-card-1 snd-usb-audio ## module options should go here options snd-hda-intel index=0 model=hp-dv5 enable_msi=1 options snd-usb-audio index=1 The second issue concerned the external speakers/headphones: when such devices were plugged in, the internal speakers were not muted and the audio stream was not redirected to the external headset (that is, there was no sound in the headphones); although I didn't check, the microphone was probably not working, too. The issue was solved by adding position-fix=1. Thus, the following line was used options snd-hda-intel index=0 model=hp-dv5 enable-msi=1 position-fix=1 I have similar issues on Ubuntu Jackalope, and it seems that on Ubuntu it is quite often that audio cards on HP Pavilions misbehave by default (no sound, no sound in headphones, etc.). Best regards Liviu [1] http://www.mail-archive.com/debian-user@lists.debian.org/msg555984.html [2] http://www.pubbs.net/debian/200909/102780/ ### Debug messages ### liviu@debian-liv:~$ cat /proc/asound/card*/codec* | grep Codec Codec: IDT 92HD71B7X Codec: ATI RS690/780 HDMI liviu@debian-liv:~$ cat /proc/asound/cards 0 [SB ]: HDA-Intel - HDA ATI SB HDA ATI SB at 0xd2400000 irq 16 1 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xd2310000 irq 29 liviu@debian-liv:~$ cat /proc/asound/devices 0: [ 0] : control 1: : sequencer 4: [ 0- 0]: hardware dependent 16: [ 0- 0]: digital audio playback 17: [ 0- 1]: digital audio playback 24: [ 0- 0]: digital audio capture 32: [ 1] : control 33: : timer 36: [ 1- 0]: hardware dependent 51: [ 1- 3]: digital audio playback liviu@debian-liv:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 1: STAC92xx Digital [STAC92xx Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 liviu@debian-liv:~$ aplay -L default:CARD=SB HDA ATI SB, STAC92xx Analog Default Audio Device front:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Analog Front speakers surround40:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=SB,DEV=0 HDA ATI SB, STAC92xx Digital IEC958 (S/PDIF) Digital Audio Output null Discard all samples (playback) or generate zero samples (capture) hdmi:CARD=HDMI HDA ATI HDMI, ATI HDMI HDMI Audio Output -Computer- Processor : 2x AMD Turion(tm) X2 Dual-Core Mobile RM-72 Memory : 3802MB (907MB used) Operating System : Debian GNU/Linux squeeze/sid User Name : liviu (Liviu) Date/Time : Tue 22 Sep 2009 12:58:23 BST -Display- Resolution : 1280x800 pixels OpenGL Renderer : ATI Radeon HD 3200 Graphics X11 Vendor : The X.Org Foundation -Multimedia- Audio Adapter : HDA-Intel - HDA ATI SB Audio Adapter : HDA-Intel - HDA ATI HDMI -Input Devices- Macintosh mouse button emulation AT Translated Set 2 keyboard Video Bus Power Button Power Button Lid Switch PC Speaker ST LIS3LV02DL Accelerometer HP Webcam SynPS/2 Synaptics TouchPad HDA Digital PCBeep HDA ATI SB Mic at Ext Front Jack HDA ATI SB HP Out at Ext Front Jack -Printers (CUPS)- psc_1200_first -Version- Kernel : Linux 2.6.30-1-amd64 (x86_64) Compiled : #1 SMP Sat Aug 15 18:09:19 UTC 2009 C Library : GNU C Library version 2.9 (stable) Default C Compiler : GNU C Compiler version 4.3.4 (Debian 4.3.4-2) Distribution : Debian GNU/Linux squeeze/sid -Current Session- Computer Name : debian-liv User Name : liviu (Liviu) Home Directory : /home/liviu Desktop Environment : XFCE 4 -Misc- Uptime : 2 hours, 19 minutes Load Average : 0.57, 0.67, 0.73 liviu@debian-liv:~$ lspci -vv 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) Subsystem: Hewlett-Packard Company Device 1506 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 Interrupt: pin ? routed to IRQ 16 Region 0: Memory at d2400000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel [..] 01:05.1 Audio device: ATI Technologies Inc RS780 Azalia controller Subsystem: ATI Technologies Inc RS780 Azalia controller Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin B routed to IRQ 19 Region 0: Memory at d2310000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel liviu@debian-liv:~$ lsmod | grep -i snd snd_hda_codec_atihdmi 3952 1 snd_hda_codec_idt 62480 1 snd_hda_intel 26680 9 snd_hda_codec 75248 3 snd_hda_codec_atihdmi,snd_hda_codec_idt,snd_hda_intel snd_hwdep 8152 1 snd_hda_codec snd_pcm_oss 37200 0 snd_mixer_oss 15072 2 snd_pcm_oss snd_pcm 78504 4 snd_hda_intel,snd_hda_codec,snd_pcm_oss snd_seq_midi 6976 0 snd_rawmidi 23008 1 snd_seq_midi snd_seq_midi_event 7712 1 snd_seq_midi snd_seq 51392 2 snd_seq_midi,snd_seq_midi_event snd_timer 21824 3 snd_pcm,snd_seq snd_seq_device 7476 3 snd_seq_midi,snd_rawmidi,snd_seq snd 63912 25 snd_hda_codec_idt,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 7984 2 snd snd_page_alloc 10000 2 snd_hda_intel,snd_pcm -- Do you know how to read? http://www.alienetworks.com/srtest.cfm Do you know how to write? http://garbl.home.comcast.net/~garbl/stylemanual/e.htm#e-mail -- Do you know how to read? http://www.alienetworks.com/srtest.cfm Do you know how to write? http://garbl.home.comcast.net/~garbl/stylemanual/e.htm#e-mail

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[alsa-devel] [PATCH 1/4] ASoC: ALSA ASoC CQ0093 Voice Codec Driver
by miguel.aguilar＠ridgerun.com 22 Sep '09

22 Sep '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This patch adds support for the Voice Codec CQ0093. This driver was tested on a DM355 EVM rev C. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cq93vc.c | 377 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cq93vc.h | 23 +++ 4 files changed, 406 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cq93vc.c create mode 100644 sound/soc/codecs/cq93vc.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0edca93..55d1287 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C + select SND_SOC_CQ0093VC if ARCH_DAVINCI_DM365 select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -96,6 +97,9 @@ config SND_SOC_AK4535 config SND_SOC_AK4642 tristate +config SND_SOC_CQ0093VC + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fb4af28..a3e713c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -6,6 +6,7 @@ snd-soc-ad73311-objs := ad73311.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o +snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -56,6 +57,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o +obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c new file mode 100644 index 0000000..e64af25 --- /dev/null +++ b/sound/soc/codecs/cq93vc.c @@ -0,0 +1,377 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver + * + * Copyright (C) 2009 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * Initial code: Hui Geng + * + * 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. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/device.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include <mach/dm365.h> + +#include "cq93vc.h" + +#define AUDIO_NAME "cq0093" +#define CQ93VC_VERSION "0.1" + +/* + * Voice codec registers + * + * The base address for the registers used by this driver is shifted + * 0x80 from the original Voice Codec base adress since the first region + * of the Voice Codec registers is used by the Voice Codec Interface driver. + */ +#define VC_REG05 0x14 /* PGA and MIC Gain Control */ +#define VC_REG09 0x24 /* Digital Soft Mute/Attenuation Control */ +#define VC_REG12 0x30 /* Voice Codec Up/Down */ + +/* Registers bits */ +#define PGA_GAIN 0x07 +#define DIG_ATTEN 0x3F +#define MUTE 0x40 +#define POWER_ALL_ON 0xFD + +/* Codec private data */ +struct cq93vc_priv { + void __iomem *base; + unsigned int sysclk; + resource_size_t pbase; + size_t base_size; +}; + +/* + * Read to the cq0093 register space + */ +static inline unsigned int cq93vc_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct cq93vc_priv *cq93vc = codec->private_data; + + return __raw_readb(cq93vc->base + reg); +} + +/* + * Write to the cq0093 register space + */ +static int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct cq93vc_priv *cq93vc = codec->private_data; + + __raw_writeb(value, cq93vc->base + reg); + return 0; +} + +static int cq93vc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + return 0; +} + +static const struct snd_kcontrol_new cq93vc_snd_controls[] = { + SOC_SINGLE("PGA Capture Volume", VC_REG05, 0, 0x03, 0), + SOC_SINGLE("Mono DAC Playback Volume", VC_REG09, 0, 0x3f, 0), +}; + +static int cq93vc_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg = cq93vc_read(codec, VC_REG09) & ~MUTE; + + if (mute) + cq93vc_write(codec, VC_REG09, reg | MUTE); + else + cq93vc_write(codec, VC_REG09, reg); + + return 0; +} + +static int cq93vc_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(cq93vc_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&cq93vc_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +static int cq93vc_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cq93vc_priv *cq93vc = codec->private_data; + + switch (freq) { + case 22579200: + case 27000000: + case 33868800: + cq93vc->sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int cq93vc_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + /* all power is driven by DAPM system */ + cq93vc_write(codec, VC_REG12, POWER_ALL_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* + * all power is driven by DAPM system, + * so output power is safe if bypass was set + */ + cq93vc_write(codec, VC_REG12, 0x0); + break; + case SND_SOC_BIAS_OFF: + /* force all power off */ + cq93vc_write(codec, VC_REG12, 0x0); + break; + } + codec->bias_level = level; + + return 0; +} + +#define CQ93VC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) +#define CQ93VC_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_ops cq93vc_dai_ops = { + .hw_params = cq93vc_hw_params, + .digital_mute = cq93vc_mute, + .set_sysclk = cq93vc_set_dai_sysclk, +}; + +struct snd_soc_dai cq93vc_dai = { + .name = "cq93vc", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .ops = &cq93vc_dai_ops, +}; +EXPORT_SYMBOL_GPL(cq93vc_dai); + +static int cq93vc_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; + + cq93vc_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cq93vc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + cq93vc_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static struct snd_soc_device *cq93vc_socdev; + +static int cq93vc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct device *dev = &pdev->dev; + struct cq93vc_priv *cq93vc; + struct snd_soc_codec *codec; + struct resource *res, *mem; + int ret; + + dev_info(dev, "CQ0093 Voice Codec %s\n", CQ93VC_VERSION); + + cq93vc = kzalloc(sizeof(struct cq93vc_priv), GFP_KERNEL); + if (cq93vc == NULL) { + dev_dbg(dev, "%s: could not allocate memory for private data\n", + pdev->name); + return -ENOMEM; + } + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) { + dev_dbg(dev, "%s: could not allocate memory for codec data\n", + pdev->name); + ret = -ENOMEM; + goto fail1; + } + + res = cq93vc_resources; + + cq93vc->pbase = res->start; + cq93vc->base_size = resource_size(res); + + mem = request_mem_region(cq93vc->pbase, + cq93vc->base_size, "cq93vc"); + if (!mem) { + dev_dbg(dev,"%s: voice codec registers at %08x are not free\n", + pdev->name, cq93vc->pbase); + ret = -EBUSY; + goto fail2; + } + + cq93vc->base = ioremap(cq93vc->pbase, cq93vc->base_size); + if (cq93vc->base == NULL) { + dev_dbg(dev,"%s: can't ioremap mem resource.\n", + pdev->name); + ret = -ENOMEM; + goto fail3; + } + + mutex_init(&codec->mutex); + codec->private_data = cq93vc; + codec->name = "cq93vc"; + codec->owner = THIS_MODULE; + codec->read = cq93vc_read; + codec->write = cq93vc_write; + codec->set_bias_level = cq93vc_set_bias_level; + codec->dai = &cq93vc_dai; + codec->num_dai = 1; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + socdev->card->codec = codec; + + /* Register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(dev, "%s: failed to create pcms\n", pdev->name); + goto fail4; + } + + /* Set the PGA Gain to 18 dB */ + cq93vc_write(codec, VC_REG05, PGA_GAIN); + + /* Set the DAC digital attenuation to 0 dB */ + cq93vc_write(codec, VC_REG09, DIG_ATTEN); + + /* Set controls */ + cq93vc_add_controls(codec); + + /* Off, with power on */ + cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Register sound card */ + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(dev, "%s: failed to register card\n", pdev->name); + goto fail5; + } + + cq93vc_socdev = socdev; + + return 0; +fail5: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +fail4: + iounmap(cq93vc->base); +fail3: + release_mem_region(cq93vc->pbase, cq93vc->base_size); +fail2: + kfree(codec); +fail1: + kfree(cq93vc); + + return ret; +} + +static int cq93vc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + struct cq93vc_priv *cq93vc = codec->private_data; + + /* Power down chip */ + if (codec->control_data) + cq93vc_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + iounmap(cq93vc->base); + release_mem_region(cq93vc->pbase, cq93vc->base_size); + + kfree(codec); + kfree(cq93vc); + + cq93vc_socdev = NULL; + + snd_soc_unregister_dai(&cq93vc_dai); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_cq93vc = { + .probe = cq93vc_probe, + .remove = cq93vc_remove, + .suspend = cq93vc_suspend, + .resume = cq93vc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_cq93vc); + +static __init int cq93vc_init(void) +{ + return snd_soc_register_dai(&cq93vc_dai); +} +module_init(cq93vc_init); + +static __exit void cq93vc_exit(void) +{ + snd_soc_unregister_dai(&cq93vc_dai); +} +module_exit(cq93vc_exit); + +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC CQ0093 Voice Codec Driver"); +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cq93vc.h b/sound/soc/codecs/cq93vc.h new file mode 100644 index 0000000..8d37763 --- /dev/null +++ b/sound/soc/codecs/cq93vc.h @@ -0,0 +1,23 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver + * + * Copyright (C) 2009 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * Initial code: Hui Geng + * + * 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. + */ + +#ifndef _CQ93VC_H +#define _CQ93VC_H + +extern struct snd_soc_dai cq93vc_dai; +extern struct snd_soc_codec_device soc_codec_dev_cq93vc; +extern struct resource cq93vc_resources[]; + +#endif -- 1.6.0.4

3 3

[alsa-devel] [PATCH 4/4] DaVinci: DM365: Add voice codec support for DM365 EVM
by miguel.aguilar＠ridgerun.com 22 Sep '09

22 Sep '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This patch enables the voice codec support for the DM365 EVM. For the DM365 EVM only one codec is able to be enabled at time, external (AIC3101) or internal (Voice Codec CQ0093). Either internal or external codec can be selected in the configuration menu. This patch was created based on the temp/asoc branch of the DaVinci git and it assumes that the patch "[PATCH v4 2/2] ASoC: Davinci: Add audio codec support for DM365 EVM" is applied. This patch was tested on the DM365 EVM rev C. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/board-dm365-evm.c | 4 +++ sound/soc/davinci/davinci-evm.c | 42 +++++++++++++++++++++++++++++- 2 files changed, 44 insertions(+), 2 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c index 77f815c..083b8d7 100644 --- a/arch/arm/mach-davinci/board-dm365-evm.c +++ b/arch/arm/mach-davinci/board-dm365-evm.c @@ -482,7 +482,11 @@ static __init void dm365_evm_init(void) /* maybe setup mmc1/etc ... _after_ mmc0 */ evm_init_cpld(); +#if defined(CONFIG_SND_DM365_EXTERNAL_CODEC) dm365_init_asp(&dm365_evm_snd_data); +#elif defined(CONFIG_SND_DM365_INTERNAL_CODEC) + dm365_init_vcif(&dm365_evm_snd_data); +#endif } static __init void dm365_evm_irq_init(void) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 7ccbe66..e826a80 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -28,9 +28,11 @@ #include <mach/mux.h> #include "../codecs/tlv320aic3x.h" +#include "../codecs/cq93vc.h" #include "../codecs/spdif_transciever.h" #include "davinci-pcm.h" #include "davinci-i2s.h" +#include "davinci-vcif.h" #include "davinci-mcasp.h" #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ @@ -151,6 +153,15 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +#ifdef CONFIG_SND_DM365_INTERNAL_CODEC +static struct snd_soc_dai_link cq93vc_evm_dai = { + .name = "CQ93VC", + .stream_name = "CQ93", + .cpu_dai = &davinci_vcif_dai, + .codec_dai = &cq93vc_dai, +}; +#endif + static struct snd_soc_dai_link dm6467_evm_dai[] = { { .name = "TLV320AIC3X", @@ -177,7 +188,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .ops = &evm_ops, }; -/* davinci dm6446, dm355 or dm365 evm audio machine driver */ +/* davinci dm6446, dm355 evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", .platform = &davinci_soc_platform, @@ -185,6 +196,19 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm365 evm audio machine driver external or internal */ +static struct snd_soc_card dm365_snd_soc_card_evm = { +#if defined(CONFIG_SND_DM365_EXTERNAL_CODEC) + .name = "DaVinci DM365 EVM", + .dai_link = &evm_dai, +#elif defined(CONFIG_SND_DM365_INTERNAL_CODEC) + .name = "On-chip Voice Codec", + .dai_link = &cq93vc_evm_dai, +#endif + .platform = &davinci_soc_platform, + .num_links = 1, +}; + /* davinci dm6467 evm audio machine driver */ static struct snd_soc_card dm6467_snd_soc_card_evm = { .name = "DaVinci DM6467 EVM", @@ -217,6 +241,17 @@ static struct snd_soc_device evm_snd_devdata = { }; /* evm audio subsystem */ +static struct snd_soc_device dm365_evm_snd_devdata = { + .card = &dm365_snd_soc_card_evm, +#if defined(CONFIG_SND_DM365_EXTERNAL_CODEC) + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &aic3x_setup, +#elif defined(CONFIG_SND_DM365_INTERNAL_CODEC) + .codec_dev = &soc_codec_dev_cq93vc, +#endif +}; + +/* evm audio subsystem */ static struct snd_soc_device dm6467_evm_snd_devdata = { .card = &dm6467_snd_soc_card_evm, .codec_dev = &soc_codec_dev_aic3x, @@ -244,12 +279,15 @@ static int __init evm_init(void) int index; int ret; - if (machine_is_davinci_evm() || machine_is_davinci_dm365_evm()) { + if (machine_is_davinci_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 0; } else if (machine_is_davinci_dm355_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 1; + } else if (machine_is_davinci_dm365_evm()) { + evm_snd_dev_data = &dm365_evm_snd_devdata; + index = 0; } else if (machine_is_davinci_dm6467_evm()) { evm_snd_dev_data = &dm6467_evm_snd_devdata; index = 0; -- 1.6.0.4

2 1

[alsa-devel] [PATCH 3/4] DaVinci: DM365: Add resources for voice codec and platform device for VCIF
by miguel.aguilar＠ridgerun.com 22 Sep '09

22 Sep '09

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> 1) Add the resources needed by the CQ0093 voice codec. 2) Add platform device for the voice codec interface. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/dm365.c | 46 ++++++++++++++++++++++++++++ arch/arm/mach-davinci/include/mach/dm365.h | 5 +++ arch/arm/mach-davinci/include/mach/mux.h | 2 + 3 files changed, 53 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c index 2674438..80dc35b 100644 --- a/arch/arm/mach-davinci/dm365.c +++ b/arch/arm/mach-davinci/dm365.c @@ -605,6 +605,8 @@ INT_CFG(DM365, INT_IMX1_DISABLE, 24, 1, 0, false) INT_CFG(DM365, INT_NSF_ENABLE, 25, 1, 1, false) INT_CFG(DM365, INT_NSF_DISABLE, 25, 1, 0, false) +EVT_CFG(DM365, EVT2_VC_TX, 0, 1, 1, false) +EVT_CFG(DM365, EVT3_VC_RX, 1, 1, 1, false) EVT_CFG(DM365, EVT2_ASP_TX, 0, 1, 0, false) EVT_CFG(DM365, EVT3_ASP_RX, 1, 1, 0, false) #endif @@ -835,6 +837,42 @@ static struct platform_device dm365_asp_device = { .resource = dm365_asp_resources, }; +static struct resource dm365_vcif_resources[] = { + /* The first are of Voice Codec registers before 0x80 belongs to VCIF */ + { + .start = DAVINCI_DM365_VC_BASE, + .end = DAVINCI_DM365_VC_BASE + (SZ_1K>>4) - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_VC_TX, + .end = DAVINCI_DMA_VC_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_VC_RX, + .end = DAVINCI_DMA_VC_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm365_vcif_device = { + .name = "voice_codec", + .id = -1, + .num_resources = ARRAY_SIZE(dm365_vcif_resources), + .resource = dm365_vcif_resources, +}; + +struct resource cq93vc_resources[] = { + /* The base address for the Voice Codec registers is shifted by 0x80 */ + { + .start = DAVINCI_DM365_VC_BASE + 0x80, + .end = DAVINCI_DM365_VC_BASE + 0x80 + (SZ_1K>>3) - 1, + .flags = IORESOURCE_MEM, + }, +}; +EXPORT_SYMBOL_GPL(cq93vc_resources); + static struct map_desc dm365_io_desc[] = { { .virtual = IO_VIRT, @@ -950,6 +988,14 @@ void __init dm365_init_asp(struct snd_platform_data *pdata) platform_device_register(&dm365_asp_device); } +void __init dm365_init_vcif(struct snd_platform_data *pdata) +{ + davinci_cfg_reg(DM365_EVT2_VC_TX); + davinci_cfg_reg(DM365_EVT3_VC_RX); + dm365_vcif_device.dev.platform_data = pdata; + platform_device_register(&dm365_vcif_device); +} + void __init dm365_init(void) { davinci_common_init(&davinci_soc_info_dm365); diff --git a/arch/arm/mach-davinci/include/mach/dm365.h b/arch/arm/mach-davinci/include/mach/dm365.h index 2291c0d..b78ec55 100644 --- a/arch/arm/mach-davinci/include/mach/dm365.h +++ b/arch/arm/mach-davinci/include/mach/dm365.h @@ -25,7 +25,12 @@ #define DM365_EMAC_MDIO_OFFSET (0x4000) #define DM365_EMAC_CNTRL_RAM_SIZE (0x2000) +#define DAVINCI_DM365_VC_BASE (0x01D0C000) +#define DAVINCI_DMA_VC_TX 2 +#define DAVINCI_DMA_VC_RX 3 + void __init dm365_init(void); void __init dm365_init_asp(struct snd_platform_data *pdata); +void __init dm365_init_vcif(struct snd_platform_data *pdata); #endif /* __ASM_ARCH_DM365_H */ diff --git a/arch/arm/mach-davinci/include/mach/mux.h b/arch/arm/mach-davinci/include/mach/mux.h index 7732832..b01b5c4 100644 --- a/arch/arm/mach-davinci/include/mach/mux.h +++ b/arch/arm/mach-davinci/include/mach/mux.h @@ -311,6 +311,8 @@ enum davinci_dm365_index { DM365_INT_NSF_DISABLE, /* EDMA event muxing */ + DM365_EVT2_VC_TX, + DM365_EVT3_VC_RX, DM365_EVT2_ASP_TX, DM365_EVT3_ASP_RX, DM365_EVT26_MMC0_RX, -- 1.6.0.4

2 1

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 22 Sep '09

22 Sep '09

The following changes since commit 0c31cf3e4af79ea18bbd365b07ef0de207673894: Chaithrika U S (1): ASoC: DaVinci: Fixes to McASP configuration are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (1): ASoC: some minor changes for AD1836 and AD1938 codec drivers Phil Vandry (1): ASoC: wm8753: fix mapping when MONOMIX is set to Stereo sound/soc/codecs/ad1836.c | 3 +-- sound/soc/codecs/ad1938.c | 2 +- sound/soc/codecs/wm8753.c | 1 + 3 files changed, 3 insertions(+), 3 deletions(-)

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[alsa-devel] [PATCH 0/4] Add Voice Codec support for DM365
by Miguel Aguilar 22 Sep '09

22 Sep '09

Hi, The following series of patches adds the following support: 1) CQ0093 Voice Codec driver. 2) Voice Codec Interface. 3) Voice Codec resources and Voice Codec Interface platform device. 4) Voice Codec on the DM365 EVM. Notes: The patch series was created based on the current temp/asoc branch of the DaVinci git. It assumes that the patch "[PATCH v4 2/2] ASoC: Davinci: Add audio codec support for DM365 EVM" is applied. This patch series was tested on a DM365 EVM rev C. Regards, -- Miguel Angel Aguilar Ulloa Embedded Software Engineer RidgeRun Embedded Solutions miguel.aguilar(a)ridgerun.com

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[alsa-devel] (AT91RM2000 - ARM) Driver that puts data over SSC to FPGA
by Michel Müller 22 Sep '09

22 Sep '09

Hello, I am trying to write an ALSA-driver that puts the audio data over the SSC channel to the FPGA. This FPGA routes the Data to the AD/DA chips. I tried to figure out how I can do this with the ASoC Interfaces but I am currently stuck. Currently there is an Atmel SSC and PCM Interface that I can probably use. If I want to use the ASoC API I need to have a Codec Driver. But I don't have direct access to the codec. How should I proceed? I have 2 possibilities I think: Writing a Codec Driver / Writing an Alsa Driver as described in the guide of Takashi Iwai. If I write a character device that transfers data to the FPGA over the SSC I can hear sound, but I want to use ALSA because the use of audio applications will be easy and flexible. Thank you in advance! Regards, Michel

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Re: [alsa-devel] Application and hw mixing for VoIP (modified)
by Takashi Iwai 22 Sep '09

22 Sep '09

At Mon, 21 Sep 2009 15:48:39 +0200, Fabien COMTE wrote: > > Hi, > I need some help please. > I have an application that does VoIP and alsa interface. > > I want to : > - send through VoIP application a mixed sound from mic and an another > application (aplay should be ok for testing). > > mic ---------- soundcard 0 -------------> > mix ---- alsa interface -----> > voip application > application 1 --- alsa interface ------> > (aplay) > > - use sound received from VoIP with HP output and an another application > (arecord should be ok for testing). > > ----- alsa interface > ------> application 2 (arecord) > voip application ----- alsa interface ----> share > ------ soundcard 0 > --------> HP output > > > And a little question : > How to define an application interface in asoundrc ? Well, this isn't so trivial as it sounds. If the application 1 (aplay) needs to be output actually, it means to split the signal to two interfaces, one to voip program, and another to the output channel. If app1 doesn't have to be heard, then you can use a loopback device (snd-aloop driver included in alsa-driver tree) and bind with the mic device using multi plugin. The latter case, split from voip to app2 and HP output, can be also done via multi plugin + aloop device. But, this kind of infrastructure could be better done with a proper daemon, such as JACK, I guess... Takashi

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