From f6f11018dc7ea62482f36846e9f6eb0f27df7c3c Mon Sep 17 00:00:00 2001 From: Stephen Rothwell Date: Mon, 11 Aug 2008 17:04:32 +1000 Subject: powerpc/drivers: Use linux/of_device.h instead of asm/of_device.h Signed-off-by: Stephen Rothwell Acked-by: Takashi Iwai Signed-off-by: Paul Mackerras --- sound/aoa/soundbus/soundbus.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/aoa/soundbus/soundbus.h b/sound/aoa/soundbus/soundbus.h index 622cd37a011..a0f223c13f6 100644 --- a/sound/aoa/soundbus/soundbus.h +++ b/sound/aoa/soundbus/soundbus.h @@ -8,7 +8,7 @@ #ifndef __SOUNDBUS_H #define __SOUNDBUS_H -#include +#include #include #include -- cgit v1.2.3 From 7a715f46012f3552294154978aed59cba9804928 Mon Sep 17 00:00:00 2001 From: "David S. Miller" Date: Wed, 27 Aug 2008 18:37:58 -0700 Subject: sparc: Make SBUS DMA interfaces take struct device. This is the first step in converting all the SBUS drivers over to generic dma_*(). Signed-off-by: David S. Miller --- sound/core/memalloc.c | 6 ++++-- sound/sparc/dbri.c | 15 +++++++++------ 2 files changed, 13 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index f5d6d8d1297..cc803972c0f 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -192,7 +192,8 @@ static void *snd_malloc_sbus_pages(struct device *dev, size_t size, snd_assert(size > 0, return NULL); snd_assert(dma_addr != NULL, return NULL); pg = get_order(size); - res = sbus_alloc_consistent(sdev, PAGE_SIZE * (1 << pg), dma_addr); + res = sbus_alloc_consistent(&sdev->ofdev.dev, PAGE_SIZE * (1 << pg), + dma_addr); if (res != NULL) inc_snd_pages(pg); return res; @@ -208,7 +209,8 @@ static void snd_free_sbus_pages(struct device *dev, size_t size, return; pg = get_order(size); dec_snd_pages(pg); - sbus_free_consistent(sdev, PAGE_SIZE * (1 << pg), ptr, dma_addr); + sbus_free_consistent(&sdev->ofdev.dev, PAGE_SIZE * (1 << pg), + ptr, dma_addr); } #endif /* CONFIG_SBUS */ diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index ee2e1b4f355..a6b32ec34bd 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2097,7 +2097,8 @@ static int snd_dbri_hw_params(struct snd_pcm_substream *substream, else direction = SBUS_DMA_FROMDEVICE; - info->dvma_buffer = sbus_map_single(dbri->sdev, + info->dvma_buffer = + sbus_map_single(&dbri->sdev->ofdev.dev, runtime->dma_area, params_buffer_bytes(hw_params), direction); @@ -2125,7 +2126,7 @@ static int snd_dbri_hw_free(struct snd_pcm_substream *substream) else direction = SBUS_DMA_FROMDEVICE; - sbus_unmap_single(dbri->sdev, info->dvma_buffer, + sbus_unmap_single(&dbri->sdev->ofdev.dev, info->dvma_buffer, substream->runtime->buffer_size, direction); info->dvma_buffer = 0; } @@ -2524,7 +2525,8 @@ static int __devinit snd_dbri_create(struct snd_card *card, dbri->sdev = sdev; dbri->irq = irq; - dbri->dma = sbus_alloc_consistent(sdev, sizeof(struct dbri_dma), + dbri->dma = sbus_alloc_consistent(&sdev->ofdev.dev, + sizeof(struct dbri_dma), &dbri->dma_dvma); memset((void *)dbri->dma, 0, sizeof(struct dbri_dma)); @@ -2537,7 +2539,7 @@ static int __devinit snd_dbri_create(struct snd_card *card, dbri->regs_size, "DBRI Registers"); if (!dbri->regs) { printk(KERN_ERR "DBRI: could not allocate registers\n"); - sbus_free_consistent(sdev, sizeof(struct dbri_dma), + sbus_free_consistent(&sdev->ofdev.dev, sizeof(struct dbri_dma), (void *)dbri->dma, dbri->dma_dvma); return -EIO; } @@ -2547,7 +2549,7 @@ static int __devinit snd_dbri_create(struct snd_card *card, if (err) { printk(KERN_ERR "DBRI: Can't get irq %d\n", dbri->irq); sbus_iounmap(dbri->regs, dbri->regs_size); - sbus_free_consistent(sdev, sizeof(struct dbri_dma), + sbus_free_consistent(&sdev->ofdev.dev, sizeof(struct dbri_dma), (void *)dbri->dma, dbri->dma_dvma); return err; } @@ -2575,7 +2577,8 @@ static void snd_dbri_free(struct snd_dbri *dbri) sbus_iounmap(dbri->regs, dbri->regs_size); if (dbri->dma) - sbus_free_consistent(dbri->sdev, sizeof(struct dbri_dma), + sbus_free_consistent(&dbri->sdev->ofdev.dev, + sizeof(struct dbri_dma), (void *)dbri->dma, dbri->dma_dvma); } -- cgit v1.2.3 From 738f2b7b813913e651f39387d007dd961755dee2 Mon Sep 17 00:00:00 2001 From: "David S. Miller" Date: Wed, 27 Aug 2008 18:09:11 -0700 Subject: sparc: Convert all SBUS drivers to dma_*() interfaces. And all the SBUS dma interfaces are deleted. A private implementation remains inside of the 32-bit sparc port which exists only for the sake of the implementation of dma_*(). Signed-off-by: David S. Miller --- sound/core/memalloc.c | 8 ++++---- sound/sparc/dbri.c | 41 +++++++++++++++++++++-------------------- 2 files changed, 25 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index cc803972c0f..ccaaac45faf 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -192,8 +192,8 @@ static void *snd_malloc_sbus_pages(struct device *dev, size_t size, snd_assert(size > 0, return NULL); snd_assert(dma_addr != NULL, return NULL); pg = get_order(size); - res = sbus_alloc_consistent(&sdev->ofdev.dev, PAGE_SIZE * (1 << pg), - dma_addr); + res = dma_alloc_coherent(&sdev->ofdev.dev, PAGE_SIZE * (1 << pg), + dma_addr, GFP_ATOMIC); if (res != NULL) inc_snd_pages(pg); return res; @@ -209,8 +209,8 @@ static void snd_free_sbus_pages(struct device *dev, size_t size, return; pg = get_order(size); dec_snd_pages(pg); - sbus_free_consistent(&sdev->ofdev.dev, PAGE_SIZE * (1 << pg), - ptr, dma_addr); + dma_free_coherent(&sdev->ofdev.dev, PAGE_SIZE * (1 << pg), + ptr, dma_addr); } #endif /* CONFIG_SBUS */ diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index a6b32ec34bd..5242ecbb91d 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -57,6 +57,7 @@ #include #include #include +#include #include #include @@ -2093,15 +2094,15 @@ static int snd_dbri_hw_params(struct snd_pcm_substream *substream, */ if (info->dvma_buffer == 0) { if (DBRI_STREAMNO(substream) == DBRI_PLAY) - direction = SBUS_DMA_TODEVICE; + direction = DMA_TO_DEVICE; else - direction = SBUS_DMA_FROMDEVICE; + direction = DMA_FROM_DEVICE; info->dvma_buffer = - sbus_map_single(&dbri->sdev->ofdev.dev, - runtime->dma_area, - params_buffer_bytes(hw_params), - direction); + dma_map_single(&dbri->sdev->ofdev.dev, + runtime->dma_area, + params_buffer_bytes(hw_params), + direction); } direction = params_buffer_bytes(hw_params); @@ -2122,12 +2123,12 @@ static int snd_dbri_hw_free(struct snd_pcm_substream *substream) */ if (info->dvma_buffer) { if (DBRI_STREAMNO(substream) == DBRI_PLAY) - direction = SBUS_DMA_TODEVICE; + direction = DMA_TO_DEVICE; else - direction = SBUS_DMA_FROMDEVICE; + direction = DMA_FROM_DEVICE; - sbus_unmap_single(&dbri->sdev->ofdev.dev, info->dvma_buffer, - substream->runtime->buffer_size, direction); + dma_unmap_single(&dbri->sdev->ofdev.dev, info->dvma_buffer, + substream->runtime->buffer_size, direction); info->dvma_buffer = 0; } if (info->pipe != -1) { @@ -2525,9 +2526,9 @@ static int __devinit snd_dbri_create(struct snd_card *card, dbri->sdev = sdev; dbri->irq = irq; - dbri->dma = sbus_alloc_consistent(&sdev->ofdev.dev, - sizeof(struct dbri_dma), - &dbri->dma_dvma); + dbri->dma = dma_alloc_coherent(&sdev->ofdev.dev, + sizeof(struct dbri_dma), + &dbri->dma_dvma, GFP_ATOMIC); memset((void *)dbri->dma, 0, sizeof(struct dbri_dma)); dprintk(D_GEN, "DMA Cmd Block 0x%p (0x%08x)\n", @@ -2539,8 +2540,8 @@ static int __devinit snd_dbri_create(struct snd_card *card, dbri->regs_size, "DBRI Registers"); if (!dbri->regs) { printk(KERN_ERR "DBRI: could not allocate registers\n"); - sbus_free_consistent(&sdev->ofdev.dev, sizeof(struct dbri_dma), - (void *)dbri->dma, dbri->dma_dvma); + dma_free_coherent(&sdev->ofdev.dev, sizeof(struct dbri_dma), + (void *)dbri->dma, dbri->dma_dvma); return -EIO; } @@ -2549,8 +2550,8 @@ static int __devinit snd_dbri_create(struct snd_card *card, if (err) { printk(KERN_ERR "DBRI: Can't get irq %d\n", dbri->irq); sbus_iounmap(dbri->regs, dbri->regs_size); - sbus_free_consistent(&sdev->ofdev.dev, sizeof(struct dbri_dma), - (void *)dbri->dma, dbri->dma_dvma); + dma_free_coherent(&sdev->ofdev.dev, sizeof(struct dbri_dma), + (void *)dbri->dma, dbri->dma_dvma); return err; } @@ -2577,9 +2578,9 @@ static void snd_dbri_free(struct snd_dbri *dbri) sbus_iounmap(dbri->regs, dbri->regs_size); if (dbri->dma) - sbus_free_consistent(&dbri->sdev->ofdev.dev, - sizeof(struct dbri_dma), - (void *)dbri->dma, dbri->dma_dvma); + dma_free_coherent(&dbri->sdev->ofdev.dev, + sizeof(struct dbri_dma), + (void *)dbri->dma, dbri->dma_dvma); } static int __devinit dbri_probe(struct of_device *of_dev, -- cgit v1.2.3 From dc8aa2e9d6c965c11ddd05fca721b2218585b9f0 Mon Sep 17 00:00:00 2001 From: "David S. Miller" Date: Wed, 27 Aug 2008 00:29:27 -0700 Subject: amd7930: Convert to pure OF driver. Signed-off-by: David S. Miller --- sound/sparc/amd7930.c | 83 +++++++++++++-------------------------------------- 1 file changed, 20 insertions(+), 63 deletions(-) (limited to 'sound') diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index 0c63e0585b1..0f82c3f6cfd 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -1,6 +1,6 @@ /* * Driver for AMD7930 sound chips found on Sparcs. - * Copyright (C) 2002 David S. Miller + * Copyright (C) 2002, 2008 David S. Miller * * Based entirely upon drivers/sbus/audio/amd7930.c which is: * Copyright (C) 1996,1997 Thomas K. Dyas (tdyas@eden.rutgers.edu) @@ -35,6 +35,8 @@ #include #include #include +#include +#include #include #include @@ -44,7 +46,6 @@ #include #include -#include #include static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ @@ -335,8 +336,8 @@ struct snd_amd7930 { int pgain; int mgain; + struct of_device *op; unsigned int irq; - unsigned int regs_size; struct snd_amd7930 *next; }; @@ -920,13 +921,16 @@ static int __devinit snd_amd7930_mixer(struct snd_amd7930 *amd) static int snd_amd7930_free(struct snd_amd7930 *amd) { + struct of_device *op = amd->op; + amd7930_idle(amd); if (amd->irq) free_irq(amd->irq, amd); if (amd->regs) - sbus_iounmap(amd->regs, amd->regs_size); + of_iounmap(&op->resource[0], amd->regs, + resource_size(&op->resource[0])); kfree(amd); @@ -945,13 +949,12 @@ static struct snd_device_ops snd_amd7930_dev_ops = { }; static int __devinit snd_amd7930_create(struct snd_card *card, - struct resource *rp, - unsigned int reg_size, + struct of_device *op, int irq, int dev, struct snd_amd7930 **ramd) { - unsigned long flags; struct snd_amd7930 *amd; + unsigned long flags; int err; *ramd = NULL; @@ -961,9 +964,10 @@ static int __devinit snd_amd7930_create(struct snd_card *card, spin_lock_init(&amd->lock); amd->card = card; - amd->regs_size = reg_size; + amd->op = op; - amd->regs = sbus_ioremap(rp, 0, amd->regs_size, "amd7930"); + amd->regs = of_ioremap(&op->resource[0], 0, + resource_size(&op->resource[0]), "amd7930"); if (!amd->regs) { snd_printk("amd7930-%d: Unable to map chip registers.\n", dev); return -EIO; @@ -1012,12 +1016,15 @@ static int __devinit snd_amd7930_create(struct snd_card *card, return 0; } -static int __devinit amd7930_attach_common(struct resource *rp, int irq) +static int __devinit amd7930_sbus_probe(struct of_device *op, const struct of_device_id *match) { + struct resource *rp = &op->resource[0]; static int dev_num; struct snd_card *card; struct snd_amd7930 *amd; - int err; + int err, irq; + + irq = op->irqs[0]; if (dev_num >= SNDRV_CARDS) return -ENODEV; @@ -1038,8 +1045,7 @@ static int __devinit amd7930_attach_common(struct resource *rp, int irq) (unsigned long long)rp->start, irq); - if ((err = snd_amd7930_create(card, rp, - (rp->end - rp->start) + 1, + if ((err = snd_amd7930_create(card, op, irq, dev_num, &amd)) < 0) goto out_err; @@ -1064,42 +1070,6 @@ out_err: return err; } -static int __devinit amd7930_obio_attach(struct device_node *dp) -{ - const struct linux_prom_registers *regs; - const struct linux_prom_irqs *irqp; - struct resource res, *rp; - int len; - - irqp = of_get_property(dp, "intr", &len); - if (!irqp) { - snd_printk("%s: Firmware node lacks IRQ property.\n", - dp->full_name); - return -ENODEV; - } - - regs = of_get_property(dp, "reg", &len); - if (!regs) { - snd_printk("%s: Firmware node lacks register property.\n", - dp->full_name); - return -ENODEV; - } - - rp = &res; - rp->start = regs->phys_addr; - rp->end = rp->start + regs->reg_size - 1; - rp->flags = IORESOURCE_IO | (regs->which_io & 0xff); - - return amd7930_attach_common(rp, irqp->pri); -} - -static int __devinit amd7930_sbus_probe(struct of_device *dev, const struct of_device_id *match) -{ - struct sbus_dev *sdev = to_sbus_device(&dev->dev); - - return amd7930_attach_common(&sdev->resource[0], sdev->irqs[0]); -} - static struct of_device_id amd7930_match[] = { { .name = "audio", @@ -1115,20 +1085,7 @@ static struct of_platform_driver amd7930_sbus_driver = { static int __init amd7930_init(void) { - struct device_node *dp; - - /* Try to find the sun4c "audio" node first. */ - dp = of_find_node_by_path("/"); - dp = dp->child; - while (dp) { - if (!strcmp(dp->name, "audio")) - amd7930_obio_attach(dp); - - dp = dp->sibling; - } - - /* Probe each SBUS for amd7930 chips. */ - return of_register_driver(&amd7930_sbus_driver, &sbus_bus_type); + return of_register_driver(&amd7930_sbus_driver, &of_bus_type); } static void __exit amd7930_exit(void) -- cgit v1.2.3 From 2bd320f89d417c370f4cf45470d102332c25ea24 Mon Sep 17 00:00:00 2001 From: "David S. Miller" Date: Wed, 27 Aug 2008 00:30:59 -0700 Subject: dbri: Convert to pure OF driver. Signed-off-by: David S. Miller --- sound/sparc/dbri.c | 59 +++++++++++++++++++++++++----------------------------- 1 file changed, 27 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 5242ecbb91d..446f985b76f 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -67,7 +67,7 @@ #include #include -#include +#include #include MODULE_AUTHOR("Rudolf Koenig, Brent Baccala and Martin Habets"); @@ -298,7 +298,7 @@ struct dbri_streaminfo { /* This structure holds the information for both chips (DBRI & CS4215) */ struct snd_dbri { int regs_size, irq; /* Needed for unload */ - struct sbus_dev *sdev; /* SBUS device info */ + struct of_device *op; /* OF device info */ spinlock_t lock; struct dbri_dma *dma; /* Pointer to our DMA block */ @@ -2099,7 +2099,7 @@ static int snd_dbri_hw_params(struct snd_pcm_substream *substream, direction = DMA_FROM_DEVICE; info->dvma_buffer = - dma_map_single(&dbri->sdev->ofdev.dev, + dma_map_single(&dbri->op->dev, runtime->dma_area, params_buffer_bytes(hw_params), direction); @@ -2127,7 +2127,7 @@ static int snd_dbri_hw_free(struct snd_pcm_substream *substream) else direction = DMA_FROM_DEVICE; - dma_unmap_single(&dbri->sdev->ofdev.dev, info->dvma_buffer, + dma_unmap_single(&dbri->op->dev, info->dvma_buffer, substream->runtime->buffer_size, direction); info->dvma_buffer = 0; } @@ -2516,17 +2516,17 @@ static void __devinit snd_dbri_proc(struct snd_card *card) static void snd_dbri_free(struct snd_dbri *dbri); static int __devinit snd_dbri_create(struct snd_card *card, - struct sbus_dev *sdev, - int irq, int dev) + struct of_device *op, + int irq, int dev) { struct snd_dbri *dbri = card->private_data; int err; spin_lock_init(&dbri->lock); - dbri->sdev = sdev; + dbri->op = op; dbri->irq = irq; - dbri->dma = dma_alloc_coherent(&sdev->ofdev.dev, + dbri->dma = dma_alloc_coherent(&op->dev, sizeof(struct dbri_dma), &dbri->dma_dvma, GFP_ATOMIC); memset((void *)dbri->dma, 0, sizeof(struct dbri_dma)); @@ -2535,12 +2535,12 @@ static int __devinit snd_dbri_create(struct snd_card *card, dbri->dma, dbri->dma_dvma); /* Map the registers into memory. */ - dbri->regs_size = sdev->reg_addrs[0].reg_size; - dbri->regs = sbus_ioremap(&sdev->resource[0], 0, - dbri->regs_size, "DBRI Registers"); + dbri->regs_size = resource_size(&op->resource[0]); + dbri->regs = of_ioremap(&op->resource[0], 0, + dbri->regs_size, "DBRI Registers"); if (!dbri->regs) { printk(KERN_ERR "DBRI: could not allocate registers\n"); - dma_free_coherent(&sdev->ofdev.dev, sizeof(struct dbri_dma), + dma_free_coherent(&op->dev, sizeof(struct dbri_dma), (void *)dbri->dma, dbri->dma_dvma); return -EIO; } @@ -2549,8 +2549,8 @@ static int __devinit snd_dbri_create(struct snd_card *card, "DBRI audio", dbri); if (err) { printk(KERN_ERR "DBRI: Can't get irq %d\n", dbri->irq); - sbus_iounmap(dbri->regs, dbri->regs_size); - dma_free_coherent(&sdev->ofdev.dev, sizeof(struct dbri_dma), + of_iounmap(&op->resource[0], dbri->regs, dbri->regs_size); + dma_free_coherent(&op->dev, sizeof(struct dbri_dma), (void *)dbri->dma, dbri->dma_dvma); return err; } @@ -2575,28 +2575,23 @@ static void snd_dbri_free(struct snd_dbri *dbri) free_irq(dbri->irq, dbri); if (dbri->regs) - sbus_iounmap(dbri->regs, dbri->regs_size); + of_iounmap(&dbri->op->resource[0], dbri->regs, dbri->regs_size); if (dbri->dma) - dma_free_coherent(&dbri->sdev->ofdev.dev, + dma_free_coherent(&dbri->op->dev, sizeof(struct dbri_dma), (void *)dbri->dma, dbri->dma_dvma); } -static int __devinit dbri_probe(struct of_device *of_dev, - const struct of_device_id *match) +static int __devinit dbri_probe(struct of_device *op, const struct of_device_id *match) { - struct sbus_dev *sdev = to_sbus_device(&of_dev->dev); struct snd_dbri *dbri; - int irq; struct resource *rp; struct snd_card *card; static int dev = 0; + int irq; int err; - dprintk(D_GEN, "DBRI: Found %s in SBUS slot %d\n", - sdev->prom_name, sdev->slot); - if (dev >= SNDRV_CARDS) return -ENODEV; if (!enable[dev]) { @@ -2604,7 +2599,7 @@ static int __devinit dbri_probe(struct of_device *of_dev, return -ENOENT; } - irq = sdev->irqs[0]; + irq = op->irqs[0]; if (irq <= 0) { printk(KERN_ERR "DBRI-%d: No IRQ.\n", dev); return -ENODEV; @@ -2617,12 +2612,12 @@ static int __devinit dbri_probe(struct of_device *of_dev, strcpy(card->driver, "DBRI"); strcpy(card->shortname, "Sun DBRI"); - rp = &sdev->resource[0]; + rp = &op->resource[0]; sprintf(card->longname, "%s at 0x%02lx:0x%016Lx, irq %d", card->shortname, rp->flags & 0xffL, (unsigned long long)rp->start, irq); - err = snd_dbri_create(card, sdev, irq, dev); + err = snd_dbri_create(card, op, irq, dev); if (err < 0) { snd_card_free(card); return err; @@ -2639,7 +2634,7 @@ static int __devinit dbri_probe(struct of_device *of_dev, /* /proc file handling */ snd_dbri_proc(card); - dev_set_drvdata(&of_dev->dev, card); + dev_set_drvdata(&op->dev, card); err = snd_card_register(card); if (err < 0) @@ -2647,7 +2642,7 @@ static int __devinit dbri_probe(struct of_device *of_dev, printk(KERN_INFO "audio%d at %p (irq %d) is DBRI(%c)+CS4215(%d)\n", dev, dbri->regs, - dbri->irq, sdev->prom_name[9], dbri->mm.version); + dbri->irq, op->node->name[9], dbri->mm.version); dev++; return 0; @@ -2658,14 +2653,14 @@ _err: return err; } -static int __devexit dbri_remove(struct of_device *dev) +static int __devexit dbri_remove(struct of_device *op) { - struct snd_card *card = dev_get_drvdata(&dev->dev); + struct snd_card *card = dev_get_drvdata(&op->dev); snd_dbri_free(card->private_data); snd_card_free(card); - dev_set_drvdata(&dev->dev, NULL); + dev_set_drvdata(&op->dev, NULL); return 0; } @@ -2692,7 +2687,7 @@ static struct of_platform_driver dbri_sbus_driver = { /* Probe for the dbri chip and then attach the driver. */ static int __init dbri_init(void) { - return of_register_driver(&dbri_sbus_driver, &sbus_bus_type); + return of_register_driver(&dbri_sbus_driver, &of_bus_type); } static void __exit dbri_exit(void) -- cgit v1.2.3 From 12b1c03df10da61e222b86f0e1ece9cdb2a909d6 Mon Sep 17 00:00:00 2001 From: "David S. Miller" Date: Wed, 27 Aug 2008 00:31:36 -0700 Subject: cs4231: Use SNDRV_DMA_TYPE_DEV instead of SNDRV_DMA_TYPE_SBUS. SBUS layer now uses dma_*() interfaces, no need for special DMA code any longer. Signed-off-by: David S. Miller --- sound/sparc/cs4231.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index 1c4797be72e..e2be131723e 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -1784,9 +1784,9 @@ static unsigned int sbus_dma_addr(struct cs4231_dma_control *dma_cont) static void sbus_dma_preallocate(struct snd_cs4231 *chip, struct snd_pcm *pcm) { - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_SBUS, - snd_dma_sbus_data(chip->dev_u.sdev), - 64 * 1024, 128 * 1024); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + &chip->dev_u.sdev->ofdev.dev, + 64 * 1024, 128 * 1024); } /* -- cgit v1.2.3 From 759ee81be6d87c150ea2b300c221b4fec8b5f646 Mon Sep 17 00:00:00 2001 From: "David S. Miller" Date: Wed, 27 Aug 2008 00:33:26 -0700 Subject: alsa: Remove special SBUS dma support code. No longer used. Signed-off-by: David S. Miller --- sound/core/memalloc.c | 50 +------------------------------------------------- 1 file changed, 1 insertion(+), 49 deletions(-) (limited to 'sound') diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index ccaaac45faf..3733351a27f 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -33,9 +33,6 @@ #include #include #include -#ifdef CONFIG_SBUS -#include -#endif MODULE_AUTHOR("Takashi Iwai , Jaroslav Kysela "); @@ -180,41 +177,6 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, } #endif /* CONFIG_HAS_DMA */ -#ifdef CONFIG_SBUS - -static void *snd_malloc_sbus_pages(struct device *dev, size_t size, - dma_addr_t *dma_addr) -{ - struct sbus_dev *sdev = (struct sbus_dev *)dev; - int pg; - void *res; - - snd_assert(size > 0, return NULL); - snd_assert(dma_addr != NULL, return NULL); - pg = get_order(size); - res = dma_alloc_coherent(&sdev->ofdev.dev, PAGE_SIZE * (1 << pg), - dma_addr, GFP_ATOMIC); - if (res != NULL) - inc_snd_pages(pg); - return res; -} - -static void snd_free_sbus_pages(struct device *dev, size_t size, - void *ptr, dma_addr_t dma_addr) -{ - struct sbus_dev *sdev = (struct sbus_dev *)dev; - int pg; - - if (ptr == NULL) - return; - pg = get_order(size); - dec_snd_pages(pg); - dma_free_coherent(&sdev->ofdev.dev, PAGE_SIZE * (1 << pg), - ptr, dma_addr); -} - -#endif /* CONFIG_SBUS */ - /* * * ALSA generic memory management @@ -249,11 +211,6 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->area = snd_malloc_pages(size, (unsigned long)device); dmab->addr = 0; break; -#ifdef CONFIG_SBUS - case SNDRV_DMA_TYPE_SBUS: - dmab->area = snd_malloc_sbus_pages(device, size, &dmab->addr); - break; -#endif #ifdef CONFIG_HAS_DMA case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); @@ -322,11 +279,6 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) case SNDRV_DMA_TYPE_CONTINUOUS: snd_free_pages(dmab->area, dmab->bytes); break; -#ifdef CONFIG_SBUS - case SNDRV_DMA_TYPE_SBUS: - snd_free_sbus_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); - break; -#endif #ifdef CONFIG_HAS_DMA case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); @@ -433,7 +385,7 @@ static int snd_mem_proc_read(struct seq_file *seq, void *offset) long pages = snd_allocated_pages >> (PAGE_SHIFT-12); struct snd_mem_list *mem; int devno; - static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG", "SBUS" }; + static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG" }; mutex_lock(&list_mutex); seq_printf(seq, "pages : %li bytes (%li pages per %likB)\n", -- cgit v1.2.3 From ae251031ad22a659b8261440b049d5981e57880c Mon Sep 17 00:00:00 2001 From: "David S. Miller" Date: Wed, 27 Aug 2008 18:24:01 -0700 Subject: cs4231: Convert SBUS side to OF driver. Signed-off-by: David S. Miller --- sound/sparc/cs4231.c | 78 +++++++++++++++++++++++++++++++++------------------- 1 file changed, 50 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index e2be131723e..271b0420f8b 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -1,6 +1,6 @@ /* * Driver for CS4231 sound chips found on Sparcs. - * Copyright (C) 2002 David S. Miller + * Copyright (C) 2002, 2008 David S. Miller * * Based entirely upon drivers/sbus/audio/cs4231.c which is: * Copyright (C) 1996, 1997, 1998 Derrick J Brashear (shadow@andrew.cmu.edu) @@ -17,7 +17,8 @@ #include #include #include - +#include +#include #include #include @@ -29,7 +30,6 @@ #ifdef CONFIG_SBUS #define SBUS_SUPPORT -#include #endif #if defined(CONFIG_PCI) && defined(CONFIG_SPARC64) @@ -116,7 +116,7 @@ struct snd_cs4231 { union { #ifdef SBUS_SUPPORT - struct sbus_dev *sdev; + struct of_device *op; #endif #ifdef EBUS_SUPPORT struct pci_dev *pdev; @@ -1785,7 +1785,7 @@ static unsigned int sbus_dma_addr(struct cs4231_dma_control *dma_cont) static void sbus_dma_preallocate(struct snd_cs4231 *chip, struct snd_pcm *pcm) { snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - &chip->dev_u.sdev->ofdev.dev, + &chip->dev_u.op->dev, 64 * 1024, 128 * 1024); } @@ -1795,11 +1795,13 @@ static void sbus_dma_preallocate(struct snd_cs4231 *chip, struct snd_pcm *pcm) static int snd_cs4231_sbus_free(struct snd_cs4231 *chip) { + struct of_device *op = chip->dev_u.op; + if (chip->irq[0]) free_irq(chip->irq[0], chip); if (chip->port) - sbus_iounmap(chip->port, chip->regs_size); + of_iounmap(&op->resource[0], chip->port, chip->regs_size); return 0; } @@ -1816,7 +1818,7 @@ static struct snd_device_ops snd_cs4231_sbus_dev_ops = { }; static int __init snd_cs4231_sbus_create(struct snd_card *card, - struct sbus_dev *sdev, + struct of_device *op, int dev) { struct snd_cs4231 *chip = card->private_data; @@ -1827,13 +1829,13 @@ static int __init snd_cs4231_sbus_create(struct snd_card *card, spin_lock_init(&chip->p_dma.sbus_info.lock); mutex_init(&chip->mce_mutex); mutex_init(&chip->open_mutex); - chip->dev_u.sdev = sdev; - chip->regs_size = sdev->reg_addrs[0].reg_size; + chip->dev_u.op = op; + chip->regs_size = resource_size(&op->resource[0]); memcpy(&chip->image, &snd_cs4231_original_image, sizeof(snd_cs4231_original_image)); - chip->port = sbus_ioremap(&sdev->resource[0], 0, - chip->regs_size, "cs4231"); + chip->port = of_ioremap(&op->resource[0], 0, + chip->regs_size, "cs4231"); if (!chip->port) { snd_printdd("cs4231-%d: Unable to map chip registers.\n", dev); return -EIO; @@ -1856,14 +1858,14 @@ static int __init snd_cs4231_sbus_create(struct snd_card *card, chip->c_dma.address = sbus_dma_addr; chip->c_dma.preallocate = sbus_dma_preallocate; - if (request_irq(sdev->irqs[0], snd_cs4231_sbus_interrupt, + if (request_irq(op->irqs[0], snd_cs4231_sbus_interrupt, IRQF_SHARED, "cs4231", chip)) { snd_printdd("cs4231-%d: Unable to grab SBUS IRQ %d\n", - dev, sdev->irqs[0]); + dev, op->irqs[0]); snd_cs4231_sbus_free(chip); return -EBUSY; } - chip->irq[0] = sdev->irqs[0]; + chip->irq[0] = op->irqs[0]; if (snd_cs4231_probe(chip) < 0) { snd_cs4231_sbus_free(chip); @@ -1880,12 +1882,16 @@ static int __init snd_cs4231_sbus_create(struct snd_card *card, return 0; } -static int __init cs4231_sbus_attach(struct sbus_dev *sdev) +static int __devinit cs4231_probe(struct of_device *op, const struct of_device_id *match) { - struct resource *rp = &sdev->resource[0]; + struct resource *rp = &op->resource[0]; struct snd_card *card; int err; + if (strcmp(op->node->parent->name, "sbus") && + strcmp(op->node->parent->name, "sbi")) + return -ENODEV; + err = cs4231_attach_begin(&card); if (err) return err; @@ -1894,9 +1900,9 @@ static int __init cs4231_sbus_attach(struct sbus_dev *sdev) card->shortname, rp->flags & 0xffL, (unsigned long long)rp->start, - sdev->irqs[0]); + op->irqs[0]); - err = snd_cs4231_sbus_create(card, sdev, dev); + err = snd_cs4231_sbus_create(card, op, dev); if (err < 0) { snd_card_free(card); return err; @@ -2101,12 +2107,25 @@ static int __init cs4231_ebus_attach(struct linux_ebus_device *edev) } #endif -static int __init cs4231_init(void) -{ #ifdef SBUS_SUPPORT - struct sbus_bus *sbus; - struct sbus_dev *sdev; +static struct of_device_id cs4231_match[] = { + { + .name = "SUNW,CS4231", + }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs4231_match); + +static struct of_platform_driver cs4231_driver = { + .name = "audio", + .match_table = cs4231_match, + .probe = cs4231_probe, +}; #endif + +static int __init cs4231_init(void) +{ #ifdef EBUS_SUPPORT struct linux_ebus *ebus; struct linux_ebus_device *edev; @@ -2116,11 +2135,10 @@ static int __init cs4231_init(void) found = 0; #ifdef SBUS_SUPPORT - for_all_sbusdev(sdev, sbus) { - if (!strcmp(sdev->prom_name, "SUNW,CS4231")) { - if (cs4231_sbus_attach(sdev) == 0) - found++; - } + { + int err = of_register_driver(&cs4231_driver, &of_bus_type); + if (err) + return err; } #endif #ifdef EBUS_SUPPORT @@ -2147,13 +2165,17 @@ static int __init cs4231_init(void) #endif - return (found > 0) ? 0 : -EIO; + return 0; } static void __exit cs4231_exit(void) { struct snd_cs4231 *p = cs4231_list; +#ifdef SBUS_SUPPORT + of_unregister_driver(&cs4231_driver); +#endif + while (p != NULL) { struct snd_cs4231 *next = p->next; -- cgit v1.2.3 From aae7fb87ec4d2df6cb551670b1765cf4e5795a3b Mon Sep 17 00:00:00 2001 From: "David S. Miller" Date: Fri, 29 Aug 2008 23:10:21 -0700 Subject: sparc: Move EBUS DMA interfaces into seperate header file. These have no dependencies on the EBUS probing layer, the clients setup the registers and all of those details. The EBUS DMA layer just programs and manages the DMA controller found in EBUS. Signed-off-by: David S. Miller --- sound/sparc/cs4231.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index 271b0420f8b..e6dba4f7169 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -36,6 +36,7 @@ #define EBUS_SUPPORT #include #include +#include #endif static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ -- cgit v1.2.3 From afc88ad6b38797aeec4d635140099ace077df587 Mon Sep 17 00:00:00 2001 From: "David S. Miller" Date: Sat, 30 Aug 2008 00:13:55 -0700 Subject: cs4231: Convert to EBUS side to pure OF driver. Signed-off-by: David S. Miller --- sound/sparc/cs4231.c | 172 ++++++++++++++++----------------------------------- 1 file changed, 54 insertions(+), 118 deletions(-) (limited to 'sound') diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index e6dba4f7169..cdbfae96bd8 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -35,7 +35,6 @@ #if defined(CONFIG_PCI) && defined(CONFIG_SPARC64) #define EBUS_SUPPORT #include -#include #include #endif @@ -71,8 +70,6 @@ struct cs4231_dma_control { int (*request)(struct cs4231_dma_control *dma_cont, dma_addr_t bus_addr, size_t len); unsigned int (*address)(struct cs4231_dma_control *dma_cont); - void (*preallocate)(struct snd_cs4231 *chip, - struct snd_pcm *pcm); #ifdef EBUS_SUPPORT struct ebus_dma_info ebus_info; #endif @@ -115,21 +112,12 @@ struct snd_cs4231 { struct mutex mce_mutex; /* mutex for mce register */ struct mutex open_mutex; /* mutex for ALSA open/close */ - union { -#ifdef SBUS_SUPPORT - struct of_device *op; -#endif -#ifdef EBUS_SUPPORT - struct pci_dev *pdev; -#endif - } dev_u; + struct of_device *op; unsigned int irq[2]; unsigned int regs_size; struct snd_cs4231 *next; }; -static struct snd_cs4231 *cs4231_list; - /* Eventually we can use sound/isa/cs423x/cs4231_lib.c directly, but for * now.... -DaveM */ @@ -268,27 +256,19 @@ static unsigned char snd_cs4231_original_image[32] = static u8 __cs4231_readb(struct snd_cs4231 *cp, void __iomem *reg_addr) { -#ifdef EBUS_SUPPORT if (cp->flags & CS4231_FLAG_EBUS) return readb(reg_addr); else -#endif -#ifdef SBUS_SUPPORT return sbus_readb(reg_addr); -#endif } static void __cs4231_writeb(struct snd_cs4231 *cp, u8 val, void __iomem *reg_addr) { -#ifdef EBUS_SUPPORT if (cp->flags & CS4231_FLAG_EBUS) return writeb(val, reg_addr); else -#endif -#ifdef SBUS_SUPPORT return sbus_writeb(val, reg_addr); -#endif } /* @@ -1259,7 +1239,9 @@ static int __init snd_cs4231_pcm(struct snd_card *card) pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX; strcpy(pcm->name, "CS4231"); - chip->p_dma.preallocate(chip, pcm); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + &chip->op->dev, + 64 * 1024, 128 * 1024); chip->pcm = pcm; @@ -1627,8 +1609,7 @@ static int __init cs4231_attach_finish(struct snd_card *card) if (err < 0) goto out_err; - chip->next = cs4231_list; - cs4231_list = chip; + dev_set_drvdata(&chip->op->dev, chip); dev++; return 0; @@ -1783,20 +1764,13 @@ static unsigned int sbus_dma_addr(struct cs4231_dma_control *dma_cont) return sbus_readl(base->regs + base->dir + APCVA); } -static void sbus_dma_preallocate(struct snd_cs4231 *chip, struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - &chip->dev_u.op->dev, - 64 * 1024, 128 * 1024); -} - /* * Init and exit routines */ static int snd_cs4231_sbus_free(struct snd_cs4231 *chip) { - struct of_device *op = chip->dev_u.op; + struct of_device *op = chip->op; if (chip->irq[0]) free_irq(chip->irq[0], chip); @@ -1830,7 +1804,7 @@ static int __init snd_cs4231_sbus_create(struct snd_card *card, spin_lock_init(&chip->p_dma.sbus_info.lock); mutex_init(&chip->mce_mutex); mutex_init(&chip->open_mutex); - chip->dev_u.op = op; + chip->op = op; chip->regs_size = resource_size(&op->resource[0]); memcpy(&chip->image, &snd_cs4231_original_image, sizeof(snd_cs4231_original_image)); @@ -1851,13 +1825,11 @@ static int __init snd_cs4231_sbus_create(struct snd_card *card, chip->p_dma.enable = sbus_dma_enable; chip->p_dma.request = sbus_dma_request; chip->p_dma.address = sbus_dma_addr; - chip->p_dma.preallocate = sbus_dma_preallocate; chip->c_dma.prepare = sbus_dma_prepare; chip->c_dma.enable = sbus_dma_enable; chip->c_dma.request = sbus_dma_request; chip->c_dma.address = sbus_dma_addr; - chip->c_dma.preallocate = sbus_dma_preallocate; if (request_irq(op->irqs[0], snd_cs4231_sbus_interrupt, IRQF_SHARED, "cs4231", chip)) { @@ -1883,16 +1855,12 @@ static int __init snd_cs4231_sbus_create(struct snd_card *card, return 0; } -static int __devinit cs4231_probe(struct of_device *op, const struct of_device_id *match) +static int __devinit cs4231_sbus_probe(struct of_device *op, const struct of_device_id *match) { struct resource *rp = &op->resource[0]; struct snd_card *card; int err; - if (strcmp(op->node->parent->name, "sbus") && - strcmp(op->node->parent->name, "sbi")) - return -ENODEV; - err = cs4231_attach_begin(&card); if (err) return err; @@ -1956,30 +1924,25 @@ static unsigned int _ebus_dma_addr(struct cs4231_dma_control *dma_cont) return ebus_dma_addr(&dma_cont->ebus_info); } -static void _ebus_dma_preallocate(struct snd_cs4231 *chip, struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->dev_u.pdev), - 64*1024, 128*1024); -} - /* * Init and exit routines */ static int snd_cs4231_ebus_free(struct snd_cs4231 *chip) { + struct of_device *op = chip->op; + if (chip->c_dma.ebus_info.regs) { ebus_dma_unregister(&chip->c_dma.ebus_info); - iounmap(chip->c_dma.ebus_info.regs); + of_iounmap(&op->resource[2], chip->c_dma.ebus_info.regs, 0x10); } if (chip->p_dma.ebus_info.regs) { ebus_dma_unregister(&chip->p_dma.ebus_info); - iounmap(chip->p_dma.ebus_info.regs); + of_iounmap(&op->resource[1], chip->p_dma.ebus_info.regs, 0x10); } if (chip->port) - iounmap(chip->port); + of_iounmap(&op->resource[0], chip->port, 0x10); return 0; } @@ -1996,7 +1959,7 @@ static struct snd_device_ops snd_cs4231_ebus_dev_ops = { }; static int __init snd_cs4231_ebus_create(struct snd_card *card, - struct linux_ebus_device *edev, + struct of_device *op, int dev) { struct snd_cs4231 *chip = card->private_data; @@ -2008,35 +1971,35 @@ static int __init snd_cs4231_ebus_create(struct snd_card *card, mutex_init(&chip->mce_mutex); mutex_init(&chip->open_mutex); chip->flags |= CS4231_FLAG_EBUS; - chip->dev_u.pdev = edev->bus->self; + chip->op = op; memcpy(&chip->image, &snd_cs4231_original_image, sizeof(snd_cs4231_original_image)); strcpy(chip->c_dma.ebus_info.name, "cs4231(capture)"); chip->c_dma.ebus_info.flags = EBUS_DMA_FLAG_USE_EBDMA_HANDLER; chip->c_dma.ebus_info.callback = snd_cs4231_ebus_capture_callback; chip->c_dma.ebus_info.client_cookie = chip; - chip->c_dma.ebus_info.irq = edev->irqs[0]; + chip->c_dma.ebus_info.irq = op->irqs[0]; strcpy(chip->p_dma.ebus_info.name, "cs4231(play)"); chip->p_dma.ebus_info.flags = EBUS_DMA_FLAG_USE_EBDMA_HANDLER; chip->p_dma.ebus_info.callback = snd_cs4231_ebus_play_callback; chip->p_dma.ebus_info.client_cookie = chip; - chip->p_dma.ebus_info.irq = edev->irqs[1]; + chip->p_dma.ebus_info.irq = op->irqs[1]; chip->p_dma.prepare = _ebus_dma_prepare; chip->p_dma.enable = _ebus_dma_enable; chip->p_dma.request = _ebus_dma_request; chip->p_dma.address = _ebus_dma_addr; - chip->p_dma.preallocate = _ebus_dma_preallocate; chip->c_dma.prepare = _ebus_dma_prepare; chip->c_dma.enable = _ebus_dma_enable; chip->c_dma.request = _ebus_dma_request; chip->c_dma.address = _ebus_dma_addr; - chip->c_dma.preallocate = _ebus_dma_preallocate; - chip->port = ioremap(edev->resource[0].start, 0x10); - chip->p_dma.ebus_info.regs = ioremap(edev->resource[1].start, 0x10); - chip->c_dma.ebus_info.regs = ioremap(edev->resource[2].start, 0x10); + chip->port = of_ioremap(&op->resource[0], 0, 0x10, "cs4231"); + chip->p_dma.ebus_info.regs = + of_ioremap(&op->resource[1], 0, 0x10, "cs4231_pdma"); + chip->c_dma.ebus_info.regs = + of_ioremap(&op->resource[2], 0, 0x10, "cs4231_cdma"); if (!chip->port || !chip->p_dma.ebus_info.regs || !chip->c_dma.ebus_info.regs) { snd_cs4231_ebus_free(chip); @@ -2084,7 +2047,7 @@ static int __init snd_cs4231_ebus_create(struct snd_card *card, return 0; } -static int __init cs4231_ebus_attach(struct linux_ebus_device *edev) +static int __devinit cs4231_ebus_probe(struct of_device *op, const struct of_device_id *match) { struct snd_card *card; int err; @@ -2095,10 +2058,10 @@ static int __init cs4231_ebus_attach(struct linux_ebus_device *edev) sprintf(card->longname, "%s at 0x%lx, irq %d", card->shortname, - edev->resource[0].start, - edev->irqs[0]); + op->resource[0].start, + op->irqs[0]); - err = snd_cs4231_ebus_create(card, edev, dev); + err = snd_cs4231_ebus_create(card, op, dev); if (err < 0) { snd_card_free(card); return err; @@ -2108,11 +2071,37 @@ static int __init cs4231_ebus_attach(struct linux_ebus_device *edev) } #endif +static int __devinit cs4231_probe(struct of_device *op, const struct of_device_id *match) +{ +#ifdef EBUS_SUPPORT + if (!strcmp(op->node->parent->name, "ebus")) + return cs4231_ebus_probe(op, match); +#endif #ifdef SBUS_SUPPORT + if (!strcmp(op->node->parent->name, "sbus") || + !strcmp(op->node->parent->name, "sbi")) + return cs4231_sbus_probe(op, match); +#endif + return -ENODEV; +} + +static int __devexit cs4231_remove(struct of_device *op) +{ + struct snd_cs4231 *chip = dev_get_drvdata(&op->dev); + + snd_card_free(chip->card); + + return 0; +} + static struct of_device_id cs4231_match[] = { { .name = "SUNW,CS4231", }, + { + .name = "audio", + .compatible = "SUNW,CS4231", + }, {}, }; @@ -2122,70 +2111,17 @@ static struct of_platform_driver cs4231_driver = { .name = "audio", .match_table = cs4231_match, .probe = cs4231_probe, + .remove = __devexit_p(cs4231_remove), }; -#endif static int __init cs4231_init(void) { -#ifdef EBUS_SUPPORT - struct linux_ebus *ebus; - struct linux_ebus_device *edev; -#endif - int found; - - found = 0; - -#ifdef SBUS_SUPPORT - { - int err = of_register_driver(&cs4231_driver, &of_bus_type); - if (err) - return err; - } -#endif -#ifdef EBUS_SUPPORT - for_each_ebus(ebus) { - for_each_ebusdev(edev, ebus) { - int match = 0; - - if (!strcmp(edev->prom_node->name, "SUNW,CS4231")) { - match = 1; - } else if (!strcmp(edev->prom_node->name, "audio")) { - const char *compat; - - compat = of_get_property(edev->prom_node, - "compatible", NULL); - if (compat && !strcmp(compat, "SUNW,CS4231")) - match = 1; - } - - if (match && - cs4231_ebus_attach(edev) == 0) - found++; - } - } -#endif - - - return 0; + return of_register_driver(&cs4231_driver, &of_bus_type); } static void __exit cs4231_exit(void) { - struct snd_cs4231 *p = cs4231_list; - -#ifdef SBUS_SUPPORT of_unregister_driver(&cs4231_driver); -#endif - - while (p != NULL) { - struct snd_cs4231 *next = p->next; - - snd_card_free(p->card); - - p = next; - } - - cs4231_list = NULL; } module_init(cs4231_init); -- cgit v1.2.3 From fd098316ef533e8441576f020ead4beab93154ce Mon Sep 17 00:00:00 2001 From: "David S. Miller" Date: Sun, 31 Aug 2008 01:23:17 -0700 Subject: sparc: Annotate of_device_id arrays with const or __initdata. As suggested by Stephen Rothwell. Signed-off-by: David S. Miller --- sound/sparc/amd7930.c | 2 +- sound/sparc/cs4231.c | 2 +- sound/sparc/dbri.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index 0f82c3f6cfd..5af5503edad 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -1070,7 +1070,7 @@ out_err: return err; } -static struct of_device_id amd7930_match[] = { +static const struct of_device_id amd7930_match[] = { { .name = "audio", }, diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index cdbfae96bd8..727438d276e 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -2094,7 +2094,7 @@ static int __devexit cs4231_remove(struct of_device *op) return 0; } -static struct of_device_id cs4231_match[] = { +static const struct of_device_id cs4231_match[] = { { .name = "SUNW,CS4231", }, diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 446f985b76f..2edb0ad3de7 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2665,7 +2665,7 @@ static int __devexit dbri_remove(struct of_device *op) return 0; } -static struct of_device_id dbri_match[] = { +static const struct of_device_id dbri_match[] = { { .name = "SUNW,DBRIe", }, -- cgit v1.2.3 From 5c73a7d0411999e3cb3c6d64225450813738ae25 Mon Sep 17 00:00:00 2001 From: Arjan van de Ven Date: Mon, 1 Sep 2008 15:25:20 -0700 Subject: hrtimer: convert sound/ to the new hrtimer apis In order to be able to do range hrtimers we need to use accessor functions to the "expire" member of the hrtimer struct. This patch converts sound/ to these accessors. Signed-off-by: Arjan van de Ven --- sound/drivers/pcsp/pcsp_lib.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index e341f3f83b6..1f42e406311 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -34,7 +34,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) chip->thalf = 0; if (!atomic_read(&chip->timer_active)) return HRTIMER_NORESTART; - hrtimer_forward(&chip->timer, chip->timer.expires, + hrtimer_forward(&chip->timer, hrtimer_get_expires(&chip->timer), ktime_set(0, chip->ns_rem)); return HRTIMER_RESTART; } @@ -118,7 +118,8 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) chip->ns_rem = PCSP_PERIOD_NS(); ns = (chip->thalf ? PCSP_CALC_NS(timer_cnt) : chip->ns_rem); chip->ns_rem -= ns; - hrtimer_forward(&chip->timer, chip->timer.expires, ktime_set(0, ns)); + hrtimer_forward(&chip->timer, hrtimer_get_expires(&chip->timer), + ktime_set(0, ns)); return HRTIMER_RESTART; exit_nr_unlock2: -- cgit v1.2.3 From 6168cda927ecdf3347537e0d01618ca2a2272007 Mon Sep 17 00:00:00 2001 From: Eric Miao Date: Fri, 5 Sep 2008 18:15:22 +0800 Subject: [ARM] pxa/corgi: use generic GPIO API for SCOOP GPIOs Original patch from Dmitry Baryshkov's inital scoop gpio conversion work at http://git.infradead.org/users/dbaryshkov/zaurus-2.6.git. Signed-off-by: Dmitry Baryshkov Signed-off-by: Eric Miao Signed-off-by: Russell King --- sound/soc/pxa/corgi.c | 34 +++++++++++++--------------------- 1 file changed, 13 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 0a53f72077f..fa69faa886e 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -18,13 +18,13 @@ #include #include #include +#include #include #include #include #include #include -#include #include #include #include @@ -54,8 +54,8 @@ static void corgi_ext_control(struct snd_soc_codec *codec) switch (corgi_jack_func) { case CORGI_HP: /* set = unmute headphone */ - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + gpio_set_value(CORGI_GPIO_MUTE_L, 1); + gpio_set_value(CORGI_GPIO_MUTE_R, 1); snd_soc_dapm_disable_pin(codec, "Mic Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); snd_soc_dapm_enable_pin(codec, "Headphone Jack"); @@ -63,24 +63,24 @@ static void corgi_ext_control(struct snd_soc_codec *codec) break; case CORGI_MIC: /* reset = mute headphone */ - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + gpio_set_value(CORGI_GPIO_MUTE_L, 0); + gpio_set_value(CORGI_GPIO_MUTE_R, 0); snd_soc_dapm_enable_pin(codec, "Mic Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); snd_soc_dapm_disable_pin(codec, "Headphone Jack"); snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_LINE: - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + gpio_set_value(CORGI_GPIO_MUTE_L, 0); + gpio_set_value(CORGI_GPIO_MUTE_R, 0); snd_soc_dapm_disable_pin(codec, "Mic Jack"); snd_soc_dapm_enable_pin(codec, "Line Jack"); snd_soc_dapm_disable_pin(codec, "Headphone Jack"); snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_HEADSET: - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + gpio_set_value(CORGI_GPIO_MUTE_L, 0); + gpio_set_value(CORGI_GPIO_MUTE_R, 1); snd_soc_dapm_enable_pin(codec, "Mic Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); snd_soc_dapm_disable_pin(codec, "Headphone Jack"); @@ -114,8 +114,8 @@ static int corgi_shutdown(struct snd_pcm_substream *substream) struct snd_soc_codec *codec = rtd->socdev->codec; /* set = unmute headphone */ - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + gpio_set_value(CORGI_GPIO_MUTE_L, 1); + gpio_set_value(CORGI_GPIO_MUTE_R, 1); return 0; } @@ -218,22 +218,14 @@ static int corgi_set_spk(struct snd_kcontrol *kcontrol, static int corgi_amp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - if (SND_SOC_DAPM_EVENT_ON(event)) - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON); - else - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON); - + gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event)); return 0; } static int corgi_mic_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - if (SND_SOC_DAPM_EVENT_ON(event)) - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS); - else - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS); - + gpio_set_value(CORGI_GPIO_MIC_BIAS, SND_SOC_DAPM_EVENT_ON(event)); return 0; } -- cgit v1.2.3 From fff147208b48680cb7b627a144113a6585828a0e Mon Sep 17 00:00:00 2001 From: Eric Miao Date: Fri, 5 Sep 2008 22:15:23 +0800 Subject: [ARM] pxa/spitz: use generic GPIO API for SCOOP1/SCOOP2 GPIOs Original patch from Dmitry Baryshkov's inital scoop gpio conversion work at http://git.infradead.org/users/dbaryshkov/zaurus-2.6.git. Signed-off-by: Dmitry Baryshkov Signed-off-by: Eric Miao Signed-off-by: Russell King --- sound/soc/pxa/spitz.c | 33 ++++++++++++++------------------- 1 file changed, 14 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 37cb768fc93..acfa712844e 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -19,13 +19,13 @@ #include #include #include +#include #include #include #include #include #include -#include #include #include #include @@ -63,8 +63,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_disable_pin(codec, "Mic Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); snd_soc_dapm_enable_pin(codec, "Headphone Jack"); - set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); - set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); + gpio_set_value(SPITZ_GPIO_MUTE_L, 1); + gpio_set_value(SPITZ_GPIO_MUTE_R, 1); break; case SPITZ_MIC: /* enable mic jack and bias, mute hp */ @@ -72,8 +72,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_disable_pin(codec, "Headset Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); snd_soc_dapm_enable_pin(codec, "Mic Jack"); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); + gpio_set_value(SPITZ_GPIO_MUTE_L, 0); + gpio_set_value(SPITZ_GPIO_MUTE_R, 0); break; case SPITZ_LINE: /* enable line jack, disable mic bias and mute hp */ @@ -81,8 +81,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_disable_pin(codec, "Headset Jack"); snd_soc_dapm_disable_pin(codec, "Mic Jack"); snd_soc_dapm_enable_pin(codec, "Line Jack"); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); + gpio_set_value(SPITZ_GPIO_MUTE_L, 0); + gpio_set_value(SPITZ_GPIO_MUTE_R, 0); break; case SPITZ_HEADSET: /* enable and unmute headset jack enable mic bias, mute L hp */ @@ -90,8 +90,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_enable_pin(codec, "Mic Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); snd_soc_dapm_enable_pin(codec, "Headset Jack"); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); - set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); + gpio_set_value(SPITZ_GPIO_MUTE_L, 0); + gpio_set_value(SPITZ_GPIO_MUTE_R, 1); break; case SPITZ_HP_OFF: @@ -100,8 +100,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_disable_pin(codec, "Headset Jack"); snd_soc_dapm_disable_pin(codec, "Mic Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); + gpio_set_value(SPITZ_GPIO_MUTE_L, 0); + gpio_set_value(SPITZ_GPIO_MUTE_R, 0); break; } snd_soc_dapm_sync(codec); @@ -215,14 +215,9 @@ static int spitz_set_spk(struct snd_kcontrol *kcontrol, static int spitz_mic_bias(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - if (machine_is_borzoi() || machine_is_spitz()) { - if (SND_SOC_DAPM_EVENT_ON(event)) - set_scoop_gpio(&spitzscoop2_device.dev, - SPITZ_SCP2_MIC_BIAS); - else - reset_scoop_gpio(&spitzscoop2_device.dev, - SPITZ_SCP2_MIC_BIAS); - } + if (machine_is_borzoi() || machine_is_spitz()) + gpio_set_value(SPITZ_GPIO_MIC_BIAS, + SND_SOC_DAPM_EVENT_ON(event)); if (machine_is_akita()) { if (SND_SOC_DAPM_EVENT_ON(event)) -- cgit v1.2.3 From f72de6638b8e55283739de174b57c0ae4203c446 Mon Sep 17 00:00:00 2001 From: Eric Miao Date: Sat, 6 Sep 2008 08:46:23 +0800 Subject: [ARM] pxa/akita: use pca953x instead of akita-ioexp Use generic pca953x which provides gpiolib interface instead of akita-specific akita-ioexp with non-standard interface to pins. Signed-off-by: Dmitry Baryshkov Cc: Richard Purdie Cc: Mark Brown Signed-off-by: Eric Miao Signed-off-by: Russell King --- sound/soc/pxa/spitz.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index acfa712844e..b89a3edd218 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -28,7 +28,6 @@ #include #include #include -#include #include #include "../codecs/wm8750.h" #include "pxa2xx-pcm.h" @@ -219,14 +218,10 @@ static int spitz_mic_bias(struct snd_soc_dapm_widget *w, gpio_set_value(SPITZ_GPIO_MIC_BIAS, SND_SOC_DAPM_EVENT_ON(event)); - if (machine_is_akita()) { - if (SND_SOC_DAPM_EVENT_ON(event)) - akita_set_ioexp(&akitaioexp_device.dev, - AKITA_IOEXP_MIC_BIAS); - else - akita_reset_ioexp(&akitaioexp_device.dev, - AKITA_IOEXP_MIC_BIAS); - } + if (machine_is_akita()) + gpio_set_value(AKITA_GPIO_MIC_BIAS, + SND_SOC_DAPM_EVENT_ON(event)); + return 0; } -- cgit v1.2.3 From a3927471c65b5ca94da853a1a46ec5db4609c502 Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 25 Sep 2008 13:30:07 +0100 Subject: [ARM] pxa: avoid needless pxa_gpio_mode() during resume The pin configurations are restored early on during resume. There's no need for drivers to re-affirm the gpio modes. Signed-off-by: Russell King --- sound/soc/pxa/pxa2xx-ac97.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index d94a495bd6b..7d7ce164836 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -293,14 +293,6 @@ static int pxa2xx_ac97_suspend(struct platform_device *pdev, static int pxa2xx_ac97_resume(struct platform_device *pdev, struct snd_soc_dai *dai) { - pxa_gpio_mode(GPIO31_SYNC_AC97_MD); - pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); - pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); - pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); -#ifdef CONFIG_PXA27x - /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); -#endif clk_enable(ac97_clk); return 0; } -- cgit v1.2.3 From 87f3dd77974cba1ba0798abd741ede50f56b3eb3 Mon Sep 17 00:00:00 2001 From: Eric Miao Date: Mon, 8 Sep 2008 15:26:43 +0800 Subject: [ARM] pxa: simplify DMA register definitions 1. DRCMRxx is no longer recommended, use DRCMR(xx) instead, and pass DRCMR index by "struct resource" if possible 2. DCSRxx, DDADRxx, DSADRxx, DTADRxx, DCMDxx is never used, use DCSR(), DDADR(), DSADR(), DTADR(), DCMD() instead Signed-off-by: Eric Miao Acked-by: Nicolas Pitre Signed-off-by: Russell King --- sound/arm/pxa2xx-ac97.c | 4 ++-- sound/soc/pxa/pxa2xx-ac97.c | 10 +++++----- sound/soc/pxa/pxa2xx-i2s.c | 4 ++-- 3 files changed, 9 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 199cca3366d..714b3baa4be 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -215,7 +215,7 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_out = { .name = "AC97 PCM out", .dev_addr = __PREG(PCDR), - .drcmr = &DRCMRTXPCDR, + .drcmr = &DRCMR(12), .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | DCMD_BURST32 | DCMD_WIDTH4, }; @@ -223,7 +223,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_out = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_in = { .name = "AC97 PCM in", .dev_addr = __PREG(PCDR), - .drcmr = &DRCMRRXPCDR, + .drcmr = &DRCMR(11), .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST32 | DCMD_WIDTH4, }; diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 7d7ce164836..ac8f227bab0 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -244,7 +244,7 @@ struct snd_ac97_bus_ops soc_ac97_ops = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = { .name = "AC97 PCM Stereo out", .dev_addr = __PREG(PCDR), - .drcmr = &DRCMRTXPCDR, + .drcmr = &DRCMR(12), .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | DCMD_BURST32 | DCMD_WIDTH4, }; @@ -252,7 +252,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = { .name = "AC97 PCM Stereo in", .dev_addr = __PREG(PCDR), - .drcmr = &DRCMRRXPCDR, + .drcmr = &DRCMR(11), .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST32 | DCMD_WIDTH4, }; @@ -260,7 +260,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = { .name = "AC97 Aux PCM (Slot 5) Mono out", .dev_addr = __PREG(MODR), - .drcmr = &DRCMRTXMODR, + .drcmr = &DRCMR(10), .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | DCMD_BURST16 | DCMD_WIDTH2, }; @@ -268,7 +268,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = { .name = "AC97 Aux PCM (Slot 5) Mono in", .dev_addr = __PREG(MODR), - .drcmr = &DRCMRRXMODR, + .drcmr = &DRCMR(9), .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST16 | DCMD_WIDTH2, }; @@ -276,7 +276,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = { .name = "AC97 Mic PCM (Slot 6) Mono in", .dev_addr = __PREG(MCDR), - .drcmr = &DRCMRRXMCDR, + .drcmr = &DRCMR(8), .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST16 | DCMD_WIDTH2, }; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index c796b188277..2dbe612fddd 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -44,7 +44,7 @@ static struct clk *clk_i2s; static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = { .name = "I2S PCM Stereo out", .dev_addr = __PREG(SADR), - .drcmr = &DRCMRTXSADR, + .drcmr = &DRCMR(3), .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | DCMD_BURST32 | DCMD_WIDTH4, }; @@ -52,7 +52,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = { static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = { .name = "I2S PCM Stereo in", .dev_addr = __PREG(SADR), - .drcmr = &DRCMRRXSADR, + .drcmr = &DRCMR(2), .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST32 | DCMD_WIDTH4, }; -- cgit v1.2.3 From 52358ba3a89012c54712c24074ceb4b1c669af52 Mon Sep 17 00:00:00 2001 From: Eric Miao Date: Mon, 8 Sep 2008 15:37:50 +0800 Subject: [ARM] pxa: move I2S register and bit definitions into pxa2xx-i2s.c Signed-off-by: Eric Miao Acked-by: Mark Brown Signed-off-by: Russell King --- sound/soc/pxa/pxa2xx-i2s.c | 40 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 40 insertions(+) (limited to 'sound') diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 2dbe612fddd..ad4c31ddb3d 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -30,6 +30,46 @@ #include "pxa2xx-pcm.h" #include "pxa2xx-i2s.h" +/* + * I2S Controller Register and Bit Definitions + */ +#define SACR0 __REG(0x40400000) /* Global Control Register */ +#define SACR1 __REG(0x40400004) /* Serial Audio I 2 S/MSB-Justified Control Register */ +#define SASR0 __REG(0x4040000C) /* Serial Audio I 2 S/MSB-Justified Interface and FIFO Status Register */ +#define SAIMR __REG(0x40400014) /* Serial Audio Interrupt Mask Register */ +#define SAICR __REG(0x40400018) /* Serial Audio Interrupt Clear Register */ +#define SADIV __REG(0x40400060) /* Audio Clock Divider Register. */ +#define SADR __REG(0x40400080) /* Serial Audio Data Register (TX and RX FIFO access Register). */ + +#define SACR0_RFTH(x) ((x) << 12) /* Rx FIFO Interrupt or DMA Trigger Threshold */ +#define SACR0_TFTH(x) ((x) << 8) /* Tx FIFO Interrupt or DMA Trigger Threshold */ +#define SACR0_STRF (1 << 5) /* FIFO Select for EFWR Special Function */ +#define SACR0_EFWR (1 << 4) /* Enable EFWR Function */ +#define SACR0_RST (1 << 3) /* FIFO, i2s Register Reset */ +#define SACR0_BCKD (1 << 2) /* Bit Clock Direction */ +#define SACR0_ENB (1 << 0) /* Enable I2S Link */ +#define SACR1_ENLBF (1 << 5) /* Enable Loopback */ +#define SACR1_DRPL (1 << 4) /* Disable Replaying Function */ +#define SACR1_DREC (1 << 3) /* Disable Recording Function */ +#define SACR1_AMSL (1 << 0) /* Specify Alternate Mode */ + +#define SASR0_I2SOFF (1 << 7) /* Controller Status */ +#define SASR0_ROR (1 << 6) /* Rx FIFO Overrun */ +#define SASR0_TUR (1 << 5) /* Tx FIFO Underrun */ +#define SASR0_RFS (1 << 4) /* Rx FIFO Service Request */ +#define SASR0_TFS (1 << 3) /* Tx FIFO Service Request */ +#define SASR0_BSY (1 << 2) /* I2S Busy */ +#define SASR0_RNE (1 << 1) /* Rx FIFO Not Empty */ +#define SASR0_TNF (1 << 0) /* Tx FIFO Not Empty */ + +#define SAICR_ROR (1 << 6) /* Clear Rx FIFO Overrun Interrupt */ +#define SAICR_TUR (1 << 5) /* Clear Tx FIFO Underrun Interrupt */ + +#define SAIMR_ROR (1 << 6) /* Enable Rx FIFO Overrun Condition Interrupt */ +#define SAIMR_TUR (1 << 5) /* Enable Tx FIFO Underrun Condition Interrupt */ +#define SAIMR_RFS (1 << 4) /* Enable Rx FIFO Service Interrupt */ +#define SAIMR_TFS (1 << 3) /* Enable Tx FIFO Service Interrupt */ + struct pxa_i2s_port { u32 sadiv; u32 sacr0; -- cgit v1.2.3 From 2f3d00250ae5b1d2727e2723da805290ec408503 Mon Sep 17 00:00:00 2001 From: Hans Verkuil Date: Sat, 23 Aug 2008 04:52:00 -0300 Subject: V4L/DVB (8777): tea575x-tuner: replace video_exclusive_open/release Move the video_exclusive_open/release functionality into the driver itself. Signed-off-by: Hans Verkuil Signed-off-by: Mauro Carvalho Chehab --- sound/i2c/other/tea575x-tuner.c | 22 ++++++++++++++++++++-- 1 file changed, 20 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 83e90057270..621d7fe3074 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -175,6 +175,23 @@ static void snd_tea575x_release(struct video_device *vfd) { } +static int snd_tea575x_exclusive_open(struct inode *inode, struct file *file) +{ + struct video_device *dev = video_devdata(file); + struct snd_tea575x *tea = video_get_drvdata(dev); + + return test_and_set_bit(0, &tea->in_use) ? -EBUSY : 0; +} + +static int snd_tea575x_exclusive_release(struct inode *inode, struct file *file) +{ + struct video_device *dev = video_devdata(file); + struct snd_tea575x *tea = video_get_drvdata(dev); + + clear_bit(0, &tea->in_use); + return 0; +} + /* * initialize all the tea575x chips */ @@ -193,9 +210,10 @@ void snd_tea575x_init(struct snd_tea575x *tea) tea->vd.release = snd_tea575x_release; video_set_drvdata(&tea->vd, tea); tea->vd.fops = &tea->fops; + tea->in_use = 0; tea->fops.owner = tea->card->module; - tea->fops.open = video_exclusive_open; - tea->fops.release = video_exclusive_release; + tea->fops.open = snd_tea575x_exclusive_open; + tea->fops.release = snd_tea575x_exclusive_release; tea->fops.ioctl = snd_tea575x_ioctl; if (video_register_device(&tea->vd, VFL_TYPE_RADIO, tea->dev_nr - 1) < 0) { snd_printk(KERN_ERR "unable to register tea575x tuner\n"); -- cgit v1.2.3 From c170ecf434bceb0e188b14a6deb3bfa3ec9ef699 Mon Sep 17 00:00:00 2001 From: Hans Verkuil Date: Sat, 23 Aug 2008 08:32:09 -0300 Subject: V4L/DVB (8788): v4l: replace video_get_drvdata(video_devdata(filp)) with video_drvdata(filp) Use the new video_drvdata(filp) function where it is safe to do so. Signed-off-by: Hans Verkuil Signed-off-by: Mauro Carvalho Chehab --- sound/i2c/other/tea575x-tuner.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 621d7fe3074..c13a178383b 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -87,8 +87,7 @@ static void snd_tea575x_set_freq(struct snd_tea575x *tea) static int snd_tea575x_ioctl(struct inode *inode, struct file *file, unsigned int cmd, unsigned long data) { - struct video_device *dev = video_devdata(file); - struct snd_tea575x *tea = video_get_drvdata(dev); + struct snd_tea575x *tea = video_drvdata(file); void __user *arg = (void __user *)data; switch(cmd) { @@ -177,16 +176,14 @@ static void snd_tea575x_release(struct video_device *vfd) static int snd_tea575x_exclusive_open(struct inode *inode, struct file *file) { - struct video_device *dev = video_devdata(file); - struct snd_tea575x *tea = video_get_drvdata(dev); + struct snd_tea575x *tea = video_drvdata(file); return test_and_set_bit(0, &tea->in_use) ? -EBUSY : 0; } static int snd_tea575x_exclusive_release(struct inode *inode, struct file *file) { - struct video_device *dev = video_devdata(file); - struct snd_tea575x *tea = video_get_drvdata(dev); + struct snd_tea575x *tea = video_drvdata(file); clear_bit(0, &tea->in_use); return 0; -- cgit v1.2.3 From 5817b52a298adce69e01acf2c131b3dcfda65d64 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 24 Sep 2008 11:23:11 +0100 Subject: ALSA: ASoC: Allow machine drivers to mark pins as not connected Add a new API call snd_soc_dapm_nc_pin() which allows machine drivers to mark pins as being permanently disabled. At present this is identical to snd_soc_dapm_disable_pin() except in terms of improving the internal documentation of machine drivers that use it. The intention is that in future it will be extended to provide additional features such as hiding controls that are only relevant to paths using the disconnected pin. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/soc-dapm.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9ca9c08610f..83fa9c47b66 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1483,6 +1483,26 @@ int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin) } EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); +/** + * snd_soc_dapm_nc_pin - permanently disable pin. + * @codec: SoC codec + * @pin: pin name + * + * Marks the specified pin as being not connected, disabling it along + * any parent or child widgets. At present this is identical to + * snd_soc_dapm_disable_pin() but in future it will be extended to do + * additional things such as disabling controls which only affect + * paths through the pin. + * + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin) +{ + return snd_soc_dapm_set_pin(codec, pin, 0); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin); + /** * snd_soc_dapm_get_pin_status - get audio pin status * @codec: audio codec -- cgit v1.2.3 From b1cbc21c8e0cb9d253dc1388f24495b68261821a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 24 Sep 2008 11:33:05 +0100 Subject: ALSA: ASoC: Use snd_soc_dapm_nc_pin() in GTA01 audio driver Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/s3c24xx/neo1973_wm8753.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 73a50e93a9a..006c36ded25 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -511,13 +511,12 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) DBG("Entered %s\n", __func__); /* set up NC codec pins */ - snd_soc_dapm_disable_pin(codec, "LOUT2"); - snd_soc_dapm_disable_pin(codec, "ROUT2"); - snd_soc_dapm_disable_pin(codec, "OUT3"); - snd_soc_dapm_disable_pin(codec, "OUT4"); - snd_soc_dapm_disable_pin(codec, "LINE1"); - snd_soc_dapm_disable_pin(codec, "LINE2"); - + snd_soc_dapm_nc_pin(codec, "LOUT2"); + snd_soc_dapm_nc_pin(codec, "ROUT2"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "OUT4"); + snd_soc_dapm_nc_pin(codec, "LINE1"); + snd_soc_dapm_nc_pin(codec, "LINE2"); /* set endpoints to default mode */ set_scenario_endpoints(codec, NEO_AUDIO_OFF); -- cgit v1.2.3 From 5cabc1a8b3acc4babd69f2c91a6ab4468dac6663 Mon Sep 17 00:00:00 2001 From: Frank Mandarino Date: Tue, 30 Sep 2008 10:42:40 -0400 Subject: ALSA: ASoC: Remove references to Endrelia ETI-B1 board The ASoC machine drivers for this board were only provided as examples for the new AT91 ASoC platform driver. Since the ETI-B1 board is proprietary and there are other AT91 ASoC machine drivers available, it makes sense to remove these drivers. Signed-off-by: Frank Mandarino Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/at91/Kconfig | 17 -- sound/soc/at91/Makefile | 5 - sound/soc/at91/eti_b1_wm8731.c | 349 ----------------------------------------- 3 files changed, 371 deletions(-) delete mode 100644 sound/soc/at91/eti_b1_wm8731.c (limited to 'sound') diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig index 905186502e0..85a883299c2 100644 --- a/sound/soc/at91/Kconfig +++ b/sound/soc/at91/Kconfig @@ -8,20 +8,3 @@ config SND_AT91_SOC config SND_AT91_SOC_SSC tristate - -config SND_AT91_SOC_ETI_B1_WM8731 - tristate "SoC Audio support for WM8731-based Endrelia ETI-B1 boards" - depends on SND_AT91_SOC && (MACH_ETI_B1 || MACH_ETI_C1) - select SND_AT91_SOC_SSC - select SND_SOC_WM8731 - help - Say Y if you want to add support for SoC audio on WM8731-based - Endrelia Technologies Inc ETI-B1 or ETI-C1 boards. - -config SND_AT91_SOC_ETI_SLAVE - bool "Run codec in slave Mode on Endrelia boards" - depends on SND_AT91_SOC_ETI_B1_WM8731 - default n - help - Say Y if you want to run with the AT91 SSC generating the BCLK - and LRC signals on Endrelia boards. diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile index f23da17cc32..b817f11df28 100644 --- a/sound/soc/at91/Makefile +++ b/sound/soc/at91/Makefile @@ -4,8 +4,3 @@ snd-soc-at91-ssc-objs := at91-ssc.o obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o - -# AT91 Machine Support -snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o - -obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c deleted file mode 100644 index 684781e4088..00000000000 --- a/sound/soc/at91/eti_b1_wm8731.c +++ /dev/null @@ -1,349 +0,0 @@ -/* - * eti_b1_wm8731 -- SoC audio for AT91RM9200-based Endrelia ETI_B1 board. - * - * Author: Frank Mandarino - * Endrelia Technologies Inc. - * Created: Mar 29, 2006 - * - * Based on corgi.c by: - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * - * Authors: Liam Girdwood - * Richard Purdie - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include -#include - -#include "../codecs/wm8731.h" -#include "at91-pcm.h" -#include "at91-ssc.h" - -#if 0 -#define DBG(x...) printk(KERN_INFO "eti_b1_wm8731: " x) -#else -#define DBG(x...) -#endif - -static struct clk *pck1_clk; -static struct clk *pllb_clk; - - -static int eti_b1_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - int ret; - - /* cpu clock is the AT91 master clock sent to the SSC */ - ret = snd_soc_dai_set_sysclk(cpu_dai, AT91_SYSCLK_MCK, - 60000000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* codec system clock is supplied by PCK1, set to 12MHz */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, - 12000000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* Start PCK1 clock. */ - clk_enable(pck1_clk); - DBG("pck1 started\n"); - - return 0; -} - -static void eti_b1_shutdown(struct snd_pcm_substream *substream) -{ - /* Stop PCK1 clock. */ - clk_disable(pck1_clk); - DBG("pck1 stopped\n"); -} - -static int eti_b1_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - int ret; - -#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE - unsigned int rate; - int cmr_div, period; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* - * The SSC clock dividers depend on the sample rate. The CMR.DIV - * field divides the system master clock MCK to drive the SSC TK - * signal which provides the codec BCLK. The TCMR.PERIOD and - * RCMR.PERIOD fields further divide the BCLK signal to drive - * the SSC TF and RF signals which provide the codec DACLRC and - * ADCLRC clocks. - * - * The dividers were determined through trial and error, where a - * CMR.DIV value is chosen such that the resulting BCLK value is - * divisible, or almost divisible, by (2 * sample rate), and then - * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. - */ - rate = params_rate(params); - - switch (rate) { - case 8000: - cmr_div = 25; /* BCLK = 60MHz/(2*25) = 1.2MHz */ - period = 74; /* LRC = BCLK/(2*(74+1)) = 8000Hz */ - break; - case 32000: - cmr_div = 7; /* BCLK = 60MHz/(2*7) ~= 4.28571428MHz */ - period = 66; /* LRC = BCLK/(2*(66+1)) = 31982.942Hz */ - break; - case 48000: - cmr_div = 13; /* BCLK = 60MHz/(2*13) ~= 2.3076923MHz */ - period = 23; /* LRC = BCLK/(2*(23+1)) = 48076.923Hz */ - break; - default: - printk(KERN_WARNING "unsupported rate %d on ETI-B1 board\n", rate); - return -EINVAL; - } - - /* set the MCK divider for BCLK */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div); - if (ret < 0) - return ret; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* set the BCLK divider for DACLRC */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - AT91SSC_TCMR_PERIOD, period); - } else { - /* set the BCLK divider for ADCLRC */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - AT91SSC_RCMR_PERIOD, period); - } - if (ret < 0) - return ret; - -#else /* CONFIG_SND_AT91_SOC_ETI_SLAVE */ - /* - * Codec in Master Mode. - */ - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - -#endif /* CONFIG_SND_AT91_SOC_ETI_SLAVE */ - - return 0; -} - -static struct snd_soc_ops eti_b1_ops = { - .startup = eti_b1_startup, - .hw_params = eti_b1_hw_params, - .shutdown = eti_b1_shutdown, -}; - - -static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), -}; - -static const struct snd_soc_dapm_route intercon[] = { - - /* speaker connected to LHPOUT */ - {"Ext Spk", NULL, "LHPOUT"}, - - /* mic is connected to Mic Jack, with WM8731 Mic Bias */ - {"MICIN", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Int Mic"}, -}; - -/* - * Logic for a wm8731 as connected on a Endrelia ETI-B1 board. - */ -static int eti_b1_wm8731_init(struct snd_soc_codec *codec) -{ - DBG("eti_b1_wm8731_init() called\n"); - - /* Add specific widgets */ - snd_soc_dapm_new_controls(codec, eti_b1_dapm_widgets, - ARRAY_SIZE(eti_b1_dapm_widgets)); - - /* Set up specific audio path interconnects */ - snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon)); - - /* not connected */ - snd_soc_dapm_disable_pin(codec, "RLINEIN"); - snd_soc_dapm_disable_pin(codec, "LLINEIN"); - - /* always connected */ - snd_soc_dapm_enable_pin(codec, "Int Mic"); - snd_soc_dapm_enable_pin(codec, "Ext Spk"); - - snd_soc_dapm_sync(codec); - - return 0; -} - -static struct snd_soc_dai_link eti_b1_dai = { - .name = "WM8731", - .stream_name = "WM8731 PCM", - .cpu_dai = &at91_ssc_dai[1], - .codec_dai = &wm8731_dai, - .init = eti_b1_wm8731_init, - .ops = &eti_b1_ops, -}; - -static struct snd_soc_machine snd_soc_machine_eti_b1 = { - .name = "ETI_B1_WM8731", - .dai_link = &eti_b1_dai, - .num_links = 1, -}; - -static struct wm8731_setup_data eti_b1_wm8731_setup = { - .i2c_bus = 0, - .i2c_address = 0x1a, -}; - -static struct snd_soc_device eti_b1_snd_devdata = { - .machine = &snd_soc_machine_eti_b1, - .platform = &at91_soc_platform, - .codec_dev = &soc_codec_dev_wm8731, - .codec_data = &eti_b1_wm8731_setup, -}; - -static struct platform_device *eti_b1_snd_device; - -static int __init eti_b1_init(void) -{ - int ret; - struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data; - - if (!request_mem_region(AT91RM9200_BASE_SSC1, SZ_16K, "soc-audio")) { - DBG("SSC1 memory region is busy\n"); - return -EBUSY; - } - - ssc->base = ioremap(AT91RM9200_BASE_SSC1, SZ_16K); - if (!ssc->base) { - DBG("SSC1 memory ioremap failed\n"); - ret = -ENOMEM; - goto fail_release_mem; - } - - ssc->pid = AT91RM9200_ID_SSC1; - - eti_b1_snd_device = platform_device_alloc("soc-audio", -1); - if (!eti_b1_snd_device) { - DBG("platform device allocation failed\n"); - ret = -ENOMEM; - goto fail_io_unmap; - } - - platform_set_drvdata(eti_b1_snd_device, &eti_b1_snd_devdata); - eti_b1_snd_devdata.dev = &eti_b1_snd_device->dev; - - ret = platform_device_add(eti_b1_snd_device); - if (ret) { - DBG("platform device add failed\n"); - platform_device_put(eti_b1_snd_device); - goto fail_io_unmap; - } - - at91_set_A_periph(AT91_PIN_PB6, 0); /* TF1 */ - at91_set_A_periph(AT91_PIN_PB7, 0); /* TK1 */ - at91_set_A_periph(AT91_PIN_PB8, 0); /* TD1 */ - at91_set_A_periph(AT91_PIN_PB9, 0); /* RD1 */ -/* at91_set_A_periph(AT91_PIN_PB10, 0);*/ /* RK1 */ - at91_set_A_periph(AT91_PIN_PB11, 0); /* RF1 */ - - /* - * Set PCK1 parent to PLLB and its rate to 12 Mhz. - */ - pllb_clk = clk_get(NULL, "pllb"); - pck1_clk = clk_get(NULL, "pck1"); - - clk_set_parent(pck1_clk, pllb_clk); - clk_set_rate(pck1_clk, 12000000); - - DBG("MCLK rate %luHz\n", clk_get_rate(pck1_clk)); - - /* assign the GPIO pin to PCK1 */ - at91_set_B_periph(AT91_PIN_PA24, 0); - -#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE - printk(KERN_INFO "eti_b1_wm8731: Codec in Slave Mode\n"); -#else - printk(KERN_INFO "eti_b1_wm8731: Codec in Master Mode\n"); -#endif - return ret; - -fail_io_unmap: - iounmap(ssc->base); -fail_release_mem: - release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K); - return ret; -} - -static void __exit eti_b1_exit(void) -{ - struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data; - - clk_put(pck1_clk); - clk_put(pllb_clk); - - platform_device_unregister(eti_b1_snd_device); - - iounmap(ssc->base); - release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K); -} - -module_init(eti_b1_init); -module_exit(eti_b1_exit); - -/* Module information */ -MODULE_AUTHOR("Frank Mandarino "); -MODULE_DESCRIPTION("ALSA SoC ETI-B1-WM8731"); -MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 0e77e78410245894e5ebde30dc68ced6daa81bce Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Sat, 27 Sep 2008 16:57:26 +0800 Subject: ALSA: ASoC codec: AD73311 audio codec driver Signed-off-by: Cliff Cai Signed-off-by: Bryan Wu Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/Kconfig | 4 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad73311.c | 107 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ad73311.h | 90 ++++++++++++++++++++++++++++++++++++++ 4 files changed, 203 insertions(+) create mode 100644 sound/soc/codecs/ad73311.c create mode 100644 sound/soc/codecs/ad73311.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e0b9869df0f..0507fcf6608 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -3,6 +3,7 @@ config SND_SOC_ALL_CODECS depends on I2C select SPI select SPI_MASTER + select SND_SOC_AD73311 select SND_SOC_AK4535 select SND_SOC_CS4270 select SND_SOC_SSM2602 @@ -34,6 +35,9 @@ config SND_SOC_AC97_CODEC config SND_SOC_AD1980 tristate +config SND_SOC_AD73311 + tristate + config SND_SOC_AK4535 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f977978a340..07318445a1f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,5 +1,6 @@ snd-soc-ac97-objs := ac97.o snd-soc-ad1980-objs := ad1980.o +snd-soc-ad73311-objs := ad73311.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o snd-soc-ssm2602-objs := ssm2602.o @@ -20,6 +21,7 @@ snd-soc-wm9713-objs := wm9713.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o +obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c new file mode 100644 index 00000000000..37af8607b00 --- /dev/null +++ b/sound/soc/codecs/ad73311.c @@ -0,0 +1,107 @@ +/* + * ad73311.c -- ALSA Soc AD73311 codec support + * + * Copyright: Analog Device Inc. + * Author: Cliff Cai + * + * 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. + * + * Revision history + * 25th Sep 2008 Initial version. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "ad73311.h" + +struct snd_soc_dai ad73311_dai = { + .name = "AD73311", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, +}; +EXPORT_SYMBOL_GPL(ad73311_dai); + +static int ad73311_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + mutex_init(&codec->mutex); + codec->name = "AD73311"; + codec->owner = THIS_MODULE; + codec->dai = &ad73311_dai; + codec->num_dai = 1; + socdev->codec = codec; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ad73311: failed to create pcms\n"); + goto pcm_err; + } + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ad73311: failed to register card\n"); + goto register_err; + } + + return ret; + +register_err: + snd_soc_free_pcms(socdev); +pcm_err: + kfree(socdev->codec); + socdev->codec = NULL; + return ret; +} + +static int ad73311_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec == NULL) + return 0; + snd_soc_free_pcms(socdev); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ad73311 = { + .probe = ad73311_soc_probe, + .remove = ad73311_soc_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311); + +MODULE_DESCRIPTION("ASoC ad73311 driver"); +MODULE_AUTHOR("Cliff Cai "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad73311.h b/sound/soc/codecs/ad73311.h new file mode 100644 index 00000000000..507ce0c30ed --- /dev/null +++ b/sound/soc/codecs/ad73311.h @@ -0,0 +1,90 @@ +/* + * File: sound/soc/codec/ad73311.h + * Based on: + * Author: Cliff Cai + * + * Created: Thur Sep 25, 2008 + * Description: definitions for AD73311 registers + * + * + * Modified: + * Copyright 2006 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * 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, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef __AD73311_H__ +#define __AD73311_H__ + +#define AD_CONTROL 0x8000 +#define AD_DATA 0x0000 +#define AD_READ 0x4000 +#define AD_WRITE 0x0000 + +/* Control register A */ +#define CTRL_REG_A (0 << 8) + +#define REGA_MODE_PRO 0x00 +#define REGA_MODE_DATA 0x01 +#define REGA_MODE_MIXED 0x03 +#define REGA_DLB 0x04 +#define REGA_SLB 0x08 +#define REGA_DEVC(x) ((x & 0x7) << 4) +#define REGA_RESET 0x80 + +/* Control register B */ +#define CTRL_REG_B (1 << 8) + +#define REGB_DIRATE(x) (x & 0x3) +#define REGB_SCDIV(x) ((x & 0x3) << 2) +#define REGB_MCDIV(x) ((x & 0x7) << 4) +#define REGB_CEE (1 << 7) + +/* Control register C */ +#define CTRL_REG_C (2 << 8) + +#define REGC_PUDEV (1 << 0) +#define REGC_PUADC (1 << 3) +#define REGC_PUDAC (1 << 4) +#define REGC_PUREF (1 << 5) +#define REGC_REFUSE (1 << 6) + +/* Control register D */ +#define CTRL_REG_D (3 << 8) + +#define REGD_IGS(x) (x & 0x7) +#define REGD_RMOD (1 << 3) +#define REGD_OGS(x) ((x & 0x7) << 4) +#define REGD_MUTE (x << 7) + +/* Control register E */ +#define CTRL_REG_E (4 << 8) + +#define REGE_DA(x) (x & 0x1f) +#define REGE_IBYP (1 << 5) + +/* Control register F */ +#define CTRL_REG_F (5 << 8) + +#define REGF_SEEN (1 << 5) +#define REGF_INV (1 << 6) +#define REGF_ALB (1 << 7) + +extern struct snd_soc_dai ad73311_dai; +extern struct snd_soc_codec_device soc_codec_dev_ad73311; +#endif -- cgit v1.2.3 From 333926803557ee43568ebd9ae17b868d60e77a62 Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Sat, 27 Sep 2008 22:30:15 +0800 Subject: ALSA: ASoC Blackfin: add I2S DAI support for AD73311 Signed-off-by: Cliff Cai Signed-off-by: Bryan Wu Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/blackfin/bf5xx-i2s.c | 47 +++++++++++++++++++++++++++++------------- 1 file changed, 33 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 43a4092eeb8..827587f0818 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -70,6 +70,13 @@ static struct sport_param sport_params[2] = { } }; +static u16 sport_req[][7] = { + { P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, + P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}, + { P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, + P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}, +}; + static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -78,6 +85,14 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* interface format:support I2S,slave mode */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: + bf5xx_i2s.tcr1 |= TFSR | TCKFE; + bf5xx_i2s.rcr1 |= RFSR | RCKFE; + bf5xx_i2s.tcr2 |= TSFSE; + bf5xx_i2s.rcr2 |= RSFSE; + break; + case SND_SOC_DAIFMT_DSP_A: + bf5xx_i2s.tcr1 |= TFSR; + bf5xx_i2s.rcr1 |= RFSR; break; case SND_SOC_DAIFMT_LEFT_J: ret = -EINVAL; @@ -127,14 +142,17 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: bf5xx_i2s.tcr2 |= 15; bf5xx_i2s.rcr2 |= 15; + sport_handle->wdsize = 2; break; case SNDRV_PCM_FORMAT_S24_LE: bf5xx_i2s.tcr2 |= 23; bf5xx_i2s.rcr2 |= 23; + sport_handle->wdsize = 3; break; case SNDRV_PCM_FORMAT_S32_LE: bf5xx_i2s.tcr2 |= 31; bf5xx_i2s.rcr2 |= 31; + sport_handle->wdsize = 4; break; } @@ -145,17 +163,17 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, * need to configure both of them at the time when the first * stream is opened. * - * CPU DAI format:I2S, slave mode. + * CPU DAI:slave mode. */ - ret = sport_config_rx(sport_handle, RFSR | RCKFE, - RSFSE|bf5xx_i2s.rcr2, 0, 0); + ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1, + bf5xx_i2s.rcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - ret = sport_config_tx(sport_handle, TFSR | TCKFE, - TSFSE|bf5xx_i2s.tcr2, 0, 0); + ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1, + bf5xx_i2s.tcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; @@ -174,13 +192,6 @@ static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream) static int bf5xx_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { - u16 sport_req[][7] = { - { P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, - P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}, - { P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, - P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}, - }; - pr_debug("%s enter\n", __func__); if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) { pr_err("Requesting Peripherals failed\n"); @@ -198,6 +209,13 @@ static int bf5xx_i2s_probe(struct platform_device *pdev, return 0; } +static void bf5xx_i2s_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + pr_debug("%s enter\n", __func__); + peripheral_free_list(&sport_req[sport_num][0]); +} + #ifdef CONFIG_PM static int bf5xx_i2s_suspend(struct platform_device *dev, struct snd_soc_dai *dai) @@ -263,15 +281,16 @@ struct snd_soc_dai bf5xx_i2s_dai = { .id = 0, .type = SND_SOC_DAI_I2S, .probe = bf5xx_i2s_probe, + .remove = bf5xx_i2s_remove, .suspend = bf5xx_i2s_suspend, .resume = bf5xx_i2s_resume, .playback = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, -- cgit v1.2.3 From 5564b14b88a5a34ea848732030fbc202a050daa6 Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Sat, 27 Sep 2008 22:31:21 +0800 Subject: ALSA: ASoC Blackfin: add asoc ad73311 driver supporting in Blackfin boards Signed-off-by: Cliff Cai Signed-off-by: Bryan Wu Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/blackfin/Kconfig | 16 +++ sound/soc/blackfin/Makefile | 3 +- sound/soc/blackfin/bf5xx-ad73311.c | 240 +++++++++++++++++++++++++++++++++++++ 3 files changed, 258 insertions(+), 1 deletion(-) create mode 100644 sound/soc/blackfin/bf5xx-ad73311.c (limited to 'sound') diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index f98331d099e..dc006206f62 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -17,6 +17,22 @@ config SND_BF5XX_SOC_SSM2602 help Say Y if you want to add support for SoC audio on BF527-EZKIT. +config SND_BF5XX_SOC_AD73311 + tristate "SoC AD73311 Audio support for Blackfin" + depends on SND_BF5XX_I2S + select SND_BF5XX_SOC_I2S + select SND_SOC_AD73311 + help + Say Y if you want to add support for AD73311 codec on Blackfin. + +config SND_BFIN_AD73311_SE + int "PF pin for AD73311L Chip Select" + depends on SND_BF5XX_SOC_AD73311 + default 4 + help + Enter the GPIO used to control AD73311's SE pin. Acceptable + values are 0 to 7 + config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" depends on BLACKFIN && SND_SOC diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 9ea8bd9e0ba..97bb37a6359 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -14,7 +14,8 @@ obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o # Blackfin Machine Support snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o - +snd-ad73311-objs := bf5xx-ad73311.o obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o +obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c new file mode 100644 index 00000000000..622c9b90953 --- /dev/null +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -0,0 +1,240 @@ +/* + * File: sound/soc/blackfin/bf5xx-ad73311.c + * Author: Cliff Cai + * + * Created: Thur Sep 25 2008 + * Description: Board driver for ad73311 sound chip + * + * Modified: + * Copyright 2008 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * 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, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include "../codecs/ad73311.h" +#include "bf5xx-sport.h" +#include "bf5xx-i2s-pcm.h" +#include "bf5xx-i2s.h" + +#if CONFIG_SND_BF5XX_SPORT_NUM == 0 +#define bfin_write_SPORT_TCR1 bfin_write_SPORT0_TCR1 +#define bfin_read_SPORT_TCR1 bfin_read_SPORT0_TCR1 +#define bfin_write_SPORT_TCR2 bfin_write_SPORT0_TCR2 +#define bfin_write_SPORT_TX16 bfin_write_SPORT0_TX16 +#define bfin_read_SPORT_STAT bfin_read_SPORT0_STAT +#else +#define bfin_write_SPORT_TCR1 bfin_write_SPORT1_TCR1 +#define bfin_read_SPORT_TCR1 bfin_read_SPORT1_TCR1 +#define bfin_write_SPORT_TCR2 bfin_write_SPORT1_TCR2 +#define bfin_write_SPORT_TX16 bfin_write_SPORT1_TX16 +#define bfin_read_SPORT_STAT bfin_read_SPORT1_STAT +#endif + +#define GPIO_SE CONFIG_SND_BFIN_AD73311_SE + +static struct snd_soc_machine bf5xx_ad73311; + +static int snd_ad73311_startup(void) +{ + pr_debug("%s enter\n", __func__); + + /* Pull up SE pin on AD73311L */ + gpio_set_value(GPIO_SE, 1); + return 0; +} + +static int snd_ad73311_configure(void) +{ + unsigned short ctrl_regs[6]; + unsigned short status = 0; + int count = 0; + + /* DMCLK = MCLK = 16.384 MHz + * SCLK = DMCLK/8 = 2.048 MHz + * Sample Rate = DMCLK/2048 = 8 KHz + */ + ctrl_regs[0] = AD_CONTROL | AD_WRITE | CTRL_REG_B | REGB_MCDIV(0) | \ + REGB_SCDIV(0) | REGB_DIRATE(0); + ctrl_regs[1] = AD_CONTROL | AD_WRITE | CTRL_REG_C | REGC_PUDEV | \ + REGC_PUADC | REGC_PUDAC | REGC_PUREF | REGC_REFUSE ; + ctrl_regs[2] = AD_CONTROL | AD_WRITE | CTRL_REG_D | REGD_OGS(2) | \ + REGD_IGS(2); + ctrl_regs[3] = AD_CONTROL | AD_WRITE | CTRL_REG_E | REGE_DA(0x1f); + ctrl_regs[4] = AD_CONTROL | AD_WRITE | CTRL_REG_F | REGF_SEEN ; + ctrl_regs[5] = AD_CONTROL | AD_WRITE | CTRL_REG_A | REGA_MODE_DATA; + + local_irq_disable(); + snd_ad73311_startup(); + udelay(1); + + bfin_write_SPORT_TCR1(TFSR); + bfin_write_SPORT_TCR2(0xF); + SSYNC(); + + /* SPORT Tx Register is a 8 x 16 FIFO, all the data can be put to + * FIFO before enable SPORT to transfer the data + */ + for (count = 0; count < 6; count++) + bfin_write_SPORT_TX16(ctrl_regs[count]); + SSYNC(); + bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() | TSPEN); + SSYNC(); + + /* When TUVF is set, the data is already send out */ + while (!(status & TUVF) && count++ < 10000) { + udelay(1); + status = bfin_read_SPORT_STAT(); + SSYNC(); + } + bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() & ~TSPEN); + SSYNC(); + local_irq_enable(); + + if (count == 10000) { + printk(KERN_ERR "ad73311: failed to configure codec\n"); + return -1; + } + return 0; +} + +static int bf5xx_probe(struct platform_device *pdev) +{ + int err; + if (gpio_request(GPIO_SE, "AD73311_SE")) { + printk(KERN_ERR "%s: Failed ro request GPIO_%d\n", __func__, GPIO_SE); + return -EBUSY; + } + + gpio_direction_output(GPIO_SE, 0); + + err = snd_ad73311_configure(); + if (err < 0) + return -EFAULT; + + return 0; +} + +static int bf5xx_ad73311_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + pr_debug("%s enter\n", __func__); + cpu_dai->private_data = sport_handle; + return 0; +} + +static int bf5xx_ad73311_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + + pr_debug("%s rate %d format %x\n", __func__, params_rate(params), + params_format(params)); + + /* set cpu DAI configuration */ + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + return 0; +} + + +static struct snd_soc_ops bf5xx_ad73311_ops = { + .startup = bf5xx_ad73311_startup, + .hw_params = bf5xx_ad73311_hw_params, +}; + +static struct snd_soc_dai_link bf5xx_ad73311_dai = { + .name = "ad73311", + .stream_name = "AD73311", + .cpu_dai = &bf5xx_i2s_dai, + .codec_dai = &ad73311_dai, + .ops = &bf5xx_ad73311_ops, +}; + +static struct snd_soc_machine bf5xx_ad73311 = { + .name = "bf5xx_ad73311", + .probe = bf5xx_probe, + .dai_link = &bf5xx_ad73311_dai, + .num_links = 1, +}; + +static struct snd_soc_device bf5xx_ad73311_snd_devdata = { + .machine = &bf5xx_ad73311, + .platform = &bf5xx_i2s_soc_platform, + .codec_dev = &soc_codec_dev_ad73311, +}; + +static struct platform_device *bf52x_ad73311_snd_device; + +static int __init bf5xx_ad73311_init(void) +{ + int ret; + + pr_debug("%s enter\n", __func__); + bf52x_ad73311_snd_device = platform_device_alloc("soc-audio", -1); + if (!bf52x_ad73311_snd_device) + return -ENOMEM; + + platform_set_drvdata(bf52x_ad73311_snd_device, &bf5xx_ad73311_snd_devdata); + bf5xx_ad73311_snd_devdata.dev = &bf52x_ad73311_snd_device->dev; + ret = platform_device_add(bf52x_ad73311_snd_device); + + if (ret) + platform_device_put(bf52x_ad73311_snd_device); + + return ret; +} + +static void __exit bf5xx_ad73311_exit(void) +{ + pr_debug("%s enter\n", __func__); + platform_device_unregister(bf52x_ad73311_snd_device); +} + +module_init(bf5xx_ad73311_init); +module_exit(bf5xx_ad73311_exit); + +/* Module information */ +MODULE_AUTHOR("Cliff Cai"); +MODULE_DESCRIPTION("ALSA SoC AD73311 Blackfin"); +MODULE_LICENSE("GPL"); + -- cgit v1.2.3 From 6b58a82121320f96513d88032dc3495a9c6f450b Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Sat, 27 Sep 2008 22:32:20 +0800 Subject: ALSA: ASoC Blackfin: fix bug - Audio Latency on AD1981 with MMAP enabled With MMAP enabled (DMA mode) on the AD1981, there is +/- 250ms of delay between writing data to alsa and audio starts coming out of the AD1981. Copy more data to local buffer before starting DMA Signed-off-by: Cliff Cai Signed-off-by: Bryan Wu Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/blackfin/bf5xx-ac97-pcm.c | 42 ++++++++++++++++++++++++++++++------- sound/soc/blackfin/bf5xx-ac97.c | 1 - sound/soc/blackfin/bf5xx-sport.h | 2 ++ 3 files changed, 37 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 51f4907c483..25e50d2ea1e 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -56,6 +56,7 @@ static void bf5xx_mmap_copy(struct snd_pcm_substream *substream, sport->tx_pos += runtime->period_size; if (sport->tx_pos >= runtime->buffer_size) sport->tx_pos %= runtime->buffer_size; + sport->tx_delay_pos = sport->tx_pos; } else { bf5xx_ac97_to_pcm( (struct ac97_frame *)sport->rx_dma_buf + sport->rx_pos, @@ -72,7 +73,15 @@ static void bf5xx_dma_irq(void *data) struct snd_pcm_substream *pcm = data; #if defined(CONFIG_SND_MMAP_SUPPORT) struct snd_pcm_runtime *runtime = pcm->runtime; + struct sport_device *sport = runtime->private_data; bf5xx_mmap_copy(pcm, runtime->period_size); + if (pcm->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (sport->once == 0) { + snd_pcm_period_elapsed(pcm); + bf5xx_mmap_copy(pcm, runtime->period_size); + sport->once = 1; + } + } #endif snd_pcm_period_elapsed(pcm); } @@ -114,6 +123,10 @@ static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + memset(runtime->dma_area, 0, runtime->buffer_size); snd_pcm_lib_free_pages(substream); return 0; } @@ -127,16 +140,11 @@ static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream) * SPORT working in TMD mode(include AC97). */ #if defined(CONFIG_SND_MMAP_SUPPORT) - size_t size = bf5xx_pcm_hardware.buffer_bytes_max - * sizeof(struct ac97_frame) / 4; - /*clean up intermediate buffer*/ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - memset(sport->tx_dma_buf, 0, size); sport_set_tx_callback(sport, bf5xx_dma_irq, substream); sport_config_tx_dma(sport, sport->tx_dma_buf, runtime->periods, runtime->period_size * sizeof(struct ac97_frame)); } else { - memset(sport->rx_dma_buf, 0, size); sport_set_rx_callback(sport, bf5xx_dma_irq, substream); sport_config_rx_dma(sport, sport->rx_dma_buf, runtime->periods, runtime->period_size * sizeof(struct ac97_frame)); @@ -164,8 +172,12 @@ static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) pr_debug("%s enter\n", __func__); switch (cmd) { case SNDRV_PCM_TRIGGER_START: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + bf5xx_mmap_copy(substream, runtime->period_size); + snd_pcm_period_elapsed(substream); + sport->tx_delay_pos = 0; sport_tx_start(sport); + } else sport_rx_start(sport); break; @@ -198,7 +210,7 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) #if defined(CONFIG_SND_MMAP_SUPPORT) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - curr = sport->tx_pos; + curr = sport->tx_delay_pos; else curr = sport->rx_pos; #else @@ -237,6 +249,21 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream) return ret; } +static int bf5xx_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + + pr_debug("%s enter\n", __func__); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + sport->once = 0; + memset(sport->tx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame)); + } else + memset(sport->rx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame)); + + return 0; +} + #ifdef CONFIG_SND_MMAP_SUPPORT static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) @@ -272,6 +299,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, struct snd_pcm_ops bf5xx_pcm_ac97_ops = { .open = bf5xx_pcm_open, + .close = bf5xx_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = bf5xx_pcm_hw_params, .hw_free = bf5xx_pcm_hw_free, diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index c782e311fd5..5e5aafb6485 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -128,7 +128,6 @@ static void enqueue_cmd(struct snd_ac97 *ac97, __u16 addr, __u16 data) int nextfrag = sport_tx_curr_frag(sport); struct ac97_frame *nextwrite; - sport_incfrag(sport, &nextfrag, 1); sport_incfrag(sport, &nextfrag, 1); nextwrite = (struct ac97_frame *)(sport->tx_buf + \ diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h index 4c163454bbf..fcadcc081f7 100644 --- a/sound/soc/blackfin/bf5xx-sport.h +++ b/sound/soc/blackfin/bf5xx-sport.h @@ -123,6 +123,8 @@ struct sport_device { int rx_pos; unsigned int tx_buffer_size; unsigned int rx_buffer_size; + int tx_delay_pos; + int once; #endif void *private_data; }; -- cgit v1.2.3 From 35f5e54db923477f71d948f30c291d31bc0de0fc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 27 Sep 2008 10:45:09 +0100 Subject: ALSA: ASoC: Use snd_soc_dapm_nc_pin() in Zaurus machine drivers Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/pxa/corgi.c | 4 ++-- sound/soc/pxa/poodle.c | 4 ++-- sound/soc/pxa/spitz.c | 14 +++++++------- sound/soc/pxa/tosa.c | 4 ++-- 4 files changed, 13 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 72b7a5140bf..0bceaf66eff 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -289,8 +289,8 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_disable_pin(codec, "LLINEIN"); - snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_nc_pin(codec, "LLINEIN"); + snd_soc_dapm_nc_pin(codec, "RLINEIN"); /* Add corgi specific controls */ for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) { diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index f84f7d8db09..e5adb0e9193 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -242,8 +242,8 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_disable_pin(codec, "LLINEIN"); - snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_nc_pin(codec, "LLINEIN"); + snd_soc_dapm_nc_pin(codec, "RLINEIN"); snd_soc_dapm_enable_pin(codec, "MICIN"); /* Add poodle specific controls */ diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 3d4738c06e7..e0bcc4250ce 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -291,13 +291,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) int i, err; /* NC codec pins */ - snd_soc_dapm_disable_pin(codec, "RINPUT1"); - snd_soc_dapm_disable_pin(codec, "LINPUT2"); - snd_soc_dapm_disable_pin(codec, "RINPUT2"); - snd_soc_dapm_disable_pin(codec, "LINPUT3"); - snd_soc_dapm_disable_pin(codec, "RINPUT3"); - snd_soc_dapm_disable_pin(codec, "OUT3"); - snd_soc_dapm_disable_pin(codec, "MONO1"); + snd_soc_dapm_nc_pin(codec, "RINPUT1"); + snd_soc_dapm_nc_pin(codec, "LINPUT2"); + snd_soc_dapm_nc_pin(codec, "RINPUT2"); + snd_soc_dapm_nc_pin(codec, "LINPUT3"); + snd_soc_dapm_nc_pin(codec, "RINPUT3"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "MONO1"); /* Add spitz specific controls */ for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) { diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 2baaa750f12..eae2a0fb45d 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -190,8 +190,8 @@ static int tosa_ac97_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_disable_pin(codec, "OUT3"); - snd_soc_dapm_disable_pin(codec, "MONOOUT"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "MONOOUT"); /* add tosa specific controls */ for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) { -- cgit v1.2.3 From 869fbb36eeb599eb284548232dce40bb413ed2e9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 27 Sep 2008 10:48:31 +0100 Subject: ALSA: ASoC: Use snd_soc_dapm_nc_pin() in N810 machine driver Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/omap/n810.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index d166b6b2a60..fae3ad36e0b 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -247,9 +247,9 @@ static int n810_aic33_init(struct snd_soc_codec *codec) int i, err; /* Not connected */ - snd_soc_dapm_disable_pin(codec, "MONO_LOUT"); - snd_soc_dapm_disable_pin(codec, "HPLCOM"); - snd_soc_dapm_disable_pin(codec, "HPRCOM"); + snd_soc_dapm_nc_pin(codec, "MONO_LOUT"); + snd_soc_dapm_nc_pin(codec, "HPLCOM"); + snd_soc_dapm_nc_pin(codec, "HPRCOM"); /* Add N810 specific controls */ for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) { -- cgit v1.2.3 From c1f27190a72e9310f1777261b33a05319ff2822c Mon Sep 17 00:00:00 2001 From: Arun KS Date: Thu, 2 Oct 2008 14:45:49 +0530 Subject: ALSA: ASoC: Add TLV320AIC23 codec driver ASoC codec driver for TLV320AIC23 device Signed-off-by: Arun KS Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic23.c | 670 +++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic23.h | 122 ++++++++ 4 files changed, 799 insertions(+) create mode 100644 sound/soc/codecs/tlv320aic23.c create mode 100644 sound/soc/codecs/tlv320aic23.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0507fcf6608..bdead2dc996 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -7,6 +7,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 select SND_SOC_CS4270 select SND_SOC_SSM2602 + select SND_SOC_TLV320AIC23 select SND_SOC_TLV320AIC26 select SND_SOC_TLV320AIC3X select SND_SOC_UDA1380 @@ -62,6 +63,10 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_SSM2602 tristate +config SND_SOC_TLV320AIC23 + tristate + depends on I2C + config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" depends on SND_SOC && SPI diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 07318445a1f..90f0a585fc7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -4,6 +4,7 @@ snd-soc-ad73311-objs := ad73311.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-uda1380-objs := uda1380.o @@ -25,6 +26,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c new file mode 100644 index 00000000000..c2d35e9de33 --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.c @@ -0,0 +1,670 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd., + * + * Based on sound/soc/codecs/wm8731.c by Richard Purdie + * + * 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. + * + * Notes: + * The AIC23 is a driver for a low power stereo audio + * codec tlv320aic23 + * + * The machine layer should disable unsupported inputs/outputs by + * snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "tlv320aic23.h" + +#define AUDIO_NAME "tlv320aic23" +#define AIC23_VERSION "0.1" + +struct tlv320aic23_srate_reg_info { + u32 sample_rate; + u8 control; /* SR3, SR2, SR1, SR0 and BOSR */ + u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ +}; + +/* + * AIC23 register cache + */ +static const u16 tlv320aic23_reg[] = { + 0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */ + 0x001A, 0x0004, 0x0007, 0x0001, /* 4 */ + 0x0020, 0x0000, 0x0000, 0x0000, /* 8 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ +}; + +/* + * read tlv320aic23 register cache + */ +static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec + *codec, unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= ARRAY_SIZE(tlv320aic23_reg)) + return -1; + return cache[reg]; +} + +/* + * write tlv320aic23 register cache + */ +static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u16 value) +{ + u16 *cache = codec->reg_cache; + if (reg >= ARRAY_SIZE(tlv320aic23_reg)) + return; + cache[reg] = value; +} + +/* + * write to the tlv320aic23 register space + */ +static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + + u8 data; + + /* TLV320AIC23 has 7 bit address and 9 bits of data + * so we need to switch one data bit into reg and rest + * of data into val + */ + + if ((reg < 0 || reg > 9) && (reg != 15)) { + printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg); + return -1; + } + + data = (reg << 1) | (value >> 8 & 0x01); + + tlv320aic23_write_reg_cache(codec, reg, value); + + if (codec->hw_write(codec->control_data, data, + (value & 0xff)) == 0) + return 0; + + printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__, + value, reg); + + return -EIO; +} + +static const char *rec_src_text[] = { "Line", "Mic" }; +static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const char *sidetone_text[] = {"-6db", "-9db", "-12db", "-18db", "0db"}; + +static const struct soc_enum rec_src_enum = + SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); + +static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls = +SOC_DAPM_ENUM("Input Select", rec_src_enum); + +static const struct soc_enum tlv320aic23_rec_src = + SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); +static const struct soc_enum tlv320aic23_deemph = + SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text); +static const struct soc_enum tlv320aic23_sidetone = + SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 6, 5, sidetone_text); + +static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0); +static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0); + +static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { + SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL, + TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv), + SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1), + SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL, + TLV320AIC23_RINVOL, 7, 1, 0), + SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL, + TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv), + SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1), + SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0), + SOC_ENUM("Sidetone Gain", tlv320aic23_sidetone), + SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph), +}; + +/* add non dapm controls */ +static int tlv320aic23_add_controls(struct snd_soc_codec *codec) +{ + + int err, i; + + for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&tlv320aic23_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; + +} + +/* PGA Mixer controls for Line and Mic switch */ +static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0), + SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1), + SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0, + &tlv320aic23_rec_src_mux_controls), + SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1, + &tlv320aic23_output_mixer_controls[0], + ARRAY_SIZE(tlv320aic23_output_mixer_controls)), + SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0), + + SND_SOC_DAPM_OUTPUT("LHPOUT"), + SND_SOC_DAPM_OUTPUT("RHPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + + SND_SOC_DAPM_INPUT("LLINEIN"), + SND_SOC_DAPM_INPUT("RLINEIN"), + + SND_SOC_DAPM_INPUT("MICIN"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Output Mixer */ + {"Output Mixer", "Line Bypass Switch", "Line Input"}, + {"Output Mixer", "Playback Switch", "DAC"}, + {"Output Mixer", "Mic Sidetone Switch", "Mic Input"}, + + /* Outputs */ + {"RHPOUT", NULL, "Output Mixer"}, + {"LHPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + {"ROUT", NULL, "Output Mixer"}, + + /* Inputs */ + {"Line Input", "NULL", "LLINEIN"}, + {"Line Input", "NULL", "RLINEIN"}, + + {"Mic Input", "NULL", "MICIN"}, + + /* input mux */ + {"Capture Source", "Line", "Line Input"}, + {"Capture Source", "Mic", "Mic Input"}, + {"ADC", NULL, "Capture Source"}, + +}; + +/* tlv320aic23 related */ +static const struct tlv320aic23_srate_reg_info srate_reg_info[] = { + {4000, 0x06, 1}, /* 4000 */ + {8000, 0x06, 0}, /* 8000 */ + {16000, 0x0C, 1}, /* 16000 */ + {22050, 0x11, 1}, /* 22050 */ + {24000, 0x00, 1}, /* 24000 */ + {32000, 0x0C, 0}, /* 32000 */ + {44100, 0x11, 0}, /* 44100 */ + {48000, 0x00, 0}, /* 48000 */ + {88200, 0x1F, 0}, /* 88200 */ + {96000, 0x0E, 0}, /* 96000 */ +}; + +static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 iface_reg, data; + u8 count = 0; + + iface_reg = + tlv320aic23_read_reg_cache(codec, + TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + + /* Search for the right sample rate */ + /* Verify what happens if the rate is not supported + * now it goes to 96Khz */ + while ((srate_reg_info[count].sample_rate != params_rate(params)) && + (count < ARRAY_SIZE(srate_reg_info))) { + count++; + } + + data = (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) | + (srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) | + TLV320AIC23_USB_CLK_ON; + + tlv320aic23_write(codec, TLV320AIC23_SRATE, data); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_reg |= (0x01 << 2); + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_reg |= (0x02 << 2); + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_reg |= (0x03 << 2); + break; + } + tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + + return 0; +} + +static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + + /* set active */ + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001); + + return 0; +} + +static void tlv320aic23_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + + /* deactivate */ + if (!codec->active) { + udelay(50); + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + } +} + +static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 reg; + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT); + if (mute) + reg |= TLV320AIC23_DACM_MUTE; + + else + reg &= ~TLV320AIC23_DACM_MUTE; + + tlv320aic23_write(codec, TLV320AIC23_DIGT, reg); + + return 0; +} + +static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface_reg; + + iface_reg = + tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface_reg |= TLV320AIC23_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface_reg |= TLV320AIC23_FOR_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg |= TLV320AIC23_FOR_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg |= TLV320AIC23_FOR_LJUST; + break; + default: + return -EINVAL; + + } + + tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + + return 0; +} + +static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + switch (freq) { + case 12000000: + return 0; + } + return -EINVAL; +} + +static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f; + + switch (level) { + case SND_SOC_BIAS_ON: + /* vref/mid, osc on, dac unmute */ + tlv320aic23_write(codec, TLV320AIC23_PWR, reg); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +#define AIC23_RATES SNDRV_PCM_RATE_8000_96000 +#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai tlv320aic23_dai = { + .name = "tlv320aic23", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .ops = { + .prepare = tlv320aic23_pcm_prepare, + .hw_params = tlv320aic23_hw_params, + .shutdown = tlv320aic23_shutdown, + }, + .dai_ops = { + .digital_mute = tlv320aic23_mute, + .set_fmt = tlv320aic23_set_dai_fmt, + .set_sysclk = tlv320aic23_set_dai_sysclk, + } +}; +EXPORT_SYMBOL_GPL(tlv320aic23_dai); + +static int tlv320aic23_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int tlv320aic23_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u16 reg; + + /* Sync reg_cache with the hardware */ + for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) { + u16 val = tlv320aic23_read_reg_cache(codec, reg); + tlv320aic23_write(codec, reg, val); + } + + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + tlv320aic23_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +/* + * initialise the AIC23 driver + * register the mixer and dsp interfaces with the kernel + */ +static int tlv320aic23_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + u16 reg; + + codec->name = "tlv320aic23"; + codec->owner = THIS_MODULE; + codec->read = tlv320aic23_read_reg_cache; + codec->write = tlv320aic23_write; + codec->set_bias_level = tlv320aic23_set_bias_level; + codec->dai = &tlv320aic23_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(tlv320aic23_reg); + codec->reg_cache = + kmemdup(tlv320aic23_reg, sizeof(tlv320aic23_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* Reset codec */ + tlv320aic23_write(codec, TLV320AIC23_RESET, 0); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); + + /* Unmute input */ + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL); + tlv320aic23_write(codec, TLV320AIC23_LINVOL, + (reg & (~TLV320AIC23_LIM_MUTED)) | + (TLV320AIC23_LRS_ENABLED)); + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL); + tlv320aic23_write(codec, TLV320AIC23_RINVOL, + (reg & (~TLV320AIC23_LIM_MUTED)) | + TLV320AIC23_LRS_ENABLED); + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG); + tlv320aic23_write(codec, TLV320AIC23_ANLG, + (reg) & (~TLV320AIC23_BYPASS_ON) & + (~TLV320AIC23_MICM_MUTED)); + + /* Default output volume */ + tlv320aic23_write(codec, TLV320AIC23_LCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & + TLV320AIC23_OUT_VOL_MASK); + tlv320aic23_write(codec, TLV320AIC23_RCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & + TLV320AIC23_OUT_VOL_MASK); + + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1); + + tlv320aic23_add_controls(codec); + tlv320aic23_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} +static struct snd_soc_device *tlv320aic23_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + * If the i2c layer weren't so broken, we could pass this kind of data + * around + */ +static int tlv320aic23_codec_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct snd_soc_device *socdev = tlv320aic23_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) + return -EINVAL; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = tlv320aic23_init(socdev); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to initialise AIC23\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} +static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c) +{ + put_device(&i2c->dev); + return 0; +} + +static const struct i2c_device_id tlv320aic23_id[] = { + {"tlv320aic23", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, tlv320aic23_id); + +static struct i2c_driver tlv320aic23_i2c_driver = { + .driver = { + .name = "tlv320aic23", + }, + .probe = tlv320aic23_codec_probe, + .remove = __exit_p(tlv320aic23_i2c_remove), + .id_table = tlv320aic23_id, +}; + +#endif + +static int tlv320aic23_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + tlv320aic23_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data; + codec->hw_read = NULL; + ret = i2c_add_driver(&tlv320aic23_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); +#endif + return ret; +} + +static int tlv320aic23_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&tlv320aic23_i2c_driver); +#endif + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} +struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = { + .probe = tlv320aic23_probe, + .remove = tlv320aic23_remove, + .suspend = tlv320aic23_suspend, + .resume = tlv320aic23_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23); + +MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); +MODULE_AUTHOR("Arun KS "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h new file mode 100644 index 00000000000..79d1faf8e57 --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.h @@ -0,0 +1,122 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd + * + * 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. + */ + +#ifndef _TLV320AIC23_H +#define _TLV320AIC23_H + +/* Codec TLV320AIC23 */ +#define TLV320AIC23_LINVOL 0x00 +#define TLV320AIC23_RINVOL 0x01 +#define TLV320AIC23_LCHNVOL 0x02 +#define TLV320AIC23_RCHNVOL 0x03 +#define TLV320AIC23_ANLG 0x04 +#define TLV320AIC23_DIGT 0x05 +#define TLV320AIC23_PWR 0x06 +#define TLV320AIC23_DIGT_FMT 0x07 +#define TLV320AIC23_SRATE 0x08 +#define TLV320AIC23_ACTIVE 0x09 +#define TLV320AIC23_RESET 0x0F + +/* Left (right) line input volume control register */ +#define TLV320AIC23_LRS_ENABLED 0x0100 +#define TLV320AIC23_LIM_MUTED 0x0080 +#define TLV320AIC23_LIV_DEFAULT 0x0017 +#define TLV320AIC23_LIV_MAX 0x001f +#define TLV320AIC23_LIV_MIN 0x0000 + +/* Left (right) channel headphone volume control register */ +#define TLV320AIC23_LZC_ON 0x0080 +#define TLV320AIC23_LHV_DEFAULT 0x0079 +#define TLV320AIC23_LHV_MAX 0x007f +#define TLV320AIC23_LHV_MIN 0x0000 + +/* Analog audio path control register */ +#define TLV320AIC23_STA_REG(x) ((x)<<6) +#define TLV320AIC23_STE_ENABLED 0x0020 +#define TLV320AIC23_DAC_SELECTED 0x0010 +#define TLV320AIC23_BYPASS_ON 0x0008 +#define TLV320AIC23_INSEL_MIC 0x0004 +#define TLV320AIC23_MICM_MUTED 0x0002 +#define TLV320AIC23_MICB_20DB 0x0001 + +/* Digital audio path control register */ +#define TLV320AIC23_DACM_MUTE 0x0008 +#define TLV320AIC23_DEEMP_32K 0x0002 +#define TLV320AIC23_DEEMP_44K 0x0004 +#define TLV320AIC23_DEEMP_48K 0x0006 +#define TLV320AIC23_ADCHP_ON 0x0001 + +/* Power control down register */ +#define TLV320AIC23_DEVICE_PWR_OFF 0x0080 +#define TLV320AIC23_CLK_OFF 0x0040 +#define TLV320AIC23_OSC_OFF 0x0020 +#define TLV320AIC23_OUT_OFF 0x0010 +#define TLV320AIC23_DAC_OFF 0x0008 +#define TLV320AIC23_ADC_OFF 0x0004 +#define TLV320AIC23_MIC_OFF 0x0002 +#define TLV320AIC23_LINE_OFF 0x0001 + +/* Digital audio interface register */ +#define TLV320AIC23_MS_MASTER 0x0040 +#define TLV320AIC23_LRSWAP_ON 0x0020 +#define TLV320AIC23_LRP_ON 0x0010 +#define TLV320AIC23_IWL_16 0x0000 +#define TLV320AIC23_IWL_20 0x0004 +#define TLV320AIC23_IWL_24 0x0008 +#define TLV320AIC23_IWL_32 0x000C +#define TLV320AIC23_FOR_I2S 0x0002 +#define TLV320AIC23_FOR_DSP 0x0003 +#define TLV320AIC23_FOR_LJUST 0x0001 + +/* Sample rate control register */ +#define TLV320AIC23_CLKOUT_HALF 0x0080 +#define TLV320AIC23_CLKIN_HALF 0x0040 +#define TLV320AIC23_BOSR_384fs 0x0002 /* BOSR_272fs in USB mode */ +#define TLV320AIC23_USB_CLK_ON 0x0001 +#define TLV320AIC23_SR_MASK 0xf +#define TLV320AIC23_CLKOUT_SHIFT 7 +#define TLV320AIC23_CLKIN_SHIFT 6 +#define TLV320AIC23_SR_SHIFT 2 +#define TLV320AIC23_BOSR_SHIFT 1 + +/* Digital interface register */ +#define TLV320AIC23_ACT_ON 0x0001 + +/* + * AUDIO related MACROS + */ + +#define TLV320AIC23_DEFAULT_OUT_VOL 0x70 +#define TLV320AIC23_DEFAULT_IN_VOLUME 0x10 + +#define TLV320AIC23_OUT_VOL_MIN TLV320AIC23_LHV_MIN +#define TLV320AIC23_OUT_VOL_MAX TLV320AIC23_LHV_MAX +#define TLV320AIC23_OUT_VO_RANGE (TLV320AIC23_OUT_VOL_MAX - \ + TLV320AIC23_OUT_VOL_MIN) +#define TLV320AIC23_OUT_VOL_MASK TLV320AIC23_OUT_VOL_MAX + +#define TLV320AIC23_IN_VOL_MIN TLV320AIC23_LIV_MIN +#define TLV320AIC23_IN_VOL_MAX TLV320AIC23_LIV_MAX +#define TLV320AIC23_IN_VOL_RANGE (TLV320AIC23_IN_VOL_MAX - \ + TLV320AIC23_IN_VOL_MIN) +#define TLV320AIC23_IN_VOL_MASK TLV320AIC23_IN_VOL_MAX + +#define TLV320AIC23_SIDETONE_MASK 0x1c0 +#define TLV320AIC23_SIDETONE_0 0x100 +#define TLV320AIC23_SIDETONE_6 0x000 +#define TLV320AIC23_SIDETONE_9 0x040 +#define TLV320AIC23_SIDETONE_12 0x080 +#define TLV320AIC23_SIDETONE_18 0x0c0 + +extern struct snd_soc_dai tlv320aic23_dai; +extern struct snd_soc_codec_device soc_codec_dev_tlv320aic23; + +#endif /* _TLV320AIC23_H */ -- cgit v1.2.3 From 17f9ecf34aaa0ade5c89aba603847309c849297c Mon Sep 17 00:00:00 2001 From: Arun KS Date: Thu, 2 Oct 2008 15:02:45 +0530 Subject: ALSA: ASoC: Add support for osk5912 Adding ASoC machine driver for osk5912 Signed-off-by: Arun KS Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/omap/Kconfig | 8 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/osk5912.c | 232 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 242 insertions(+) create mode 100644 sound/soc/omap/osk5912.c (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index aea27e70043..8b7766b998d 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -13,3 +13,11 @@ config SND_OMAP_SOC_N810 select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on Nokia N810. + +config SND_OMAP_SOC_OSK5912 + tristate "SoC Audio support for omap osk5912" + depends on SND_OMAP_SOC && MACH_OMAP_OSK + select SND_OMAP_SOC_MCBSP + select SND_SOC_TLV320AIC23 + help + Say Y if you want to add support for SoC audio on osk5912. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index d8d8d58075e..e09d1f297f6 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -7,5 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o # OMAP Machine Support snd-soc-n810-objs := n810.o +snd-soc-osk5912-objs := osk5912.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o +obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c new file mode 100644 index 00000000000..0fe73379689 --- /dev/null +++ b/sound/soc/omap/osk5912.c @@ -0,0 +1,232 @@ +/* + * osk5912.c -- SoC audio for OSK 5912 + * + * Copyright (C) 2008 Mistral Solutions + * + * Contact: Arun KS + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/tlv320aic23.h" + +#define CODEC_CLOCK 12000000 + +static struct clk *tlv320aic23_mclk; + +static int osk_startup(struct snd_pcm_substream *substream) +{ + return clk_enable(tlv320aic23_mclk); +} + +static void osk_shutdown(struct snd_pcm_substream *substream) +{ + clk_disable(tlv320aic23_mclk); +} + +static int osk_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err; + + /* Set codec DAI configuration */ + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return err; + } + + /* Set cpu DAI configuration */ + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return err; + } + + /* Set the codec system clock for DAC and ADC */ + err = + snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); + + if (err < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return err; + } + + return err; +} + +static struct snd_soc_ops osk_ops = { + .startup = osk_startup, + .hw_params = osk_hw_params, + .shutdown = osk_shutdown, +}; + +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "LHPOUT"}, + {"Headphone Jack", NULL, "RHPOUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, + + {"MICIN", NULL, "Mic Jack"}, +}; + +static int osk_tlv320aic23_init(struct snd_soc_codec *codec) +{ + + /* Add osk5912 specific widgets */ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* Set up osk5912 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link osk_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &tlv320aic23_dai, + .init = osk_tlv320aic23_init, + .ops = &osk_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_osk = { + .name = "OSK5912", + .dai_link = &osk_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device osk_snd_devdata = { + .machine = &snd_soc_machine_osk, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static struct platform_device *osk_snd_device; + +static int __init osk_soc_init(void) +{ + int err; + u32 curRate; + struct device *dev; + + if (!(machine_is_omap_osk())) + return -ENODEV; + + osk_snd_device = platform_device_alloc("soc-audio", -1); + if (!osk_snd_device) + return -ENOMEM; + + platform_set_drvdata(osk_snd_device, &osk_snd_devdata); + osk_snd_devdata.dev = &osk_snd_device->dev; + *(unsigned int *)osk_dai.cpu_dai->private_data = 0; /* McBSP1 */ + err = platform_device_add(osk_snd_device); + if (err) + goto err1; + + dev = &osk_snd_device->dev; + + tlv320aic23_mclk = clk_get(dev, "mclk"); + if (IS_ERR(tlv320aic23_mclk)) { + printk(KERN_ERR "Could not get mclk clock\n"); + return -ENODEV; + } + + if (clk_get_usecount(tlv320aic23_mclk) > 0) { + /* MCLK is already in use */ + printk(KERN_WARNING + "MCLK in use at %d Hz. We change it to %d Hz\n", + (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK); + } + + /* + * Configure 12 MHz output on MCLK. + */ + curRate = (uint) clk_get_rate(tlv320aic23_mclk); + if (curRate != CODEC_CLOCK) { + if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) { + printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n"); + err = -ECANCELED; + goto err1; + } + } + + printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n", + (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK, + clk_get_usecount(tlv320aic23_mclk)); + + return 0; +err1: + clk_put(tlv320aic23_mclk); + platform_device_del(osk_snd_device); + platform_device_put(osk_snd_device); + + return err; + +} + +static void __exit osk_soc_exit(void) +{ + platform_device_unregister(osk_snd_device); +} + +module_init(osk_soc_init); +module_exit(osk_soc_exit); + +MODULE_AUTHOR("Arun KS "); +MODULE_DESCRIPTION("ALSA SoC OSK 5912"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 3336c5b548b71dcc106a0d862675b30fdf58b3f1 Mon Sep 17 00:00:00 2001 From: Arun KS Date: Thu, 2 Oct 2008 15:07:06 +0530 Subject: ALSA: ASoC: Add DSP DAI format support to the OMAP McBSP driver Enables DSP DAI format for McBSP in OMAP platform driver Signed-off-by: Arun KS Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/omap/omap-mcbsp.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 35310e16d7f..fb920e1b551 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -245,6 +245,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); break; + case SND_SOC_DAIFMT_DSP_A: + /* 0-bit data delay */ + regs->rcr2 |= RDATDLY(0); + regs->xcr2 |= XDATDLY(0); + break; default: /* Unsupported data format */ return -EINVAL; -- cgit v1.2.3 From df91ddf178481e68b8517bed0813d032d493efa0 Mon Sep 17 00:00:00 2001 From: Arun KS Date: Fri, 3 Oct 2008 17:07:30 +0530 Subject: ALSA: ASoC: Add custom SOC_SINGLE_TLV for tlv320aic23 codec Replaces SOC_ENUM with custom SOC_SINGLE_TLV for Sidetone volume Signed-off-by: Arun KS Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/tlv320aic23.c | 53 ++++++++++++++++++++++++++++++++++++++---- 1 file changed, 49 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index c2d35e9de33..bb7cfb80ed4 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -113,7 +113,6 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, static const char *rec_src_text[] = { "Line", "Mic" }; static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; -static const char *sidetone_text[] = {"-6db", "-9db", "-12db", "-18db", "0db"}; static const struct soc_enum rec_src_enum = SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); @@ -125,11 +124,56 @@ static const struct soc_enum tlv320aic23_rec_src = SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); static const struct soc_enum tlv320aic23_deemph = SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text); -static const struct soc_enum tlv320aic23_sidetone = - SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 6, 5, sidetone_text); static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0); static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(sidetone_vol_tlv, -1800, 300, 0); + +static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u16 val, reg; + + val = (ucontrol->value.integer.value[0] & 0x07); + + /* linear conversion to userspace + * 000 = -6db + * 001 = -9db + * 010 = -12db + * 011 = -18db (Min) + * 100 = 0db (Max) + */ + val = (val >= 4) ? 4 : (3 - val); + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0); + tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); + + return 0; +} + +static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u16 val; + + val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0); + val = val >> 6; + val = (val >= 4) ? 4 : (3 - val); + ucontrol->value.integer.value[0] = val; + return 0; + +} + +#define SOC_TLV320AIC23_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, .get = snd_soc_tlv320aic23_get_volsw,\ + .put = snd_soc_tlv320aic23_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL, @@ -141,7 +185,8 @@ static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv), SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1), SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0), - SOC_ENUM("Sidetone Gain", tlv320aic23_sidetone), + SOC_TLV320AIC23_SINGLE_TLV("Sidetone Volume", TLV320AIC23_ANLG, + 6, 4, 0, sidetone_vol_tlv), SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph), }; -- cgit v1.2.3 From dd0c0c805d932f34e87ee3c2db9eaee0974bfef8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 6 Oct 2008 16:54:34 +0100 Subject: ALSA: ASoC: Add WM8753 SPI support Implement SPI support for WM8753, cut'n'pasting from the support for WM8731 contributed by Cliff Cai and Alan Horstmann since the wire format is the same for both codecs. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8753.c | 71 +++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/wm8753.h | 1 + 2 files changed, 70 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 8c4df44f334..83ba4199c9c 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -40,6 +40,7 @@ #include #include #include +#include #include #include #include @@ -1719,6 +1720,63 @@ err_driver: } #endif +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8753_spi_probe(struct spi_device *spi) +{ + struct snd_soc_device *socdev = wm8753_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + codec->control_data = spi; + + ret = wm8753_init(socdev); + if (ret < 0) + dev_err(&spi->dev, "failed to initialise WM8753\n"); + + return ret; +} + +static int __devexit wm8753_spi_remove(struct spi_device *spi) +{ + return 0; +} + +static struct spi_driver wm8753_spi_driver = { + .driver = { + .name = "wm8753", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8753_spi_probe, + .remove = __devexit_p(wm8753_spi_remove), +}; + +static int wm8753_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#endif + + static int wm8753_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -1753,8 +1811,14 @@ static int wm8753_probe(struct platform_device *pdev) codec->hw_write = (hw_write_t)i2c_master_send; ret = wm8753_add_i2c_device(pdev, setup); } -#else - /* Add other interfaces here */ +#endif +#if defined(CONFIG_SPI_MASTER) + if (setup->spi) { + codec->hw_write = (hw_write_t)wm8753_spi_write; + ret = spi_register_driver(&wm8753_spi_driver); + if (ret != 0) + printk(KERN_ERR "can't add spi driver"); + } #endif if (ret != 0) { @@ -1797,6 +1861,9 @@ static int wm8753_remove(struct platform_device *pdev) #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8753_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8753_spi_driver); #endif kfree(codec->private_data); kfree(codec); diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h index 7defde069f1..6678379c0a2 100644 --- a/sound/soc/codecs/wm8753.h +++ b/sound/soc/codecs/wm8753.h @@ -79,6 +79,7 @@ #define WM8753_ADCTL2 0x3f struct wm8753_setup_data { + int spi; int i2c_bus; unsigned short i2c_address; }; -- cgit v1.2.3 From 5e357952b186555afa0ff4da87431c16503a8ad7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Oct 2008 11:56:20 +0100 Subject: ALSA: ASoC: Add WM8510 SPI support Implement SPI support for WM8510, cut'n'pasting from the support for WM8731 contributed by Cliff Cai and Alan Horstmann since the wire format is the same for both codecs. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8510.c | 70 +++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/wm8510.h | 1 + 2 files changed, 69 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 9a37c8d95ed..16768a5acc4 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -747,6 +748,62 @@ err_driver: } #endif +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8510_spi_probe(struct spi_device *spi) +{ + struct snd_soc_device *socdev = wm8510_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + codec->control_data = spi; + + ret = wm8510_init(socdev); + if (ret < 0) + dev_err(&spi->dev, "failed to initialise WM8510\n"); + + return ret; +} + +static int __devexit wm8510_spi_remove(struct spi_device *spi) +{ + return 0; +} + +static struct spi_driver wm8510_spi_driver = { + .driver = { + .name = "wm8510", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8510_spi_probe, + .remove = __devexit_p(wm8510_spi_remove), +}; + +static int wm8510_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#endif /* CONFIG_SPI_MASTER */ + static int wm8510_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -772,8 +829,14 @@ static int wm8510_probe(struct platform_device *pdev) codec->hw_write = (hw_write_t)i2c_master_send; ret = wm8510_add_i2c_device(pdev, setup); } -#else - /* Add other interfaces here */ +#endif +#if defined(CONFIG_SPI_MASTER) + if (setup->spi) { + codec->hw_write = (hw_write_t)wm8510_spi_write; + ret = spi_register_driver(&wm8510_spi_driver); + if (ret != 0) + printk(KERN_ERR "can't add spi driver"); + } #endif if (ret != 0) @@ -795,6 +858,9 @@ static int wm8510_remove(struct platform_device *pdev) #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8510_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8510_spi_driver); #endif kfree(codec); diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h index c5368396045..bdefcf5c69f 100644 --- a/sound/soc/codecs/wm8510.h +++ b/sound/soc/codecs/wm8510.h @@ -94,6 +94,7 @@ #define WM8510_MCLKDIV_12 (7 << 5) struct wm8510_setup_data { + int spi; int i2c_bus; unsigned short i2c_address; }; -- cgit v1.2.3 From e78cc18d91f23edd9c5319bc1b15a540e351d942 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Tue, 7 Oct 2008 14:49:23 +0300 Subject: ALSA: ASoC: tlv320aic3x: Use uniform tlv320aic naming Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/tlv320aic3x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 566a427c928..57fc5aff054 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -991,7 +991,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected); SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) struct snd_soc_dai aic3x_dai = { - .name = "aic3x", + .name = "tlv320aic3x", .playback = { .stream_name = "Playback", .channels_min = 1, @@ -1055,7 +1055,7 @@ static int aic3x_init(struct snd_soc_device *socdev) struct aic3x_setup_data *setup = socdev->codec_data; int reg, ret = 0; - codec->name = "aic3x"; + codec->name = "tlv320aic3x"; codec->owner = THIS_MODULE; codec->read = aic3x_read_reg_cache; codec->write = aic3x_write; -- cgit v1.2.3 From 3ab57fbe91994e5d6fb371a34390520c6c905bee Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Tue, 7 Oct 2008 14:49:22 +0300 Subject: ALSA: ASoC: Remove unused AUDIO_NAME define from codec drivers Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/ssm2602.c | 1 - sound/soc/codecs/tlv320aic23.c | 1 - sound/soc/codecs/tlv320aic3x.c | 1 - sound/soc/codecs/uda1380.c | 1 - sound/soc/codecs/wm8510.c | 1 - sound/soc/codecs/wm8580.c | 1 - sound/soc/codecs/wm8731.c | 1 - sound/soc/codecs/wm8750.c | 1 - sound/soc/codecs/wm8753.c | 1 - sound/soc/codecs/wm8971.c | 1 - sound/soc/codecs/wm8990.c | 1 - 12 files changed, 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 088cf992772..2a89b5888e1 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -28,7 +28,6 @@ #include "ak4535.h" -#define AUDIO_NAME "ak4535" #define AK4535_VERSION "0.3" struct snd_soc_codec_device soc_codec_dev_ak4535; diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 940ce1c3522..44ef0dacd56 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -42,7 +42,6 @@ #include "ssm2602.h" -#define AUDIO_NAME "ssm2602" #define SSM2602_VERSION "0.1" struct snd_soc_codec_device soc_codec_dev_ssm2602; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index bb7cfb80ed4..bac7815e00f 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -35,7 +35,6 @@ #include "tlv320aic23.h" -#define AUDIO_NAME "tlv320aic23" #define AIC23_VERSION "0.1" struct tlv320aic23_srate_reg_info { diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 57fc5aff054..05336ed7e49 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -48,7 +48,6 @@ #include "tlv320aic3x.h" -#define AUDIO_NAME "aic3x" #define AIC3X_VERSION "0.2" /* codec private data */ diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index d206d7f892b..a69ee72a7af 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -36,7 +36,6 @@ #include "uda1380.h" #define UDA1380_VERSION "0.6" -#define AUDIO_NAME "uda1380" /* * uda1380 register cache diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 16768a5acc4..142c49bf18e 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -28,7 +28,6 @@ #include "wm8510.h" -#define AUDIO_NAME "wm8510" #define WM8510_VERSION "0.6" struct snd_soc_codec_device soc_codec_dev_wm8510; diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index df1ffbe305b..056787f800b 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -36,7 +36,6 @@ #include "wm8580.h" -#define AUDIO_NAME "wm8580" #define WM8580_VERSION "0.1" struct pll_state { diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 7b64d9a7ff7..7f8a7e36b33 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -29,7 +29,6 @@ #include "wm8731.h" -#define AUDIO_NAME "wm8731" #define WM8731_VERSION "0.13" struct snd_soc_codec_device soc_codec_dev_wm8731; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 4892e398a59..9b7296ee5b0 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -29,7 +29,6 @@ #include "wm8750.h" -#define AUDIO_NAME "WM8750" #define WM8750_VERSION "0.12" /* codec private data */ diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 83ba4199c9c..63dbc56a303 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -52,7 +52,6 @@ #include "wm8753.h" -#define AUDIO_NAME "wm8753" #define WM8753_VERSION "0.16" static int caps_charge = 2000; diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 974a4cd0f3f..f41a578ddd4 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -29,7 +29,6 @@ #include "wm8971.h" -#define AUDIO_NAME "wm8971" #define WM8971_VERSION "0.9" #define WM8971_REG_COUNT 43 diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 63410d7b5ef..572d22b0880 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -30,7 +30,6 @@ #include "wm8990.h" -#define AUDIO_NAME "wm8990" #define WM8990_VERSION "0.2" /* codec private data */ -- cgit v1.2.3 From 09af98b08f72471ea53efe26494eef0947a6a10d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Oct 2008 13:04:58 +0100 Subject: ALSA: ASoC: Implement WM8510 bias level control The WM8510 bias level configuration blindly overwrites the power management registers, interfering with the operation of DAPM. Only adjust the specific bits required, implementing use of the VMID resistor string configuration control as we go. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8510.c | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 142c49bf18e..94cab495d5f 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -55,6 +55,9 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { 0x0001, }; +#define WM8510_POWER1_BIASEN 0x08 +#define WM8510_POWER1_BUFIOEN 0x10 + /* * read wm8510 register cache */ @@ -526,23 +529,35 @@ static int wm8510_mute(struct snd_soc_dai *dai, int mute) static int wm8510_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + u16 power1 = wm8510_read_reg_cache(codec, WM8510_POWER1) & ~0x3; switch (level) { case SND_SOC_BIAS_ON: - wm8510_write(codec, WM8510_POWER1, 0x1ff); - wm8510_write(codec, WM8510_POWER2, 0x1ff); - wm8510_write(codec, WM8510_POWER3, 0x1ff); - break; case SND_SOC_BIAS_PREPARE: + power1 |= 0x1; /* VMID 50k */ + wm8510_write(codec, WM8510_POWER1, power1); + break; + case SND_SOC_BIAS_STANDBY: + power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN; + + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Initial cap charge at VMID 5k */ + wm8510_write(codec, WM8510_POWER1, power1 | 0x3); + mdelay(100); + } + + power1 |= 0x2; /* VMID 500k */ + wm8510_write(codec, WM8510_POWER1, power1); break; + case SND_SOC_BIAS_OFF: - /* everything off, dac mute, inactive */ - wm8510_write(codec, WM8510_POWER1, 0x0); - wm8510_write(codec, WM8510_POWER2, 0x0); - wm8510_write(codec, WM8510_POWER3, 0x0); + wm8510_write(codec, WM8510_POWER1, 0); + wm8510_write(codec, WM8510_POWER2, 0); + wm8510_write(codec, WM8510_POWER3, 0); break; } + codec->bias_level = level; return 0; } @@ -640,6 +655,7 @@ static int wm8510_init(struct snd_soc_device *socdev) } /* power on device */ + codec->bias_level = SND_SOC_BIAS_OFF; wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); wm8510_add_controls(codec); wm8510_add_widgets(codec); -- cgit v1.2.3 From 2b5f34c5556fc6480bcace016fc35d9d2921c38f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Oct 2008 16:13:50 +0100 Subject: ALSA: ASoC: Make WM8510 microphone input a DAPM mixer The WM8510 microphone input PGA was represented as a DAPM PGA but in DAPM terms the functionality is that of a mixer since it takes three switchable inputs and produces one output. Representing it as an input was causing its controls to be misinterpreted as gain controls and would cause some required DAPM updates to be missed. Reported-by: Jukka Hynninen Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8510.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 94cab495d5f..ea524c4ce9f 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -227,9 +227,9 @@ SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0), SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0), SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0), -SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0, - &wm8510_micpga_controls[0], - ARRAY_SIZE(wm8510_micpga_controls)), +SND_SOC_DAPM_MIXER("Mic PGA", WM8510_POWER2, 2, 0, + &wm8510_micpga_controls[0], + ARRAY_SIZE(wm8510_micpga_controls)), SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0, &wm8510_boost_controls[0], ARRAY_SIZE(wm8510_boost_controls)), -- cgit v1.2.3 From 446e0f69101baa59de2473f7deba05a730acfe6e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 Oct 2008 08:26:57 +0200 Subject: ALSA: ASoC - clean up Kconfig for TLV320AIC2 Removed unnecessary dependency. Also, make it uninteractive, as it's only for selection by other configs. Signed-off-by: Takashi Iwai --- sound/soc/codecs/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bdead2dc996..11eebce97ae 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -68,8 +68,8 @@ config SND_SOC_TLV320AIC23 depends on I2C config SND_SOC_TLV320AIC26 - tristate "TI TLV320AIC26 Codec support" - depends on SND_SOC && SPI + tristate + depends on SPI config SND_SOC_TLV320AIC3X tristate -- cgit v1.2.3 From 9296bb43f1a3b60ab2e9c4ff48a296cacff117a9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 10 Oct 2008 12:32:16 +0100 Subject: ALSA: ASoC: Make TLV320AIC26 user-visible The TLV320AIC26 Kconfig option is unusual in that it supports the OpenFirmware machine driver which doesn't have a hard binding to the codec driver but discovers the codec via the device tree. This makes it meaningful to select the codec without a machine driver. Ideally there would be a proxy entry so that this option was only visible on OpenFirmware systems. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 11eebce97ae..4975d8573e4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -68,7 +68,7 @@ config SND_SOC_TLV320AIC23 depends on I2C config SND_SOC_TLV320AIC26 - tristate + tristate "TI TLV320AIC26 Codec support" depends on SPI config SND_SOC_TLV320AIC3X -- cgit v1.2.3 From 8def464dddd61686e00e96db714a2930a08ef272 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 9 Oct 2008 15:57:22 +0300 Subject: ALSA: ASoC: OMAP: Add multilink support to McBSP DAI driver Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/omap/omap-mcbsp.c | 72 ++++++++++++++++++++++++++------------------- sound/soc/omap/omap-mcbsp.h | 16 ++++++---- 2 files changed, 53 insertions(+), 35 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index fb920e1b551..d32eb4703cf 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -381,37 +381,49 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return err; } -struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS] = { -{ - .name = "omap-mcbsp-dai", - .id = 0, - .type = SND_SOC_DAI_I2S, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = OMAP_MCBSP_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = OMAP_MCBSP_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = { - .startup = omap_mcbsp_dai_startup, - .shutdown = omap_mcbsp_dai_shutdown, - .trigger = omap_mcbsp_dai_trigger, - .hw_params = omap_mcbsp_dai_hw_params, - }, - .dai_ops = { - .set_fmt = omap_mcbsp_dai_set_dai_fmt, - .set_clkdiv = omap_mcbsp_dai_set_clkdiv, - .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, - }, - .private_data = &mcbsp_data[0].bus_id, -}, +#define OMAP_MCBSP_DAI_BUILDER(link_id) \ +{ \ + .name = "omap-mcbsp-dai-(link_id)", \ + .id = (link_id), \ + .type = SND_SOC_DAI_I2S, \ + .playback = { \ + .channels_min = 2, \ + .channels_max = 2, \ + .rates = OMAP_MCBSP_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ + .capture = { \ + .channels_min = 2, \ + .channels_max = 2, \ + .rates = OMAP_MCBSP_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ + .ops = { \ + .startup = omap_mcbsp_dai_startup, \ + .shutdown = omap_mcbsp_dai_shutdown, \ + .trigger = omap_mcbsp_dai_trigger, \ + .hw_params = omap_mcbsp_dai_hw_params, \ + }, \ + .dai_ops = { \ + .set_fmt = omap_mcbsp_dai_set_dai_fmt, \ + .set_clkdiv = omap_mcbsp_dai_set_clkdiv, \ + .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, \ + }, \ + .private_data = &mcbsp_data[(link_id)].bus_id, \ +} + +struct snd_soc_dai omap_mcbsp_dai[] = { + OMAP_MCBSP_DAI_BUILDER(0), + OMAP_MCBSP_DAI_BUILDER(1), +#if NUM_LINKS >= 3 + OMAP_MCBSP_DAI_BUILDER(2), +#endif +#if NUM_LINKS == 5 + OMAP_MCBSP_DAI_BUILDER(3), + OMAP_MCBSP_DAI_BUILDER(4), +#endif }; + EXPORT_SYMBOL_GPL(omap_mcbsp_dai); MODULE_AUTHOR("Jarkko Nikula "); diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index ed8afb55067..df7ad13ba73 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -38,11 +38,17 @@ enum omap_mcbsp_div { OMAP_MCBSP_CLKGDV, /* Sample rate generator divider */ }; -/* - * REVISIT: Preparation for the ASoC v2. Let the number of available links to - * be same than number of McBSP ports found in OMAP(s) we are compiling for. - */ -#define NUM_LINKS 1 +#if defined(CONFIG_ARCH_OMAP2420) +#define NUM_LINKS 2 +#endif +#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX) +#undef NUM_LINKS +#define NUM_LINKS 3 +#endif +#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX) +#undef NUM_LINKS +#define NUM_LINKS 5 +#endif extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS]; -- cgit v1.2.3 From 406e2c48cf716411c07aecf2a0e5331ae9521efe Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 9 Oct 2008 15:57:20 +0300 Subject: ALSA: ASoC: OMAP: Add support for OMAP2430 and OMAP34xx in McBSP DAI driver Thanks to Arun KS for fixing one typo in original version of this patch. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/omap/omap-mcbsp.c | 95 ++++++++++++++++++++++++++++++++++++--------- 1 file changed, 77 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d32eb4703cf..e97e6b28b8a 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -84,11 +84,22 @@ static const unsigned long omap1_mcbsp_port[][2] = { static const int omap1_dma_reqs[][2] = {}; static const unsigned long omap1_mcbsp_port[][2] = {}; #endif -#if defined(CONFIG_ARCH_OMAP2420) -static const int omap2420_dma_reqs[][2] = { + +#if defined(CONFIG_ARCH_OMAP24XX) || defined(CONFIG_ARCH_OMAP34XX) +static const int omap24xx_dma_reqs[][2] = { { OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX }, { OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX }, +#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX) + { OMAP24XX_DMA_MCBSP3_TX, OMAP24XX_DMA_MCBSP3_RX }, + { OMAP24XX_DMA_MCBSP4_TX, OMAP24XX_DMA_MCBSP4_RX }, + { OMAP24XX_DMA_MCBSP5_TX, OMAP24XX_DMA_MCBSP5_RX }, +#endif }; +#else +static const int omap24xx_dma_reqs[][2] = {}; +#endif + +#if defined(CONFIG_ARCH_OMAP2420) static const unsigned long omap2420_mcbsp_port[][2] = { { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1, OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 }, @@ -96,10 +107,43 @@ static const unsigned long omap2420_mcbsp_port[][2] = { OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 }, }; #else -static const int omap2420_dma_reqs[][2] = {}; static const unsigned long omap2420_mcbsp_port[][2] = {}; #endif +#if defined(CONFIG_ARCH_OMAP2430) +static const unsigned long omap2430_mcbsp_port[][2] = { + { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, + OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR, + OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DXR, + OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DXR, + OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DXR, + OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DRR }, +}; +#else +static const unsigned long omap2430_mcbsp_port[][2] = {}; +#endif + +#if defined(CONFIG_ARCH_OMAP34XX) +static const unsigned long omap34xx_mcbsp_port[][2] = { + { OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DRR }, +}; +#else +static const unsigned long omap34xx_mcbsp_port[][2] = {}; +#endif + static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -167,12 +211,15 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, dma = omap1_dma_reqs[bus_id][substream->stream]; port = omap1_mcbsp_port[bus_id][substream->stream]; } else if (cpu_is_omap2420()) { - dma = omap2420_dma_reqs[bus_id][substream->stream]; + dma = omap24xx_dma_reqs[bus_id][substream->stream]; port = omap2420_mcbsp_port[bus_id][substream->stream]; + } else if (cpu_is_omap2430()) { + dma = omap24xx_dma_reqs[bus_id][substream->stream]; + port = omap2430_mcbsp_port[bus_id][substream->stream]; + } else if (cpu_is_omap343x()) { + dma = omap24xx_dma_reqs[bus_id][substream->stream]; + port = omap34xx_mcbsp_port[bus_id][substream->stream]; } else { - /* - * TODO: Add support for 2430 and 3430 - */ return -ENODEV; } omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma; @@ -315,7 +362,7 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data, int clk_id) { int sel_bit; - u16 reg; + u16 reg, reg_devconf1 = OMAP243X_CONTROL_DEVCONF1; if (cpu_class_is_omap1()) { /* OMAP1's can use only external source clock */ @@ -325,6 +372,12 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data, return 0; } + if (cpu_is_omap2420() && mcbsp_data->bus_id > 1) + return -EINVAL; + + if (cpu_is_omap343x()) + reg_devconf1 = OMAP343X_CONTROL_DEVCONF1; + switch (mcbsp_data->bus_id) { case 0: reg = OMAP2_CONTROL_DEVCONF0; @@ -334,20 +387,26 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data, reg = OMAP2_CONTROL_DEVCONF0; sel_bit = 6; break; - /* TODO: Support for ports 3 - 5 in OMAP2430 and OMAP34xx */ + case 2: + reg = reg_devconf1; + sel_bit = 0; + break; + case 3: + reg = reg_devconf1; + sel_bit = 2; + break; + case 4: + reg = reg_devconf1; + sel_bit = 4; + break; default: return -EINVAL; } - if (cpu_class_is_omap2()) { - if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK) { - omap_ctrl_writel(omap_ctrl_readl(reg) & - ~(1 << sel_bit), reg); - } else { - omap_ctrl_writel(omap_ctrl_readl(reg) | - (1 << sel_bit), reg); - } - } + if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK) + omap_ctrl_writel(omap_ctrl_readl(reg) & ~(1 << sel_bit), reg); + else + omap_ctrl_writel(omap_ctrl_readl(reg) | (1 << sel_bit), reg); return 0; } -- cgit v1.2.3 From 2e89713a8396ab07b9cccc83e50e55646c235342 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 9 Oct 2008 15:57:21 +0300 Subject: ALSA: ASoC: OMAP: Set DMA stream name at runtime in McBSP DAI driver This suits better when adding support for multiple links and different link formats. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/omap/omap-mcbsp.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index e97e6b28b8a..0a063a98a66 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -59,12 +59,7 @@ static struct omap_mcbsp_data mcbsp_data[NUM_LINKS]; * Stream DMA parameters. DMA request line and port address are set runtime * since they are different between OMAP1 and later OMAPs */ -static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2] = { -{ - { .name = "I2S PCM Stereo out", }, - { .name = "I2S PCM Stereo in", }, -}, -}; +static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2]; #if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX) static const int omap1_dma_reqs[][2] = { @@ -222,6 +217,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, } else { return -ENODEV; } + omap_mcbsp_dai_dma_params[id][substream->stream].name = + substream->stream ? "Audio Capture" : "Audio Playback"; omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma; omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port; cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream]; -- cgit v1.2.3 From 5715952b39ebded49407ff02e58fe2d90904b991 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 24 Sep 2008 10:47:02 +0100 Subject: ALSA: ASoC: Fix inverted input PGA mute bits in WM8903 Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8903.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index a3f54ec4226..ce40d787760 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -653,14 +653,14 @@ static const struct snd_kcontrol_new wm8903_snd_controls[] = { /* Input PGAs - No TLV since the scale depends on PGA mode */ SOC_SINGLE("Left Input PGA Switch", WM8903_ANALOGUE_LEFT_INPUT_0, - 7, 1, 0), + 7, 1, 1), SOC_SINGLE("Left Input PGA Volume", WM8903_ANALOGUE_LEFT_INPUT_0, 0, 31, 0), SOC_SINGLE("Left Input PGA Common Mode Switch", WM8903_ANALOGUE_LEFT_INPUT_1, 6, 1, 0), SOC_SINGLE("Right Input PGA Switch", WM8903_ANALOGUE_RIGHT_INPUT_0, - 7, 1, 0), + 7, 1, 1), SOC_SINGLE("Right Input PGA Volume", WM8903_ANALOGUE_RIGHT_INPUT_0, 0, 31, 0), SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1, -- cgit v1.2.3 From e8089948d65911c78bcd72960dd419ec636d6f0b Mon Sep 17 00:00:00 2001 From: Jonas Bonn Date: Wed, 1 Oct 2008 18:17:12 +0100 Subject: ALSA: ASoC: Add widgets before setting endpoints on GTA01 This prevents error messages at startup where the endpoints are being set before the widgets/controls have even been added. Signed-off-by: Jonas Bonn Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/s3c24xx/neo1973_wm8753.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 006c36ded25..9eda86259e6 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -518,13 +518,13 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) snd_soc_dapm_nc_pin(codec, "LINE1"); snd_soc_dapm_nc_pin(codec, "LINE2"); - /* set endpoints to default mode */ - set_scenario_endpoints(codec, NEO_AUDIO_OFF); - /* Add neo1973 specific widgets */ snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, ARRAY_SIZE(wm8753_dapm_widgets)); + /* set endpoints to default mode */ + set_scenario_endpoints(codec, NEO_AUDIO_OFF); + /* add neo1973 specific controls */ for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) { err = snd_ctl_add(codec->card, -- cgit v1.2.3 From df20cf92cae5640568ee3d48bf7a32987c057413 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 24 Sep 2008 11:57:27 +0100 Subject: ALSA: ASoC: Fix build of GTA01 audio driver Fix a couple of thinkos introduced during the I2C API update. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/s3c24xx/neo1973_wm8753.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 9eda86259e6..f7fc231e238 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -649,7 +649,7 @@ static void lm4857_shutdown(struct i2c_client *dev) } static const struct i2c_device_id lm4857_i2c_id[] = { - { "neo1973_lm4857", 0 } + { "neo1973_lm4857", 0 }, { } }; @@ -736,7 +736,7 @@ static int __init neo1973_init(void) } ret = neo1973_add_lm4857_device(neo1973_snd_device, - neo1973_wm8753_setup, 0x7C); + 0, 0x7C); if (ret != 0) platform_device_unregister(neo1973_snd_device); -- cgit v1.2.3 From f9d1ab39e8c993f183c39a9724ca5ad29b6336e9 Mon Sep 17 00:00:00 2001 From: Jonas Bonn Date: Wed, 1 Oct 2008 21:47:19 +0200 Subject: ALSA: ASoC: Drop device registration from GTA01 lm4857 driver Device registration should be handled at the machine level and not in the driver code itself. This patch removes the device registration from the driver code in preparation for moving it to the machine definition. [Squashed down two parts to this patch for bisectability - there's also a third part adding registration of the device to the out of tree GTA01 machine driver -- broonie] Signed-off-by: Jonas Bonn Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/s3c24xx/neo1973_wm8753.c | 51 +++++--------------------------------- 1 file changed, 6 insertions(+), 45 deletions(-) (limited to 'sound') diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index f7fc231e238..87ddfefcc2f 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -602,6 +602,8 @@ static int lm4857_i2c_probe(struct i2c_client *client, { DBG("Entered %s\n", __func__); + i2c = client; + lm4857_write_regs(); return 0; } @@ -610,6 +612,8 @@ static int lm4857_i2c_remove(struct i2c_client *client) { DBG("Entered %s\n", __func__); + i2c = NULL; + return 0; } @@ -667,48 +671,6 @@ static struct i2c_driver lm4857_i2c_driver = { }; static struct platform_device *neo1973_snd_device; -static struct i2c_client *lm4857_client; - -static int __init neo1973_add_lm4857_device(struct platform_device *pdev, - int i2c_bus, - unsigned short i2c_address) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - int ret; - - ret = i2c_add_driver(&lm4857_i2c_driver); - if (ret != 0) { - dev_err(&pdev->dev, "can't add lm4857 driver\n"); - return ret; - } - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = i2c_address; - strlcpy(info.type, "neo1973_lm4857", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(i2c_bus); - if (!adapter) { - dev_err(&pdev->dev, "can't get i2c adapter %d\n", i2c_bus); - goto err_driver; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - dev_err(&pdev->dev, "can't add lm4857 device at 0x%x\n", - (unsigned int)info.addr); - goto err_driver; - } - - lm4857_client = client; - return 0; - -err_driver: - i2c_del_driver(&lm4857_i2c_driver); - return -ENODEV; -} static int __init neo1973_init(void) { @@ -735,8 +697,8 @@ static int __init neo1973_init(void) return ret; } - ret = neo1973_add_lm4857_device(neo1973_snd_device, - 0, 0x7C); + ret = i2c_add_driver(&lm4857_i2c_driver); + if (ret != 0) platform_device_unregister(neo1973_snd_device); @@ -747,7 +709,6 @@ static void __exit neo1973_exit(void) { DBG("Entered %s\n", __func__); - i2c_unregister_device(lm4857_client); i2c_del_driver(&lm4857_i2c_driver); platform_device_unregister(neo1973_snd_device); } -- cgit v1.2.3 From 9d37484c8ce06d95c53c2bbadfc205faaff834bc Mon Sep 17 00:00:00 2001 From: Arun KS Date: Tue, 30 Sep 2008 15:35:16 +0530 Subject: ALSA: ASoC: Add destination and source port for DMA on OMAP1 Adds destination and source port for dma in platform driver as required by OMAP1 Signed-off-by: Arun KS Acked-by: Jarkko Nikula Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/omap/omap-pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 690bfeaec4a..e9084fdd208 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -97,7 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, prtd->dma_data = dma_data; err = omap_request_dma(dma_data->dma_req, dma_data->name, omap_pcm_dma_irq, substream, &prtd->dma_ch); - if (!cpu_is_omap1510()) { + if (!err & !cpu_is_omap1510()) { /* * Link channel with itself so DMA doesn't need any * reprogramming while looping the buffer @@ -147,12 +147,14 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_or_dst_synch = OMAP_DMA_DST_SYNC; dma_params.src_start = runtime->dma_addr; dma_params.dst_start = dma_data->port_addr; + dma_params.dst_port = OMAP_DMA_PORT_MPUI; } else { dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT; dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC; dma_params.src_or_dst_synch = OMAP_DMA_SRC_SYNC; dma_params.src_start = dma_data->port_addr; dma_params.dst_start = runtime->dma_addr; + dma_params.src_port = OMAP_DMA_PORT_MPUI; } /* * Set DMA transfer frame size equal to ALSA period size and frame -- cgit v1.2.3 From d331124dc2923ec0966a82e3428c532cee8da95f Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Sun, 12 Oct 2008 13:17:36 +0100 Subject: ALSA: ASoC: update email address for Liam Girdwood Update the contact information for Liam Girdwood in ASoC core and drivers as my old email address is no longer valid. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai --- sound/oss/ac97_codec.c | 2 +- sound/pci/ac97/ac97_patch.c | 2 +- sound/soc/at91/at91-ssc.c | 2 +- sound/soc/codecs/ac97.c | 3 +-- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8753.c | 3 +-- sound/soc/codecs/wm8753.h | 3 +-- sound/soc/codecs/wm9712.c | 3 +-- sound/soc/codecs/wm9713.c | 3 +-- sound/soc/pxa/corgi.c | 2 +- sound/soc/pxa/em-x270.c | 2 +- sound/soc/pxa/poodle.c | 2 +- sound/soc/pxa/pxa2xx-i2s.c | 4 ++-- sound/soc/pxa/spitz.c | 2 +- sound/soc/pxa/tosa.c | 2 +- sound/soc/soc-core.c | 5 ++--- sound/soc/soc-dapm.c | 5 ++--- 17 files changed, 20 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/oss/ac97_codec.c b/sound/oss/ac97_codec.c index b63839e8f9b..456a1b4d783 100644 --- a/sound/oss/ac97_codec.c +++ b/sound/oss/ac97_codec.c @@ -30,7 +30,7 @@ ************************************************************************** * * History - * May 02, 2003 Liam Girdwood + * May 02, 2003 Liam Girdwood * Removed non existant WM9700 * Added support for WM9705, WM9708, WM9709, WM9710, WM9711 * WM9712 and WM9717 diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 6ce3cbe98a6..6e831aff1bd 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -476,7 +476,7 @@ static int patch_yamaha_ymf753(struct snd_ac97 * ac97) } /* - * May 2, 2003 Liam Girdwood + * May 2, 2003 Liam Girdwood * removed broken wolfson00 patch. * added support for WM9705,WM9708,WM9709,WM9710,WM9711,WM9712 and WM9717. */ diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c index a5b1a79ebff..1b61cc46126 100644 --- a/sound/soc/at91/at91-ssc.c +++ b/sound/soc/at91/at91-ssc.c @@ -5,7 +5,7 @@ * Endrelia Technologies Inc. * * Based on pxa2xx Platform drivers by - * Liam Girdwood + * Liam Girdwood * * 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 diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 61fd96ca7bc..bd1ebdc6c86 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -2,8 +2,7 @@ * ac97.c -- ALSA Soc AC97 codec support * * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood * * 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 diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index ea524c4ce9f..d8ca2da8d63 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -3,7 +3,7 @@ * * Copyright 2006 Wolfson Microelectronics PLC. * - * Author: Liam Girdwood + * Author: Liam Girdwood * * 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 diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 63dbc56a303..d426eaa2218 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -2,8 +2,7 @@ * wm8753.c -- WM8753 ALSA Soc Audio driver * * Copyright 2003 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood * * 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 diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h index 6678379c0a2..f55704ce931 100644 --- a/sound/soc/codecs/wm8753.h +++ b/sound/soc/codecs/wm8753.h @@ -2,8 +2,7 @@ * wm8753.h -- audio driver for WM8753 * * Copyright 2003 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood * * 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 diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 2f1c91b1d55..ffb471e420e 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -2,8 +2,7 @@ * wm9712.c -- ALSA Soc WM9712 codec support * * Copyright 2006 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood * * 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 diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 441d0580db1..aba402b3c99 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -2,8 +2,7 @@ * wm9713.c -- ALSA Soc WM9713 codec support * * Copyright 2006 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood * * 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 diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 0bceaf66eff..dd7fa0b329c 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood + * Authors: Liam Girdwood * Richard Purdie * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index d9c3f7b28be..e6ff6929ab4 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -9,7 +9,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood + * Authors: Liam Girdwood * Richard Purdie * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index e5adb0e9193..4d9930c5278 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood + * Authors: Liam Girdwood * Richard Purdie * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 39d19212f6d..64057b1d220 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -3,7 +3,7 @@ * * Copyright 2005 Wolfson Microelectronics PLC. * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * lrg@slimlogic.co.uk * * 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 @@ -366,6 +366,6 @@ module_init(pxa2xx_i2s_init); module_exit(pxa2xx_i2s_exit); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk"); MODULE_DESCRIPTION("pxa2xx I2S SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index e0bcc4250ce..8f89188e541 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood + * Authors: Liam Girdwood * Richard Purdie * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index eae2a0fb45d..afefe41b8c4 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood + * Authors: Liam Girdwood * Richard Purdie * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad381138fc2..462e635dfc7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4,8 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood * with code, comments and ideas from :- * Richard Purdie * @@ -1886,7 +1885,7 @@ module_init(snd_soc_init); module_exit(snd_soc_exit); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk"); MODULE_DESCRIPTION("ALSA SoC Core"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:soc-audio"); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 83fa9c47b66..efbd0b37810 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2,8 +2,7 @@ * soc-dapm.c -- ALSA SoC Dynamic Audio Power Management * * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood * * 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 @@ -1541,6 +1540,6 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev) EXPORT_SYMBOL_GPL(snd_soc_dapm_free); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk"); MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC"); MODULE_LICENSE("GPL"); -- cgit v1.2.3 From a7e54e6de3b01d9085202fdbf0110da425f4af38 Mon Sep 17 00:00:00 2001 From: Huang Weiyi Date: Sun, 12 Oct 2008 23:12:56 +0800 Subject: ALSA: ASoC codec: remove unused #include The files below do not use LINUX_VERSION_CODE nor KERNEL_VERSION. sound/soc/codecs/ad1980.c sound/soc/codecs/wm8580.c sound/soc/codecs/wm8900.c This patch removes the said #include . Signed-off-by: Huang Weiyi Signed-off-by: Takashi Iwai --- sound/soc/codecs/ad1980.c | 1 - sound/soc/codecs/wm8580.c | 1 - sound/soc/codecs/wm8900.c | 1 - 3 files changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 4e09c1f2c06..1397b8e06c0 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -13,7 +13,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 056787f800b..627ebfb4209 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -18,7 +18,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 0b8c6d38b48..3b326c9b558 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -18,7 +18,6 @@ #include #include -#include #include #include #include -- cgit v1.2.3 From 4aa02396f934b355a4002ea9bc524aa42d6b53d6 Mon Sep 17 00:00:00 2001 From: Arun KS Date: Mon, 13 Oct 2008 15:47:25 +0530 Subject: ALSA: ASoC: Fix compile-time warning for tlv320aic23.c Fixes this warning: sound/soc/codecs/tlv320aic23.c: In function 'tlv320aic23_write': sound/soc/codecs/tlv320aic23.c:104: warning: passing argument 2 of 'codec->hw_write' makes pointer from integer without a cast Replaces i2c smbus write function with standard i2c write function Signed-off-by: Arun KS Signed-off-by: Takashi Iwai --- sound/soc/codecs/tlv320aic23.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index bac7815e00f..44308dac9e1 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -84,7 +84,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - u8 data; + u8 data[2]; /* TLV320AIC23 has 7 bit address and 9 bits of data * so we need to switch one data bit into reg and rest @@ -96,12 +96,12 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, return -1; } - data = (reg << 1) | (value >> 8 & 0x01); + data[0] = (reg << 1) | (value >> 8 & 0x01); + data[1] = value & 0xff; tlv320aic23_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, - (value & 0xff)) == 0) + if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__, @@ -674,7 +674,7 @@ static int tlv320aic23_probe(struct platform_device *pdev) tlv320aic23_socdev = socdev; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data; + codec->hw_write = (hw_write_t) i2c_master_send; codec->hw_read = NULL; ret = i2c_add_driver(&tlv320aic23_i2c_driver); if (ret != 0) -- cgit v1.2.3 From d5d8d83773165b951d190717637bfbc1eb0111a0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 13 Oct 2008 19:16:14 +0100 Subject: ALSA: ASoC: Hide TLV320AIC26 configuration option for non-OpenFirwmare users Make the visibility of the tristate conditional on having the OpenFirmware helper code enabed so that users who can't use it don't see the visible option. Kconfig ignores dependencies for select so other users are unaffected. Thanks to Takashi for the suggestion. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4975d8573e4..38a0e3b620a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -68,7 +68,7 @@ config SND_SOC_TLV320AIC23 depends on I2C config SND_SOC_TLV320AIC26 - tristate "TI TLV320AIC26 Codec support" + tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE depends on SPI config SND_SOC_TLV320AIC3X -- cgit v1.2.3 From 8825e8e8d09c1fe6352f94c70f6ff73db449ff56 Mon Sep 17 00:00:00 2001 From: Marc Zyngier Date: Tue, 14 Oct 2008 09:57:05 +0100 Subject: ALSA: Fix pxa2xx-ac97-lib.c compilation The last ALSA merge broke pxa2xx-ac97-lib.c, as it brought back references to cpu_is_pxa21x that Eric Miao removed in commit 0ffcbfd54ea81ca24c0749f55ca4fcf3e2bdc23e: [ARM] pxa: make cpu_is_pxa2* macros more consistent This patch gets rid of those references, and only keeps cpu_is_pxa25x(). Signed-off-by: Marc Zyngier Acked-by: Eric Miao Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/arm/pxa2xx-ac97-lib.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 99026dfb81e..34c1d94f921 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -50,7 +50,7 @@ unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg) mutex_lock(&car_mutex); /* set up primary or secondary codec space */ - if ((cpu_is_pxa21x() || cpu_is_pxa25x()) && reg == AC97_GPIO_STATUS) + if (cpu_is_pxa25x() && reg == AC97_GPIO_STATUS) reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE; else reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; @@ -90,7 +90,7 @@ void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, mutex_lock(&car_mutex); /* set up primary or secondary codec space */ - if ((cpu_is_pxa21x() || cpu_is_pxa25x()) && reg == AC97_GPIO_STATUS) + if (cpu_is_pxa25x() && reg == AC97_GPIO_STATUS) reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE; else reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; @@ -200,7 +200,7 @@ static inline void pxa_ac97_cold_pxa3xx(void) bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) { #ifdef CONFIG_PXA25x - if (cpu_is_pxa21x() || cpu_is_pxa25x()) + if (cpu_is_pxa25x()) pxa_ac97_warm_pxa25x(); else #endif @@ -230,7 +230,7 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_warm_reset); bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) { #ifdef CONFIG_PXA25x - if (cpu_is_pxa21x() || cpu_is_pxa25x()) + if (cpu_is_pxa25x()) pxa_ac97_cold_pxa25x(); else #endif @@ -301,7 +301,7 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_suspend); int pxa2xx_ac97_hw_resume(void) { - if (cpu_is_pxa21x() || cpu_is_pxa25x() || cpu_is_pxa27x()) { + if (cpu_is_pxa25x() || cpu_is_pxa27x()) { pxa_gpio_mode(GPIO31_SYNC_AC97_MD); pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); @@ -325,7 +325,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (ret < 0) goto err; - if (cpu_is_pxa21x() || cpu_is_pxa25x() || cpu_is_pxa27x()) { + if (cpu_is_pxa25x() || cpu_is_pxa27x()) { pxa_gpio_mode(GPIO31_SYNC_AC97_MD); pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); -- cgit v1.2.3 From 428ffb7151a4078994b5c01ecbf845954843c1ec Mon Sep 17 00:00:00 2001 From: Li Zefan Date: Wed, 15 Oct 2008 10:07:23 +0800 Subject: ALSA: us122l: fix missing unlock in usb_stream_hwdep_vm_fault() Should unlock us122l->mutex before returning VM_FAULT_SIGBUS. Signed-off-by: Li Zefan Signed-off-by: Takashi Iwai --- sound/usb/usx2y/us122l.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index b441fe2cd19..c2515b680f9 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -118,12 +118,11 @@ static int usb_stream_hwdep_vm_fault(struct vm_area_struct *area, void *vaddr; struct us122l *us122l = area->vm_private_data; struct usb_stream *s; - int vm_f = VM_FAULT_SIGBUS; mutex_lock(&us122l->mutex); s = us122l->sk.s; if (!s) - goto out; + goto unlock; offset = vmf->pgoff << PAGE_SHIFT; if (offset < PAGE_ALIGN(s->read_size)) @@ -131,7 +130,7 @@ static int usb_stream_hwdep_vm_fault(struct vm_area_struct *area, else { offset -= PAGE_ALIGN(s->read_size); if (offset >= PAGE_ALIGN(s->write_size)) - goto out; + goto unlock; vaddr = us122l->sk.write_page + offset; } @@ -141,9 +140,11 @@ static int usb_stream_hwdep_vm_fault(struct vm_area_struct *area, mutex_unlock(&us122l->mutex); vmf->page = page; - vm_f = 0; -out: - return vm_f; + + return 0; +unlock: + mutex_unlock(&us122l->mutex); + return VM_FAULT_SIGBUS; } static void usb_stream_hwdep_vm_close(struct vm_area_struct *area) -- cgit v1.2.3 From 9a3f371e9962749e5bdcf45a3bebdba555651a2b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 15 Oct 2008 17:07:47 +0100 Subject: ALSA: Handle NULL jacks in snd_jack_report() Facilitate drivers that wish to carry on if they can't create a jack input device by handling attempts to report the state of a NULL jack, removing the need to check for initialisation before use. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/core/jack.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/core/jack.c b/sound/core/jack.c index 8133a2b173a..bd2d9e6b55e 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -147,6 +147,9 @@ EXPORT_SYMBOL(snd_jack_set_parent); */ void snd_jack_report(struct snd_jack *jack, int status) { + if (!jack) + return; + if (jack->type & SND_JACK_HEADPHONE) input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT, status & SND_JACK_HEADPHONE); -- cgit v1.2.3 From 1c85cc64456c97f3b265788abafec5c482c6a908 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Wed, 15 Oct 2008 14:38:40 -0700 Subject: ALSA: kernel docs: fix sound/core/ kernel-doc Add kernel-doc function short descriptions to sound/core functions that are missing this short description. Mostly this involves moving some of the function description onto the @funcname line. Also correct a few variable names and fix other kernel-doc notation. Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 48 ++++++++++++++++++++++++------------------------ sound/core/pcm_native.c | 24 ++++++++---------------- 2 files changed, 32 insertions(+), 40 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6ea5cfb8399..921691080f3 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -908,12 +908,12 @@ int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, EXPORT_SYMBOL(snd_pcm_hw_rule_add); /** - * snd_pcm_hw_constraint_mask + * snd_pcm_hw_constraint_mask - apply the given bitmap mask constraint * @runtime: PCM runtime instance * @var: hw_params variable to apply the mask * @mask: the bitmap mask * - * Apply the constraint of the given bitmap mask to a mask parameter. + * Apply the constraint of the given bitmap mask to a 32-bit mask parameter. */ int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, u_int32_t mask) @@ -928,12 +928,12 @@ int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param } /** - * snd_pcm_hw_constraint_mask64 + * snd_pcm_hw_constraint_mask64 - apply the given bitmap mask constraint * @runtime: PCM runtime instance * @var: hw_params variable to apply the mask * @mask: the 64bit bitmap mask * - * Apply the constraint of the given bitmap mask to a mask parameter. + * Apply the constraint of the given bitmap mask to a 64-bit mask parameter. */ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, u_int64_t mask) @@ -949,7 +949,7 @@ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_par } /** - * snd_pcm_hw_constraint_integer + * snd_pcm_hw_constraint_integer - apply an integer constraint to an interval * @runtime: PCM runtime instance * @var: hw_params variable to apply the integer constraint * @@ -964,7 +964,7 @@ int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_pa EXPORT_SYMBOL(snd_pcm_hw_constraint_integer); /** - * snd_pcm_hw_constraint_minmax + * snd_pcm_hw_constraint_minmax - apply a min/max range constraint to an interval * @runtime: PCM runtime instance * @var: hw_params variable to apply the range * @min: the minimal value @@ -995,7 +995,7 @@ static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params, /** - * snd_pcm_hw_constraint_list + * snd_pcm_hw_constraint_list - apply a list of constraints to a parameter * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the list constraint @@ -1031,7 +1031,7 @@ static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params, } /** - * snd_pcm_hw_constraint_ratnums + * snd_pcm_hw_constraint_ratnums - apply ratnums constraint to a parameter * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the ratnums constraint @@ -1064,7 +1064,7 @@ static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params, } /** - * snd_pcm_hw_constraint_ratdens + * snd_pcm_hw_constraint_ratdens - apply ratdens constraint to a parameter * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the ratdens constraint @@ -1095,7 +1095,7 @@ static int snd_pcm_hw_rule_msbits(struct snd_pcm_hw_params *params, } /** - * snd_pcm_hw_constraint_msbits + * snd_pcm_hw_constraint_msbits - add a hw constraint msbits rule * @runtime: PCM runtime instance * @cond: condition bits * @width: sample bits width @@ -1123,7 +1123,7 @@ static int snd_pcm_hw_rule_step(struct snd_pcm_hw_params *params, } /** - * snd_pcm_hw_constraint_step + * snd_pcm_hw_constraint_step - add a hw constraint step rule * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the step constraint @@ -1154,7 +1154,7 @@ static int snd_pcm_hw_rule_pow2(struct snd_pcm_hw_params *params, struct snd_pcm } /** - * snd_pcm_hw_constraint_pow2 + * snd_pcm_hw_constraint_pow2 - add a hw constraint power-of-2 rule * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the power-of-2 constraint @@ -1202,13 +1202,13 @@ void _snd_pcm_hw_params_any(struct snd_pcm_hw_params *params) EXPORT_SYMBOL(_snd_pcm_hw_params_any); /** - * snd_pcm_hw_param_value + * snd_pcm_hw_param_value - return @params field @var value * @params: the hw_params instance * @var: parameter to retrieve - * @dir: pointer to the direction (-1,0,1) or NULL + * @dir: pointer to the direction (-1,0,1) or %NULL * - * Return the value for field PAR if it's fixed in configuration space - * defined by PARAMS. Return -EINVAL otherwise + * Return the value for field @var if it's fixed in configuration space + * defined by @params. Return -%EINVAL otherwise. */ int snd_pcm_hw_param_value(const struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var, int *dir) @@ -1271,13 +1271,13 @@ static int _snd_pcm_hw_param_first(struct snd_pcm_hw_params *params, /** - * snd_pcm_hw_param_first + * snd_pcm_hw_param_first - refine config space and return minimum value * @pcm: PCM instance * @params: the hw_params instance * @var: parameter to retrieve - * @dir: pointer to the direction (-1,0,1) or NULL + * @dir: pointer to the direction (-1,0,1) or %NULL * - * Inside configuration space defined by PARAMS remove from PAR all + * Inside configuration space defined by @params remove from @var all * values > minimum. Reduce configuration space accordingly. * Return the minimum. */ @@ -1317,13 +1317,13 @@ static int _snd_pcm_hw_param_last(struct snd_pcm_hw_params *params, /** - * snd_pcm_hw_param_last + * snd_pcm_hw_param_last - refine config space and return maximum value * @pcm: PCM instance * @params: the hw_params instance * @var: parameter to retrieve - * @dir: pointer to the direction (-1,0,1) or NULL + * @dir: pointer to the direction (-1,0,1) or %NULL * - * Inside configuration space defined by PARAMS remove from PAR all + * Inside configuration space defined by @params remove from @var all * values < maximum. Reduce configuration space accordingly. * Return the maximum. */ @@ -1345,11 +1345,11 @@ int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm, EXPORT_SYMBOL(snd_pcm_hw_param_last); /** - * snd_pcm_hw_param_choose + * snd_pcm_hw_param_choose - choose a configuration defined by @params * @pcm: PCM instance * @params: the hw_params instance * - * Choose one configuration from configuration space defined by PARAMS + * Choose one configuration from configuration space defined by @params. * The configuration chosen is that obtained fixing in this order: * first access, first format, first subformat, min channels, * min rate, min period time, max buffer size, min tick time diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index e61e12506de..aef18682c03 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -875,10 +875,8 @@ static struct action_ops snd_pcm_action_start = { }; /** - * snd_pcm_start + * snd_pcm_start - start all linked streams * @substream: the PCM substream instance - * - * Start all linked streams. */ int snd_pcm_start(struct snd_pcm_substream *substream) { @@ -926,12 +924,11 @@ static struct action_ops snd_pcm_action_stop = { }; /** - * snd_pcm_stop + * snd_pcm_stop - try to stop all running streams in the substream group * @substream: the PCM substream instance * @state: PCM state after stopping the stream * - * Try to stop all running streams in the substream group. - * The state of each stream is changed to the given value after that unconditionally. + * The state of each stream is then changed to the given state unconditionally. */ int snd_pcm_stop(struct snd_pcm_substream *substream, int state) { @@ -941,11 +938,10 @@ int snd_pcm_stop(struct snd_pcm_substream *substream, int state) EXPORT_SYMBOL(snd_pcm_stop); /** - * snd_pcm_drain_done + * snd_pcm_drain_done - stop the DMA only when the given stream is playback * @substream: the PCM substream * - * Stop the DMA only when the given stream is playback. - * The state is changed to SETUP. + * After stopping, the state is changed to SETUP. * Unlike snd_pcm_stop(), this affects only the given stream. */ int snd_pcm_drain_done(struct snd_pcm_substream *substream) @@ -1065,10 +1061,9 @@ static struct action_ops snd_pcm_action_suspend = { }; /** - * snd_pcm_suspend + * snd_pcm_suspend - trigger SUSPEND to all linked streams * @substream: the PCM substream * - * Trigger SUSPEND to all linked streams. * After this call, all streams are changed to SUSPENDED state. */ int snd_pcm_suspend(struct snd_pcm_substream *substream) @@ -1088,10 +1083,9 @@ int snd_pcm_suspend(struct snd_pcm_substream *substream) EXPORT_SYMBOL(snd_pcm_suspend); /** - * snd_pcm_suspend_all + * snd_pcm_suspend_all - trigger SUSPEND to all substreams in the given pcm * @pcm: the PCM instance * - * Trigger SUSPEND to all substreams in the given pcm. * After this call, all streams are changed to SUSPENDED state. */ int snd_pcm_suspend_all(struct snd_pcm *pcm) @@ -1313,11 +1307,9 @@ static struct action_ops snd_pcm_action_prepare = { }; /** - * snd_pcm_prepare + * snd_pcm_prepare - prepare the PCM substream to be triggerable * @substream: the PCM substream instance * @file: file to refer f_flags - * - * Prepare the PCM substream to be triggerable. */ static int snd_pcm_prepare(struct snd_pcm_substream *substream, struct file *file) -- cgit v1.2.3 From cdbdd1676a5379f1d5cbd4d476f5e349f445befe Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Oct 2008 11:00:06 +0100 Subject: ALSA: ASoC: Convert playpaq_wm8510 to bulk route registration API Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/at32/playpaq_wm8510.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index 98a2d5826a8..b1966e4dfcd 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -304,7 +304,7 @@ static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { -static const char *intercon[][3] = { +static const struct snd_soc_dapm_route intercon[] = { /* speaker connected to SPKOUT */ {"Ext Spk", NULL, "SPKOUTP"}, {"Ext Spk", NULL, "SPKOUTN"}, @@ -312,9 +312,6 @@ static const char *intercon[][3] = { {"Mic Bias", NULL, "Int Mic"}, {"MICN", NULL, "Mic Bias"}, {"MICP", NULL, "Mic Bias"}, - - /* Terminator */ - {NULL, NULL, NULL}, }; @@ -334,11 +331,8 @@ static int playpaq_wm8510_init(struct snd_soc_codec *codec) /* * Setup audio path interconnects */ - for (i = 0; intercon[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, - intercon[i][0], - intercon[i][1], intercon[i][2]); - } + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + /* always connected pins */ -- cgit v1.2.3 From abe9ab8f62203ced11119fb96acc3b8dd107ebc4 Mon Sep 17 00:00:00 2001 From: Greg Kroah-Hartman Date: Mon, 21 Jul 2008 20:03:34 -0700 Subject: device create: sound: convert device_create_drvdata to device_create Now that device_create() has been audited, rename things back to the original call to be sane. Cc: Jaroslav Kysela Signed-off-by: Greg Kroah-Hartman --- sound/core/init.c | 6 +++--- sound/core/sound.c | 5 ++--- sound/oss/soundcard.c | 15 +++++++-------- sound/sound_core.c | 5 ++--- 4 files changed, 14 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/core/init.c b/sound/core/init.c index 8af467df924..ef2352c2e45 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -549,9 +549,9 @@ int snd_card_register(struct snd_card *card) return -EINVAL; #ifndef CONFIG_SYSFS_DEPRECATED if (!card->card_dev) { - card->card_dev = device_create_drvdata(sound_class, card->dev, - MKDEV(0, 0), NULL, - "card%i", card->number); + card->card_dev = device_create(sound_class, card->dev, + MKDEV(0, 0), NULL, + "card%i", card->number); if (IS_ERR(card->card_dev)) card->card_dev = NULL; } diff --git a/sound/core/sound.c b/sound/core/sound.c index c0685e2f0af..44a69bb8d4f 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -274,9 +274,8 @@ int snd_register_device_for_dev(int type, struct snd_card *card, int dev, return minor; } snd_minors[minor] = preg; - preg->dev = device_create_drvdata(sound_class, device, - MKDEV(major, minor), - private_data, "%s", name); + preg->dev = device_create(sound_class, device, MKDEV(major, minor), + private_data, "%s", name); if (IS_ERR(preg->dev)) { snd_minors[minor] = NULL; mutex_unlock(&sound_mutex); diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index 7d89c081a08..61aaedae6b7 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -560,19 +560,18 @@ static int __init oss_init(void) sound_dmap_flag = (dmabuf > 0 ? 1 : 0); for (i = 0; i < ARRAY_SIZE(dev_list); i++) { - device_create_drvdata(sound_class, NULL, - MKDEV(SOUND_MAJOR, dev_list[i].minor), - NULL, "%s", dev_list[i].name); + device_create(sound_class, NULL, + MKDEV(SOUND_MAJOR, dev_list[i].minor), NULL, + "%s", dev_list[i].name); if (!dev_list[i].num) continue; for (j = 1; j < *dev_list[i].num; j++) - device_create_drvdata(sound_class, NULL, - MKDEV(SOUND_MAJOR, - dev_list[i].minor + (j*0x10)), - NULL, - "%s%d", dev_list[i].name, j); + device_create(sound_class, NULL, + MKDEV(SOUND_MAJOR, + dev_list[i].minor + (j*0x10)), + NULL, "%s%d", dev_list[i].name, j); } if (sound_nblocks >= 1024) diff --git a/sound/sound_core.c b/sound/sound_core.c index 4ae07e236b3..faef87a9bc3 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -220,9 +220,8 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati else sprintf(s->name, "sound/%s%d", name, r / SOUND_STEP); - device_create_drvdata(sound_class, dev, - MKDEV(SOUND_MAJOR, s->unit_minor), - NULL, s->name+6); + device_create(sound_class, dev, MKDEV(SOUND_MAJOR, s->unit_minor), + NULL, s->name+6); return r; fail: -- cgit v1.2.3 From 7c2dfee84863628f9af109131edd4344ce567d16 Mon Sep 17 00:00:00 2001 From: Zhaolei Date: Fri, 17 Oct 2008 21:04:55 +0800 Subject: ALSA: Fix debugfs_create_dir's error checking method for sound/soc/ debugfs_create_dir() returns NULL if an error occurs, returns -ENODEV when debugfs is not enabled in the kernel. Signed-off-by: Zhao Lei Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index efbd0b37810..7351db9606e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -831,7 +831,7 @@ int snd_soc_dapm_sys_add(struct device *dev) return ret; asoc_debugfs = debugfs_create_dir("asoc", NULL); - if (!IS_ERR(asoc_debugfs)) + if (!IS_ERR(asoc_debugfs) && asoc_debugfs) debugfs_create_u32("dapm_pop_time", 0744, asoc_debugfs, &pop_time); else -- cgit v1.2.3 From a66547f3a1295d782fc0923d872b309994cea32a Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Fri, 17 Oct 2008 11:28:11 -0700 Subject: sound: add missing pcm kernel-doc Fix alsa kernel-doc warning in linux-next: Warning(linux-next-20081016//sound/core/pcm_misc.c:327): No description found for parameter 'samples' Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai --- sound/core/pcm_misc.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 89b7f549beb..ea2bf82c937 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -319,6 +319,7 @@ EXPORT_SYMBOL(snd_pcm_format_physical_width); /** * snd_pcm_format_size - return the byte size of samples on the given format * @format: the format to check + * @samples: sampling rate * * Returns the byte size of the given samples for the format, or a * negative error code if unknown format. -- cgit v1.2.3 From e78521f3212d5d3931442819cbf0910fe1b28beb Mon Sep 17 00:00:00 2001 From: Mariusz Kozlowski Date: Sun, 19 Oct 2008 10:34:22 +0200 Subject: ALSA: misc typo fixes Fixed typos in disabled codes via #if 0. Signed-off-by: Mariusz Kozlowski Signed-off-by: Takashi Iwai --- sound/drivers/dummy.c | 2 +- sound/pci/ca0106/ca0106_main.c | 1 - 2 files changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index e5e749f3e0e..73be7e14a60 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -51,7 +51,7 @@ static int emu10k1_playback_constraints(struct snd_pcm_runtime *runtime) if (err < 0) return err; err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 256, UINT_MAX); - if (err) < 0) + if (err < 0) return err; return 0; } diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index a7d89662acf..88fbf285d2b 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -759,7 +759,6 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | SPCS_GENERATIONSTATUS | 0x00001200 | 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT ); - } #endif return 0; -- cgit v1.2.3 From 7639a4541f7e7abb1295ff8ab39cc2f5842239ae Mon Sep 17 00:00:00 2001 From: Paul Mundt Date: Mon, 20 Oct 2008 13:02:48 +0900 Subject: sh: Migrate common board headers to mach-common/. Signed-off-by: Paul Mundt --- sound/oss/sh_dac_audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/oss/sh_dac_audio.c b/sound/oss/sh_dac_audio.c index b493660deb3..e5d42399491 100644 --- a/sound/oss/sh_dac_audio.c +++ b/sound/oss/sh_dac_audio.c @@ -26,7 +26,7 @@ #include #include #include -#include +#include #include #define MODNAME "sh_dac_audio" -- cgit v1.2.3 From 1ee2a322b058f6399dc900603f9ebb392037ff77 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2008 08:06:39 +0200 Subject: ALSA: ps3: Add support for SPDIF/HDMI passthru Add support for SPDIF/HDMI pass-through support of PS3 audio driver. Signed-off-by: Masakazu Mokuno Signed-off-by: Takashi Iwai --- sound/ppc/snd_ps3.c | 96 +++++++++++++++++++++++++++++++++++++++++++++++++---- sound/ppc/snd_ps3.h | 1 + 2 files changed, 91 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 20d0e328288..8f9e3859c37 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -666,6 +666,7 @@ static int snd_ps3_init_avsetting(struct snd_ps3_card_info *card) card->avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_16; card->avs.avs_audio_format = PS3AV_CMD_AUDIO_FORMAT_PCM; card->avs.avs_audio_source = PS3AV_CMD_AUDIO_SOURCE_SERIAL; + memcpy(card->avs.avs_cs_info, ps3av_mode_cs_info, 8); ret = snd_ps3_change_avsetting(card); @@ -685,6 +686,7 @@ static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream) { struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream); struct snd_ps3_avsetting_info avs; + int ret; avs = card->avs; @@ -729,19 +731,92 @@ static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream) return 1; } - if ((card->avs.avs_audio_width != avs.avs_audio_width) || - (card->avs.avs_audio_rate != avs.avs_audio_rate)) { - card->avs = avs; - snd_ps3_change_avsetting(card); + memcpy(avs.avs_cs_info, ps3av_mode_cs_info, 8); + if (memcmp(&card->avs, &avs, sizeof(avs))) { pr_debug("%s: after freq=%d width=%d\n", __func__, card->avs.avs_audio_rate, card->avs.avs_audio_width); - return 0; + card->avs = avs; + snd_ps3_change_avsetting(card); + ret = 0; } else + ret = 1; + + /* check CS non-audio bit and mute accordingly */ + if (avs.avs_cs_info[0] & 0x02) + ps3av_audio_mute_analog(1); /* mute if non-audio */ + else + ps3av_audio_mute_analog(0); + + return ret; +} + +/* + * SPDIF status bits controls + */ +static int snd_ps3_spdif_mask_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + return 0; +} + +/* FIXME: ps3av_set_audio_mode() assumes only consumer mode */ +static int snd_ps3_spdif_cmask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + memset(ucontrol->value.iec958.status, 0xff, 8); + return 0; +} + +static int snd_ps3_spdif_pmask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return 0; +} + +static int snd_ps3_spdif_default_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + memcpy(ucontrol->value.iec958.status, ps3av_mode_cs_info, 8); + return 0; +} + +static int snd_ps3_spdif_default_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (memcmp(ps3av_mode_cs_info, ucontrol->value.iec958.status, 8)) { + memcpy(ps3av_mode_cs_info, ucontrol->value.iec958.status, 8); return 1; + } + return 0; } +static struct snd_kcontrol_new spdif_ctls[] = { + { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), + .info = snd_ps3_spdif_mask_info, + .get = snd_ps3_spdif_cmask_get, + }, + { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK), + .info = snd_ps3_spdif_mask_info, + .get = snd_ps3_spdif_pmask_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), + .info = snd_ps3_spdif_mask_info, + .get = snd_ps3_spdif_default_get, + .put = snd_ps3_spdif_default_put, + }, +}; static int snd_ps3_map_mmio(void) @@ -842,7 +917,7 @@ static void snd_ps3_audio_set_base_addr(uint64_t ioaddr_start) static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev) { - int ret; + int i, ret; u64 lpar_addr, lpar_size; BUG_ON(!firmware_has_feature(FW_FEATURE_PS3_LV1)); @@ -903,6 +978,15 @@ static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev) strcpy(the_card.card->driver, "PS3"); strcpy(the_card.card->shortname, "PS3"); strcpy(the_card.card->longname, "PS3 sound"); + + /* create control elements */ + for (i = 0; i < ARRAY_SIZE(spdif_ctls); i++) { + ret = snd_ctl_add(the_card.card, + snd_ctl_new1(&spdif_ctls[i], &the_card)); + if (ret < 0) + goto clean_card; + } + /* create PCM devices instance */ /* NOTE:this driver works assuming pcm:substream = 1:1 */ ret = snd_pcm_new(the_card.card, diff --git a/sound/ppc/snd_ps3.h b/sound/ppc/snd_ps3.h index 4b7e6fbbe50..326fb29e82d 100644 --- a/sound/ppc/snd_ps3.h +++ b/sound/ppc/snd_ps3.h @@ -51,6 +51,7 @@ struct snd_ps3_avsetting_info { uint32_t avs_audio_width; uint32_t avs_audio_format; /* fixed */ uint32_t avs_audio_source; /* fixed */ + unsigned char avs_cs_info[8]; }; /* * PS3 audio 'card' instance -- cgit v1.2.3 From ba9d0fd0f0af3eb7ec463847b409dd88f8e5c031 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 20 Oct 2008 15:29:59 +0300 Subject: ALSA: ASoC: OMAP: Fix DSP DAI format in McBSP DAI driver Fix word clock length which must equal to one bit clock cycle in DSP mode. Surprisingly McBSP is able synchronize into wrong length when it's slave but e.g. TLV320AIC33 codec in slave configuration is outputting some amount of noise if word clock length is longer than one bit clock cycle. Fix also bit clock and frame sync polarities in DSP mode since they are opposite from I2S. Signed-off-by: Jarkko Nikula Cc: Arun KS Signed-off-by: Takashi Iwai --- sound/soc/omap/omap-mcbsp.c | 24 ++++++++++++++++++++---- 1 file changed, 20 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 0a063a98a66..853b33ae343 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -43,6 +43,7 @@ struct omap_mcbsp_data { unsigned int bus_id; struct omap_mcbsp_reg_cfg regs; + unsigned int fmt; /* * Flags indicating is the bus already activated and configured by * another substream @@ -200,6 +201,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; + int wlen; unsigned long port; if (cpu_class_is_omap1()) { @@ -244,19 +246,29 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: /* Set word lengths */ + wlen = 16; regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_16); regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_16); regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_16); regs->xcr1 |= XWDLEN1(OMAP_MCBSP_WORD_16); - /* Set FS period and length in terms of bit clock periods */ - regs->srgr2 |= FPER(16 * 2 - 1); - regs->srgr1 |= FWID(16 - 1); break; default: /* Unsupported PCM format */ return -EINVAL; } + /* Set FS period and length in terms of bit clock periods */ + switch (mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + regs->srgr2 |= FPER(wlen * 2 - 1); + regs->srgr1 |= FWID(wlen - 1); + break; + case SND_SOC_DAIFMT_DSP_A: + regs->srgr2 |= FPER(wlen * 2 - 1); + regs->srgr1 |= FWID(0); + break; + } + omap_mcbsp_config(bus_id, &mcbsp_data->regs); mcbsp_data->configured = 1; @@ -272,10 +284,12 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; + unsigned int temp_fmt = fmt; if (mcbsp_data->configured) return 0; + mcbsp_data->fmt = fmt; memset(regs, 0, sizeof(*regs)); /* Generic McBSP register settings */ regs->spcr2 |= XINTM(3) | FREE; @@ -293,6 +307,8 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); regs->xcr2 |= XDATDLY(0); + /* Invert bit clock and FS polarity configuration for DSP_A */ + temp_fmt ^= SND_SOC_DAIFMT_IB_IF; break; default: /* Unsupported data format */ @@ -316,7 +332,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* Set bit clock (CLKX/CLKR) and FS polarities */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: /* * Normal BCLK + FS. -- cgit v1.2.3 From 2f5ad54ea6e2e38156bfb889964deee991f3087a Mon Sep 17 00:00:00 2001 From: Arjan van de Ven Date: Sun, 28 Sep 2008 16:20:09 -0700 Subject: pci: use pci_ioremap_bar() in sound/ Use the newly introduced pci_ioremap_bar() function in sound/. pci_ioremap_bar() just takes a pci device and a bar number, with the goal of making it really hard to get wrong, while also having a central place to stick sanity checks. Signed-off-by: Arjan van de Ven Signed-off-by: Takashi Iwai --- sound/pci/ad1889.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/au88x0/au88x0.c | 3 +-- sound/pci/bt87x.c | 3 +-- sound/pci/cs4281.c | 4 ++-- sound/pci/cs5530.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- sound/pci/mixart/mixart.c | 3 +-- 9 files changed, 10 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 92f3a976ef2..a7f38e63303 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -932,7 +932,7 @@ snd_ad1889_create(struct snd_card *card, goto free_and_ret; chip->bar = pci_resource_start(pci, 0); - chip->iobase = ioremap_nocache(chip->bar, pci_resource_len(pci, 0)); + chip->iobase = pci_ioremap_bar(pci, 0); if (chip->iobase == NULL) { printk(KERN_ERR PFX "unable to reserve region.\n"); err = -EBUSY; diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 085a52b8c80..226fe8237d3 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1609,7 +1609,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card, return err; } chip->addr = pci_resource_start(pci, 0); - chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci, 0)); + chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR "AC'97 space ioremap problem\n"); snd_atiixp_free(chip); diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 2f106306c7f..0e6e5cc1c50 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -1252,7 +1252,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card, return err; } chip->addr = pci_resource_start(pci, 0); - chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci, 0)); + chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR "AC'97 space ioremap problem\n"); snd_atiixp_free(chip); diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index 68368e49007..a36d4d1fd41 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -180,8 +180,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) if ((err = pci_request_regions(pci, CARD_NAME_SHORT)) != 0) goto regions_out; - chip->mmio = ioremap_nocache(pci_resource_start(pci, 0), - pci_resource_len(pci, 0)); + chip->mmio = pci_ioremap_bar(pci, 0); if (!chip->mmio) { printk(KERN_ERR "MMIO area remap failed.\n"); err = -ENOMEM; diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 3aa8d973540..1aa1c040254 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -749,8 +749,7 @@ static int __devinit snd_bt87x_create(struct snd_card *card, pci_disable_device(pci); return err; } - chip->mmio = ioremap_nocache(pci_resource_start(pci, 0), - pci_resource_len(pci, 0)); + chip->mmio = pci_ioremap_bar(pci, 0); if (!chip->mmio) { snd_printk(KERN_ERR "cannot remap io memory\n"); err = -ENOMEM; diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index ef9308f7c45..192e7842e18 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -1382,8 +1382,8 @@ static int __devinit snd_cs4281_create(struct snd_card *card, chip->ba0_addr = pci_resource_start(pci, 0); chip->ba1_addr = pci_resource_start(pci, 1); - chip->ba0 = ioremap_nocache(chip->ba0_addr, pci_resource_len(pci, 0)); - chip->ba1 = ioremap_nocache(chip->ba1_addr, pci_resource_len(pci, 1)); + chip->ba0 = pci_ioremap_bar(pci, 0); + chip->ba1 = pci_ioremap_bar(pci, 1); if (!chip->ba0 || !chip->ba1) { snd_cs4281_free(chip); return -ENOMEM; diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index 7ff8b68e997..4d9378d8120 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -132,7 +132,7 @@ static int __devinit snd_cs5530_create(struct snd_card *card, } chip->pci_base = pci_resource_start(pci, 0); - mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0)); + mem = pci_ioremap_bar(pci, 0); if (mem == NULL) { kfree(chip); pci_disable_device(pci); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9f316c1b279..f080f8ce0ec 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2158,7 +2158,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } chip->addr = pci_resource_start(pci, 0); - chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci,0)); + chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR SFX "ioremap error\n"); err = -ENXIO; diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 2d0dce649a6..ae7601f353a 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1314,8 +1314,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, } for (i = 0; i < 2; i++) { mgr->mem[i].phys = pci_resource_start(pci, i); - mgr->mem[i].virt = ioremap_nocache(mgr->mem[i].phys, - pci_resource_len(pci, i)); + mgr->mem[i].virt = pci_ioremap_bar(pci, i); if (!mgr->mem[i].virt) { printk(KERN_ERR "unable to remap resource 0x%lx\n", mgr->mem[i].phys); -- cgit v1.2.3 From aeb5d727062a0238a2f96c9c380fbd2be4640c6f Mon Sep 17 00:00:00 2001 From: Al Viro Date: Tue, 2 Sep 2008 15:28:45 -0400 Subject: [PATCH] introduce fmode_t, do annotations Signed-off-by: Al Viro --- sound/core/oss/pcm_oss.c | 2 +- sound/oss/au1550_ac97.c | 2 +- sound/oss/dmasound/dmasound.h | 4 ++-- sound/oss/dmasound/dmasound_atari.c | 4 ++-- sound/oss/dmasound/dmasound_core.c | 10 +++++----- sound/oss/msnd.h | 2 +- sound/oss/sound_config.h | 20 ++++++-------------- sound/oss/swarm_cs4297a.c | 2 +- sound/oss/vwsnd.c | 2 +- 9 files changed, 20 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 1af62b8b86c..e17836680f4 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2283,7 +2283,7 @@ static int snd_pcm_oss_open_file(struct file *file, int idx, err; struct snd_pcm_oss_file *pcm_oss_file; struct snd_pcm_substream *substream; - unsigned int f_mode = file->f_mode; + fmode_t f_mode = file->f_mode; if (rpcm_oss_file) *rpcm_oss_file = NULL; diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c index 23018a7c063..81e1f443d09 100644 --- a/sound/oss/au1550_ac97.c +++ b/sound/oss/au1550_ac97.c @@ -93,7 +93,7 @@ static struct au1550_state { spinlock_t lock; struct mutex open_mutex; struct mutex sem; - mode_t open_mode; + fmode_t open_mode; wait_queue_head_t open_wait; struct dmabuf { diff --git a/sound/oss/dmasound/dmasound.h b/sound/oss/dmasound/dmasound.h index d978b009656..1cb13fe56ec 100644 --- a/sound/oss/dmasound/dmasound.h +++ b/sound/oss/dmasound/dmasound.h @@ -129,7 +129,7 @@ typedef struct { int (*mixer_ioctl)(u_int, u_long); /* optional */ int (*write_sq_setup)(void); /* optional */ int (*read_sq_setup)(void); /* optional */ - int (*sq_open)(mode_t); /* optional */ + int (*sq_open)(fmode_t); /* optional */ int (*state_info)(char *, size_t); /* optional */ void (*abort_read)(void); /* optional */ int min_dsp_speed; @@ -235,7 +235,7 @@ struct sound_queue { */ int active; wait_queue_head_t action_queue, open_queue, sync_queue; - int open_mode; + fmode_t open_mode; int busy, syncing, xruns, died; }; diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c index 285239d64b8..4d45bd63718 100644 --- a/sound/oss/dmasound/dmasound_atari.c +++ b/sound/oss/dmasound/dmasound_atari.c @@ -143,7 +143,7 @@ static int AtaMixerIoctl(u_int cmd, u_long arg); static int TTMixerIoctl(u_int cmd, u_long arg); static int FalconMixerIoctl(u_int cmd, u_long arg); static int AtaWriteSqSetup(void); -static int AtaSqOpen(mode_t mode); +static int AtaSqOpen(fmode_t mode); static int TTStateInfo(char *buffer, size_t space); static int FalconStateInfo(char *buffer, size_t space); @@ -1461,7 +1461,7 @@ static int AtaWriteSqSetup(void) return 0 ; } -static int AtaSqOpen(mode_t mode) +static int AtaSqOpen(fmode_t mode) { write_sq_ignore_int = 1; return 0 ; diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index 95fc5c68175..b8239f3168f 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -212,7 +212,7 @@ static int irq_installed; #endif /* MODULE */ /* control over who can modify resources shared between play/record */ -static mode_t shared_resource_owner; +static fmode_t shared_resource_owner; static int shared_resources_initialised; /* @@ -668,7 +668,7 @@ static inline void sq_init_waitqueue(struct sound_queue *sq) #if 0 /* blocking open() */ static inline void sq_wake_up(struct sound_queue *sq, struct file *file, - mode_t mode) + fmode_t mode) { if (file->f_mode & mode) { sq->busy = 0; /* CHECK: IS THIS OK??? */ @@ -677,7 +677,7 @@ static inline void sq_wake_up(struct sound_queue *sq, struct file *file, } #endif -static int sq_open2(struct sound_queue *sq, struct file *file, mode_t mode, +static int sq_open2(struct sound_queue *sq, struct file *file, fmode_t mode, int numbufs, int bufsize) { int rc = 0; @@ -891,10 +891,10 @@ static int sq_release(struct inode *inode, struct file *file) is the owner - if we have problems. */ -static int shared_resources_are_mine(mode_t md) +static int shared_resources_are_mine(fmode_t md) { if (shared_resource_owner) - return (shared_resource_owner & md ) ; + return (shared_resource_owner & md) != 0; else { shared_resource_owner = md ; return 1 ; diff --git a/sound/oss/msnd.h b/sound/oss/msnd.h index 61b3955481c..c8be47ec2b7 100644 --- a/sound/oss/msnd.h +++ b/sound/oss/msnd.h @@ -211,7 +211,7 @@ typedef struct multisound_dev { /* State variables */ enum { msndClassic, msndPinnacle } type; - mode_t mode; + fmode_t mode; unsigned long flags; #define F_RESETTING 0 #define F_HAVEDIGITAL 1 diff --git a/sound/oss/sound_config.h b/sound/oss/sound_config.h index 1a00a321061..55271fbe7f4 100644 --- a/sound/oss/sound_config.h +++ b/sound/oss/sound_config.h @@ -110,24 +110,16 @@ struct channel_info { #define OPEN_WRITE PCM_ENABLE_OUTPUT #define OPEN_READWRITE (OPEN_READ|OPEN_WRITE) -#if OPEN_READ == FMODE_READ && OPEN_WRITE == FMODE_WRITE - -static inline int translate_mode(struct file *file) -{ - return file->f_mode; -} - -#else - static inline int translate_mode(struct file *file) { - return ((file->f_mode & FMODE_READ) ? OPEN_READ : 0) | - ((file->f_mode & FMODE_WRITE) ? OPEN_WRITE : 0); + if (OPEN_READ == (__force int)FMODE_READ && + OPEN_WRITE == (__force int)FMODE_WRITE) + return (__force int)(file->f_mode & (FMODE_READ | FMODE_WRITE)); + else + return ((file->f_mode & FMODE_READ) ? OPEN_READ : 0) | + ((file->f_mode & FMODE_WRITE) ? OPEN_WRITE : 0); } -#endif - - #include "sound_calls.h" #include "dev_table.h" diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c index 044453a4ee5..41562ecde5b 100644 --- a/sound/oss/swarm_cs4297a.c +++ b/sound/oss/swarm_cs4297a.c @@ -295,7 +295,7 @@ struct cs4297a_state { struct mutex open_mutex; struct mutex open_sem_adc; struct mutex open_sem_dac; - mode_t open_mode; + fmode_t open_mode; wait_queue_head_t open_wait; wait_queue_head_t open_wait_adc; wait_queue_head_t open_wait_dac; diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index dcbb3f739e6..78b8acc7c3b 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -1509,7 +1509,7 @@ typedef struct vwsnd_dev { struct mutex open_mutex; struct mutex io_mutex; struct mutex mix_mutex; - mode_t open_mode; + fmode_t open_mode; wait_queue_head_t open_wait; lithium_t lith; -- cgit v1.2.3 From 13095c37b29b90ec96070cb63400a79737b4a185 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Oct 2008 13:27:49 +0100 Subject: ALSA: Ensure PXA runtime data is initialised The rest of the code relies on the runtime data being zero initialised so we need to use kzalloc() to allocate it. Reported-by: Oliver Ford Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/arm/pxa2xx-pcm-lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 1c93eb77cb9..75a0d746fb6 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -194,7 +194,7 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) goto out; ret = -ENOMEM; - rtd = kmalloc(sizeof(*rtd), GFP_KERNEL); + rtd = kzalloc(sizeof(*rtd), GFP_KERNEL); if (!rtd) goto out; rtd->dma_desc_array = -- cgit v1.2.3 From da6320becf31c40b60d4b1dc6b339c9a766b671c Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 22 Oct 2008 15:00:29 +0300 Subject: ALSA: ASoC: OMAP: Continue fixing DSP DAI format in McBSP DAI driver Fix "ASoC: OMAP: Fix DSP DAI format in McBSP DAI driver" was not correct due misunderstanding of DSP_A format and similar error in TLV320AIC33 codec which was used to test the original fix. This patch corrects now DSP_A format in OMAP McBSP DAI driver and is verified with TLV320AIC23 codec that's implementing DSP_A correctly. Signed-off-by: Jarkko Nikula Cc: Arun KS Signed-off-by: Takashi Iwai --- sound/soc/omap/omap-mcbsp.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 853b33ae343..8485a8a9d0f 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -265,7 +265,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, break; case SND_SOC_DAIFMT_DSP_A: regs->srgr2 |= FPER(wlen * 2 - 1); - regs->srgr1 |= FWID(0); + regs->srgr1 |= FWID(wlen * 2 - 2); break; } @@ -284,7 +284,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; - unsigned int temp_fmt = fmt; if (mcbsp_data->configured) return 0; @@ -307,8 +306,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); regs->xcr2 |= XDATDLY(0); - /* Invert bit clock and FS polarity configuration for DSP_A */ - temp_fmt ^= SND_SOC_DAIFMT_IB_IF; break; default: /* Unsupported data format */ @@ -332,7 +329,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* Set bit clock (CLKX/CLKR) and FS polarities */ - switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) { + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: /* * Normal BCLK + FS. -- cgit v1.2.3 From 4b7d283150b35db6e5e10f72606f603ff424c92a Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 23 Oct 2008 14:27:03 +0300 Subject: ALSA: ASoC: tlv320aic3x: Fix DSP DAI format and signal polarities matching - Codec doesn't support to configure bit clock and frame sync polarities - Codec doesn't support DSP_A format but DSP_B with inverted bit clock polarity - Match also other formats with their signal polarities Signed-off-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/tlv320aic3x.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 05336ed7e49..cff276ee261 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -863,17 +863,21 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - /* interface format */ - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: + /* + * match both interface format and signal polarities since they + * are fixed + */ + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_INV_MASK)) { + case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): break; - case SND_SOC_DAIFMT_DSP_A: + case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF): iface_breg |= (0x01 << 6); break; - case SND_SOC_DAIFMT_RIGHT_J: + case (SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF): iface_breg |= (0x02 << 6); break; - case SND_SOC_DAIFMT_LEFT_J: + case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF): iface_breg |= (0x03 << 6); break; default: -- cgit v1.2.3 From 9f50bbad8fc3b0f9fd453ab1e2716b3106d89b13 Mon Sep 17 00:00:00 2001 From: Johannes Berg Date: Thu, 23 Oct 2008 13:57:39 +0200 Subject: ALSA: aoa i2sbus: don't overwrite module parameter We shouldn't modify a global variable here. Signed-off-by: Johannes Berg Signed-off-by: Takashi Iwai --- sound/aoa/soundbus/i2sbus/i2sbus-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-core.c b/sound/aoa/soundbus/i2sbus/i2sbus-core.c index e6beb92c693..b4590df0746 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus-core.c +++ b/sound/aoa/soundbus/i2sbus/i2sbus-core.c @@ -159,7 +159,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, struct i2sbus_dev *dev; struct device_node *child = NULL, *sound = NULL; struct resource *r; - int i, layout = 0, rlen; + int i, layout = 0, rlen, ok = force; static const char *rnames[] = { "i2sbus: %s (control)", "i2sbus: %s (tx)", "i2sbus: %s (rx)" }; @@ -192,7 +192,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, layout = *layout_id; snprintf(dev->sound.modalias, 32, "sound-layout-%d", layout); - force = 1; + ok = 1; } } /* for the time being, until we can handle non-layout-id @@ -201,7 +201,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, * When there are two i2s busses and only one has a layout-id, * then this depends on the order, but that isn't important * either as the second one in that case is just a modem. */ - if (!force) { + if (!ok) { kfree(dev); return -ENODEV; } -- cgit v1.2.3 From 2f1e593d4209d0194f9639c5d11aa91171435963 Mon Sep 17 00:00:00 2001 From: Alan Cox Date: Mon, 27 Oct 2008 15:21:19 +0000 Subject: sound: use a common working email address Signed-off-by: Alan Cox Signed-off-by: Takashi Iwai --- sound/oss/kahlua.c | 2 +- sound/pci/cs5530.c | 2 +- sound/sound_core.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c index eb9bc365530..c180598f171 100644 --- a/sound/oss/kahlua.c +++ b/sound/oss/kahlua.c @@ -1,7 +1,7 @@ /* * Initialisation code for Cyrix/NatSemi VSA1 softaudio * - * (C) Copyright 2003 Red Hat Inc + * (C) Copyright 2003 Red Hat Inc * * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems. * The older version (VSA1) provides fairly good soundblaster emulation diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index 7ff8b68e997..4ba2db2f425 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -2,7 +2,7 @@ * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio * * (C) Copyright 2007 Ash Willis - * (C) Copyright 2003 Red Hat Inc + * (C) Copyright 2003 Red Hat Inc * * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did * mess with it a bit. The chip seems to have to have trouble with full duplex diff --git a/sound/sound_core.c b/sound/sound_core.c index 4ae07e236b3..ee66d43f848 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -57,7 +57,7 @@ module_exit(cleanup_soundcore); /* * OSS sound core handling. Breaks out sound functions to submodules * - * Author: Alan Cox + * Author: Alan Cox * * Fixes: * -- cgit v1.2.3 From c3e5203bed1999df716e3c7119f6749523eb952f Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Mon, 27 Oct 2008 17:09:25 +0800 Subject: ALSA: ASoC: Blackfin: update SPORT0 port selector (v2) - Setting the TFS pin selector for SPORT 0 based on whether the selected port id F or G. If the port is F then no conflict should exist for the TFS. When Port G is selected and EMAC then there is a conflict between the PHY interrupt line and TFS. Current settings prevent the conflict by ignoring the TFS pin when Port G is selected. This allows both ssm2602 using Port G and EMAC concurrently. - some code cleanup Signed-off-by: Cliff Cai Signed-off-by: Bryan Wu Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/blackfin/bf5xx-i2s.c | 34 ++++++++++++++++++++++------------ 1 file changed, 22 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 827587f0818..e020c160ee4 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -70,12 +70,24 @@ static struct sport_param sport_params[2] = { } }; -static u16 sport_req[][7] = { - { P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, - P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}, - { P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, - P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}, -}; +/* + * Setting the TFS pin selector for SPORT 0 based on whether the selected + * port id F or G. If the port is F then no conflict should exist for the + * TFS. When Port G is selected and EMAC then there is a conflict between + * the PHY interrupt line and TFS. Current settings prevent the conflict + * by ignoring the TFS pin when Port G is selected. This allows both + * ssm2602 using Port G and EMAC concurrently. + */ +#ifdef CONFIG_BF527_SPORT0_PORTF +#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) +#else +#define LOCAL_SPORT0_TFS (0) +#endif + +static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, + P_SPORT0_DRPRI, P_SPORT0_RSCLK, LOCAL_SPORT0_TFS, 0}, + {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, P_SPORT1_DRPRI, + P_SPORT1_RSCLK, P_SPORT1_TFS, 0} }; static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) @@ -98,23 +110,21 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, ret = -EINVAL; break; default: + printk(KERN_ERR "%s: Unknown DAI format type\n", __func__); ret = -EINVAL; break; } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - ret = -EINVAL; - break; - case SND_SOC_DAIFMT_CBM_CFS: - ret = -EINVAL; - break; case SND_SOC_DAIFMT_CBM_CFM: break; + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBM_CFS: case SND_SOC_DAIFMT_CBS_CFM: ret = -EINVAL; break; default: + printk(KERN_ERR "%s: Unknown DAI master type\n", __func__); ret = -EINVAL; break; } -- cgit v1.2.3 From 78fad343a4fd4c837e3034dcbca13269a783e0f6 Mon Sep 17 00:00:00 2001 From: Bastien Nocera Date: Wed, 29 Oct 2008 12:59:05 +0000 Subject: ALSA: intel8x0 - add Dell Optiplex GX620 (AD1981B) to AC97 clock whitelist alsa-info.sh output at: https://bugzilla.redhat.com/show_bug.cgi?id=441087#c49 Signed-off-by: Bastien Nocera Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index c88d1eace1c..19d3391e229 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2702,6 +2702,7 @@ static struct snd_pci_quirk intel8x0_clock_list[] __devinitdata = { SND_PCI_QUIRK(0x0e11, 0x008a, "AD1885", 41000), SND_PCI_QUIRK(0x1028, 0x00be, "AD1885", 44100), SND_PCI_QUIRK(0x1028, 0x0177, "AD1980", 48000), + SND_PCI_QUIRK(0x1028, 0x01ad, "AD1981B", 48000), SND_PCI_QUIRK(0x1043, 0x80f3, "AD1985", 48000), { } /* terminator */ }; -- cgit v1.2.3 From 366840d7e1ece4331bb3ccc2a53e3bc355529f21 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 29 Oct 2008 14:40:30 +0000 Subject: ALSA: Warn when control names are truncated This is likely to confuse user interfaces since the end of the control name is interpreted (eg, "Volume", "Switch"). Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/core/control.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index 6d71f9a7ccb..b0bf4269104 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -225,8 +225,13 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, kctl.id.iface = ncontrol->iface; kctl.id.device = ncontrol->device; kctl.id.subdevice = ncontrol->subdevice; - if (ncontrol->name) + if (ncontrol->name) { strlcpy(kctl.id.name, ncontrol->name, sizeof(kctl.id.name)); + if (strcmp(ncontrol->name, kctl.id.name) != 0) + snd_printk(KERN_WARNING + "Control name '%s' truncated to '%s'\n", + ncontrol->name, kctl.id.name); + } kctl.id.index = ncontrol->index; kctl.count = ncontrol->count ? ncontrol->count : 1; access = ncontrol->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : -- cgit v1.2.3 From be376649344ba2c3d00021f8bbf64392aa01ad55 Mon Sep 17 00:00:00 2001 From: FUJITA Tomonori Date: Wed, 29 Oct 2008 15:34:39 -0700 Subject: dbri: check dma_alloc_coherent errors Needs to check for dma_alloc_coherent() allocation failure. Signed-off-by: FUJITA Tomonori Signed-off-by: David S. Miller --- sound/sparc/dbri.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index c257ad8bdfb..23ed6f04a71 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2534,6 +2534,8 @@ static int __devinit snd_dbri_create(struct snd_card *card, dbri->dma = dma_alloc_coherent(&op->dev, sizeof(struct dbri_dma), &dbri->dma_dvma, GFP_ATOMIC); + if (!dbri->dma) + return -ENOMEM; memset((void *)dbri->dma, 0, sizeof(struct dbri_dma)); dprintk(D_GEN, "DMA Cmd Block 0x%p (0x%08x)\n", -- cgit v1.2.3 From 096a957169ff1c31d9d8fbf4e59e45497cfae10d Mon Sep 17 00:00:00 2001 From: Kay Sievers Date: Thu, 30 Oct 2008 02:27:20 +0100 Subject: sound: struct device - replace bus_id with dev_name(), dev_set_name() Acked-by: Greg Kroah-Hartman Signed-off-By: Kay Sievers Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 6704acbca8c..bd510eceff1 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1927,9 +1927,9 @@ static int snd_ac97_dev_register(struct snd_device *device) ac97->dev.bus = &ac97_bus_type; ac97->dev.parent = ac97->bus->card->dev; ac97->dev.release = ac97_device_release; - snprintf(ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s", - ac97->bus->card->number, ac97->num, - snd_ac97_get_short_name(ac97)); + dev_set_name(&ac97->dev, "%d-%d:%s", + ac97->bus->card->number, ac97->num, + snd_ac97_get_short_name(ac97)); if ((err = device_register(&ac97->dev)) < 0) { snd_printk(KERN_ERR "Can't register ac97 bus\n"); ac97->dev.bus = NULL; -- cgit v1.2.3 From e2736261b4c85e36f7c8a66dd082ec0751230460 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2008 16:07:45 +0200 Subject: ALSA: usb - Add quirk for Edirol UA-25EX advanced modes Added the quirk for UA-25EX advanced modes. UA-25EX is almost compatible with UA-25. Tested-by: Serge Perinsky Signed-off-by: Takashi Iwai --- sound/usb/usbquirks.h | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) (limited to 'sound') diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 69689e79bf7..92115755d98 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1480,6 +1480,36 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* Advanced modes of the Edirol UA-25EX. + * For the standard mode, UA-25EX has ID 0582:00e7, which + * offers only 16-bit PCM at 44.1 kHz and no MIDI. + */ + USB_DEVICE_VENDOR_SPEC(0x0582, 0x00e6), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "EDIROL", + .product_name = "UA-25EX", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = -1 + } + } + } +}, /* Guillemot devices */ { -- cgit v1.2.3 From 8cd2b264d92bb949ddccfb6db6a82254bb07af75 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Ilpo=20J=C3=A4rvinen?= Date: Thu, 30 Oct 2008 13:09:55 +0200 Subject: ALSA: sound/ice1712: indentation & braces disagree - add braces MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Neither has any significance currently to the flow because err is checked for the same condition before the place of disagreement. Signed-off-by: Ilpo Järvinen Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1712.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 5b442383fcd..58d7cda03de 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2688,12 +2688,13 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, return err; } - if (ice_has_con_ac97(ice)) + if (ice_has_con_ac97(ice)) { err = snd_ice1712_pcm(ice, pcm_dev++, NULL); if (err < 0) { snd_card_free(card); return err; } + } err = snd_ice1712_ac97_mixer(ice); if (err < 0) { @@ -2715,12 +2716,13 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, } } - if (ice_has_con_ac97(ice)) + if (ice_has_con_ac97(ice)) { err = snd_ice1712_pcm_ds(ice, pcm_dev++, NULL); if (err < 0) { snd_card_free(card); return err; } + } if (!c->no_mpu401) { err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712, -- cgit v1.2.3 From 762b8df748d83c14070afbf0c6f8c0f4a91a13bf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 30 Oct 2008 12:37:08 +0000 Subject: ALSA: ASoC: Fix mono controls after conversion to support full int masks When ASoC was converted to support full int width masks SOC_SINGLE_VALUE() omitted the assignment of rshift, causing the control operatins to report some mono controls as stereo. This happened to work some of the time due to a confusion between shift and min in snd_soc_info_volsw(). Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 462e635dfc7..a3adbf06b1e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1462,7 +1462,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; int max = mc->max; - unsigned int shift = mc->min; + unsigned int shift = mc->shift; unsigned int rshift = mc->rshift; if (max == 1) -- cgit v1.2.3 From f06c8fd73e3658a659d5e5b56c44bd80fe244268 Mon Sep 17 00:00:00 2001 From: Jon Smirl Date: Thu, 30 Oct 2008 12:37:09 +0000 Subject: ALSA: ASoC: Fix some minor errors in mpc5200 psc i2s driver Fix missing unsigned for irqsave flags in psc i2s driver Make attribute visiblity static Collect all sysfs errors before checking status [Word wrapped DEVICE_ATTR() lines for 80 columns -- broonie] Signed-off-by: Jon Smirl Acked-by: Grant Likely Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/fsl/mpc5200_psc_i2s.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 86923299bc1..94a02eaa482 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -277,7 +277,7 @@ static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd) struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs; u16 imr; u8 psc_cmd; - long flags; + unsigned long flags; if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) s = &psc_i2s->capture; @@ -699,9 +699,11 @@ static ssize_t psc_i2s_stat_store(struct device *dev, return count; } -DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL); -DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store); -DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store); +static DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL); +static DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show, + psc_i2s_stat_store); +static DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, + psc_i2s_stat_store); /* --------------------------------------------------------------------- * OF platform bus binding code: @@ -819,8 +821,8 @@ static int __devinit psc_i2s_of_probe(struct of_device *op, /* Register the SYSFS files */ rc = device_create_file(psc_i2s->dev, &dev_attr_status); - rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun); - rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun); + rc |= device_create_file(psc_i2s->dev, &dev_attr_capture_overrun); + rc |= device_create_file(psc_i2s->dev, &dev_attr_playback_underrun); if (rc) dev_info(psc_i2s->dev, "error creating sysfs files\n"); -- cgit v1.2.3 From 13d622b33dfbd99205a7add5751ba47445e480a6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 30 Oct 2008 12:37:10 +0000 Subject: ALSA: ASoC: Fix WM9713 ALC Decay Time name The control had an extra space at the end of the name. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm9713.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index aba402b3c99..945b32ed988 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -140,7 +140,7 @@ SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0), SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), -SOC_SINGLE("ALC Decay Time ", AC97_CODEC_CLASS_REV, 4, 15, 0), +SOC_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0), SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0), SOC_ENUM("ALC Function", wm9713_enum[6]), SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0), -- cgit v1.2.3 From 233e70f4228e78eb2f80dc6650f65d3ae3dbf17c Mon Sep 17 00:00:00 2001 From: Al Viro Date: Fri, 31 Oct 2008 23:28:30 +0000 Subject: saner FASYNC handling on file close As it is, all instances of ->release() for files that have ->fasync() need to remember to evict file from fasync lists; forgetting that creates a hole and we actually have a bunch that *does* forget. So let's keep our lives simple - let __fput() check FASYNC in file->f_flags and call ->fasync() there if it's been set. And lose that crap in ->release() instances - leaving it there is still valid, but we don't have to bother anymore. Signed-off-by: Al Viro Signed-off-by: Linus Torvalds --- sound/core/control.c | 1 - sound/core/init.c | 5 ++++- sound/core/pcm_native.c | 1 - sound/core/timer.c | 1 - 4 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index b0bf4269104..636b3b52ef8 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -113,7 +113,6 @@ static int snd_ctl_release(struct inode *inode, struct file *file) unsigned int idx; ctl = file->private_data; - fasync_helper(-1, file, 0, &ctl->fasync); file->private_data = NULL; card = ctl->card; write_lock_irqsave(&card->ctl_files_rwlock, flags); diff --git a/sound/core/init.c b/sound/core/init.c index ef2352c2e45..b47ff8b44be 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -264,8 +264,11 @@ static int snd_disconnect_release(struct inode *inode, struct file *file) } spin_unlock(&shutdown_lock); - if (likely(df)) + if (likely(df)) { + if ((file->f_flags & FASYNC) && df->disconnected_f_op->fasync) + df->disconnected_f_op->fasync(-1, file, 0); return df->disconnected_f_op->release(inode, file); + } panic("%s(%p, %p) failed!", __func__, inode, file); } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index aef18682c03..a789efc9df3 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2169,7 +2169,6 @@ static int snd_pcm_release(struct inode *inode, struct file *file) if (snd_BUG_ON(!substream)) return -ENXIO; pcm = substream->pcm; - fasync_helper(-1, file, 0, &substream->runtime->fasync); mutex_lock(&pcm->open_mutex); snd_pcm_release_substream(substream); kfree(pcm_file); diff --git a/sound/core/timer.c b/sound/core/timer.c index e582face89d..c584408c9f1 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1263,7 +1263,6 @@ static int snd_timer_user_release(struct inode *inode, struct file *file) if (file->private_data) { tu = file->private_data; file->private_data = NULL; - fasync_helper(-1, file, 0, &tu->fasync); if (tu->timeri) snd_timer_close(tu->timeri); kfree(tu->queue); -- cgit v1.2.3 From 4b30fbde910dd40b831528e916cb8a91823f618f Mon Sep 17 00:00:00 2001 From: Al Viro Date: Sat, 1 Nov 2008 18:19:49 +0000 Subject: oss: fix O_NONBLOCK in dmasound_core We broke O_NONBLOCK handling in OSS dmasound_core in 2.3.11-pre3 - the original code copied f_flags to open_mode and then checked for O_NONBLOCK in there, but that got changed to copying f_mode and O_NONBLOCK has not reached that field in any kernel version. Since we do not care for any other bits, the fix is obvious... Signed-off-by: Al Viro Signed-off-by: Linus Torvalds --- sound/oss/dmasound/dmasound.h | 2 +- sound/oss/dmasound/dmasound_core.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/oss/dmasound/dmasound.h b/sound/oss/dmasound/dmasound.h index 1cb13fe56ec..1308d8d3418 100644 --- a/sound/oss/dmasound/dmasound.h +++ b/sound/oss/dmasound/dmasound.h @@ -235,7 +235,7 @@ struct sound_queue { */ int active; wait_queue_head_t action_queue, open_queue, sync_queue; - fmode_t open_mode; + int non_blocking; int busy, syncing, xruns, died; }; diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index b8239f3168f..793b7f47843 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -603,7 +603,7 @@ static ssize_t sq_write(struct file *file, const char __user *src, size_t uLeft, while (uLeft) { while (write_sq.count >= write_sq.max_active) { sq_play(); - if (write_sq.open_mode & O_NONBLOCK) + if (write_sq.non_blocking) return uWritten > 0 ? uWritten : -EAGAIN; SLEEP(write_sq.action_queue); if (signal_pending(current)) @@ -718,7 +718,7 @@ static int sq_open2(struct sound_queue *sq, struct file *file, fmode_t mode, return rc; } - sq->open_mode = file->f_mode; + sq->non_blocking = file->f_flags & O_NONBLOCK; } return rc; } -- cgit v1.2.3 From 219df32faec97349516c29f33008fea59a46e99a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 3 Nov 2008 08:17:05 +0100 Subject: ALSA: rawmidi - Add open check in rawmidi callbacks The drivers (e.g. mtpav) may call rawmidi functions in irq handlers even though the streams are not opened. This results in Oops or panic. This patch adds the rawmidi state check before actually operating the rawmidi buffers. Tested-by: Ingo Molnar Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index c4995c9f573..39672f68ce5 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -148,6 +148,8 @@ static int snd_rawmidi_runtime_free(struct snd_rawmidi_substream *substream) static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *substream,int up) { + if (!substream->opened) + return; if (up) { tasklet_hi_schedule(&substream->runtime->tasklet); } else { @@ -158,6 +160,8 @@ static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *subs static void snd_rawmidi_input_trigger(struct snd_rawmidi_substream *substream, int up) { + if (!substream->opened) + return; substream->ops->trigger(substream, up); if (!up && substream->runtime->event) tasklet_kill(&substream->runtime->tasklet); @@ -857,6 +861,8 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, int result = 0, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; + if (!substream->opened) + return -EBADFD; if (runtime->buffer == NULL) { snd_printd("snd_rawmidi_receive: input is not active!!!\n"); return -EINVAL; @@ -1126,6 +1132,8 @@ int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count) int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream, unsigned char *buffer, int count) { + if (!substream->opened) + return -EBADFD; count = snd_rawmidi_transmit_peek(substream, buffer, count); if (count < 0) return count; -- cgit v1.2.3 From b02555c3845f02924b8224ff1fd9a44f2c144dbb Mon Sep 17 00:00:00 2001 From: Zoltan Devai Date: Mon, 3 Nov 2008 00:30:28 +0100 Subject: ALSA: Fix PIT lockup on some chipsets when using the PC-Speaker Fix PIT lockup on some chipsets when using the PC-Speaker. Signed-off-by: Zoltan Devai Signed-off-by: Takashi Iwai --- sound/drivers/pcsp/pcsp_input.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/drivers/pcsp/pcsp_input.c b/sound/drivers/pcsp/pcsp_input.c index cd9b83e7f7d..0444cdeb4be 100644 --- a/sound/drivers/pcsp/pcsp_input.c +++ b/sound/drivers/pcsp/pcsp_input.c @@ -24,13 +24,13 @@ static void pcspkr_do_sound(unsigned int count) spin_lock_irqsave(&i8253_lock, flags); if (count) { - /* enable counter 2 */ - outb_p(inb_p(0x61) | 3, 0x61); /* set command for counter 2, 2 byte write */ outb_p(0xB6, 0x43); /* select desired HZ */ outb_p(count & 0xff, 0x42); outb((count >> 8) & 0xff, 0x42); + /* enable counter 2 */ + outb_p(inb_p(0x61) | 3, 0x61); } else { /* disable counter 2 */ outb(inb_p(0x61) & 0xFC, 0x61); -- cgit v1.2.3 From bb072bf0980abbe4fc905df91857a8ad34ca673c Mon Sep 17 00:00:00 2001 From: Kay Sievers Date: Sun, 2 Nov 2008 03:50:35 +0100 Subject: sound: struct device - replace bus_id with dev_name(), dev_set_name() [stripped sound/isa/* changes, replaced with the next patch -- tiwai] Signed-off-by: Kay Sievers Signed-off-by: Takashi Iwai --- sound/aoa/soundbus/core.c | 2 +- sound/drivers/ml403-ac97cr.c | 4 ++-- sound/soc/soc-core.c | 4 ++-- 3 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c index f84f3e50578..fa8ab2815a9 100644 --- a/sound/aoa/soundbus/core.c +++ b/sound/aoa/soundbus/core.c @@ -176,7 +176,7 @@ int soundbus_add_one(struct soundbus_dev *dev) return -EINVAL; } - snprintf(dev->ofdev.dev.bus_id, BUS_ID_SIZE, "soundbus:%x", ++devcount); + dev_set_name(&dev->ofdev.dev, "soundbus:%x", ++devcount); dev->ofdev.dev.bus = &soundbus_bus_type; return of_device_register(&dev->ofdev); } diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index ecdbeb6d360..7783843ca9a 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1153,7 +1153,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, /* get irq */ irq = platform_get_irq(pfdev, 0); if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED, - pfdev->dev.bus_id, (void *)ml403_ac97cr)) { + dev_name(&pfdev->dev), (void *)ml403_ac97cr)) { snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " "unable to grab IRQ %d\n", irq); @@ -1166,7 +1166,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, ml403_ac97cr->irq); irq = platform_get_irq(pfdev, 1); if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED, - pfdev->dev.bus_id, (void *)ml403_ac97cr)) { + dev_name(&pfdev->dev), (void *)ml403_ac97cr)) { snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " "unable to grab IRQ %d\n", irq); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 462e635dfc7..0af3bbde5bb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -95,8 +95,8 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) codec->ac97->dev.parent = NULL; codec->ac97->dev.release = soc_ac97_device_release; - snprintf(codec->ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s", - codec->card->number, 0, codec->name); + dev_set_name(&codec->ac97->dev, "%d-%d:%s", + codec->card->number, 0, codec->name); err = device_register(&codec->ac97->dev); if (err < 0) { snd_printk(KERN_ERR "Can't register ac97 bus\n"); -- cgit v1.2.3 From 0418ff0c8e48aae4e5e0cbcd3e19a057bea2e55a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 3 Nov 2008 08:51:33 +0100 Subject: ALSA: remove direct access of dev->bus_id in sound/isa/* Removed the direct accesses of dev->bus_id in sound/isa/* by replacement with dev_err() or dev_warn() functions. Signed-off-by: Takashi Iwai --- sound/isa/ad1848/ad1848.c | 6 +++--- sound/isa/adlib.c | 12 ++++++------ sound/isa/cs423x/cs4231.c | 8 ++++---- sound/isa/cs423x/cs4236.c | 8 ++++---- sound/isa/es1688/es1688.c | 9 +++------ sound/isa/gus/gusclassic.c | 13 +++++-------- sound/isa/gus/gusextreme.c | 19 +++++++------------ sound/isa/sb/sb8.c | 4 ++-- 8 files changed, 34 insertions(+), 45 deletions(-) (limited to 'sound') diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c index b68d20edc20..223a6c03881 100644 --- a/sound/isa/ad1848/ad1848.c +++ b/sound/isa/ad1848/ad1848.c @@ -70,15 +70,15 @@ static int __devinit snd_ad1848_match(struct device *dev, unsigned int n) return 0; if (port[n] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id); + dev_err(dev, "please specify port\n"); return 0; } if (irq[n] == SNDRV_AUTO_IRQ) { - snd_printk(KERN_ERR "%s: please specify irq\n", dev->bus_id); + dev_err(dev, "please specify irq\n"); return 0; } if (dma1[n] == SNDRV_AUTO_DMA) { - snd_printk(KERN_ERR "%s: please specify dma1\n", dev->bus_id); + dev_err(dev, "please specify dma1\n"); return 0; } return 1; diff --git a/sound/isa/adlib.c b/sound/isa/adlib.c index efa8c80d05b..374b7177e11 100644 --- a/sound/isa/adlib.c +++ b/sound/isa/adlib.c @@ -36,7 +36,7 @@ static int __devinit snd_adlib_match(struct device *dev, unsigned int n) return 0; if (port[n] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id); + dev_err(dev, "please specify port\n"); return 0; } return 1; @@ -55,13 +55,13 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n) card = snd_card_new(index[n], id[n], THIS_MODULE, 0); if (!card) { - snd_printk(KERN_ERR "%s: could not create card\n", dev->bus_id); + dev_err(dev, "could not create card\n"); return -EINVAL; } card->private_data = request_region(port[n], 4, CRD_NAME); if (!card->private_data) { - snd_printk(KERN_ERR "%s: could not grab ports\n", dev->bus_id); + dev_err(dev, "could not grab ports\n"); error = -EBUSY; goto out; } @@ -73,13 +73,13 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n) error = snd_opl3_create(card, port[n], port[n] + 2, OPL3_HW_AUTO, 1, &opl3); if (error < 0) { - snd_printk(KERN_ERR "%s: could not create OPL\n", dev->bus_id); + dev_err(dev, "could not create OPL\n"); goto out; } error = snd_opl3_hwdep_new(opl3, 0, 0, NULL); if (error < 0) { - snd_printk(KERN_ERR "%s: could not create FM\n", dev->bus_id); + dev_err(dev, "could not create FM\n"); goto out; } @@ -87,7 +87,7 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n) error = snd_card_register(card); if (error < 0) { - snd_printk(KERN_ERR "%s: could not register card\n", dev->bus_id); + dev_err(dev, "could not register card\n"); goto out; } diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c index ddd289120aa..f019d449e2d 100644 --- a/sound/isa/cs423x/cs4231.c +++ b/sound/isa/cs423x/cs4231.c @@ -74,15 +74,15 @@ static int __devinit snd_cs4231_match(struct device *dev, unsigned int n) return 0; if (port[n] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id); + dev_err(dev, "please specify port\n"); return 0; } if (irq[n] == SNDRV_AUTO_IRQ) { - snd_printk(KERN_ERR "%s: please specify irq\n", dev->bus_id); + dev_err(dev, "please specify irq\n"); return 0; } if (dma1[n] == SNDRV_AUTO_DMA) { - snd_printk(KERN_ERR "%s: please specify dma1\n", dev->bus_id); + dev_err(dev, "please specify dma1\n"); return 0; } return 1; @@ -133,7 +133,7 @@ static int __devinit snd_cs4231_probe(struct device *dev, unsigned int n) mpu_port[n], 0, mpu_irq[n], mpu_irq[n] >= 0 ? IRQF_DISABLED : 0, NULL) < 0) - printk(KERN_WARNING "%s: MPU401 not detected\n", dev->bus_id); + dev_warn(dev, "MPU401 not detected\n"); } snd_card_set_dev(card, dev); diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 91f9c15d3e3..019c9401663 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -488,19 +488,19 @@ static int __devinit snd_cs423x_isa_match(struct device *pdev, return 0; if (port[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "%s: please specify port\n", pdev->bus_id); + dev_err(pdev, "please specify port\n"); return 0; } if (cport[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "%s: please specify cport\n", pdev->bus_id); + dev_err(pdev, "please specify cport\n"); return 0; } if (irq[dev] == SNDRV_AUTO_IRQ) { - snd_printk(KERN_ERR "%s: please specify irq\n", pdev->bus_id); + dev_err(pdev, "please specify irq\n"); return 0; } if (dma1[dev] == SNDRV_AUTO_DMA) { - snd_printk(KERN_ERR "%s: please specify dma1\n", pdev->bus_id); + dev_err(pdev, "please specify dma1\n"); return 0; } return 1; diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index f88639ea64b..b46377139cf 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -88,16 +88,14 @@ static int __devinit snd_es1688_legacy_create(struct snd_card *card, if (irq[n] == SNDRV_AUTO_IRQ) { irq[n] = snd_legacy_find_free_irq(possible_irqs); if (irq[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free IRQ\n", - dev->bus_id); + dev_err(dev, "unable to find a free IRQ\n"); return -EBUSY; } } if (dma8[n] == SNDRV_AUTO_DMA) { dma8[n] = snd_legacy_find_free_dma(possible_dmas); if (dma8[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free DMA\n", - dev->bus_id); + dev_err(dev, "unable to find a free DMA\n"); return -EBUSY; } } @@ -147,8 +145,7 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n) if (snd_opl3_create(card, chip->port, chip->port + 2, OPL3_HW_OPL3, 0, &opl3) < 0) - printk(KERN_WARNING "%s: opl3 not detected at 0x%lx\n", - dev->bus_id, chip->port); + dev_warn(dev, "opl3 not detected at 0x%lx\n", chip->port); else { error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (error < 0) diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c index 8f914b37bf8..426532a4d73 100644 --- a/sound/isa/gus/gusclassic.c +++ b/sound/isa/gus/gusclassic.c @@ -90,24 +90,21 @@ static int __devinit snd_gusclassic_create(struct snd_card *card, if (irq[n] == SNDRV_AUTO_IRQ) { irq[n] = snd_legacy_find_free_irq(possible_irqs); if (irq[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free IRQ\n", - dev->bus_id); + dev_err(dev, "unable to find a free IRQ\n"); return -EBUSY; } } if (dma1[n] == SNDRV_AUTO_DMA) { dma1[n] = snd_legacy_find_free_dma(possible_dmas); if (dma1[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free DMA1\n", - dev->bus_id); + dev_err(dev, "unable to find a free DMA1\n"); return -EBUSY; } } if (dma2[n] == SNDRV_AUTO_DMA) { dma2[n] = snd_legacy_find_free_dma(possible_dmas); if (dma2[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free DMA2\n", - dev->bus_id); + dev_err(dev, "unable to find a free DMA2\n"); return -EBUSY; } } @@ -174,8 +171,8 @@ static int __devinit snd_gusclassic_probe(struct device *dev, unsigned int n) error = -ENODEV; if (gus->max_flag || gus->ess_flag) { - snd_printk(KERN_ERR "%s: GUS Classic or ACE soundcard was " - "not detected at 0x%lx\n", dev->bus_id, gus->gf1.port); + dev_err(dev, "GUS Classic or ACE soundcard was " + "not detected at 0x%lx\n", gus->gf1.port); goto out; } diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index da13185eb0a..7ad4c3b41a8 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -106,16 +106,14 @@ static int __devinit snd_gusextreme_es1688_create(struct snd_card *card, if (irq[n] == SNDRV_AUTO_IRQ) { irq[n] = snd_legacy_find_free_irq(possible_irqs); if (irq[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free IRQ " - "for ES1688\n", dev->bus_id); + dev_err(dev, "unable to find a free IRQ for ES1688\n"); return -EBUSY; } } if (dma8[n] == SNDRV_AUTO_DMA) { dma8[n] = snd_legacy_find_free_dma(possible_dmas); if (dma8[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free DMA " - "for ES1688\n", dev->bus_id); + dev_err(dev, "unable to find a free DMA for ES1688\n"); return -EBUSY; } } @@ -143,16 +141,14 @@ static int __devinit snd_gusextreme_gus_card_create(struct snd_card *card, if (gf1_irq[n] == SNDRV_AUTO_IRQ) { gf1_irq[n] = snd_legacy_find_free_irq(possible_irqs); if (gf1_irq[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free IRQ " - "for GF1\n", dev->bus_id); + dev_err(dev, "unable to find a free IRQ for GF1\n"); return -EBUSY; } } if (dma1[n] == SNDRV_AUTO_DMA) { dma1[n] = snd_legacy_find_free_dma(possible_dmas); if (dma1[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free DMA " - "for GF1\n", dev->bus_id); + dev_err(dev, "unable to find a free DMA for GF1\n"); return -EBUSY; } } @@ -278,8 +274,8 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n) error = -ENODEV; if (!gus->ess_flag) { - snd_printk(KERN_ERR "%s: GUS Extreme soundcard was not " - "detected at 0x%lx\n", dev->bus_id, gus->gf1.port); + dev_err(dev, "GUS Extreme soundcard was not " + "detected at 0x%lx\n", gus->gf1.port); goto out; } gus->codec_flag = 1; @@ -310,8 +306,7 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n) if (snd_opl3_create(card, es1688->port, es1688->port + 2, OPL3_HW_OPL3, 0, &opl3) < 0) - printk(KERN_ERR "%s: opl3 not detected at 0x%lx\n", - dev->bus_id, es1688->port); + dev_warn(dev, "opl3 not detected at 0x%lx\n", es1688->port); else { error = snd_opl3_hwdep_new(opl3, 0, 2, NULL); if (error < 0) diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index 336a3427790..667eccc676a 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -85,11 +85,11 @@ static int __devinit snd_sb8_match(struct device *pdev, unsigned int dev) if (!enable[dev]) return 0; if (irq[dev] == SNDRV_AUTO_IRQ) { - snd_printk(KERN_ERR "%s: please specify irq\n", pdev->bus_id); + dev_err(pdev, "please specify irq\n"); return 0; } if (dma8[dev] == SNDRV_AUTO_DMA) { - snd_printk(KERN_ERR "%s: please specify dma8\n", pdev->bus_id); + dev_err(pdev, "please specify dma8\n"); return 0; } return 1; -- cgit v1.2.3 From 55e03a68d2489d116a5c5e8111ecef3f69831ed6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 3 Nov 2008 10:21:36 +0100 Subject: ALSA: emu10k1 - Add more invert_shared_spdif flag to Audigy models Reported in Novell bnc#440862: https://bugzilla.novell.com/show_bug.cgi?id=440862 Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_main.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 2f283ea6ad9..de5ee8f097f 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1464,6 +1464,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0151_chip = 1, .spk71 = 1, .spdif_bug = 1, + .invert_shared_spdif = 1, /* digital/analog switch swapped */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20021102, .driver = "Audigy2", .name = "Audigy 2 ZS [SB0350]", @@ -1473,6 +1474,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0151_chip = 1, .spk71 = 1, .spdif_bug = 1, + .invert_shared_spdif = 1, /* digital/analog switch swapped */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20011102, .driver = "Audigy2", .name = "Audigy 2 ZS [2001]", @@ -1482,6 +1484,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0151_chip = 1, .spk71 = 1, .spdif_bug = 1, + .invert_shared_spdif = 1, /* digital/analog switch swapped */ .ac97_chip = 1} , /* Audigy 2 */ /* Tested by James@superbug.co.uk 3rd July 2005 */ -- cgit v1.2.3 From 00c9ddd1d4cc73aa0077f379279d716cb0ab0ba5 Mon Sep 17 00:00:00 2001 From: Tim Blechmann Date: Sun, 9 Nov 2008 12:50:52 +0100 Subject: ALSA: HDSP: check for io box before uploading firmware currently the hdsp driver tries to upload the firmware, even if the io box is not connected. this patch adds a check for the io box before trying to upload the firmware. thus instead of messages complaining about the fifo status and firmware loading failure, the driver gives a message that no multiface or digiface is connected. [A minor coding-style fix by tiwai] Signed-off-by: Tim Blechmann Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index d723543bead..d680114b3e4 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5045,6 +5045,10 @@ static int __devinit snd_hdsp_create(struct snd_card *card, /* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */ ssleep(2); + err = hdsp_check_for_iobox(hdsp); + if (err < 0) + return err; + if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { #ifdef HDSP_FW_LOADER if ((err = hdsp_request_fw_loader(hdsp)) < 0) @@ -5057,7 +5061,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card, /* init is complete, we return */ return 0; #endif - /* no iobox connected, we defer initialization */ + /* we defer initialization */ snd_printk(KERN_INFO "Hammerfall-DSP: card initialization pending : waiting for firmware\n"); if ((err = snd_hdsp_create_hwdep(card, hdsp)) < 0) return err; -- cgit v1.2.3 From 3ae7e2e22900b90bbe1d19454405950893c110be Mon Sep 17 00:00:00 2001 From: Tim Blechmann Date: Sat, 8 Nov 2008 14:42:18 +0100 Subject: ALSA: hdsp: check for iobox and upload firmware during ioctl currently, the error message when trying to run hdspmixer or hdspconf if the breakout box is not connected is somehow misleading, since it asks the user to upload the firmware. this patch adds a test, whether the breakout box is connected and tries to upload the firmware in the case, that it is not present, e.g. because of power failures of the breakout box. [Minor coding-style fixes by tiwai] Signed-off-by: Tim Blechmann Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 21 +++++++++++++++++---- 1 file changed, 17 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index d680114b3e4..736246f98ac 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -4548,11 +4548,20 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne { struct hdsp *hdsp = (struct hdsp *)hw->private_data; void __user *argp = (void __user *)arg; + int err; switch (cmd) { case SNDRV_HDSP_IOCTL_GET_PEAK_RMS: { struct hdsp_peak_rms __user *peak_rms = (struct hdsp_peak_rms __user *)arg; + err = hdsp_check_for_iobox(hdsp); + if (err < 0) + return err; + + err = hdsp_check_for_firmware(hdsp, 1); + if (err < 0) + return err; + if (!(hdsp->state & HDSP_FirmwareLoaded)) { snd_printk(KERN_ERR "Hammerfall-DSP: firmware needs to be uploaded to the card.\n"); return -EINVAL; @@ -4572,10 +4581,14 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne unsigned long flags; int i; - if (!(hdsp->state & HDSP_FirmwareLoaded)) { - snd_printk(KERN_ERR "Hammerfall-DSP: Firmware needs to be uploaded to the card.\n"); - return -EINVAL; - } + err = hdsp_check_for_iobox(hdsp); + if (err < 0) + return err; + + err = hdsp_check_for_firmware(hdsp, 1); + if (err < 0) + return err; + spin_lock_irqsave(&hdsp->lock, flags); info.pref_sync_ref = (unsigned char)hdsp_pref_sync_ref(hdsp); info.wordclock_sync_check = (unsigned char)hdsp_wc_sync_check(hdsp); -- cgit v1.2.3 From 0f101fa6bccdc797ddcee40fd4c55e879155a40f Mon Sep 17 00:00:00 2001 From: Ville Syrjala Date: Sun, 9 Nov 2008 20:32:40 +0200 Subject: ALSA: gusextreme: Fix build errors gusextreme depends on opl3 support. Add the approriate select to Kconfig. Also remove the unnecessary hwdep select. Relevant build errors: ERROR: "snd_opl3_hwdep_new" [sound/isa/gus/snd-gusextreme.ko] undefined! ERROR: "snd_opl3_create" [sound/isa/gus/snd-gusextreme.ko] undefined! Signed-off-by: Ville Syrjala Signed-off-by: Takashi Iwai --- sound/isa/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 660beb41f76..ce0aa044e27 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -211,7 +211,7 @@ config SND_GUSCLASSIC config SND_GUSEXTREME tristate "Gravis UltraSound Extreme" - select SND_HWDEP + select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM help -- cgit v1.2.3 From 847cdf42d589882aca683b6fb65b2c7832e92231 Mon Sep 17 00:00:00 2001 From: Grant Likely Date: Fri, 14 Nov 2008 05:19:00 -0700 Subject: powerpc/mpc5200: fix bestcomm Kconfig dependencies Without this patch it is possible to select drivers which require bestcomm support without bestcomm support being selected. This patch reworks the bestcomm dependencies to ensure the correct bestcomm tasks are always enabled. Reported-by: Hans Lehmann Signed-off-by: Grant Likely --- sound/soc/fsl/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index bba9546ba5f..8d73edc5610 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -20,7 +20,8 @@ config SND_SOC_MPC8610_HPCD config SND_SOC_MPC5200_I2S tristate "Freescale MPC5200 PSC in I2S mode driver" + depends on SND_SOC && PPC_MPC52xx && PPC_BESTCOMM select SND_SOC_OF_SIMPLE - depends on SND_SOC && PPC_MPC52xx + select PPC_BESTCOMM_GEN_BD help Say Y here to support the MPC5200 PSCs in I2S mode. -- cgit v1.2.3 From d6f35e3f0da14a5eb02114708d88ae2f59dc8283 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Fri, 14 Nov 2008 19:08:18 +0100 Subject: ALSA: sound/pci/pcxhr/pcxhr.c: introduce missing kfree and pci_disable_device Error handling code following a kzalloc should free the allocated data. The error handling code is adjusted to call pci_disable_device(pci); as well, as done later in the function The semantic match that finds the problem is as follows: (http://www.emn.fr/x-info/coccinelle/) // @r exists@ local idexpression x; statement S; expression E; identifier f,l; position p1,p2; expression *ptr != NULL; @@ ( if ((x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...)) == NULL) S | x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...); ... if (x == NULL) S ) <... when != x when != if (...) { <+...x...+> } x->f = E ...> ( return \(0\|<+...x...+>\|ptr\); | return@p2 ...; ) @script:python@ p1 << r.p1; p2 << r.p2; @@ print "* file: %s kmalloc %s return %s" % (p1[0].file,p1[0].line,p2[0].line) // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/pci/pcxhr/pcxhr.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 0e06c6c9fcc..73de6e989b3 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1229,8 +1229,11 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id return -ENOMEM; } - if (snd_BUG_ON(pci_id->driver_data >= PCI_ID_LAST)) + if (snd_BUG_ON(pci_id->driver_data >= PCI_ID_LAST)) { + kfree(mgr); + pci_disable_device(pci); return -ENODEV; + } card_name = pcxhr_board_params[pci_id->driver_data].board_name; mgr->playback_chips = pcxhr_board_params[pci_id->driver_data].playback_chips; mgr->capture_chips = pcxhr_board_params[pci_id->driver_data].capture_chips; -- cgit v1.2.3 From a39c4ad1089be34d8dc66e926e93a52c44993a0a Mon Sep 17 00:00:00 2001 From: Hannes Eder Date: Thu, 20 Nov 2008 21:25:25 +0100 Subject: sound/sound_core: Fix sparse warnings Fix the following sparse warnings: sound/sound_core.c:460:2: warning: returning void-valued expression sound/sound_core.c:477:2: warning: returning void-valued expression sound/sound_core.c:510:5: warning: symbol 'soundcore_open' was not declared. Should it be static? Signed-off-by: Hannes Eder Signed-off-by: Takashi Iwai --- sound/sound_core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/sound_core.c b/sound/sound_core.c index a75b289a5d7..10ba4218161 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -457,7 +457,7 @@ EXPORT_SYMBOL(unregister_sound_mixer); void unregister_sound_midi(int unit) { - return sound_remove_unit(&chains[2], unit); + sound_remove_unit(&chains[2], unit); } EXPORT_SYMBOL(unregister_sound_midi); @@ -474,7 +474,7 @@ EXPORT_SYMBOL(unregister_sound_midi); void unregister_sound_dsp(int unit) { - return sound_remove_unit(&chains[3], unit); + sound_remove_unit(&chains[3], unit); } @@ -507,7 +507,7 @@ static struct sound_unit *__look_for_unit(int chain, int unit) return NULL; } -int soundcore_open(struct inode *inode, struct file *file) +static int soundcore_open(struct inode *inode, struct file *file) { int chain; int unit = iminor(inode); -- cgit v1.2.3 From e669dae6141ff97d3c7566207f5de3b487dcf837 Mon Sep 17 00:00:00 2001 From: Al Viro Date: Sat, 22 Nov 2008 17:35:04 +0000 Subject: of_platform_driver noise on sparce switch to __init for those; unlike powerpc sparc has no hotplug support for that stuff and their ->probe() tends to call __init functions while being declared __devinit. Signed-off-by: Al Viro Signed-off-by: Linus Torvalds --- sound/sparc/cs4231.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index d44bf98e965..6c427bbac32 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -1856,7 +1856,7 @@ static int __init snd_cs4231_sbus_create(struct snd_card *card, return 0; } -static int __devinit cs4231_sbus_probe(struct of_device *op, const struct of_device_id *match) +static int __init cs4231_sbus_probe(struct of_device *op, const struct of_device_id *match) { struct resource *rp = &op->resource[0]; struct snd_card *card; @@ -2048,7 +2048,7 @@ static int __init snd_cs4231_ebus_create(struct snd_card *card, return 0; } -static int __devinit cs4231_ebus_probe(struct of_device *op, const struct of_device_id *match) +static int __init cs4231_ebus_probe(struct of_device *op, const struct of_device_id *match) { struct snd_card *card; int err; @@ -2072,7 +2072,7 @@ static int __devinit cs4231_ebus_probe(struct of_device *op, const struct of_dev } #endif -static int __devinit cs4231_probe(struct of_device *op, const struct of_device_id *match) +static int __init cs4231_probe(struct of_device *op, const struct of_device_id *match) { #ifdef EBUS_SUPPORT if (!strcmp(op->node->parent->name, "ebus")) @@ -2086,7 +2086,7 @@ static int __devinit cs4231_probe(struct of_device *op, const struct of_device_i return -ENODEV; } -static int __devexit cs4231_remove(struct of_device *op) +static int __exit cs4231_remove(struct of_device *op) { struct snd_cs4231 *chip = dev_get_drvdata(&op->dev); @@ -2112,7 +2112,7 @@ static struct of_platform_driver cs4231_driver = { .name = "audio", .match_table = cs4231_match, .probe = cs4231_probe, - .remove = __devexit_p(cs4231_remove), + .remove = __exit_p(cs4231_remove), }; static int __init cs4231_init(void) -- cgit v1.2.3 From 33b07db9f38fe73b3895f8d4db8fdee03e3afec3 Mon Sep 17 00:00:00 2001 From: Linus Torvalds Date: Mon, 1 Dec 2008 07:55:14 -0800 Subject: Revert "of_platform_driver noise on sparce" This reverts commit e669dae6141ff97d3c7566207f5de3b487dcf837, since it is incomplete, and clashes with fuller patches and the sparc 32/64 unification effort. Requested-by: David Miller Acked-by: Al Viro Signed-off-by: Linus Torvalds --- sound/sparc/cs4231.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index 6c427bbac32..d44bf98e965 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -1856,7 +1856,7 @@ static int __init snd_cs4231_sbus_create(struct snd_card *card, return 0; } -static int __init cs4231_sbus_probe(struct of_device *op, const struct of_device_id *match) +static int __devinit cs4231_sbus_probe(struct of_device *op, const struct of_device_id *match) { struct resource *rp = &op->resource[0]; struct snd_card *card; @@ -2048,7 +2048,7 @@ static int __init snd_cs4231_ebus_create(struct snd_card *card, return 0; } -static int __init cs4231_ebus_probe(struct of_device *op, const struct of_device_id *match) +static int __devinit cs4231_ebus_probe(struct of_device *op, const struct of_device_id *match) { struct snd_card *card; int err; @@ -2072,7 +2072,7 @@ static int __init cs4231_ebus_probe(struct of_device *op, const struct of_device } #endif -static int __init cs4231_probe(struct of_device *op, const struct of_device_id *match) +static int __devinit cs4231_probe(struct of_device *op, const struct of_device_id *match) { #ifdef EBUS_SUPPORT if (!strcmp(op->node->parent->name, "ebus")) @@ -2086,7 +2086,7 @@ static int __init cs4231_probe(struct of_device *op, const struct of_device_id * return -ENODEV; } -static int __exit cs4231_remove(struct of_device *op) +static int __devexit cs4231_remove(struct of_device *op) { struct snd_cs4231 *chip = dev_get_drvdata(&op->dev); @@ -2112,7 +2112,7 @@ static struct of_platform_driver cs4231_driver = { .name = "audio", .match_table = cs4231_match, .probe = cs4231_probe, - .remove = __exit_p(cs4231_remove), + .remove = __devexit_p(cs4231_remove), }; static int __init cs4231_init(void) -- cgit v1.2.3 From e28d83223a1e5672174dcdc6b73c1be3fa3de877 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 17 Dec 2008 13:48:29 +0100 Subject: ALSA: hda - Fix silent HP output on D975 Some desktops seems to have no HP/mic jack detection on the front panel, which results in the silent output in the recent driver, because the driver mutes the output (to save power) when no plug is detected. This patch adds a new model that disables the jack-detection. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 5dd3e89f620..d6fa401791c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -127,6 +127,7 @@ enum { }; enum { + STAC_D965_REF_NO_JD, /* no jack-detection */ STAC_D965_REF, STAC_D965_3ST, STAC_D965_5ST, @@ -2027,6 +2028,7 @@ static unsigned int dell_3st_pin_configs[14] = { }; static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = { + [STAC_D965_REF_NO_JD] = ref927x_pin_configs, [STAC_D965_REF] = ref927x_pin_configs, [STAC_D965_3ST] = d965_3st_pin_configs, [STAC_D965_5ST] = d965_5st_pin_configs, @@ -2035,6 +2037,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = { }; static const char *stac927x_models[STAC_927X_MODELS] = { + [STAC_D965_REF_NO_JD] = "ref-no-jd", [STAC_D965_REF] = "ref", [STAC_D965_3ST] = "3stack", [STAC_D965_5ST] = "5stack", @@ -4899,6 +4902,10 @@ static int patch_stac927x(struct hda_codec *codec) */ codec->bus->needs_damn_long_delay = 1; + /* no jack detecion for ref-no-jd model */ + if (spec->board_config == STAC_D965_REF_NO_JD) + spec->hp_detect = 0; + return 0; } -- cgit v1.2.3 From fa620e97feae731f3d461baaac4da0b127b6fd8e Mon Sep 17 00:00:00 2001 From: Joerg Schirottke Date: Fri, 19 Dec 2008 08:13:49 +0100 Subject: ALSA: hda - Add quirk for Dell Studio 17 Added the matching model=dell-m6 for Dell Studio 17 laptop. Signed-off-by: Joerg Schirottke Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d6fa401791c..0bb3caecb86 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1641,6 +1641,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { "unknown Dell", STAC_DELL_M6_DMIC), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x029f, "Dell Studio 1537", STAC_DELL_M6_DMIC), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02a0, + "Dell Studio 17", STAC_DELL_M6_DMIC), {} /* terminator */ }; -- cgit v1.2.3 From 85f13b673f6a59d29529b2a60b88306c68e7b56c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 19 Dec 2008 08:20:38 +0100 Subject: ALSA: Revert "ALSA: hda: removed unneeded hp_nid references" This reverts commit 07f455f779acfb3eba4921fd1399761559b10fa9. ALSA: hda: removed unneeded hp_nid references Removed unneeded hp_nid references for 92hd73xx codec family. This caused the silent output on some Intel desktops due to missing routing of widget 0x0a and 0x0d. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0bb3caecb86..5fcaae676e2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4259,14 +4259,17 @@ again: switch (spec->multiout.num_dacs) { case 0x3: /* 6 Channel */ + spec->multiout.hp_nid = 0x17; spec->mixer = stac92hd73xx_6ch_mixer; spec->init = stac92hd73xx_6ch_core_init; break; case 0x4: /* 8 Channel */ + spec->multiout.hp_nid = 0x18; spec->mixer = stac92hd73xx_8ch_mixer; spec->init = stac92hd73xx_8ch_core_init; break; case 0x5: /* 10 Channel */ + spec->multiout.hp_nid = 0x19; spec->mixer = stac92hd73xx_10ch_mixer; spec->init = stac92hd73xx_10ch_core_init; }; -- cgit v1.2.3 From 9e43f0de690211cf7153b5f3ec251bc315647ada Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 17 Dec 2008 14:51:01 +0100 Subject: ALSA: hda - Add no-jd model for IDT 92HD73xx Added the model without the jack-detection for some desktops that have really no jack-detection. The recent driver caused regressions regarding the sound output on such machines. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 5fcaae676e2..d7c622f944f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -69,6 +69,7 @@ enum { }; enum { + STAC_92HD73XX_NO_JD, /* no jack-detection */ STAC_92HD73XX_REF, STAC_DELL_M6_AMIC, STAC_DELL_M6_DMIC, @@ -1612,6 +1613,7 @@ static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { }; static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { + [STAC_92HD73XX_NO_JD] = "no-jd", [STAC_92HD73XX_REF] = "ref", [STAC_DELL_M6_AMIC] = "dell-m6-amic", [STAC_DELL_M6_DMIC] = "dell-m6-dmic", @@ -4359,6 +4361,9 @@ again: return err; } + if (spec->board_config == STAC_92HD73XX_NO_JD) + spec->hp_detect = 0; + codec->patch_ops = stac92xx_patch_ops; return 0; -- cgit v1.2.3