April 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 10/11] ASoC: S3C64XX: IISv4: Add CPU driver
by Jassi Brar 10 May '10

10 May '10

Add the CPU driver for the IISv4 block found on S3C6410. For now, the driver is almost a copy of s3c64xx-i2s.c but it should diverge as more IISv4 specific stuff is added. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 5 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/s3c64xx-i2s-v4.c | 209 ++++++++++++++++++++++++++++++++++++ sound/soc/s3c24xx/s3c64xx-i2s.h | 1 + 4 files changed, 217 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/s3c64xx-i2s-v4.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 15fe57e..c0c7edf 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -24,6 +24,11 @@ config SND_S3C64XX_SOC_I2S select SND_S3C_I2SV2_SOC select S3C64XX_DMA +config SND_S3C64XX_SOC_I2S_V4 + tristate + select SND_S3C_I2SV2_SOC + select S3C64XX_DMA + config SND_S3C_SOC_PCM tristate diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index df071a3..81d8dc5 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -4,6 +4,7 @@ snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o snd-soc-s3c-ac97-objs := s3c-ac97.o +snd-soc-s3c64xx-i2s-v4-objs := s3c64xx-i2s-v4.o snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o snd-soc-s3c-pcm-objs := s3c-pcm.o @@ -12,6 +13,7 @@ obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_S3C_SOC_AC97) += snd-soc-s3c-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o +obj-$(CONFIG_SND_S3C64XX_SOC_I2S_V4) += snd-soc-s3c64xx-i2s-v4.o obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o obj-$(CONFIG_SND_S3C_SOC_PCM) += snd-soc-s3c-pcm.o diff --git a/sound/soc/s3c24xx/s3c64xx-i2s-v4.c b/sound/soc/s3c24xx/s3c64xx-i2s-v4.c new file mode 100644 index 0000000..06db130 --- /dev/null +++ b/sound/soc/s3c24xx/s3c64xx-i2s-v4.c @@ -0,0 +1,209 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s-v4.c + * + * ALSA SoC Audio Layer - S3C64XX I2Sv4 driver + * Copyright (c) 2010 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.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. + */ + +#include <linux/clk.h> +#include <linux/gpio.h> +#include <linux/io.h> + +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include <mach/gpio-bank-c.h> +#include <mach/gpio-bank-h.h> +#include <plat/gpio-cfg.h> + +#include <mach/map.h> +#include <mach/dma.h> + +#include "s3c-dma.h" +#include "regs-i2s-v2.h" +#include "s3c64xx-i2s.h" + +static struct s3c2410_dma_client s3c64xx_dma_client_out = { + .name = "I2Sv4 PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c64xx_dma_client_in = { + .name = "I2Sv4 PCM Stereo in" +}; + +static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_out; +static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_in; +static struct s3c_i2sv2_info s3c64xx_i2sv4; + +struct snd_soc_dai s3c64xx_i2s_v4_dai; +EXPORT_SYMBOL_GPL(s3c64xx_i2s_v4_dai); + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static int s3c64xx_i2sv4_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + /* configure GPIO for i2s port */ + s3c_gpio_cfgpin(S3C64XX_GPC(4), S3C64XX_GPC4_I2S_V40_DO0); + s3c_gpio_cfgpin(S3C64XX_GPC(5), S3C64XX_GPC5_I2S_V40_DO1); + s3c_gpio_cfgpin(S3C64XX_GPC(7), S3C64XX_GPC7_I2S_V40_DO2); + s3c_gpio_cfgpin(S3C64XX_GPH(6), S3C64XX_GPH6_I2S_V40_BCLK); + s3c_gpio_cfgpin(S3C64XX_GPH(7), S3C64XX_GPH7_I2S_V40_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPH(8), S3C64XX_GPH8_I2S_V40_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPH(9), S3C64XX_GPH9_I2S_V40_DI); + + return 0; +} + +static int s3c_i2sv4_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + struct s3c_dma_params *dma_data; + u32 iismod; + + dev_dbg(cpu_dai->dev, "Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = i2s->dma_playback; + else + dma_data = i2s->dma_capture; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + + iismod = readl(i2s->regs + S3C2412_IISMOD); + dev_dbg(cpu_dai->dev, "%s: r: IISMOD: %x\n", __func__, iismod); + + iismod &= ~S3C64XX_IISMOD_BLC_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod |= S3C64XX_IISMOD_BLC_8BIT; + break; + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S24_LE: + iismod |= S3C64XX_IISMOD_BLC_24BIT; + break; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + dev_dbg(cpu_dai->dev, "%s: w: IISMOD: %x\n", __func__, iismod); + + return 0; +} + +static struct snd_soc_dai_ops s3c64xx_i2sv4_dai_ops = { + .hw_params = s3c_i2sv4_hw_params, +}; + +static __devinit int s3c64xx_i2sv4_dev_probe(struct platform_device *pdev) +{ + struct s3c_i2sv2_info *i2s; + struct snd_soc_dai *dai; + int ret; + + i2s = &s3c64xx_i2sv4; + dai = &s3c64xx_i2s_v4_dai; + + if (dai->dev) { + dev_dbg(dai->dev, "%s: \ + I2Sv4 instance already registered!\n", __func__); + return -EBUSY; + } + + dai->dev = &pdev->dev; + dai->name = "s3c64xx-i2s-v4"; + dai->id = 0; + dai->symmetric_rates = 1; + dai->playback.channels_min = 2; + dai->playback.channels_max = 2; + dai->playback.rates = S3C64XX_I2S_RATES; + dai->playback.formats = S3C64XX_I2S_FMTS; + dai->capture.channels_min = 2; + dai->capture.channels_max = 2; + dai->capture.rates = S3C64XX_I2S_RATES; + dai->capture.formats = S3C64XX_I2S_FMTS; + dai->probe = s3c64xx_i2sv4_probe; + dai->ops = &s3c64xx_i2sv4_dai_ops; + + i2s->feature |= S3C_FEATURE_CDCLKCON; + + i2s->dma_capture = &s3c64xx_i2sv4_pcm_stereo_in; + i2s->dma_playback = &s3c64xx_i2sv4_pcm_stereo_out; + + i2s->dma_capture->channel = DMACH_HSI_I2SV40_RX; + i2s->dma_capture->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISRXD; + i2s->dma_playback->channel = DMACH_HSI_I2SV40_TX; + i2s->dma_playback->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISTXD; + + i2s->dma_capture->client = &s3c64xx_dma_client_in; + i2s->dma_capture->dma_size = 4; + i2s->dma_playback->client = &s3c64xx_dma_client_out; + i2s->dma_playback->dma_size = 4; + + i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus"); + if (IS_ERR(i2s->iis_cclk)) { + dev_err(&pdev->dev, "failed to get audio-bus\n"); + ret = PTR_ERR(i2s->iis_cclk); + goto err; + } + + clk_enable(i2s->iis_cclk); + + ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); + if (ret) + goto err_clk; + + ret = s3c_i2sv2_register_dai(dai); + if (ret != 0) + goto err_i2sv2; + + return 0; + +err_i2sv2: + /* Not implemented for I2Sv2 core yet */ +err_clk: + clk_put(i2s->iis_cclk); +err: + return ret; +} + +static __devexit int s3c64xx_i2sv4_dev_remove(struct platform_device *pdev) +{ + dev_err(&pdev->dev, "Device removal not yet supported\n"); + return 0; +} + +static struct platform_driver s3c64xx_i2sv4_driver = { + .probe = s3c64xx_i2sv4_dev_probe, + .remove = s3c64xx_i2sv4_dev_remove, + .driver = { + .name = "s3c64xx-iis-v4", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c64xx_i2sv4_init(void) +{ + return platform_driver_register(&s3c64xx_i2sv4_driver); +} +module_init(s3c64xx_i2sv4_init); + +static void __exit s3c64xx_i2sv4_exit(void) +{ + platform_driver_unregister(&s3c64xx_i2sv4_driver); +} +module_exit(s3c64xx_i2sv4_exit); + +/* Module information */ +MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); +MODULE_DESCRIPTION("S3C64XX I2Sv4 SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h index f27ed50..7a40f43 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.h +++ b/sound/soc/s3c24xx/s3c64xx-i2s.h @@ -37,5 +37,6 @@ struct clk; SNDRV_PCM_FMTBIT_S24_LE) extern struct snd_soc_dai s3c64xx_i2s_dai[]; +extern struct snd_soc_dai s3c64xx_i2s_v4_dai; #endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */ -- 1.6.2.5

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[alsa-devel] [Patches_v2] Reforming S3C I2S towards supporting I2Sv4
by Jassi Brar 07 May '10

07 May '10

I am going to resubmit pending last 11 patches of the original series http://mailman.alsa-project.org/pipermail/alsa-devel/2010-March/025920.html No changes have been made other than moving dma_data from snd_soc_dai to snd_soc_pcm_stream in the new I2Sv4 controller driver.

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[alsa-devel] [PATCH 1/2] snd-maestro3: Make hardware volume buttons an input device (rev2)
by Hans de Goede 06 May '10

06 May '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 and let userspace decide what to with them. This patch adds a Kconfig option which when enabled reports the volume buttons as keypresses using an input device. When enabled this option also gets 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/Kconfig | 10 ++++ sound/pci/maestro3.c | 116 +++++++++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 124 insertions(+), 2 deletions(-) diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 351654c..2bca3d6 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -655,6 +655,16 @@ config SND_MAESTRO3 To compile this driver as a module, choose M here: the module will be called snd-maestro3. +config SND_MAESTRO3_INPUT + bool "Enable input device for maestro3 volume buttons" + depends on SND_MAESTRO3 + depends on INPUT=y || INPUT=SND_MAESTRO3 + help + If you say Y here, you will get an input device which reports + keypresses for the volume buttons connected to the maestro3 chip. + If you say N the buttons will directly control the master volume. + It is recommended to say Y. + config SND_MIXART tristate "Digigram miXart" select SND_HWDEP diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 0ed634a..1070250 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> @@ -844,11 +845,17 @@ struct snd_m3 { struct m3_dma *substreams; spinlock_t reg_lock; - spinlock_t ac97_lock; +#ifdef CONFIG_SND_MAESTRO3_INPUT + struct input_dev *input_dev; + char phys[64]; /* physical device path */ +#else + spinlock_t ac97_lock; struct snd_kcontrol *master_switch; struct snd_kcontrol *master_volume; struct tasklet_struct hwvol_tq; +#endif + unsigned int in_suspend; #ifdef CONFIG_PM @@ -1606,7 +1613,9 @@ static void snd_m3_update_hw_volume(unsigned long private_data) { struct snd_m3 *chip = (struct snd_m3 *) private_data; int x, val; +#ifndef CONFIG_SND_MAESTRO3_INPUT unsigned long flags; +#endif /* Figure out which volume control button was pushed, based on differences from the default register @@ -1632,6 +1641,7 @@ static void snd_m3_update_hw_volume(unsigned long private_data) if (chip->in_suspend) return; +#ifndef CONFIG_SND_MAESTRO3_INPUT if (!chip->master_switch || !chip->master_volume) return; @@ -1677,6 +1687,35 @@ static void snd_m3_update_hw_volume(unsigned long private_data) break; } spin_unlock_irqrestore(&chip->ac97_lock, flags); +#else + if (!chip->input_dev) + return; + + val = 0; + 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 = KEY_MUTE; + break; + case 0xaa: + /* counters increased by 1 -> volume up */ + val = KEY_VOLUMEUP; + break; + case 0x66: + /* counters decreased by 1 -> volume down */ + val = KEY_VOLUMEDOWN; + break; + } + + if (val) { + input_report_key(chip->input_dev, val, 1); + input_sync(chip->input_dev); + input_report_key(chip->input_dev, val, 0); + input_sync(chip->input_dev); + } +#endif } static irqreturn_t snd_m3_interrupt(int irq, void *dev_id) @@ -1691,7 +1730,11 @@ static irqreturn_t snd_m3_interrupt(int irq, void *dev_id) return IRQ_NONE; if (status & HV_INT_PENDING) +#ifdef CONFIG_SND_MAESTRO3_INPUT + snd_m3_update_hw_volume((unsigned long)chip); +#else tasklet_schedule(&chip->hwvol_tq); +#endif /* * ack an assp int if its running @@ -1957,18 +2000,24 @@ static unsigned short snd_m3_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { struct snd_m3 *chip = ac97->private_data; +#ifndef CONFIG_SND_MAESTRO3_INPUT unsigned long flags; +#endif unsigned short data = 0xffff; if (snd_m3_ac97_wait(chip)) goto fail; +#ifndef CONFIG_SND_MAESTRO3_INPUT spin_lock_irqsave(&chip->ac97_lock, flags); +#endif snd_m3_outb(chip, 0x80 | (reg & 0x7f), CODEC_COMMAND); if (snd_m3_ac97_wait(chip)) goto fail_unlock; data = snd_m3_inw(chip, CODEC_DATA); fail_unlock: +#ifndef CONFIG_SND_MAESTRO3_INPUT spin_unlock_irqrestore(&chip->ac97_lock, flags); +#endif fail: return data; } @@ -1977,14 +2026,20 @@ static void snd_m3_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { struct snd_m3 *chip = ac97->private_data; +#ifndef CONFIG_SND_MAESTRO3_INPUT unsigned long flags; +#endif if (snd_m3_ac97_wait(chip)) return; +#ifndef CONFIG_SND_MAESTRO3_INPUT spin_lock_irqsave(&chip->ac97_lock, flags); +#endif snd_m3_outw(chip, val, CODEC_DATA); snd_m3_outb(chip, reg & 0x7f, CODEC_COMMAND); +#ifndef CONFIG_SND_MAESTRO3_INPUT spin_unlock_irqrestore(&chip->ac97_lock, flags); +#endif } @@ -2091,7 +2146,9 @@ static int __devinit snd_m3_mixer(struct snd_m3 *chip) { struct snd_ac97_bus *pbus; struct snd_ac97_template ac97; +#ifndef CONFIG_SND_MAESTRO3_INPUT struct snd_ctl_elem_id elem_id; +#endif int err; static struct snd_ac97_bus_ops ops = { .write = snd_m3_ac97_write, @@ -2111,6 +2168,7 @@ 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); +#ifndef CONFIG_SND_MAESTRO3_INPUT memset(&elem_id, 0, sizeof(elem_id)); elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; strcpy(elem_id.name, "Master Playback Switch"); @@ -2119,6 +2177,7 @@ static int __devinit snd_m3_mixer(struct snd_m3 *chip) 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); +#endif return 0; } @@ -2398,6 +2457,11 @@ static int snd_m3_free(struct snd_m3 *chip) struct m3_dma *s; int i; +#ifdef CONFIG_SND_MAESTRO3_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 +2588,41 @@ static int m3_resume(struct pci_dev *pci) } #endif /* CONFIG_PM */ +#ifdef CONFIG_SND_MAESTRO3_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 +2666,9 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, } spin_lock_init(&chip->reg_lock); +#ifndef CONFIG_SND_MAESTRO3_INPUT spin_lock_init(&chip->ac97_lock); +#endif switch (pci->device) { case PCI_DEVICE_ID_ESS_ALLEGRO: @@ -2650,7 +2751,9 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, snd_m3_hv_init(chip); +#ifndef CONFIG_SND_MAESTRO3_INPUT tasklet_init(&chip->hwvol_tq, snd_m3_update_hw_volume, (unsigned long)chip); +#endif if (request_irq(pci->irq, snd_m3_interrupt, IRQF_SHARED, card->driver, chip)) { @@ -2682,7 +2785,16 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, if ((err = snd_m3_pcm(chip, 0)) < 0) return err; - + +#ifdef CONFIG_SND_MAESTRO3_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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[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 05 May '10

05 May '10

The following changes since commit cb7b76961f73e4ae934d44f7b2e7ba974442f2fe: Takashi Iwai (1): Merge branch 'for-2.6.35' of git://git.kernel.org/.../lrg/asoc-2.6 into topic/asoc are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Mark Brown (2): ASoC: Remove redundant WM8960 SYSCLKSEL clkdiv option ASoC: Add WM9090 amplifier driver include/sound/wm9090.h | 28 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8960.c | 4 - sound/soc/codecs/wm8960.h | 1 - sound/soc/codecs/wm9090.c | 773 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm9090.h | 715 +++++++++++++++++++++++++++++++++++++++++ 7 files changed, 1522 insertions(+), 5 deletions(-) create mode 100644 include/sound/wm9090.h create mode 100644 sound/soc/codecs/wm9090.c create mode 100644 sound/soc/codecs/wm9090.h

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[alsa-devel] [PATCH RFC] ALSA: hda - Add quirk for Dell Inspiron 19T using a Conexant CX20582
by Anisse Astier 05 May '10

05 May '10

Add a quirk for all-in-one computer Dell Inspiron One 19 Touch to have proper HP and Mic support. Signed-off-by: Anisse Astier <anisse(a)astier.eu> --- Hi, I have a Dell 19T all-in-one computer, which has audio problems by default: - speakers works OK - no jack detection, no audio in HP - internal or external Mic not working If I specify model=ideapad, the problems go away. But Mic boost appears as an "Analog Mic Boost Capture Enum" between 0 and 40dB instead of a normal bar control in alsamixer. I also tested model=dell-vostro, laptop, and dell-laptop, and they give worse results. There's no BIOS update available from vendor ATM. I'm using kernel tree sound-2.6 on branch master. I'm attaching alsa-info.sh output with and without model=ideapad in addition to the patch. Regards, Anisse sound/pci/hda/patch_conexant.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 509a104..8446273 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2967,6 +2967,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { CXT5066_DELL_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), + SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), {} -- 1.7.0.6

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[alsa-devel] sound/core/timer.c: inconsistent lock state reported for tu->qlock
by Stefan Richter 05 May '10

05 May '10

Hi, I got the following log from a rarely used PC that I just updated from 2.6.34-rc4 to 2.6.34-rc5. I don't know though under which kernel I last used the soundcard of it; could have been 2.6.33, could have been a .34-rc. It is an x86 Mac mini with 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Kernel is vanilla except for various firewire patches, which should not matter to this issue. ================================= [ INFO: inconsistent lock state ] 2.6.34-rc5 #5 --------------------------------- inconsistent {HARDIRQ-ON-W} -> {IN-HARDIRQ-W} usage. dolphin/4003 [HC1[1]:SC0[0]:HE0:SE1] takes: (&(&tu->qlock)->rlock){?.+...}, at: [<f84ec472>] snd_timer_user_tinterrupt+0x28/0x132 [snd_timer] {HARDIRQ-ON-W} state was registered at: [<c1048de9>] __lock_acquire+0x654/0x1482 [<c1049c73>] lock_acquire+0x5c/0x73 [<c125ac3e>] _raw_spin_lock+0x25/0x34 [<f84ec370>] snd_timer_user_ccallback+0x55/0x95 [snd_timer] [<f84ecc4b>] snd_timer_notify1+0x53/0xca [snd_timer] [<f84ecdbe>] _snd_timer_stop+0xfc/0x106 [snd_timer] [<f84eceab>] snd_timer_stop+0x13/0x42 [snd_timer] [<f84ee2dc>] snd_timer_user_ioctl+0xa27/0xaf5 [snd_timer] [<c1092c2f>] vfs_ioctl+0x22/0x87 [<c10931aa>] do_vfs_ioctl+0x472/0x4ac [<c1093210>] sys_ioctl+0x2c/0x45 [<c1002850>] sysenter_do_call+0x12/0x36 irq event stamp: 2042234 hardirqs last enabled at (2042233): [<c1002883>] sysenter_exit+0xf/0x18 hardirqs last disabled at (2042234): [<c1002d67>] common_interrupt+0x27/0x34 softirqs last enabled at (2042226): [<c102cfc3>] __do_softirq+0x102/0x10a softirqs last disabled at (2042221): [<c102cff6>] do_softirq+0x2b/0x43 other info that might help us debug this: 2 locks held by dolphin/4003: #0: (snd_pcm_link_rwlock){.-....}, at: [<f8eb0a35>] snd_pcm_period_elapsed+0x2a/0xaf [snd_pcm] #1: (&(&substream->self_group.lock)->rlock){-.....}, at: [<f8eb0a3f>] snd_pcm_period_elapsed+0x34/0xaf [snd_pcm] stack backtrace: Pid: 4003, comm: dolphin Tainted: G W 2.6.34-rc5 #5 Call Trace: [<c125882d>] ? printk+0xf/0x12 [<c104761b>] print_usage_bug+0x14e/0x15a [<c10478c8>] mark_lock+0x2a1/0x4a7 [<c1048066>] ? check_usage_forwards+0x0/0xb0 [<c1048d69>] __lock_acquire+0x5d4/0x1482 [<c1049c08>] ? __lock_acquire+0x1473/0x1482 [<c1049c73>] lock_acquire+0x5c/0x73 [<f84ec472>] ? snd_timer_user_tinterrupt+0x28/0x132 [snd_timer] [<c125ac3e>] _raw_spin_lock+0x25/0x34 [<f84ec472>] ? snd_timer_user_tinterrupt+0x28/0x132 [snd_timer] [<f84ec472>] snd_timer_user_tinterrupt+0x28/0x132 [snd_timer] [<f84ec44a>] ? snd_timer_user_tinterrupt+0x0/0x132 [snd_timer] [<f84ed2e1>] snd_timer_interrupt+0x213/0x264 [snd_timer] [<c125ac45>] ? _raw_spin_lock+0x2c/0x34 [<f8eb0a7d>] snd_pcm_period_elapsed+0x72/0xaf [snd_pcm] [<f8f733bb>] azx_interrupt+0x8d/0x109 [snd_hda_intel] [<c1057fb0>] handle_IRQ_event+0x1d/0xa2 [<c105980b>] handle_fasteoi_irq+0x79/0xb2 [<c10046ff>] handle_irq+0x3b/0x48 [<c1003f3d>] do_IRQ+0x45/0x9f [<c1002d6e>] common_interrupt+0x2e/0x34 [<c1250000>] ? piix_init_one+0x1d2/0x628 -- Stefan Richter -=====-==-=- -=-- ==-== http://arcgraph.de/sr/

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[alsa-devel] timestamps
by Jonas Holmberg 04 May '10

04 May '10

How can I get the CLOCK_MONOTONIC timestamp for the first sample in the data I get from snd_pcm_readi()? /Jonas

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[alsa-devel] Playing multiple sounds at the same time
by Andreas Falkenhahn 04 May '10

04 May '10

Hi, I'm new to ALSA programming so please bear with me if the following question is all too obvious. I did some research on my question before but I didn't come across a clear answer. What I want to do is really simple: Playing multiple sounds in different formats at the same time. E.g. Sound 1: 12345 hz, 8-bit signed, mono Sound 2: 44100 hz, 16-bit unsigned, stereo Sound 3: 8000 hz, 8-bit unsigned, stereo Now I want to play sounds 1 to 3 at the same time. My question is now: Can ALSA do automatic mixing for me here or do I have to do the mixing on my own? In other words, can I simply feed all these three sound streams to ALSA or do I have to mix them into a fourth audio stream first and feeding this new fourth stream to ALSA then? Thanks, Andreas

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[alsa-devel] safe support for rewind in ALSA
by pl bossart 04 May '10

04 May '10

All, I'd like to reopen a thread on concerns with the current implementation of rewind in ALSA. This was already discussed on the mailing list last year without any progress. (see http://thread.gmane.org/gmane.linux.alsa.devel/65759/focus=65773 and http://article.gmane.org/gmane.linux.alsa.devel/66902) The problem is that snd_pcm_rewind() allows in theory to go back to the next sample to be rendered. This however isn't possible on any existing hardware. DMAs do prefetch samples ahead of time, and output interfaces have FIFOs on top of this. This isn't too bad in the PC world with HDAudio, since the prefetching is limited. However in the embedded space it has become a headache. To reduce power consumption, most of the recent embedded interfaces do embed some local internal memory (5kB for TI's McBSP3, a lot more for others), and we should provide a means for driver developers to set a bound on how much can be rewound safely when power-saving features are enabled. Note that there already are a set of definitions for FIFOs, but they are either obsolete or no one uses them (actually HDAudio makes use of this value internally in azx_dev but doesn't export it). My suggestion would be to add a 'fifo' parameter to the runtime structure (so that it's visible to the core), let the driver developer fill the value in the .open routine, and use this parameter in rewind to prevent rewinding too much. This parameter would include both hardware FIFOs and DMA buffers. Comments welcome. - Pierre

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Re: [alsa-devel] Ooops when working with USB MIDI (2.6.33.1)
by Takashi Iwai 03 May '10

03 May '10

At Fri, 9 Apr 2010 07:51:35 +0100, Tvrtko Ursulin wrote: > > On Thursday 08 Apr 2010 13:22:36 Takashi Iwai wrote: > > > Takashi, do you remember what the original problem was? > > > > Well, I have only a vague memory -- it's a similar scenario that some app > > still accessing after disconnection. The URB can't be handled after > > the disconnection is finished. > > > > I think the patch below might fix in this case. You can try it > > instead of reverting the commit above. > > > > > > thanks, > > > > Takashi > > > > --- > > diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c > > index 2c59afd..81c8d85 100644 > > --- a/sound/usb/usbmidi.c > > +++ b/sound/usb/usbmidi.c > > @@ -986,6 +986,8 @@ static void snd_usbmidi_output_drain(struct > > snd_rawmidi_substream *substream) DEFINE_WAIT(wait); > > long timeout = msecs_to_jiffies(50); > > > > + if (ep->umidi->disconnected) > > + return; > > /* > > * The substream buffer is empty, but some data might still be in the > > * currently active URBs, so we have to wait for those to complete. > > @@ -1275,6 +1277,11 @@ void snd_usbmidi_disconnect(struct list_head* p) > > snd_usbmidi_in_endpoint_delete(ep->in); > > ep->in = NULL; > > } > > + ep->active_urbs = 0; > > + if (ep->drain_urbs) { > > + ep->drain_urbs = 0; > > + wake_up(&ep->drain_wait); > > + } > > } > > del_timer_sync(&umidi->error_timer); > > } > > For the second hunk, do you think ep->out->... and so on? That would be more > in-line with code present in 2.6.33. Ah, crap. Sorry, that's just messing up. The revised (compiled but untested) patch is below. thanks, Takashi --- diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 2c59afd..9e28b20 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -986,6 +986,8 @@ static void snd_usbmidi_output_drain(struct snd_rawmidi_substream *substream) DEFINE_WAIT(wait); long timeout = msecs_to_jiffies(50); + if (ep->umidi->disconnected) + return; /* * The substream buffer is empty, but some data might still be in the * currently active URBs, so we have to wait for those to complete. @@ -1123,14 +1125,21 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi, * Frees an output endpoint. * May be called when ep hasn't been initialized completely. */ -static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint* ep) +static void snd_usbmidi_out_endpoint_clear(struct snd_usb_midi_out_endpoint *ep) { unsigned int i; for (i = 0; i < OUTPUT_URBS; ++i) - if (ep->urbs[i].urb) + if (ep->urbs[i].urb) { free_urb_and_buffer(ep->umidi, ep->urbs[i].urb, ep->max_transfer); + ep->urbs[i].urb = NULL; + } +} + +static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint *ep) +{ + snd_usbmidi_out_endpoint_clear(ep); kfree(ep); } @@ -1262,15 +1271,18 @@ void snd_usbmidi_disconnect(struct list_head* p) usb_kill_urb(ep->out->urbs[j].urb); if (umidi->usb_protocol_ops->finish_out_endpoint) umidi->usb_protocol_ops->finish_out_endpoint(ep->out); + ep->out->active_urbs = 0; + if (ep->out->drain_urbs) { + ep->out->drain_urbs = 0; + wake_up(&ep->out->drain_wait); + } } if (ep->in) for (j = 0; j < INPUT_URBS; ++j) usb_kill_urb(ep->in->urbs[j]); /* free endpoints here; later call can result in Oops */ - if (ep->out) { - snd_usbmidi_out_endpoint_delete(ep->out); - ep->out = NULL; - } + if (ep->out) + snd_usbmidi_out_endpoint_clear(ep->out); if (ep->in) { snd_usbmidi_in_endpoint_delete(ep->in); ep->in = NULL;

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