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[alsa-devel] [PATCH] Merge es1688 and es968 drivers
by Krzysztof Helt 01 May '10

01 May '10

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The ESS ES968 chip is nothing more then a PnP companion for non-PnP audio chips. It was paired with non-PnP ESS' chips: ES688 and ES1688. The ESS' audio chips are handled by the es1688 driver in native mode. The PnP cards are handled by the ES968 driver in SB compatible mode. Move the ES968 chip handling to the es1688 driver so the driver can handle both PnP and non-PnP cards. The es968 is removed. Also, a new PnP id is added for the card I acquired (the change was tested on this card). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Documentation/sound/alsa/ALSA-Configuration.txt | 21 +-- include/sound/es1688.h | 8 +- sound/isa/Kconfig | 16 +-- sound/isa/es1688/es1688.c | 204 ++++++++++++++++--- sound/isa/es1688/es1688_lib.c | 26 +-- sound/isa/sb/Makefile | 2 - sound/isa/sb/es968.c | 248 ----------------------- 7 files changed, 198 insertions(+), 327 deletions(-) delete mode 100644 sound/isa/sb/es968.c diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index bfcbbf8..448cff9 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -622,28 +622,23 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. The power-management is supported. - Module snd-es968 - ---------------- - - Module for sound cards based on ESS ES968 chip (PnP only). - - This module supports multiple cards, PnP and autoprobe. - - The power-management is supported. - Module snd-es1688 ----------------- Module for ESS AudioDrive ES-1688 and ES-688 sound cards. - port - port # for ES-1688 chip (0x220,0x240,0x260) - fm_port - port # for OPL3 (option; share the same port as default) + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default) - irq - IRQ # for ES-1688 chip (5,7,9,10) mpu_irq - IRQ # for MPU-401 port (5,7,9,10) + fm_port - port # for OPL3 (option; share the same port as default) + + with isapnp=0, the following additional options are available: + port - port # for ES-1688 chip (0x220,0x240,0x260) + irq - IRQ # for ES-1688 chip (5,7,9,10) dma8 - DMA # for ES-1688 chip (0,1,3) - This module supports multiple cards and autoprobe (without MPU-401 port). + This module supports multiple cards and autoprobe (without MPU-401 port) + and PnP with the ES968 chip. Module snd-es18xx ----------------- diff --git a/include/sound/es1688.h b/include/sound/es1688.h index 10fcf14..10d7951 100644 --- a/include/sound/es1688.h +++ b/include/sound/es1688.h @@ -44,7 +44,6 @@ struct snd_es1688 { unsigned char pad; unsigned int dma_size; - struct snd_card *card; struct snd_pcm *pcm; struct snd_pcm_substream *playback_substream; struct snd_pcm_substream *capture_substream; @@ -113,9 +112,8 @@ int snd_es1688_create(struct snd_card *card, int irq, int mpu_irq, int dma8, - unsigned short hardware, - struct snd_es1688 ** rchip); -int snd_es1688_pcm(struct snd_es1688 *chip, int device, struct snd_pcm ** rpcm); -int snd_es1688_mixer(struct snd_es1688 *chip); + unsigned short hardware); +int snd_es1688_pcm(struct snd_card *card, int device, struct snd_pcm **rpcm); +int snd_es1688_mixer(struct snd_card *card); #endif /* __SOUND_ES1688_H */ diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 755a0a5..c6990c6 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -128,26 +128,14 @@ config SND_CS4236 To compile this driver as a module, choose M here: the module will be called snd-cs4236. -config SND_ES968 - tristate "Generic ESS ES968 driver" - depends on PNP - select ISAPNP - select SND_MPU401_UART - select SND_SB8_DSP - help - Say Y here to include support for ESS AudioDrive ES968 chips. - - To compile this driver as a module, choose M here: the module - will be called snd-es968. - config SND_ES1688 - tristate "Generic ESS ES688/ES1688 driver" + tristate "Generic ESS ES688/ES1688 and ES968 PnP driver" select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM help Say Y here to include support for ESS AudioDrive ES688 or - ES1688 chips. + ES1688 chips. Also, this module support cards with ES968 PnP chip. To compile this driver as a module, choose M here: the module will be called snd-es1688. diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index 07df201..4b5cc8c 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -22,6 +22,7 @@ #include <linux/init.h> #include <linux/err.h> #include <linux/isa.h> +#include <linux/isapnp.h> #include <linux/time.h> #include <linux/wait.h> #include <linux/moduleparam.h> @@ -47,6 +48,9 @@ MODULE_SUPPORTED_DEVICE("{{ESS,ES688 PnP AudioDrive,pnp:ESS0100}," static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +#ifdef CONFIG_PNP +static int isapnp[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; +#endif static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260 */ static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* Usually 0x388 */ @@ -60,6 +64,10 @@ MODULE_PARM_DESC(index, "Index value for " CRD_NAME " soundcard."); module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard."); module_param_array(enable, bool, NULL, 0444); +#ifdef CONFIG_PNP +module_param_array(isapnp, bool, NULL, 0444); +MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard."); +#endif MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); module_param_array(port, long, NULL, 0444); MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); @@ -74,13 +82,19 @@ MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); module_param_array(dma8, int, NULL, 0444); MODULE_PARM_DESC(dma8, "8-bit DMA # for " CRD_NAME " driver."); +#ifdef CONFIG_PNP +#define is_isapnp_selected(dev) isapnp[dev] +#else +#define is_isapnp_selected(dev) 0 +#endif + static int __devinit snd_es1688_match(struct device *dev, unsigned int n) { - return enable[n]; + return enable[n] && !is_isapnp_selected(n); } -static int __devinit snd_es1688_legacy_create(struct snd_card *card, - struct device *dev, unsigned int n, struct snd_es1688 **rchip) +static int __devinit snd_es1688_legacy_create(struct snd_card *card, + struct device *dev, unsigned int n) { static long possible_ports[] = {0x220, 0x240, 0x260}; static int possible_irqs[] = {5, 9, 10, 7, -1}; @@ -105,41 +119,32 @@ static int __devinit snd_es1688_legacy_create(struct snd_card *card, if (port[n] != SNDRV_AUTO_PORT) return snd_es1688_create(card, port[n], mpu_port[n], irq[n], - mpu_irq[n], dma8[n], ES1688_HW_AUTO, rchip); + mpu_irq[n], dma8[n], ES1688_HW_AUTO); i = 0; do { port[n] = possible_ports[i]; error = snd_es1688_create(card, port[n], mpu_port[n], irq[n], - mpu_irq[n], dma8[n], ES1688_HW_AUTO, rchip); + mpu_irq[n], dma8[n], ES1688_HW_AUTO); } while (error < 0 && ++i < ARRAY_SIZE(possible_ports)); return error; } -static int __devinit snd_es1688_probe(struct device *dev, unsigned int n) +static int __devinit snd_es1688_probe(struct snd_card *card, unsigned int n) { - struct snd_card *card; - struct snd_es1688 *chip; + struct snd_es1688 *chip = card->private_data; struct snd_opl3 *opl3; struct snd_pcm *pcm; int error; - error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card); + error = snd_es1688_pcm(card, 0, &pcm); if (error < 0) return error; - error = snd_es1688_legacy_create(card, dev, n, &chip); - if (error < 0) - goto out; - - error = snd_es1688_pcm(chip, 0, &pcm); - if (error < 0) - goto out; - - error = snd_es1688_mixer(chip); + error = snd_es1688_mixer(card); if (error < 0) - goto out; + return error; strcpy(card->driver, "ES1688"); strcpy(card->shortname, pcm->name); @@ -152,12 +157,12 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n) if (fm_port[n] > 0) { if (snd_opl3_create(card, fm_port[n], fm_port[n] + 2, OPL3_HW_OPL3, 0, &opl3) < 0) - dev_warn(dev, + dev_warn(card->dev, "opl3 not detected at 0x%lx\n", fm_port[n]); else { error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (error < 0) - goto out; + return error; } } @@ -167,23 +172,41 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n) chip->mpu_port, 0, mpu_irq[n], IRQF_DISABLED, NULL); if (error < 0) - goto out; + return error; } + return snd_card_register(card); +} + +static int __devinit snd_es1688_isa_probe(struct device *dev, unsigned int n) +{ + struct snd_card *card; + int error; + + error = snd_card_create(index[n], id[n], THIS_MODULE, + sizeof(struct snd_es1688), &card); + if (error < 0) + return error; + + error = snd_es1688_legacy_create(card, dev, n); + if (error < 0) + goto out; + snd_card_set_dev(card, dev); - error = snd_card_register(card); + error = snd_es1688_probe(card, n); if (error < 0) goto out; dev_set_drvdata(dev, card); - return 0; -out: snd_card_free(card); + return 0; +out: + snd_card_free(card); return error; } -static int __devexit snd_es1688_remove(struct device *dev, unsigned int n) +static int __devexit snd_es1688_isa_remove(struct device *dev, unsigned int n) { snd_card_free(dev_get_drvdata(dev)); dev_set_drvdata(dev, NULL); @@ -192,8 +215,8 @@ static int __devexit snd_es1688_remove(struct device *dev, unsigned int n) static struct isa_driver snd_es1688_driver = { .match = snd_es1688_match, - .probe = snd_es1688_probe, - .remove = __devexit_p(snd_es1688_remove), + .probe = snd_es1688_isa_probe, + .remove = __devexit_p(snd_es1688_isa_remove), #if 0 /* FIXME */ .suspend = snd_es1688_suspend, .resume = snd_es1688_resume, @@ -203,14 +226,137 @@ static struct isa_driver snd_es1688_driver = { } }; +static int snd_es968_pnp_is_probed; + +#ifdef CONFIG_PNP +static int __devinit snd_card_es968_pnp(struct snd_card *card, unsigned int n, + struct pnp_card_link *pcard, + const struct pnp_card_device_id *pid) +{ + struct pnp_dev *pdev; + int error; + + pdev = pnp_request_card_device(pcard, pid->devs[0].id, NULL); + if (pdev == NULL) + return -ENODEV; + + error = pnp_activate_dev(pdev); + if (error < 0) { + snd_printk(KERN_ERR "ES968 pnp configure failure\n"); + return error; + } + port[n] = pnp_port_start(pdev, 0); + dma8[n] = pnp_dma(pdev, 0); + irq[n] = pnp_irq(pdev, 0); + + return snd_es1688_create(card, port[n], mpu_port[n], irq[n], + mpu_irq[n], dma8[n], ES1688_HW_AUTO); +} + +static int __devinit snd_es968_pnp_detect(struct pnp_card_link *pcard, + const struct pnp_card_device_id *pid) +{ + struct snd_card *card; + static unsigned int dev; + int error; + struct snd_es1688 *chip; + + if (snd_es968_pnp_is_probed) + return -EBUSY; + for ( ; dev < SNDRV_CARDS; dev++) { + if (enable[dev] && isapnp[dev]) + break; + } + + error = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_es1688), &card); + if (error < 0) + return error; + chip = card->private_data; + + error = snd_card_es968_pnp(card, dev, pcard, pid); + if (error < 0) { + snd_card_free(card); + return error; + } + snd_card_set_dev(card, &pcard->card->dev); + error = snd_es1688_probe(card, dev); + if (error < 0) + return error; + pnp_set_card_drvdata(pcard, card); + snd_es968_pnp_is_probed = 1; + return 0; +} + +static void __devexit snd_es968_pnp_remove(struct pnp_card_link * pcard) +{ + snd_card_free(pnp_get_card_drvdata(pcard)); + pnp_set_card_drvdata(pcard, NULL); + snd_es968_pnp_is_probed = 0; +} + +#ifdef CONFIG_PM +static int snd_es968_pnp_suspend(struct pnp_card_link *pcard, + pm_message_t state) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + struct snd_es1688 *chip = card->private_data; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(chip->pcm); + return 0; +} + +static int snd_es968_pnp_resume(struct pnp_card_link *pcard) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +#endif + +static struct pnp_card_device_id snd_es968_pnpids[] = { + { .id = "ESS0968", .devs = { { "@@@0968" }, } }, + { .id = "ESS0968", .devs = { { "ESS0968" }, } }, + { .id = "", } /* end */ +}; + +MODULE_DEVICE_TABLE(pnp_card, snd_es968_pnpids); + +static struct pnp_card_driver es968_pnpc_driver = { + .flags = PNP_DRIVER_RES_DISABLE, + .name = DEV_NAME " PnP", + .id_table = snd_es968_pnpids, + .probe = snd_es968_pnp_detect, + .remove = __devexit_p(snd_es968_pnp_remove), +#ifdef CONFIG_PM + .suspend = snd_es968_pnp_suspend, + .resume = snd_es968_pnp_resume, +#endif +}; +#endif + static int __init alsa_card_es1688_init(void) { +#ifdef CONFIG_PNP + pnp_register_card_driver(&es968_pnpc_driver); + if (snd_es968_pnp_is_probed) + return 0; + pnp_unregister_card_driver(&es968_pnpc_driver); +#endif return isa_register_driver(&snd_es1688_driver, SNDRV_CARDS); } static void __exit alsa_card_es1688_exit(void) { - isa_unregister_driver(&snd_es1688_driver); + if (!snd_es968_pnp_is_probed) { + isa_unregister_driver(&snd_es1688_driver); + return; + } +#ifdef CONFIG_PNP + pnp_unregister_card_driver(&es968_pnpc_driver); +#endif } module_init(alsa_card_es1688_init); diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index c76bb00..9632dda 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -643,20 +643,14 @@ int snd_es1688_create(struct snd_card *card, int irq, int mpu_irq, int dma8, - unsigned short hardware, - struct snd_es1688 **rchip) + unsigned short hardware) { static struct snd_device_ops ops = { .dev_free = snd_es1688_dev_free, }; - - struct snd_es1688 *chip; + struct snd_es1688 *chip = card->private_data; int err; - *rchip = NULL; - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) - return -ENOMEM; chip->irq = -1; chip->dma8 = -1; @@ -680,7 +674,6 @@ int snd_es1688_create(struct snd_card *card, spin_lock_init(&chip->reg_lock); spin_lock_init(&chip->mixer_lock); - chip->card = card; chip->port = port; mpu_port &= ~0x000f; if (mpu_port < 0x300 || mpu_port > 0x330) @@ -704,7 +697,6 @@ int snd_es1688_create(struct snd_card *card, return err; } - *rchip = chip; return 0; } @@ -730,12 +722,14 @@ static struct snd_pcm_ops snd_es1688_capture_ops = { .pointer = snd_es1688_capture_pointer, }; -int snd_es1688_pcm(struct snd_es1688 * chip, int device, struct snd_pcm ** rpcm) +int snd_es1688_pcm(struct snd_card *card, int device, struct snd_pcm **rpcm) { struct snd_pcm *pcm; + struct snd_es1688 *chip = card->private_data; int err; - if ((err = snd_pcm_new(chip->card, "ESx688", device, 1, 1, &pcm)) < 0) + err = snd_pcm_new(card, "ESx688", device, 1, 1, &pcm); + if (err < 0) return err; snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_es1688_playback_ops); @@ -1009,17 +1003,17 @@ static unsigned char snd_es1688_init_table[][2] = { { ES1688_REC_DEV, 0x17 } }; -int snd_es1688_mixer(struct snd_es1688 *chip) +int snd_es1688_mixer(struct snd_card *card) { - struct snd_card *card; + struct snd_es1688 *chip; unsigned int idx; int err; unsigned char reg, val; - if (snd_BUG_ON(!chip || !chip->card)) + if (snd_BUG_ON(!card || !card->private_data)) return -EINVAL; - card = chip->card; + chip = card->private_data; strcpy(card->mixername, snd_es1688_chip_id(chip)); diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile index af36696..08b9fb9 100644 --- a/sound/isa/sb/Makefile +++ b/sound/isa/sb/Makefile @@ -11,7 +11,6 @@ snd-sb8-objs := sb8.o snd-sb16-objs := sb16.o snd-sbawe-objs := sbawe.o emu8000.o snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o -snd-es968-objs := es968.o snd-jazz16-objs := jazz16.o # Toplevel Module Dependency @@ -21,7 +20,6 @@ obj-$(CONFIG_SND_SB8_DSP) += snd-sb8-dsp.o obj-$(CONFIG_SND_SB8) += snd-sb8.o obj-$(CONFIG_SND_SB16) += snd-sb16.o obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o -obj-$(CONFIG_SND_ES968) += snd-es968.o obj-$(CONFIG_SND_JAZZ16) += snd-jazz16.o ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o diff --git a/sound/isa/sb/es968.c b/sound/isa/sb/es968.c deleted file mode 100644 index ff18286..0000000 --- a/sound/isa/sb/es968.c +++ /dev/null @@ -1,248 +0,0 @@ - -/* - card-es968.c - driver for ESS AudioDrive ES968 based soundcards. - Copyright (C) 1999 by Massimo Piccioni <dafastidio(a)libero.it> - - Thanks to Pierfrancesco 'qM2' Passerini. - - This program is free software; you can redistribute it and/or modify - it under the terms of the GNU General Public License as published by - the Free Software Foundation; either version 2 of the License, or - (at your option) any later version. - - This program is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - GNU General Public License for more details. - - You should have received a copy of the GNU General Public License - along with this program; if not, write to the Free Software - Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -*/ - -#include <linux/init.h> -#include <linux/time.h> -#include <linux/pnp.h> -#include <linux/moduleparam.h> -#include <sound/core.h> -#include <sound/initval.h> -#include <sound/sb.h> - -#define PFX "es968: " - -MODULE_AUTHOR("Massimo Piccioni <dafastidio(a)libero.it>"); -MODULE_DESCRIPTION("ESS AudioDrive ES968"); -MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{ESS,AudioDrive ES968}}"); - -static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ -static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ -static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */ -static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* Pnp setup */ -static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */ - -module_param_array(index, int, NULL, 0444); -MODULE_PARM_DESC(index, "Index value for es968 based soundcard."); -module_param_array(id, charp, NULL, 0444); -MODULE_PARM_DESC(id, "ID string for es968 based soundcard."); -module_param_array(enable, bool, NULL, 0444); -MODULE_PARM_DESC(enable, "Enable es968 based soundcard."); - -struct snd_card_es968 { - struct pnp_dev *dev; - struct snd_sb *chip; -}; - -static struct pnp_card_device_id snd_es968_pnpids[] = { - { .id = "ESS0968", .devs = { { "@@@0968" }, } }, - { .id = "", } /* end */ -}; - -MODULE_DEVICE_TABLE(pnp_card, snd_es968_pnpids); - -#define DRIVER_NAME "snd-card-es968" - -static irqreturn_t snd_card_es968_interrupt(int irq, void *dev_id) -{ - struct snd_sb *chip = dev_id; - - if (chip->open & SB_OPEN_PCM) { - return snd_sb8dsp_interrupt(chip); - } else { - return snd_sb8dsp_midi_interrupt(chip); - } -} - -static int __devinit snd_card_es968_pnp(int dev, struct snd_card_es968 *acard, - struct pnp_card_link *card, - const struct pnp_card_device_id *id) -{ - struct pnp_dev *pdev; - int err; - - acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) - return -ENODEV; - - pdev = acard->dev; - - err = pnp_activate_dev(pdev); - if (err < 0) { - snd_printk(KERN_ERR PFX "AUDIO pnp configure failure\n"); - return err; - } - port[dev] = pnp_port_start(pdev, 0); - dma8[dev] = pnp_dma(pdev, 0); - irq[dev] = pnp_irq(pdev, 0); - - return 0; -} - -static int __devinit snd_card_es968_probe(int dev, - struct pnp_card_link *pcard, - const struct pnp_card_device_id *pid) -{ - int error; - struct snd_sb *chip; - struct snd_card *card; - struct snd_card_es968 *acard; - - error = snd_card_create(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_card_es968), &card); - if (error < 0) - return error; - acard = card->private_data; - if ((error = snd_card_es968_pnp(dev, acard, pcard, pid))) { - snd_card_free(card); - return error; - } - snd_card_set_dev(card, &pcard->card->dev); - - if ((error = snd_sbdsp_create(card, port[dev], - irq[dev], - snd_card_es968_interrupt, - dma8[dev], - -1, - SB_HW_AUTO, &chip)) < 0) { - snd_card_free(card); - return error; - } - acard->chip = chip; - - if ((error = snd_sb8dsp_pcm(chip, 0, NULL)) < 0) { - snd_card_free(card); - return error; - } - - if ((error = snd_sbmixer_new(chip)) < 0) { - snd_card_free(card); - return error; - } - - if ((error = snd_sb8dsp_midi(chip, 0, NULL)) < 0) { - snd_card_free(card); - return error; - } - - strcpy(card->driver, "ES968"); - strcpy(card->shortname, "ESS ES968"); - sprintf(card->longname, "%s soundcard, %s at 0x%lx, irq %d, dma %d", - card->shortname, chip->name, chip->port, irq[dev], dma8[dev]); - - if ((error = snd_card_register(card)) < 0) { - snd_card_free(card); - return error; - } - pnp_set_card_drvdata(pcard, card); - return 0; -} - -static unsigned int __devinitdata es968_devices; - -static int __devinit snd_es968_pnp_detect(struct pnp_card_link *card, - const struct pnp_card_device_id *id) -{ - static int dev; - int res; - - for ( ; dev < SNDRV_CARDS; dev++) { - if (!enable[dev]) - continue; - res = snd_card_es968_probe(dev, card, id); - if (res < 0) - return res; - dev++; - es968_devices++; - return 0; - } - return -ENODEV; -} - -static void __devexit snd_es968_pnp_remove(struct pnp_card_link * pcard) -{ - snd_card_free(pnp_get_card_drvdata(pcard)); - pnp_set_card_drvdata(pcard, NULL); -} - -#ifdef CONFIG_PM -static int snd_es968_pnp_suspend(struct pnp_card_link *pcard, pm_message_t state) -{ - struct snd_card *card = pnp_get_card_drvdata(pcard); - struct snd_card_es968 *acard = card->private_data; - struct snd_sb *chip = acard->chip; - - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - snd_pcm_suspend_all(chip->pcm); - snd_sbmixer_suspend(chip); - return 0; -} - -static int snd_es968_pnp_resume(struct pnp_card_link *pcard) -{ - struct snd_card *card = pnp_get_card_drvdata(pcard); - struct snd_card_es968 *acard = card->private_data; - struct snd_sb *chip = acard->chip; - - snd_sbdsp_reset(chip); - snd_sbmixer_resume(chip); - snd_power_change_state(card, SNDRV_CTL_POWER_D0); - return 0; -} -#endif - -static struct pnp_card_driver es968_pnpc_driver = { - .flags = PNP_DRIVER_RES_DISABLE, - .name = "es968", - .id_table = snd_es968_pnpids, - .probe = snd_es968_pnp_detect, - .remove = __devexit_p(snd_es968_pnp_remove), -#ifdef CONFIG_PM - .suspend = snd_es968_pnp_suspend, - .resume = snd_es968_pnp_resume, -#endif -}; - -static int __init alsa_card_es968_init(void) -{ - int err = pnp_register_card_driver(&es968_pnpc_driver); - if (err) - return err; - - if (!es968_devices) { - pnp_unregister_card_driver(&es968_pnpc_driver); -#ifdef MODULE - snd_printk(KERN_ERR "no ES968 based soundcards found\n"); -#endif - return -ENODEV; - } - return 0; -} - -static void __exit alsa_card_es968_exit(void) -{ - pnp_unregister_card_driver(&es968_pnpc_driver); -} - -module_init(alsa_card_es968_init) -module_exit(alsa_card_es968_exit) -- 1.6.4 ---------------------------------------------------------------------- Szukasz pracy? 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[alsa-devel] [PATCH 1/2] powerpc: add platform registration for ALSA SoC drivers
by Timur Tabi 01 May '10

01 May '10

An upcoming change in the architecture of ALSA SoC (ASoC) will require the MPC8610 HPCD's ASoC fabric driver to register as a standard platform driver. Therefore, we need to call platform_device_register_simple() from the board's platform code. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- Kumar, the ASoC mainters are willing to pick up this patch, but they want an ACK from you first. Or, you could pick it up, since by itself it's harmless. arch/powerpc/platforms/86xx/mpc8610_hpcd.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/arch/powerpc/platforms/86xx/mpc8610_hpcd.c b/arch/powerpc/platforms/86xx/mpc8610_hpcd.c index 5abe137..66afff3 100644 --- a/arch/powerpc/platforms/86xx/mpc8610_hpcd.c +++ b/arch/powerpc/platforms/86xx/mpc8610_hpcd.c @@ -98,6 +98,9 @@ static int __init mpc8610_declare_of_platform_devices(void) /* Without this call, the SSI device driver won't get probed. */ of_platform_bus_probe(NULL, mpc8610_ids, NULL); + /* Register the platform device for the ASoC fabric driver */ + platform_device_register_simple("snd-soc-mpc8610-hpcd", 0, NULL, 0); + return 0; } machine_device_initcall(mpc86xx_hpcd, mpc8610_declare_of_platform_devices); -- 1.6.5

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[alsa-devel] Relation between SND_PCM_IOPLUG_HW_PERIODS and SND_PCM_IOPLUG_HW_BUFFER_BYTES ?
by Stefan Schoenleitner 30 Apr '10

30 Apr '10

Hi, what is the relation between SND_PCM_IOPLUG_HW_PERIODS and SND_PCM_IOPLUG_HW_BUFFER_BYTES ? While SND_PCM_IOPLUG_HW_BUFFER_BYTES is the setting of the hardware buffer size, I'm not sure exactly what SND_PCM_IOPLUG_HW_PERIODS is. I suppose it is the number of periods the hardware buffer can take ? Hence the SND_PCM_IOPLUG_HW_PERIODS value is actually derived from the size of the hardware buffer ? Or is it rather the number of periods the ioplugin can receive at once ? Please enlighten me, cheers, Stefan

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[alsa-devel] [PATCH 0/4] ASoC: tlv320dac33: Support for turning codec off
by Peter Ujfalusi 30 Apr '10

30 Apr '10

Hello, The following series adds support for turning the tlv320dac33 codec off in BIAS_STANDBY (and obviously in BIAS_OFF). Since the driver already had mechanics for supporting turned off codec in register writes, the needed changes are mainly in the BIAS/stream handling. As part of the series further optimization has been done to codec initialization, resume. The following scenarios has been tested, and verified (I think these are covering all cases): 1. Analog bypass caused BIAS_STANDBY -> BIAS_ON We need to power on the codec, and do the chip init, but we does not need to execute the playback related configuration 2. Playback caused BIAS_STANDBY -> BIAS_ON We need to power on the codec, and do the chip init, and also we need to execute the playback related configuration. 3. Playback start, while Analog bypass is on (BIAS_ON -> BIAS_ON) We need to execute the playback related configuration. The codec is already on. 4. Analog bypass enable, while playback (BIAS_ON -> BIAS_ON) Nothing need to be done. 5. Playback start withing soc power down timeout (BIAS_ON -> BIAS_ON) We need to execute the playback related configuration. The codec is still on. 6. Analog bypass enable withing soc power down timeout (BIAS_ON -> BIAS_ON) Nothing need to be done. This series applies on top of Liam's: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.35 branch --- Peter Ujfalusi (4): ASoC: tlv320dac33: Optimize power up, and restore ASoC: tlv320dac33: Revised module loading, and DAC33 ID read ASoC: tlv320dac33: Manage a pointer for snd_pcm_substream in private structure ASoC: tlv320dac33: Support for turning off the codec in BIAS_STANDBY sound/soc/codecs/tlv320dac33.c | 236 +++++++++++++++++++++++----------------- 1 files changed, 137 insertions(+), 99 deletions(-)

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[alsa-devel] PCM multi A/V Sync bug on SMP system
by Keith Chew 30 Apr '10

30 Apr '10

Hi I have a SB Audigy Value card with 5.1 channel output. This is defined in my asound.conf: --------------------------- pcm.multi { type multi slaves { a.pcm "dmixerrear" a.channels 2 b.pcm "dmixerfront" b.channels 2 c.pcm "dmixercenter" c.channels 2 } bindings { 0.slave a 0.channel 0 1.slave a 1.channel 1 2.slave b 2.channel 0 3.slave b 3.channel 1 4.slave c 4.channel 0 5.slave c 5.channel 1 } } pcm.dmixerfront { type dmix ipc_key 1024 slave { pcm "hw:0,0" channels 2 } } pcm.dmixerrear { type dmix ipc_key 1025 slave { pcm "hw:0,1" channels 2 } } pcm.dmixercenter { type dmix ipc_key 1026 slave { pcm "hw:0,2" channels 2 } } --------------------------- Playing a video with "mplayer -ao alsa:device=multi ..." works fine, I can hear the 6-channel audio without any A/V sync issues. However, this is on a uniprocessor system. On an SMP system, the A/V sync gets out of sync from the start and drifts quite quickly. I have narrowed it down to the pcm_multi component giving this problem. On an SMP system, using just "mplayer -ao alsa ..." gives no A/V sync issues (because it is not using multi) on the 6 channels. Moreover, I have also tried "mplayer -ao alsa:device=dmixerfront ...", "mplayer -ao alsa:device=dmixercenter ..." and "mplayer -ao alsa:device=dmixerrear ...", all giving no A/V sync issues on the individual devices. Please advise how can I track down this problem in code, happy to make changes and test. PS: I need multi to work, so that I can add additional options like softvol on top of it. Regards Keith

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[alsa-devel] How to retrieve the final data being played by HW
by Yu Hang 30 Apr '10

30 Apr '10

I'm studying ALSA lib and trying to write a plugin that can perform AEC function. I have to retrieve the data being played and use it as reference. I've read pcm_dmix.c and found dmix just simply mixes data into HW mmap buffer (destination buffer). My problem is, there might be several dmix instances, which doesn't know if there is other dmix instance. So how can I know the mixed data is exactly the data being played. For example, the 1st instance had mixes the data into destination buffer, but 2nd instance haven't complete the mix operation. Would the data in destination buffer be playing by HW? Or there is some mechanism to guarantee the data will be played after all mix operation is done.

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[alsa-devel] asoc: problem with snd_soc_dai_set_fmt()
by Timur Tabi 30 Apr '10

30 Apr '10

There is a problem with snd_soc_dai_set_fmt(): int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { if (dai->driver && dai->driver->ops->set_fmt) return dai->driver->ops->set_fmt(dai, fmt); else return -EINVAL; } Let's say the DAI driver has not defined a .set_fmt() function. This means that if the fabric driver does this: ret = snd_soc_dai_set_fmt(cpu_dai, machine_data->dai_format); if (ret < 0) { dev_err(dev, "could not set CPU driver audio format\n"); return ret; } It's going to think that the DAI driver *rejected* the DAI format. What this means is that I cannot make this function optional. I have to define this function in my CPU driver. May I propose this: int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { if (dai->driver && dai->driver->ops->set_fmt) return dai->driver->ops->set_fmt(dai, fmt); else return 0; } -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] Wrote MIDIcat program, to connect ALSA MIDI to standard input/output
by Josh Lehan 30 Apr '10

30 Apr '10

Hi! A few months ago, I posted about a program to send untimed MIDI data to the ALSA sequencer. Using amidi(1) wasn't acceptable, since it only talks to "RawMIDI" devices. Using aplaymidi(1) wasn't acceptable either, because it required .MID files, and I simply wanted to send untimed MIDI data, similar to what would be produced if you hooked up a hardware MIDI keyboard and just started playing. I needed something more general, that would simply connect to the ALSA sequencer, and let me send arbitrary MIDI bytes to a sequencer destination of my choice. Unfortunately, no program existed that would simply do this. I'm pleased to announce that I've written such a program, It's called MIDIcat, as it works on the same principle as "cat" and other similar programs (Netcat, and so on). It hooks up standard input, and standard output, to the ALSA sequencer. This makes it easy to pipe data around. It's an ALSA sequencer client. Any data that is received on standard input will be forwarded into ALSA. You can choose another client as a destination, or you can just start the program passively and use ALSA's "subscription" mechanism to route the data later. Similarly, any MIDI data that comes from ALSA, will be forwarded along, and provided on standard output. This makes it easy to make many small "one-liner" command lines that are useful for testing and playing around with MIDI data in general. As an example, this plays Middle C: echo "90 3C 7F" | midicat --hex --port "TiMidity" The --hex option makes things more human-readable: standard input and output are hex digits, separated by spaces, instead of just raw binary data. The --port option has the usual ALSA sequencer meaning. Change it to target your synthesizer (I'm using the TiMidity softsynth). Here's another example for output (both of these commands block, so run them in two separate terminal windows): midicat --verbose --hex vkeybd --addr 129:0 The --verbose option prints out some more useful information at startup, such as the ALSA sequencer client:port number it got. That can later be passed into another program, such as vkeybd. If you play notes with vkeybd, you see the corresponding MIDI data appear on standard output. Standard input and standard output can both run at the same time (it's multithreaded). Here's some more elaborate examples: MIDI "panic button", sends "All Sound Off" to each channel: perl -e 'use bytes;for($i=0;$i<16;$i++){print chr(176+$i).chr(120).chr(0);}' | midicat --port "TiMidity" Another MIDI "panic button", sends individual "Note Off" commands: perl -e 'use bytes;for($i=0;$i<16;$i++){for($j=0;$j<128;$j++){print chr(128+$i).chr($j).chr(127);}}' | midicat --delay 10 --port "TiMidity" The --delay option inserts a slight delay between each MIDI command, to avoid flooding ALSA with too much data at once. Here's a fun one, sending random data into the MIDI subsystem and seeing what happens. DO NOT run this on a real synthesizer, there's the risk of composing a random SysEx command that could mess up your settings! It's fun to watch with aseqview(1): aseqview & cat /dev/urandom | midicat --delay 500 --port "MIDI Viewer" If you sent random data to a device that plays audio, you'll probably next want to use the "panic button" commands above...! Hope this illustrates some of the examples and possibilities. I'm working on writing some more documentation. The --help option is supported, there's a help output that shows you all of the available options. For now, the program is available here, on my Web server: http://krellan.com/midicat If you're interested, try it, and let me know what you think! Josh Lehan

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[alsa-devel] My understanding of ALSA playback mechanism
by Yu Hang 30 Apr '10

30 Apr '10

Hi, I'm a newbie of ALSA. I've wrote some audio applications under Windows OS with DirectSound and Waveform Audio API. And I also have some knowledge about Sound Architecture of Windows Vista/Windows 7. All my experience about audio architecture was from Windows. I'm currently working on a ALSA plugin for AEC function. What I have to do is to retrieve audio data being played (after mixing) as reference to cancel the echo captured by MIC. According to my previous experience about Sound Architecture of Win7/Vista, the audio data is sending from OS component to my own component: no matter for capture or playback. All I have to do is just simply modify the input data and put them into output buffer. If my component is under OS mixer, the received audio data is after mixing. When I read source code of ALSA lib, I found the mechanism between Windows and ALSA lib are quite different. Following was my understanding about ALSA lib mechanism during playback. There is a hardware buffer. ALSA driver takes data from the buffer and sends it to HW. The driver also update the play position on the fly. dmix receive the data from application and mix them into a shared memory, and then copy it into hardware buffer. Each of dmix instance keep a pointer to indicate current mixing position which might different from other instance. [play pos] [1st mixed data] | | +----------------------------+ hardware buffer +----------------------------+ | [2nd mixed data] The figure shows, the data mixed from 1st dmix instance lasts to [1st mixed data] pos, and data mixed from 2nd dmix instance lasts to [2nd mixed data]. The buffer between [1st mixed data] and [2nd mixed data] contains only data of 2nd dmix instance, but it does not matter, ALSA just guarantees the data of 1st dmix instance can be mixed up before the [play pos] reaches [1st mixed data] pos. Of course the hardware buffer is not unlimited (it's a ring buffer), so before dmix mix data to hardware buffer, it has to check if there is enough space. Is my understanding of ALSA playback mechanism correct?

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Re: [alsa-devel] [RFC] disabling ALSA period interrupts
by Raymond Yau 30 Apr '10

30 Apr '10

2010/4/30 pl bossart <bossart.nospam(a)gmail.com> > > do you mean that it only work on HDaudio and not intel8x0 ? > > > > How about other drivers ? > > It's driver/hardware specific, I guess everyone would have to enable > this in their drivers. I don't understand what 8x0 does...You're > welcome to test and do the same for 8x0 if you have the specs and know > how to disable interrupts. > > Any side effect of this disabling interrupt ? Do you mean that all applications must use pulse api instead of alsa-pulse plugin ? Did your test included those voip application which require low latency ? did your test only play one audio stream to the sound server ? Do those alsa application still work with "hw" device when pulseaudio is disabled ?

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