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[alsa-devel] [PATCHv2 0/5] *** SUBJECT HERE ***ASoC: tlv320dac33: FIFO caused delay reporting and fixes
by Peter Ujfalusi 26 Apr '10

26 Apr '10

Hello, Changelog: - Calculated delay instead of msleep(2) in patch 1 - ns changed to us in patch 5 (in the macro for calculation) The main feature of this series is the FIFO caused delay reporting on the DAC33 codec. The main patch is quite big, but I don't think it can be divided further. Short details of the actual implementation is in the commit message(s), and I will not copy it here. The values reported by the delay call has been verified by using one GPIO as a signal, and printout from the kernel at the same time with the delay, than based on the time scale on the scope several (>100) cases has been hand calculated and the results were compared to the kernel reported values. The difference were around (or less) the 10 samples, but during the bursts it is much harder to count, but the reported value was reasonable accurate there as well. --- Peter Ujfalusi (5): ASoC: tlv320dac33: Fix for early interrupt in FIFO Mode1 ASoC: tlv320dac33: Skip calculations in FIFO Bypass mode ASoC: tlv320dac33: Change magic numbers used in Mode7 ASoC: tlv320dac33: Calculate the interface speed during bursts ASoC: tlv320dac33: FIFO caused delay reporting sound/soc/codecs/tlv320dac33.c | 250 ++++++++++++++++++++++++++++++++++++++-- 1 files changed, 238 insertions(+), 12 deletions(-)

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[alsa-devel] [PATCH] ASoC: UDA134X: Add UDA1345 CODEC support
by Vladimir Zapolskiy 26 Apr '10

26 Apr '10

This patch adds support for Philips UDA1345 CODEC. The CODEC has only volume control, de-emphasis, mute, DC filtering and power control features. Signed-off-by: Vladimir Zapolskiy <vzapolskiy(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> --- include/sound/uda134x.h | 1 + sound/soc/codecs/uda134x.c | 13 +++++++++++++ 2 files changed, 14 insertions(+), 0 deletions(-) diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h index 475ef8b..509efb0 100644 --- a/include/sound/uda134x.h +++ b/include/sound/uda134x.h @@ -21,6 +21,7 @@ struct uda134x_platform_data { #define UDA134X_UDA1340 1 #define UDA134X_UDA1341 2 #define UDA134X_UDA1344 3 +#define UDA134X_UDA1345 4 }; #endif /* _UDA134X_H */ diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 20deaca..2f4d728 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -431,6 +431,14 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), }; +static const struct snd_kcontrol_new uda1345_snd_controls[] = { +SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1), + +SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), + +SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), +}; + static struct snd_soc_dai_ops uda134x_dai_ops = { .startup = uda134x_startup, .shutdown = uda134x_shutdown, @@ -486,6 +494,7 @@ static int uda134x_soc_probe(struct platform_device *pdev) case UDA134X_UDA1340: case UDA134X_UDA1341: case UDA134X_UDA1344: + case UDA134X_UDA1345: break; default: printk(KERN_ERR "UDA134X SoC codec: " @@ -551,6 +560,10 @@ static int uda134x_soc_probe(struct platform_device *pdev) ret = snd_soc_add_controls(codec, uda1341_snd_controls, ARRAY_SIZE(uda1341_snd_controls)); break; + case UDA134X_UDA1345: + ret = snd_soc_add_controls(codec, uda1345_snd_controls, + ARRAY_SIZE(uda1345_snd_controls)); + break; default: printk(KERN_ERR "%s unknown codec type: %d", __func__, pd->model); -- 1.7.0.4

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[alsa-devel] ASoC - Support for multiple components
by Liam Girdwood 26 Apr '10

26 Apr '10

Currently ASoC is designed around a single CODEC and a single platform DMA engine in every sound card instance. This is fine for most embedded devices but current smart phone and STB designs are starting to outgrow this architecture. I'm currently working on adding ASoC support for multiple different CODECs and Platforms (DMA, Audio Engines) into a single sound card instance. This work will allow ASoC to support N CODECs and N platforms per sound card instance and also allow better integration of audio components from GSM MODEMs, BT, FM transceivers and PCM DSPs. I've now split each ASoC component (i.e. DMA, CODEC, DAI) into driver and device structures. This now means each ASoC component is a regular kernel device and can have private device data and platform_data. It should also provide better support for open firmware and flattened device tree. I've CC'ed folks on this mail who have either contributed or maintain ASoC architecture code. Please have a look at your architecture and test if you can. I only have access to OMAP and pxa3xx hardware and as such have only tested the changes on these two architectures only. I'd appreciated anyone else testing on the other architectures too. The changes are all purely mechanical to component registration and component private data only. However, some architectures (txx9, imx, s3c) required some extra effort to use the device model as they had (to varying degrees) coupled their DMA and DAI code more tightly. The fsl platform also required extra work around the open firmware interface. Grant/Timur do we still need soc-of-simple now that all components are regular devices ? The code is in my topic/multi-component branch here :- git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git It's currently a patch for each component for easier review atm but will be rebased into a single commit when it's ready for upstream so it won't break bisect. If the testing can be done quickly then we can go for 2-6.35 otherwise we are looking at 2.6.36. Thanks Liam -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

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[alsa-devel] CA0110 bug report
by Guillaume Chevaleyre 26 Apr '10

26 Apr '10

Hello, I don't know if it's the good procedure, but anyway, maybe this could help the team. bash-4.1# lspci 02:00.0 PCI bridge: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG PCI to PCIe Bridge 03:00.0 Audio device: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG bash-4.1# modprobe snd-ctxfi bash-4.1# cat /proc/asound/modules 0 snd_hda_intel Track: http://www.alsa-project.org/db/?f=5dd8c336ee72b09e158cb32f08749a8db64715d2 Hope it could help.

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[alsa-devel] [PATCH] es968: fix wrong PnP dma index
by Krzysztof Helt 26 Apr '10

26 Apr '10

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> There is only one dma for the ESS ES968 based board. Its index is 0 and not 1. This make the es968 card working. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/sb/es968.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/isa/sb/es968.c b/sound/isa/sb/es968.c index cafc3a7..ff18286 100644 --- a/sound/isa/sb/es968.c +++ b/sound/isa/sb/es968.c @@ -93,7 +93,7 @@ static int __devinit snd_card_es968_pnp(int dev, struct snd_card_es968 *acard, return err; } port[dev] = pnp_port_start(pdev, 0); - dma8[dev] = pnp_dma(pdev, 1); + dma8[dev] = pnp_dma(pdev, 0); irq[dev] = pnp_irq(pdev, 0); return 0; -- 1.6.4

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[alsa-devel] [PATCH] ALSA: hda: Use STAC_DELL_M6_BOTH quirk for Dell Studio 1558
by Daniel Chen 23 Apr '10

23 Apr '10

Hi, Attached is a patch against sound-2.6 master HEAD making audio work by default for the OR's Dell Studio 1558. Best, -Dan

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[alsa-devel] [PATCH] ALSA: hda: Use LPIB quirk for DG965OT board version AAD63733-203
by Daniel Chen 23 Apr '10

23 Apr '10

Hi, Attached is a patch against sound-2.6 master HEAD that makes the OR's audio work by default. Best, -Dan

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[alsa-devel] [PATCH 0/5] ASoC: tlv320dac33: FIFO caused delay reporting and fixes
by Peter Ujfalusi 23 Apr '10

23 Apr '10

Hello, The main feature of this series is the FIFO caused delay reporting on the DAC33 codec. The main patch is quite big, but I don't think it can be divided further. Short details of the actual implementation is in the commit message(s), and I will not copy it here. The values reported by the delay call has been verified by using one GPIO as a signal, and printout from the kernel at the same time with the delay, than based on the time scale on the scope several (>100) cases has been hand calculated and the results were compared to the kernel reported values. The difference were around (or less) the 10 samples, but during the bursts it is much harder to count, but the reported value was reasonable accurate there as well. --- Peter Ujfalusi (5): ASoC: tlv320dac33: Fix for early interrupt in FIFO Mode1 ASoC: tlv320dac33: Skip calculations in FIFO Bypass mode ASoC: tlv320dac33: Change magic numbers used in Mode7 ASoC: tlv320dac33: Calculate the interface speed during bursts ASoC: tlv320dac33: FIFO caused delay reporting sound/soc/codecs/tlv320dac33.c | 249 ++++++++++++++++++++++++++++++++++++++-- 1 files changed, 237 insertions(+), 12 deletions(-)

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[alsa-devel] [PATCH 1/4] snd-meastro3: Add amp_gpio quirk for Compaq EVO N600C
by Hans de Goede 22 Apr '10

22 Apr '10

Without this quirk sound stops working after suspend resume. With this quirk, one still needs to manually unmute the master volume control after a suspend / / resume cycle. That is fixed in another patch in this set. Note that this patch was submitted to the alsa bug tracker a long time ago: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4319 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/pci/maestro3.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 75283fb..491a2fc 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -884,6 +884,7 @@ static struct pci_device_id snd_m3_ids[] = { MODULE_DEVICE_TABLE(pci, snd_m3_ids); static struct snd_pci_quirk m3_amp_quirk_list[] __devinitdata = { + SND_PCI_QUIRK(0x0E11, 0x0094, "Compaq Evo N600c", 0x0c), SND_PCI_QUIRK(0x10f7, 0x833e, "Panasonic CF-28", 0x0d), SND_PCI_QUIRK(0x10f7, 0x833d, "Panasonic CF-72", 0x0d), SND_PCI_QUIRK(0x1033, 0x80f1, "NEC LM800J/7", 0x03), -- 1.7.0.1

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[alsa-devel] [PATCH 1/2] snd-maestro3: Make hardware volume buttons an input device
by Hans de Goede 22 Apr '10

22 Apr '10

While working on the sound suspend / resume problems with my laptop I noticed that the hardware volume handling code in essence just detects key presses, and then does some hardcoded modification of the master volume based on which key is pressed. This made me think that clearly the right thing to do here is just report these keypresses to userspace as keypresses using an input device and let userspace decide what to with them. This patch does this, getting rid of the ugly direct ac97 writes from the tasklet, the ac97lock and the need for using a tasklet in general. As an added bonus the keys now work identical to volume keys on a (usb) keyboard with multimedia keys, providing visual feedback of the volume level change, and a better range of the volume control (with a properly configured desktop environment). Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/pci/maestro3.c | 133 +++++++++++++++++++++++++++----------------------- 1 files changed, 72 insertions(+), 61 deletions(-) diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 0ed634a..4ee1b67 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -41,6 +41,7 @@ #include <linux/vmalloc.h> #include <linux/moduleparam.h> #include <linux/firmware.h> +#include <linux/input.h> #include <sound/core.h> #include <sound/info.h> #include <sound/control.h> @@ -806,6 +807,10 @@ struct m3_dma { struct snd_m3 { struct snd_card *card; +#ifdef CONFIG_INPUT + struct input_dev *input_dev; + char phys[64]; /* physical device path */ +#endif unsigned long iobase; @@ -844,11 +849,7 @@ struct snd_m3 { struct m3_dma *substreams; spinlock_t reg_lock; - spinlock_t ac97_lock; - struct snd_kcontrol *master_switch; - struct snd_kcontrol *master_volume; - struct tasklet_struct hwvol_tq; unsigned int in_suspend; #ifdef CONFIG_PM @@ -1602,11 +1603,9 @@ static void snd_m3_update_ptr(struct snd_m3 *chip, struct m3_dma *s) (without wrap around) in response to volume button presses and then generating an interrupt. The pair of counters is stored in bits 1-3 and 5-7 of a byte wide register. The meaning of bits 0 and 4 is unknown. */ -static void snd_m3_update_hw_volume(unsigned long private_data) +static void snd_m3_update_hw_volume(struct snd_m3 *chip) { - struct snd_m3 *chip = (struct snd_m3 *) private_data; - int x, val; - unsigned long flags; + int x, key = 0; /* Figure out which volume control button was pushed, based on differences from the default register @@ -1632,51 +1631,34 @@ static void snd_m3_update_hw_volume(unsigned long private_data) if (chip->in_suspend) return; - if (!chip->master_switch || !chip->master_volume) +#ifdef CONFIG_INPUT + if (!chip->input_dev) return; - /* FIXME: we can't call snd_ac97_* functions since here is in tasklet. */ - spin_lock_irqsave(&chip->ac97_lock, flags); - - val = chip->ac97->regs[AC97_MASTER_VOL]; switch (x) { case 0x88: /* The counters have not changed, yet we've received a HV interrupt. According to tests run by various people this happens when pressing the mute button. */ - val ^= 0x8000; - chip->ac97->regs[AC97_MASTER_VOL] = val; - outw(val, chip->iobase + CODEC_DATA); - outb(AC97_MASTER_VOL, chip->iobase + CODEC_COMMAND); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, - &chip->master_switch->id); + key = KEY_MUTE; break; case 0xaa: /* counters increased by 1 -> volume up */ - if ((val & 0x7f) > 0) - val--; - if ((val & 0x7f00) > 0) - val -= 0x0100; - chip->ac97->regs[AC97_MASTER_VOL] = val; - outw(val, chip->iobase + CODEC_DATA); - outb(AC97_MASTER_VOL, chip->iobase + CODEC_COMMAND); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, - &chip->master_volume->id); + key = KEY_VOLUMEUP; break; case 0x66: /* counters decreased by 1 -> volume down */ - if ((val & 0x7f) < 0x1f) - val++; - if ((val & 0x7f00) < 0x1f00) - val += 0x0100; - chip->ac97->regs[AC97_MASTER_VOL] = val; - outw(val, chip->iobase + CODEC_DATA); - outb(AC97_MASTER_VOL, chip->iobase + CODEC_COMMAND); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, - &chip->master_volume->id); + key = KEY_VOLUMEDOWN; break; } - spin_unlock_irqrestore(&chip->ac97_lock, flags); + + if (key) { + input_report_key(chip->input_dev, key, 1); + input_sync(chip->input_dev); + input_report_key(chip->input_dev, key, 0); + input_sync(chip->input_dev); + } +#endif } static irqreturn_t snd_m3_interrupt(int irq, void *dev_id) @@ -1691,7 +1673,7 @@ static irqreturn_t snd_m3_interrupt(int irq, void *dev_id) return IRQ_NONE; if (status & HV_INT_PENDING) - tasklet_schedule(&chip->hwvol_tq); + snd_m3_update_hw_volume(chip); /* * ack an assp int if its running @@ -1957,18 +1939,14 @@ static unsigned short snd_m3_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { struct snd_m3 *chip = ac97->private_data; - unsigned long flags; unsigned short data = 0xffff; if (snd_m3_ac97_wait(chip)) goto fail; - spin_lock_irqsave(&chip->ac97_lock, flags); snd_m3_outb(chip, 0x80 | (reg & 0x7f), CODEC_COMMAND); if (snd_m3_ac97_wait(chip)) - goto fail_unlock; + goto fail; data = snd_m3_inw(chip, CODEC_DATA); -fail_unlock: - spin_unlock_irqrestore(&chip->ac97_lock, flags); fail: return data; } @@ -1977,14 +1955,11 @@ static void snd_m3_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { struct snd_m3 *chip = ac97->private_data; - unsigned long flags; if (snd_m3_ac97_wait(chip)) return; - spin_lock_irqsave(&chip->ac97_lock, flags); snd_m3_outw(chip, val, CODEC_DATA); snd_m3_outb(chip, reg & 0x7f, CODEC_COMMAND); - spin_unlock_irqrestore(&chip->ac97_lock, flags); } @@ -2091,7 +2066,6 @@ static int __devinit snd_m3_mixer(struct snd_m3 *chip) { struct snd_ac97_bus *pbus; struct snd_ac97_template ac97; - struct snd_ctl_elem_id elem_id; int err; static struct snd_ac97_bus_ops ops = { .write = snd_m3_ac97_write, @@ -2111,15 +2085,6 @@ static int __devinit snd_m3_mixer(struct snd_m3 *chip) schedule_timeout_uninterruptible(msecs_to_jiffies(100)); snd_ac97_write(chip->ac97, AC97_PCM, 0); - memset(&elem_id, 0, sizeof(elem_id)); - elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strcpy(elem_id.name, "Master Playback Switch"); - chip->master_switch = snd_ctl_find_id(chip->card, &elem_id); - memset(&elem_id, 0, sizeof(elem_id)); - elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strcpy(elem_id.name, "Master Playback Volume"); - chip->master_volume = snd_ctl_find_id(chip->card, &elem_id); - return 0; } @@ -2398,6 +2363,11 @@ static int snd_m3_free(struct snd_m3 *chip) struct m3_dma *s; int i; +#ifdef CONFIG_INPUT + if (chip->input_dev) + input_unregister_device(chip->input_dev); +#endif + if (chip->substreams) { spin_lock_irq(&chip->reg_lock); for (i = 0; i < chip->num_substreams; i++) { @@ -2524,6 +2494,41 @@ static int m3_resume(struct pci_dev *pci) } #endif /* CONFIG_PM */ +#ifdef CONFIG_INPUT +static int __devinit snd_m3_input_register(struct snd_m3 *chip) +{ + struct input_dev *input_dev; + int err; + + input_dev = input_allocate_device(); + if (!input_dev) + return -ENOMEM; + + snprintf(chip->phys, sizeof(chip->phys), "pci-%s/input0", + pci_name(chip->pci)); + + input_dev->name = chip->card->driver; + input_dev->phys = chip->phys; + input_dev->id.bustype = BUS_PCI; + input_dev->id.vendor = chip->pci->vendor; + input_dev->id.product = chip->pci->device; + input_dev->dev.parent = &chip->pci->dev; + + __set_bit(EV_KEY, input_dev->evbit); + __set_bit(KEY_MUTE, input_dev->keybit); + __set_bit(KEY_VOLUMEDOWN, input_dev->keybit); + __set_bit(KEY_VOLUMEUP, input_dev->keybit); + + err = input_register_device(input_dev); + if (err) { + input_free_device(input_dev); + return err; + } + + chip->input_dev = input_dev; + return 0; +} +#endif /* CONFIG_INPUT */ /* */ @@ -2567,7 +2572,6 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, } spin_lock_init(&chip->reg_lock); - spin_lock_init(&chip->ac97_lock); switch (pci->device) { case PCI_DEVICE_ID_ESS_ALLEGRO: @@ -2650,8 +2654,6 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, snd_m3_hv_init(chip); - tasklet_init(&chip->hwvol_tq, snd_m3_update_hw_volume, (unsigned long)chip); - if (request_irq(pci->irq, snd_m3_interrupt, IRQF_SHARED, card->driver, chip)) { snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq); @@ -2682,7 +2684,16 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, if ((err = snd_m3_pcm(chip, 0)) < 0) return err; - + +#ifdef CONFIG_INPUT + if (chip->hv_config & HV_CTRL_ENABLE) { + err = snd_m3_input_register(chip); + if (err) + snd_printk(KERN_WARNING "Input device registration " + "failed with error %i", err); + } +#endif + snd_m3_enable_ints(chip); snd_m3_assp_continue(chip); -- 1.7.0.1

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