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[alsa-devel] ALSA 1.0.19 release
by Jaroslav Kysela 19 Jan '09

19 Jan '09

Hello, I released ALSA 1.0.19 packages (except alsa-oss and alsa-python - no changes). Description of changes is available at URL bellow: http://www.alsa-project.org/main/index.php/Changes_v1.0.18_v1.0.19 Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] [PATCH] hda: fix invalid power mapping masks
by Matthew Ranostay 18 Jan '09

18 Jan '09

Fixed invalid power mappings for ports 0xd and 0xe on 93hd83xxx codecs. They were shifted right one too many bits. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index faef1ca..a4d4afe 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -334,7 +334,7 @@ static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = { }; static unsigned int stac92hd83xxx_pwr_mapping[4] = { - 0x03, 0x0c, 0x10, 0x40, + 0x03, 0x0c, 0x20, 0x80, }; static hda_nid_t stac92hd83xxx_amp_nids[1] = {

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[alsa-devel] hda_codec: Unknown model for ALC268
by Stefanie 17 Jan '09

17 Jan '09

Hello! I have problems with the sound chip of my notebook. It is a Medion akoya WIM 2220. Actually, there is a driver for the chip, but somehow, I can't get it to work. I'm desperate and don't know what to do. My friend helped my to gather the following information: This is from lspci: 00:1b.0 0403: 8086:284b (rev 04) 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 04) Subsystem: Wistron Corp. Device 4085 Flags: fast devsel, IRQ 22 Memory at fa200000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Count=1/1 Enable- Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel modules: snd-hda-intelk <?> I have found this in my syslog while trying to load snd-hda-intel: Jan 10 21:30:16 Mescalin HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 Jan 10 21:30:16 Mescalin HDA Intel 0000:00:1b.0: setting latency timer to 64 Jan 10 21:30:16 Mescalin hda_codec: Unknown model for ALC268, trying auto-probe from BIOS... Jan 10 21:30:16 Mescalin hda_codec: Cannot set up configuration from BIOS. Using base mode... Jan 10 21:30:16 Mescalin SPDIF already defined Jan 10 21:30:16 Mescalin HDA Intel 0000:00:1b.0: PCI INT A disabled Jan 10 21:30:16 Mescalin HDA Intel: probe of 0000:00:1b.0 failed with error -16 Thanks in advance for your help. Stefanie

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[alsa-devel] [Pull request] Support for wm9705 codec and two machines that use it.
by Ian Molton 17 Jan '09

17 Jan '09

The following changes since commit 6d3455ecb7d564789e8dd493fb876f4a8706e3b2: Mark Brown (1): Fix leftover merge issue. are available in the git repository at: git://git.mnementh.co.uk/linux-2.6-im.git asoc Ian Molton (3): ASoc: codec: Driver for wm9705 AC97 codec. ASoC: machine driver for Toshiba e750 ASoC: machine driver for Toshiba e800 arch/arm/mach-pxa/e750.c | 5 + arch/arm/mach-pxa/include/mach/eseries-gpio.h | 10 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm9705.c | 401 ++++++++++++++++++ sound/soc/codecs/wm9705.h | 14 + sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/e750_wm9705.c | 185 ++++++++++++ sound/soc/pxa/e800_wm9712.c | 115 +++++++- 10 files changed, 733 insertions(+), 14 deletions(-) create mode 100644 sound/soc/codecs/wm9705.c create mode 100644 sound/soc/codecs/wm9705.h create mode 100644 sound/soc/pxa/e750_wm9705.c

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[alsa-devel] [Pull request] ASoC: machine support for Toshiba e740 PDA
by Ian Molton 17 Jan '09

17 Jan '09

Hi folks, This is complete support for the wm9705 codec on the e740. Everything works. Please review. Patch attached below the pull request. Mark, if you want the pull request minus the other patchset in my tree (caching stuff) then I can redo. Otherwise, feel free to cherrypick. Thanks, -Ian The following changes since commit 7ff27c9c1f7a7b7c86d3138060b1b406ad5dd6af: Ian Molton (1): ASoC: fixes to caching implementations are available in the git repository at: git://git.mnementh.co.uk/linux-2.6-im.git asoc-pu Ian Molton (1): ASoC: machine support for Toshiba e740 PDA arch/arm/mach-pxa/e740.c | 5 + arch/arm/mach-pxa/include/mach/eseries-gpio.h | 5 + sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/e740_wm9705.c | 213 +++++++++++++++++++++++++ 5 files changed, 234 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/e740_wm9705.c From 45bf08e654a94cfc8933e57b7c275cf565783fe3 Mon Sep 17 00:00:00 2001 From: Ian Molton <ian(a)mnementh.co.uk> Date: Sat, 17 Jan 2009 15:11:06 +0000 Subject: [PATCH] ASoC: machine support for Toshiba e740 PDA This patch provides suupport for the wm9705 AC97 codec on the Toshiba e740. Note: The e740 has a hard headphone switch that turns the speaker off and is not software detectable or controlable. Also both headphone and speaker amps share a common output enable. Signed-off-by: Ian Molton <ian(a)mnementh.co.uk> --- arch/arm/mach-pxa/e740.c | 5 + arch/arm/mach-pxa/include/mach/eseries-gpio.h | 5 + sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/e740_wm9705.c | 213 +++++++++++++++++++++++++ 5 files changed, 234 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/e740_wm9705.c diff --git a/arch/arm/mach-pxa/e740.c b/arch/arm/mach-pxa/e740.c index 6d48e00..a6fff78 100644 --- a/arch/arm/mach-pxa/e740.c +++ b/arch/arm/mach-pxa/e740.c @@ -135,6 +135,11 @@ static unsigned long e740_pin_config[] __initdata = { /* IrDA */ GPIO38_GPIO | MFP_LPM_DRIVE_HIGH, + /* Audio power control */ + GPIO16_GPIO, /* AC97 codec AVDD2 supply (analogue power) */ + GPIO40_GPIO, /* Mic amp power */ + GPIO41_GPIO, /* Headphone amp power */ + /* PC Card */ GPIO8_GPIO, /* CD0 */ GPIO44_GPIO, /* CD1 */ diff --git a/arch/arm/mach-pxa/include/mach/eseries-gpio.h b/arch/arm/mach-pxa/include/mach/eseries-gpio.h index 6d6e4d8..f3e5509 100644 --- a/arch/arm/mach-pxa/include/mach/eseries-gpio.h +++ b/arch/arm/mach-pxa/include/mach/eseries-gpio.h @@ -45,6 +45,11 @@ /* e7xx IrDA power control */ #define GPIO_E7XX_IR_OFF 38 +/* e740 audio control GPIOs */ +#define GPIO_E740_WM9705_nAVDD2 16 +#define GPIO_E740_MIC_ON 40 +#define GPIO_E740_AMP_ON 41 + /* e750 audio control GPIOs */ #define GPIO_E750_HP_AMP_OFF 4 #define GPIO_E750_SPK_AMP_OFF 7 diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index b9b1a3f..958ac3f 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -61,6 +61,15 @@ config SND_PXA2XX_SOC_TOSA Say Y if you want to add support for SoC audio on Sharp Zaurus SL-C6000x models (Tosa). +config SND_PXA2XX_SOC_E740 + tristate "SoC AC97 Audio support for e740" + depends on SND_PXA2XX_SOC && MACH_E740 + select SND_SOC_WM9705 + select SND_PXA2XX_SOC_AC97 + help + Say Y if you want to add support for SoC audio on the + toshiba e740 PDA + config SND_PXA2XX_SOC_E750 tristate "SoC AC97 Audio support for e750" depends on SND_PXA2XX_SOC && MACH_E750 diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index c7d4cce..97a51a8 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -13,6 +13,7 @@ obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o snd-soc-corgi-objs := corgi.o snd-soc-poodle-objs := poodle.o snd-soc-tosa-objs := tosa.o +snd-soc-e740-objs := e740_wm9705.o snd-soc-e750-objs := e750_wm9705.o snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o @@ -23,6 +24,7 @@ snd-soc-zylonite-objs := zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o +obj-$(CONFIG_SND_PXA2XX_SOC_E740) += snd-soc-e740.o obj-$(CONFIG_SND_PXA2XX_SOC_E750) += snd-soc-e750.o obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c new file mode 100644 index 0000000..ac36176 --- /dev/null +++ b/sound/soc/pxa/e740_wm9705.c @@ -0,0 +1,213 @@ +/* + * e740-wm9705.c -- SoC audio for e740 + * + * Copyright 2007 (c) Ian Molton <spyro(a)f2s.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; version 2 ONLY. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/pxa-regs.h> +#include <mach/hardware.h> +#include <mach/audio.h> +#include <mach/eseries-gpio.h> + +#include <asm/mach-types.h> + +#include "../codecs/wm9705.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" + + +#define E740_AUDIO_OUT 1 +#define E740_AUDIO_IN 2 + +static int e740_audio_power; + +static void e740_sync_audio_power(int status) +{ + gpio_set_value(GPIO_E740_WM9705_nAVDD2, !status); + gpio_set_value(GPIO_E740_AMP_ON, (status & E740_AUDIO_OUT) ? 1 : 0); + gpio_set_value(GPIO_E740_MIC_ON, (status & E740_AUDIO_IN) ? 1 : 0); +} + +static int e740_mic_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + e740_audio_power |= E740_AUDIO_IN; + else if (event & SND_SOC_DAPM_POST_PMD) + e740_audio_power &= ~E740_AUDIO_IN; + + e740_sync_audio_power(e740_audio_power); + + return 0; +} + +static int e740_output_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + e740_audio_power |= E740_AUDIO_OUT; + else if (event & SND_SOC_DAPM_POST_PMD) + e740_audio_power &= ~E740_AUDIO_OUT; + + e740_sync_audio_power(e740_audio_power); + + return 0; +} + +static const struct snd_soc_dapm_widget e740_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic (Internal)", NULL), + SND_SOC_DAPM_PGA_E("Output Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e740_output_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("Mic Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e740_mic_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Output Amp", NULL, "LOUT"}, + {"Output Amp", NULL, "ROUT"}, + {"Output Amp", NULL, "MONOOUT"}, + + {"Speaker", NULL, "Output Amp"}, + {"Headphone Jack", NULL, "Output Amp"}, + + {"MIC1", NULL, "Mic Amp"}, + {"Mic Amp", NULL, "Mic (Internal)"}, +}; + +static int e740_ac97_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_nc_pin(codec, "HPOUTL"); + snd_soc_dapm_nc_pin(codec, "HPOUTR"); + snd_soc_dapm_nc_pin(codec, "PHONE"); + snd_soc_dapm_nc_pin(codec, "LINEINL"); + snd_soc_dapm_nc_pin(codec, "LINEINR"); + snd_soc_dapm_nc_pin(codec, "CDINL"); + snd_soc_dapm_nc_pin(codec, "CDINR"); + snd_soc_dapm_nc_pin(codec, "PCBEEP"); + snd_soc_dapm_nc_pin(codec, "MIC2"); + + snd_soc_dapm_new_controls(codec, e740_dapm_widgets, + ARRAY_SIZE(e740_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link e740_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9705_dai[WM9705_DAI_AC97_HIFI], + .init = e740_ac97_init, + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9705_dai[WM9705_DAI_AC97_AUX], + }, +}; + +static struct snd_soc_card e740 = { + .name = "Toshiba e740", + .platform = &pxa2xx_soc_platform, + .dai_link = e740_dai, + .num_links = ARRAY_SIZE(e740_dai), +}; + +static struct snd_soc_device e740_snd_devdata = { + .card = &e740, + .codec_dev = &soc_codec_dev_wm9705, +}; + +static struct platform_device *e740_snd_device; + +static int __init e740_init(void) +{ + int ret; + + if (!machine_is_e740()) + return -ENODEV; + + ret = gpio_request(GPIO_E740_MIC_ON, "Mic amp"); + if (ret) + return ret; + + ret = gpio_request(GPIO_E740_AMP_ON, "Output amp"); + if (ret) + goto free_mic_amp_gpio; + + ret = gpio_request(GPIO_E740_WM9705_nAVDD2, "Audio power"); + if (ret) + goto free_op_amp_gpio; + + /* Disable audio */ + ret = gpio_direction_output(GPIO_E740_MIC_ON, 0); + if (ret) + goto free_apwr_gpio; + ret = gpio_direction_output(GPIO_E740_AMP_ON, 0); + if (ret) + goto free_apwr_gpio; + ret = gpio_direction_output(GPIO_E740_WM9705_nAVDD2, 1); + if (ret) + goto free_apwr_gpio; + + e740_snd_device = platform_device_alloc("soc-audio", -1); + if (!e740_snd_device) { + ret = -ENOMEM; + goto free_apwr_gpio; + } + + platform_set_drvdata(e740_snd_device, &e740_snd_devdata); + e740_snd_devdata.dev = &e740_snd_device->dev; + ret = platform_device_add(e740_snd_device); + + if (!ret) + return 0; + +/* Fail gracefully */ + platform_device_put(e740_snd_device); +free_apwr_gpio: + gpio_free(GPIO_E740_WM9705_nAVDD2); +free_op_amp_gpio: + gpio_free(GPIO_E740_AMP_ON); +free_mic_amp_gpio: + gpio_free(GPIO_E740_MIC_ON); + + return ret; +} + +static void __exit e740_exit(void) +{ + platform_device_unregister(e740_snd_device); +} + +module_init(e740_init); +module_exit(e740_exit); + +/* Module information */ +MODULE_AUTHOR("Ian Molton <spyro(a)f2s.com>"); +MODULE_DESCRIPTION("ALSA SoC driver for e740"); +MODULE_LICENSE("GPL v2"); -- 1.5.6.5

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[alsa-devel] [PATCH] sscape: fix incorrect timeout after microcode upload
by Krzysztof Helt 17 Jan '09

17 Jan '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> A comment states that one should wait up to 5 secs while a waiting loop waits only 5 system ticks. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Ted tested that patch and it allowed him uploading a microcode into the Soundscape Elite. However, midi does not work for him yet. sound/isa/sscape.c | 20 ++++++++++---------- 1 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index bc44916..347a880 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -393,19 +393,19 @@ static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, unsigned ti */ static int obp_startup_ack(struct soundscape *s, unsigned timeout) { - while (timeout != 0) { + unsigned long end_time = jiffies + msecs_to_jiffies(timeout); + + while (time_before(jiffies, end_time)) { unsigned long flags; unsigned char x; - schedule_timeout_uninterruptible(1); - spin_lock_irqsave(&s->lock, flags); x = inb(HOST_DATA_IO(s->io_base)); spin_unlock_irqrestore(&s->lock, flags); if ((x & 0xfe) == 0xfe) return 1; - --timeout; + msleep(10); } /* while */ return 0; @@ -420,19 +420,19 @@ static int obp_startup_ack(struct soundscape *s, unsigned timeout) */ static int host_startup_ack(struct soundscape *s, unsigned timeout) { - while (timeout != 0) { + unsigned long end_time = jiffies + msecs_to_jiffies(timeout); + + while (time_before(jiffies, end_time)) { unsigned long flags; unsigned char x; - schedule_timeout_uninterruptible(1); - spin_lock_irqsave(&s->lock, flags); x = inb(HOST_DATA_IO(s->io_base)); spin_unlock_irqrestore(&s->lock, flags); if (x == 0xfe) return 1; - --timeout; + msleep(10); } /* while */ return 0; @@ -529,10 +529,10 @@ static int upload_dma_data(struct soundscape *s, * give it 5 seconds (max) ... */ ret = 0; - if (!obp_startup_ack(s, 5)) { + if (!obp_startup_ack(s, 5000)) { snd_printk(KERN_ERR "sscape: No response from on-board processor after upload\n"); ret = -EAGAIN; - } else if (!host_startup_ack(s, 5)) { + } else if (!host_startup_ack(s, 5000)) { snd_printk(KERN_ERR "sscape: SoundScape failed to initialise\n"); ret = -EAGAIN; } -- 1.5.2.2 ----------------------------------------------------------------------- Promocja w Speak Up. Kwartal angielskiego za darmo. 3 miesiace nauki gratis. Sprawdz teraz! >> http://link.interia.pl/f2019

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[alsa-devel] [PATCH] wss-lib: remove "pops" before each played sound
by Krzysztof Helt 17 Jan '09

17 Jan '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> A WSS codec is autocalibrated each time before playing sound. Do only one calibration during codec initialization. Complete snd_wss_calibrate_mute to mute loopback volume as well. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/wss/wss_lib.c | 41 +++++++++++++---------------------------- 1 files changed, 13 insertions(+), 28 deletions(-) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 13299ae..f0c0be5 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -181,25 +181,6 @@ static void snd_wss_wait(struct snd_wss *chip) udelay(100); } -static void snd_wss_outm(struct snd_wss *chip, unsigned char reg, - unsigned char mask, unsigned char value) -{ - unsigned char tmp = (chip->image[reg] & mask) | value; - - snd_wss_wait(chip); -#ifdef CONFIG_SND_DEBUG - if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("outm: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); -#endif - chip->image[reg] = tmp; - if (!chip->calibrate_mute) { - wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); - wmb(); - wss_outb(chip, CS4231P(REG), tmp); - mb(); - } -} - static void snd_wss_dout(struct snd_wss *chip, unsigned char reg, unsigned char value) { @@ -587,7 +568,15 @@ static void snd_wss_calibrate_mute(struct snd_wss *chip, int mute) chip->image[CS4231_RIGHT_INPUT]); snd_wss_dout(chip, CS4231_LOOPBACK, chip->image[CS4231_LOOPBACK]); + } else { + snd_wss_dout(chip, CS4231_LEFT_INPUT, + 0); + snd_wss_dout(chip, CS4231_RIGHT_INPUT, + 0); + snd_wss_dout(chip, CS4231_LOOPBACK, + 0xfd); } + snd_wss_dout(chip, CS4231_AUX1_LEFT_INPUT, mute | chip->image[CS4231_AUX1_LEFT_INPUT]); snd_wss_dout(chip, CS4231_AUX1_RIGHT_INPUT, @@ -630,7 +619,6 @@ static void snd_wss_playback_format(struct snd_wss *chip, int full_calib = 1; mutex_lock(&chip->mce_mutex); - snd_wss_calibrate_mute(chip, 1); if (chip->hardware == WSS_HW_CS4231A || (chip->hardware & WSS_HW_CS4232_MASK)) { spin_lock_irqsave(&chip->reg_lock, flags); @@ -681,7 +669,6 @@ static void snd_wss_playback_format(struct snd_wss *chip, udelay(100); /* this seems to help */ snd_wss_mce_down(chip); } - snd_wss_calibrate_mute(chip, 0); mutex_unlock(&chip->mce_mutex); } @@ -693,7 +680,6 @@ static void snd_wss_capture_format(struct snd_wss *chip, int full_calib = 1; mutex_lock(&chip->mce_mutex); - snd_wss_calibrate_mute(chip, 1); if (chip->hardware == WSS_HW_CS4231A || (chip->hardware & WSS_HW_CS4232_MASK)) { spin_lock_irqsave(&chip->reg_lock, flags); @@ -750,7 +736,6 @@ static void snd_wss_capture_format(struct snd_wss *chip, spin_unlock_irqrestore(&chip->reg_lock, flags); snd_wss_mce_down(chip); } - snd_wss_calibrate_mute(chip, 0); mutex_unlock(&chip->mce_mutex); } @@ -807,6 +792,7 @@ static void snd_wss_init(struct snd_wss *chip) { unsigned long flags; + snd_wss_calibrate_mute(chip, 1); snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE @@ -830,6 +816,8 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); + chip->image[CS4231_IFACE_CTRL] &= ~CS4231_AUTOCALIB; + snd_wss_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); snd_wss_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1]); spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -863,6 +851,7 @@ static void snd_wss_init(struct snd_wss *chip) chip->image[CS4231_REC_FORMAT]); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_wss_mce_down(chip); + snd_wss_calibrate_mute(chip, 0); #ifdef SNDRV_DEBUG_MCE snd_printk("init: (5)\n"); @@ -921,8 +910,6 @@ static void snd_wss_close(struct snd_wss *chip, unsigned int mode) mutex_unlock(&chip->open_mutex); return; } - snd_wss_calibrate_mute(chip, 1); - /* disable IRQ */ spin_lock_irqsave(&chip->reg_lock, flags); if (!(chip->hardware & WSS_HW_AD1848_MASK)) @@ -955,8 +942,6 @@ static void snd_wss_close(struct snd_wss *chip, unsigned int mode) wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_wss_calibrate_mute(chip, 0); - chip->mode = 0; mutex_unlock(&chip->open_mutex); } @@ -1149,7 +1134,7 @@ irqreturn_t snd_wss_interrupt(int irq, void *dev_id) if (chip->hardware & WSS_HW_AD1848_MASK) wss_outb(chip, CS4231P(STATUS), 0); else - snd_wss_outm(chip, CS4231_IRQ_STATUS, status, 0); + snd_wss_out(chip, CS4231_IRQ_STATUS, status); spin_unlock(&chip->reg_lock); return IRQ_HANDLED; } -- 1.5.2.2 ---------------------------------------------------------------------- Speak Up. Angielski szybko i skutecznie. 3 miesiace nauki gratis. Sprawdz. >> http://link.interia.pl/f2019

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[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 16 Jan '09

16 Jan '09

The following changes since commit c0106d72b8d71696dbe9dc80e2c77d4ac63f7531: Takashi Iwai (1): Merge branch 'topic/asoc' into next/asoc are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Ben Nizette (1): ASoC: atmel_pcm: Remove non-existant header Hugo Villeneuve (1): ASoC: DaVinci: Fix SFFSDR compilation error. Ian Molton (3): ASoC: Driver for the WM9705 AC97 codec. ASoC: machine driver for Toshiba e750 ASoC: machine driver for Toshiba e800 Mark Brown (1): Merge branch 'for-2.6.29' into for-2.6.30 arch/arm/mach-pxa/e750.c | 5 + arch/arm/mach-pxa/include/mach/eseries-gpio.h | 10 + sound/soc/atmel/atmel-pcm.c | 2 - sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 3 + sound/soc/codecs/wm9705.c | 410 +++++++++++++++++++++++++ sound/soc/codecs/wm9705.h | 14 + sound/soc/davinci/davinci-sffsdr.c | 20 +- sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/e750_wm9705.c | 189 ++++++++++++ sound/soc/pxa/e800_wm9712.c | 116 ++++++- 12 files changed, 765 insertions(+), 19 deletions(-) create mode 100644 sound/soc/codecs/wm9705.c create mode 100644 sound/soc/codecs/wm9705.h create mode 100644 sound/soc/pxa/e750_wm9705.c

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[alsa-devel] ASoC update for 2.6.29
by Mark Brown 16 Jan '09

16 Jan '09

The following changes since commit eff317d0834ad1ff03f747f6bc2d76b9a9c95160: Peter Ujfalusi (1): ASoC: Fix the power update function for snd_soc_dapm_value_mux are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.29 Ben Nizette (1): ASoC: atmel_pcm: Remove non-existant header sound/soc/atmel/atmel-pcm.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-)

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[alsa-devel] [PATCH 1/1] ASoC: Fix compilation error with davinci-sffsdr module
by Hugo Villeneuve 16 Jan '09

16 Jan '09

The SFFSDR audio codec sampling frequency clock is generated by a FPGA, and set by the FPGA driver. The FPGA driver is not yet in the mainline kernel, so that is why an extern declaration is used in the meantime. Signed-off-by: Hugo Villeneuve <hugo(a)hugovil.com> --- sound/soc/davinci/davinci-sffsdr.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 1bbde3e..0a03b41 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -25,7 +25,6 @@ #include <asm/gpio.h> #include <asm/dma.h> #include <asm/mach-types.h> -#include <asm/plat-sffsdr/sffsdr-fpga.h> #include <mach/asp.h> #include <mach/edma.h> @@ -34,6 +33,11 @@ #include "davinci-pcm.h" #include "davinci-i2s.h" +/* The SFFSDR audio codec sampling frequency clock is generated by a FPGA. + * The FPGA driver is not yet in the mainline kernel, so that is + * why an extern declaration is used in the meantime. */ +extern int sffsdr_fpga_set_codec_fs(int fs); + static int sffsdr_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { -- 1.5.4.5

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