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author | Takashi Iwai <tiwai@suse.de> | 2009-12-21 11:21:15 +0100 |
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committer | Takashi Iwai <tiwai@suse.de> | 2009-12-21 11:21:15 +0100 |
commit | de8853bc38ceab1fa7e7f723b21430d4aad60fea (patch) | |
tree | 5084ef51866fd1767324f8dc8eb36e97c55350f5 /sound | |
parent | f5de24b06aa46427500d0fdbe8616b73a71d8c28 (diff) | |
parent | 440b004cf953bec2bc8cd91c64ae707fd7e25327 (diff) | |
download | op-kernel-dev-de8853bc38ceab1fa7e7f723b21430d4aad60fea.zip op-kernel-dev-de8853bc38ceab1fa7e7f723b21430d4aad60fea.tar.gz |
Merge remote branch 'alsa/fixes' into fix/hda
Diffstat (limited to 'sound')
201 files changed, 9545 insertions, 3409 deletions
diff --git a/sound/Kconfig b/sound/Kconfig index 439e15c..fcad760f 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -1,6 +1,3 @@ -# sound/Config.in -# - menuconfig SOUND tristate "Sound card support" depends on HAS_IOMEM @@ -58,7 +55,7 @@ config SOUND_OSS_CORE_PRECLAIM Please read Documentation/feature-removal-schedule.txt for details. - If unusre, say Y. + If unsure, say Y. source "sound/oss/dmasound/Kconfig" @@ -136,4 +133,3 @@ config AC97_BUS sound subsystem and other function drivers completely unrelated to sound although they're sharing the AC97 bus. Concerned drivers should "select" this. - diff --git a/sound/arm/Makefile b/sound/arm/Makefile index 5a549ed..8c0c851 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -3,7 +3,7 @@ # obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o -snd-aaci-objs := aaci.o devdma.o +snd-aaci-objs := aaci.o obj-$(CONFIG_SND_PXA2XX_PCM) += snd-pxa2xx-pcm.o snd-pxa2xx-pcm-objs := pxa2xx-pcm.o diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 1f0f821..1497dce 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -18,10 +18,7 @@ #include <linux/interrupt.h> #include <linux/err.h> #include <linux/amba/bus.h> - -#include <asm/io.h> -#include <asm/irq.h> -#include <asm/sizes.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/initval.h> @@ -30,7 +27,6 @@ #include <sound/pcm_params.h> #include "aaci.h" -#include "devdma.h" #define DRIVER_NAME "aaci-pl041" @@ -492,7 +488,7 @@ static int aaci_pcm_hw_free(struct snd_pcm_substream *substream) /* * Clear out the DMA and any allocated buffers. */ - devdma_hw_free(NULL, substream); + snd_pcm_lib_free_pages(substream); return 0; } @@ -504,21 +500,19 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, int err; aaci_pcm_hw_free(substream); + if (aacirun->pcm_open) { + snd_ac97_pcm_close(aacirun->pcm); + aacirun->pcm_open = 0; + } - err = devdma_hw_alloc(NULL, substream, - params_buffer_bytes(params)); + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(params)); if (err < 0) goto out; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params), - params_channels(params), - aacirun->pcm->r[0].slots); - else - err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params), - params_channels(params), - aacirun->pcm->r[1].slots); - + err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params), + params_channels(params), + aacirun->pcm->r[0].slots); if (err) goto out; @@ -534,7 +528,7 @@ static int aaci_pcm_prepare(struct snd_pcm_substream *substream) struct aaci_runtime *aacirun = runtime->private_data; aacirun->start = (void *)runtime->dma_area; - aacirun->end = aacirun->start + runtime->dma_bytes; + aacirun->end = aacirun->start + snd_pcm_lib_buffer_bytes(substream); aacirun->ptr = aacirun->start; aacirun->period = aacirun->bytes = frames_to_bytes(runtime, runtime->period_size); @@ -551,11 +545,6 @@ static snd_pcm_uframes_t aaci_pcm_pointer(struct snd_pcm_substream *substream) return bytes_to_frames(runtime, bytes); } -static int aaci_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) -{ - return devdma_mmap(NULL, substream, vma); -} - /* * Playback specific ALSA stuff @@ -722,7 +711,6 @@ static struct snd_pcm_ops aaci_playback_ops = { .prepare = aaci_pcm_prepare, .trigger = aaci_pcm_playback_trigger, .pointer = aaci_pcm_pointer, - .mmap = aaci_pcm_mmap, }; static int aaci_pcm_capture_hw_params(struct snd_pcm_substream *substream, @@ -850,7 +838,6 @@ static struct snd_pcm_ops aaci_capture_ops = { .prepare = aaci_pcm_capture_prepare, .trigger = aaci_pcm_capture_trigger, .pointer = aaci_pcm_pointer, - .mmap = aaci_pcm_mmap, }; /* @@ -1040,6 +1027,8 @@ static int __devinit aaci_init_pcm(struct aaci *aaci) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &aaci_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &aaci_capture_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + NULL, 0, 64 * 104); } return ret; diff --git a/sound/arm/devdma.c b/sound/arm/devdma.c deleted file mode 100644 index 9d1e666..0000000 --- a/sound/arm/devdma.c +++ /dev/null @@ -1,80 +0,0 @@ -/* - * linux/sound/arm/devdma.c - * - * Copyright (C) 2003-2004 Russell King, All rights reserved. - * - * 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. - * - * ARM DMA shim for ALSA. - */ -#include <linux/device.h> -#include <linux/dma-mapping.h> - -#include <sound/core.h> -#include <sound/pcm.h> - -#include "devdma.h" - -void devdma_hw_free(struct device *dev, struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_dma_buffer *buf = runtime->dma_buffer_p; - - if (runtime->dma_area == NULL) - return; - - if (buf != &substream->dma_buffer) { - dma_free_coherent(buf->dev.dev, buf->bytes, buf->area, buf->addr); - kfree(runtime->dma_buffer_p); - } - - snd_pcm_set_runtime_buffer(substream, NULL); -} - -int devdma_hw_alloc(struct device *dev, struct snd_pcm_substream *substream, size_t size) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_dma_buffer *buf = runtime->dma_buffer_p; - int ret = 0; - - if (buf) { - if (buf->bytes >= size) - goto out; - devdma_hw_free(dev, substream); - } - - if (substream->dma_buffer.area != NULL && substream->dma_buffer.bytes >= size) { - buf = &substream->dma_buffer; - } else { - buf = kmalloc(sizeof(struct snd_dma_buffer), GFP_KERNEL); - if (!buf) - goto nomem; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = dev; - buf->area = dma_alloc_coherent(dev, size, &buf->addr, GFP_KERNEL); - buf->bytes = size; - buf->private_data = NULL; - - if (!buf->area) - goto free; - } - snd_pcm_set_runtime_buffer(substream, buf); - ret = 1; - out: - runtime->dma_bytes = size; - return ret; - - free: - kfree(buf); - nomem: - return -ENOMEM; -} - -int devdma_mmap(struct device *dev, struct snd_pcm_substream *substream, struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - return dma_mmap_coherent(dev, vma, runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); -} diff --git a/sound/arm/devdma.h b/sound/arm/devdma.h deleted file mode 100644 index d025329..0000000 --- a/sound/arm/devdma.h +++ /dev/null @@ -1,3 +0,0 @@ -void devdma_hw_free(struct device *dev, struct snd_pcm_substream *substream); -int devdma_hw_alloc(struct device *dev, struct snd_pcm_substream *substream, size_t size); -int devdma_mmap(struct device *dev, struct snd_pcm_substream *substream, struct vm_area_struct *vma); diff --git a/sound/core/control.c b/sound/core/control.c index a8b7fab..268ab74 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -75,7 +75,7 @@ static int snd_ctl_open(struct inode *inode, struct file *file) ctl->card = card; ctl->prefer_pcm_subdevice = -1; ctl->prefer_rawmidi_subdevice = -1; - ctl->pid = current->pid; + ctl->pid = get_pid(task_pid(current)); file->private_data = ctl; write_lock_irqsave(&card->ctl_files_rwlock, flags); list_add_tail(&ctl->list, &card->ctl_files); @@ -125,6 +125,7 @@ static int snd_ctl_release(struct inode *inode, struct file *file) control->vd[idx].owner = NULL; up_write(&card->controls_rwsem); snd_ctl_empty_read_queue(ctl); + put_pid(ctl->pid); kfree(ctl); module_put(card->module); snd_card_file_remove(card, file); @@ -672,7 +673,7 @@ static int snd_ctl_elem_info(struct snd_ctl_file *ctl, info->access |= SNDRV_CTL_ELEM_ACCESS_LOCK; if (vd->owner == ctl) info->access |= SNDRV_CTL_ELEM_ACCESS_OWNER; - info->owner = vd->owner_pid; + info->owner = pid_vnr(vd->owner->pid); } else { info->owner = -1; } @@ -827,7 +828,6 @@ static int snd_ctl_elem_lock(struct snd_ctl_file *file, result = -EBUSY; else { vd->owner = file; - vd->owner_pid = current->pid; result = 0; } } @@ -858,7 +858,6 @@ static int snd_ctl_elem_unlock(struct snd_ctl_file *file, result = -EPERM; else { vd->owner = NULL; - vd->owner_pid = 0; result = 0; } } @@ -1120,7 +1119,7 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file, goto __kctl_end; } if (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { - if (file && vd->owner != NULL && vd->owner != file) { + if (vd->owner != NULL && vd->owner != file) { err = -EPERM; goto __kctl_end; } diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c index 34c7d48..7f4d744 100644 --- a/sound/core/hrtimer.c +++ b/sound/core/hrtimer.c @@ -37,14 +37,22 @@ static unsigned int resolution; struct snd_hrtimer { struct snd_timer *timer; struct hrtimer hrt; + atomic_t running; }; static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) { struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt); struct snd_timer *t = stime->timer; + + if (!atomic_read(&stime->running)) + return HRTIMER_NORESTART; + hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution)); snd_timer_interrupt(stime->timer, t->sticks); + + if (!atomic_read(&stime->running)) + return HRTIMER_NORESTART; return HRTIMER_RESTART; } @@ -58,6 +66,7 @@ static int snd_hrtimer_open(struct snd_timer *t) hrtimer_init(&stime->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL); stime->timer = t; stime->hrt.function = snd_hrtimer_callback; + atomic_set(&stime->running, 0); t->private_data = stime; return 0; } @@ -78,16 +87,18 @@ static int snd_hrtimer_start(struct snd_timer *t) { struct snd_hrtimer *stime = t->private_data; + atomic_set(&stime->running, 0); + hrtimer_cancel(&stime->hrt); hrtimer_start(&stime->hrt, ns_to_ktime(t->sticks * resolution), HRTIMER_MODE_REL); + atomic_set(&stime->running, 1); return 0; } static int snd_hrtimer_stop(struct snd_timer *t) { struct snd_hrtimer *stime = t->private_data; - - hrtimer_cancel(&stime->hrt); + atomic_set(&stime->running, 0); return 0; } diff --git a/sound/core/pcm.c b/sound/core/pcm.c index c69c60b..6884ae0 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -435,6 +435,7 @@ static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, return; } snd_iprintf(buffer, "state: %s\n", snd_pcm_state_name(status.state)); + snd_iprintf(buffer, "owner_pid : %d\n", pid_vnr(substream->pid)); snd_iprintf(buffer, "trigger_time: %ld.%09ld\n", status.trigger_tstamp.tv_sec, status.trigger_tstamp.tv_nsec); snd_iprintf(buffer, "tstamp : %ld.%09ld\n", @@ -809,7 +810,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, card = pcm->card; read_lock(&card->ctl_files_rwlock); list_for_each_entry(kctl, &card->ctl_files, list) { - if (kctl->pid == current->pid) { + if (kctl->pid == task_pid(current)) { prefer_subdevice = kctl->prefer_pcm_subdevice; if (prefer_subdevice != -1) break; @@ -900,6 +901,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, substream->private_data = pcm->private_data; substream->ref_count = 1; substream->f_flags = file->f_flags; + substream->pid = get_pid(task_pid(current)); pstr->substream_opened++; *rsubstream = substream; return 0; @@ -921,6 +923,8 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) kfree(runtime->hw_constraints.rules); kfree(runtime); substream->runtime = NULL; + put_pid(substream->pid); + substream->pid = NULL; substream->pstr->substream_opened--; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ab73edf..29ab46a1 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -26,6 +26,7 @@ #include <linux/time.h> #include <linux/pm_qos_params.h> #include <linux/uio.h> +#include <linux/dma-mapping.h> #include <sound/core.h> #include <sound/control.h> #include <sound/info.h> @@ -3061,6 +3062,27 @@ static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file } #endif /* coherent mmap */ +static inline struct page * +snd_pcm_default_page_ops(struct snd_pcm_substream *substream, unsigned long ofs) +{ + void *vaddr = substream->runtime->dma_area + ofs; +#if defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) + return virt_to_page(CAC_ADDR(vaddr)); +#endif +#if defined(CONFIG_PPC32) && defined(CONFIG_NOT_COHERENT_CACHE) + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) { + dma_addr_t addr = substream->runtime->dma_addr + ofs; + addr -= get_dma_offset(substream->dma_buffer.dev.dev); + /* assume dma_handle set via pfn_to_phys() in + * mm/dma-noncoherent.c + */ + return pfn_to_page(addr >> PAGE_SHIFT); + } +#endif + return virt_to_page(vaddr); +} + /* * fault callback for mmapping a RAM page */ @@ -3071,7 +3093,6 @@ static int snd_pcm_mmap_data_fault(struct vm_area_struct *area, struct snd_pcm_runtime *runtime; unsigned long offset; struct page * page; - void *vaddr; size_t dma_bytes; if (substream == NULL) @@ -3081,36 +3102,53 @@ static int snd_pcm_mmap_data_fault(struct vm_area_struct *area, dma_bytes = PAGE_ALIGN(runtime->dma_bytes); if (offset > dma_bytes - PAGE_SIZE) return VM_FAULT_SIGBUS; - if (substream->ops->page) { + if (substream->ops->page) page = substream->ops->page(substream, offset); - if (!page) - return VM_FAULT_SIGBUS; - } else { - vaddr = runtime->dma_area + offset; - page = virt_to_page(vaddr); - } + else + page = snd_pcm_default_page_ops(substream, offset); + if (!page) + return VM_FAULT_SIGBUS; get_page(page); vmf->page = page; return 0; } -static const struct vm_operations_struct snd_pcm_vm_ops_data = -{ +static const struct vm_operations_struct snd_pcm_vm_ops_data = { + .open = snd_pcm_mmap_data_open, + .close = snd_pcm_mmap_data_close, +}; + +static const struct vm_operations_struct snd_pcm_vm_ops_data_fault = { .open = snd_pcm_mmap_data_open, .close = snd_pcm_mmap_data_close, .fault = snd_pcm_mmap_data_fault, }; +#ifndef ARCH_HAS_DMA_MMAP_COHERENT +/* This should be defined / handled globally! */ +#ifdef CONFIG_ARM +#define ARCH_HAS_DMA_MMAP_COHERENT +#endif +#endif + /* * mmap the DMA buffer on RAM */ static int snd_pcm_default_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *area) { - area->vm_ops = &snd_pcm_vm_ops_data; - area->vm_private_data = substream; area->vm_flags |= VM_RESERVED; - atomic_inc(&substream->mmap_count); +#ifdef ARCH_HAS_DMA_MMAP_COHERENT + if (!substream->ops->page && + substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) + return dma_mmap_coherent(substream->dma_buffer.dev.dev, + area, + substream->runtime->dma_area, + substream->runtime->dma_addr, + area->vm_end - area->vm_start); +#endif /* ARCH_HAS_DMA_MMAP_COHERENT */ + /* mmap with fault handler */ + area->vm_ops = &snd_pcm_vm_ops_data_fault; return 0; } @@ -3118,12 +3156,6 @@ static int snd_pcm_default_mmap(struct snd_pcm_substream *substream, * mmap the DMA buffer on I/O memory area */ #if SNDRV_PCM_INFO_MMAP_IOMEM -static const struct vm_operations_struct snd_pcm_vm_ops_data_mmio = -{ - .open = snd_pcm_mmap_data_open, - .close = snd_pcm_mmap_data_close, -}; - int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_struct *area) { @@ -3133,8 +3165,6 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, #ifdef pgprot_noncached area->vm_page_prot = pgprot_noncached(area->vm_page_prot); #endif - area->vm_ops = &snd_pcm_vm_ops_data_mmio; - area->vm_private_data = substream; area->vm_flags |= VM_IO; size = area->vm_end - area->vm_start; offset = area->vm_pgoff << PAGE_SHIFT; @@ -3142,7 +3172,6 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, (substream->runtime->dma_addr + offset) >> PAGE_SHIFT, size, area->vm_page_prot)) return -EAGAIN; - atomic_inc(&substream->mmap_count); return 0; } @@ -3159,6 +3188,7 @@ int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, long size; unsigned long offset; size_t dma_bytes; + int err; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (!(area->vm_flags & (VM_WRITE|VM_READ))) @@ -3183,10 +3213,15 @@ int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, if (offset > dma_bytes - size) return -EINVAL; + area->vm_ops = &snd_pcm_vm_ops_data; + area->vm_private_data = substream; if (substream->ops->mmap) - return substream->ops->mmap(substream, area); + err = substream->ops->mmap(substream, area); else - return snd_pcm_default_mmap(substream, area); + err = snd_pcm_default_mmap(substream, area); + if (!err) + atomic_inc(&substream->mmap_count); + return err; } EXPORT_SYMBOL(snd_pcm_mmap_data); diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index c0adc14..0f5a194 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -242,13 +242,12 @@ static int assign_substream(struct snd_rawmidi *rmidi, int subdevice, return -ENXIO; if (subdevice >= 0 && subdevice >= s->substream_count) return -ENODEV; - if (s->substream_opened >= s->substream_count) - return -EAGAIN; list_for_each_entry(substream, &s->substreams, list) { if (substream->opened) { if (stream == SNDRV_RAWMIDI_STREAM_INPUT || - !(mode & SNDRV_RAWMIDI_LFLG_APPEND)) + !(mode & SNDRV_RAWMIDI_LFLG_APPEND) || + !substream->append) continue; } if (subdevice < 0 || subdevice == substream->number) { @@ -266,18 +265,23 @@ static int open_substream(struct snd_rawmidi *rmidi, { int err; - err = snd_rawmidi_runtime_create(substream); - if (err < 0) - return err; - err = substream->ops->open(substream); - if (err < 0) - return err; - substream->opened = 1; - if (substream->use_count++ == 0) + if (substream->use_count == 0) { + err = snd_rawmidi_runtime_create(substream); + if (err < 0) + return err; + err = substream->ops->open(substream); + if (err < 0) { + snd_rawmidi_runtime_free(substream); + return err; + } + substream->opened = 1; substream->active_sensing = 0; - if (mode & SNDRV_RAWMIDI_LFLG_APPEND) - substream->append = 1; - rmidi->streams[substream->stream].substream_opened++; + if (mode & SNDRV_RAWMIDI_LFLG_APPEND) + substream->append = 1; + substream->pid = get_pid(task_pid(current)); + rmidi->streams[substream->stream].substream_opened++; + } + substream->use_count++; return 0; } @@ -297,27 +301,27 @@ static int rawmidi_open_priv(struct snd_rawmidi *rmidi, int subdevice, int mode, SNDRV_RAWMIDI_STREAM_INPUT, mode, &sinput); if (err < 0) - goto __error; + return err; } if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) { err = assign_substream(rmidi, subdevice, SNDRV_RAWMIDI_STREAM_OUTPUT, mode, &soutput); if (err < 0) - goto __error; + return err; } if (sinput) { err = open_substream(rmidi, sinput, mode); if (err < 0) - goto __error; + return err; } if (soutput) { err = open_substream(rmidi, soutput, mode); if (err < 0) { if (sinput) close_substream(rmidi, sinput, 0); - goto __error; + return err; } } @@ -325,13 +329,6 @@ static int rawmidi_open_priv(struct snd_rawmidi *rmidi, int subdevice, int mode, rfile->input = sinput; rfile->output = soutput; return 0; - - __error: - if (sinput && sinput->runtime) - snd_rawmidi_runtime_free(sinput); - if (soutput && soutput->runtime) - snd_rawmidi_runtime_free(soutput); - return err; } /* called from sound/core/seq/seq_midi.c */ @@ -415,7 +412,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) subdevice = -1; read_lock(&card->ctl_files_rwlock); list_for_each_entry(kctl, &card->ctl_files, list) { - if (kctl->pid == current->pid) { + if (kctl->pid == task_pid(current)) { subdevice = kctl->prefer_rawmidi_subdevice; if (subdevice != -1) break; @@ -468,7 +465,6 @@ static void close_substream(struct snd_rawmidi *rmidi, struct snd_rawmidi_substream *substream, int cleanup) { - rmidi->streams[substream->stream].substream_opened--; if (--substream->use_count) return; @@ -493,6 +489,9 @@ static void close_substream(struct snd_rawmidi *rmidi, snd_rawmidi_runtime_free(substream); substream->opened = 0; substream->append = 0; + put_pid(substream->pid); + substream->pid = NULL; + rmidi->streams[substream->stream].substream_opened--; } static void rawmidi_release_priv(struct snd_rawmidi_file *rfile) @@ -1258,7 +1257,7 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, break; count -= count1; } - if (file->f_flags & O_SYNC) { + if (file->f_flags & O_DSYNC) { spin_lock_irq(&runtime->lock); while (runtime->avail != runtime->buffer_size) { wait_queue_t wait; @@ -1340,6 +1339,9 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, substream->number, (unsigned long) substream->bytes); if (substream->opened) { + snd_iprintf(buffer, + " Owner PID : %d\n", + pid_vnr(substream->pid)); runtime = substream->runtime; snd_iprintf(buffer, " Mode : %s\n" @@ -1361,6 +1363,9 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, substream->number, (unsigned long) substream->bytes); if (substream->opened) { + snd_iprintf(buffer, + " Owner PID : %d\n", + pid_vnr(substream->pid)); runtime = substream->runtime; snd_iprintf(buffer, " Buffer size : %lu\n" diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 146ef00..252e04c 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -165,7 +165,7 @@ MODULE_PARM_DESC(enable, "Enable this dummy soundcard."); module_param_array(pcm_devs, int, NULL, 0444); MODULE_PARM_DESC(pcm_devs, "PCM devices # (0-4) for dummy driver."); module_param_array(pcm_substreams, int, NULL, 0444); -MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-16) for dummy driver."); +MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-128) for dummy driver."); //module_param_array(midi_devs, int, NULL, 0444); //MODULE_PARM_DESC(midi_devs, "MIDI devices # (0-2) for dummy driver."); module_param(fake_buffer, bool, 0444); diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c index 020a5d5..04ae870 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -23,6 +23,7 @@ #include <linux/slab.h> #include <linux/delay.h> #include <linux/init.h> +#include <linux/bitrev.h> #include <asm/unaligned.h> #include <sound/core.h> #include <sound/control.h> @@ -55,18 +56,6 @@ struct cs8427 { struct cs8427_stream capture; }; -static unsigned char swapbits(unsigned char val) -{ - int bit; - unsigned char res = 0; - for (bit = 0; bit < 8; bit++) { - res <<= 1; - res |= val & 1; - val >>= 1; - } - return res; -} - int snd_cs8427_reg_write(struct snd_i2c_device *device, unsigned char reg, unsigned char val) { @@ -149,7 +138,7 @@ static int snd_cs8427_send_corudata(struct snd_i2c_device *device, } data[0] = CS8427_REG_AUTOINC | CS8427_REG_CORU_DATABUF; for (idx = 0; idx < count; idx++) - data[idx + 1] = swapbits(ndata[idx]); + data[idx + 1] = bitrev8(ndata[idx]); if (snd_i2c_sendbytes(device, data, count + 1) != count + 1) return -EIO; return 1; diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index d31c373..c4c6ef7 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -225,7 +225,7 @@ static int vidioc_s_ctrl(struct file *file, void *priv, case V4L2_CID_AUDIO_MUTE: if (tea->ops->mute) { tea->ops->mute(tea, ctrl->value); - tea->mute = 1; + tea->mute = ctrl->value; return 0; } } diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index a076a6c..cc15d1d 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -177,7 +177,7 @@ static struct pnp_card_device_id snd_cs423x_pnpids[] = { { .id = "CSC0437", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } }, /* Digital PC 5000 Onboard - CS4236B */ { .id = "CSC0735", .devs = { { "CSC0000" }, { "CSC0010" } } }, - /* some uknown CS4236B */ + /* some unknown CS4236B */ { .id = "CSC0b35", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } }, /* Intel PR440FX Onboard sound */ { .id = "CSC0b36", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } }, @@ -394,21 +394,15 @@ static int __devinit snd_cs423x_probe(struct snd_card *card, int dev) return -EBUSY; } - err = snd_wss_create(card, port[dev], cport[dev], + err = snd_cs4236_create(card, port[dev], cport[dev], irq[dev], dma1[dev], dma2[dev], WSS_HW_DETECT3, 0, &chip); if (err < 0) return err; + + acard->chip = chip; if (chip->hardware & WSS_HW_CS4236B_MASK) { - snd_wss_free(chip); - err = snd_cs4236_create(card, - port[dev], cport[dev], - irq[dev], dma1[dev], dma2[dev], - WSS_HW_DETECT, 0, &chip); - if (err < 0) - return err; - acard->chip = chip; err = snd_cs4236_pcm(chip, 0, &pcm); if (err < 0) @@ -418,7 +412,6 @@ static int __devinit snd_cs423x_probe(struct snd_card *card, int dev) if (err < 0) return err; } else { - acard->chip = chip; err = snd_wss_pcm(chip, 0, &pcm); if (err < 0) return err; diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 38835f3..c5adca3 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -87,6 +87,8 @@ #include <sound/core.h> #include <sound/wss.h> #include <sound/asoundef.h> +#include <sound/initval.h> +#include <sound/tlv.h> /* * @@ -264,7 +266,10 @@ static void snd_cs4236_resume(struct snd_wss *chip) } #endif /* CONFIG_PM */ - +/* + * This function does no fail if the chip is not CS4236B or compatible. + * It just an equivalent to the snd_wss_create() then. + */ int snd_cs4236_create(struct snd_card *card, unsigned long port, unsigned long cport, @@ -281,21 +286,17 @@ int snd_cs4236_create(struct snd_card *card, *rchip = NULL; if (hardware == WSS_HW_DETECT) hardware = WSS_HW_DETECT3; - if (cport < 0x100) { - snd_printk(KERN_ERR "please, specify control port " - "for CS4236+ chips\n"); - return -ENODEV; - } + err = snd_wss_create(card, port, cport, irq, dma1, dma2, hardware, hwshare, &chip); if (err < 0) return err; - if (!(chip->hardware & WSS_HW_CS4236B_MASK)) { - snd_printk(KERN_ERR "CS4236+: MODE3 and extended registers " - "not available, hardware=0x%x\n", chip->hardware); - snd_device_free(card, chip); - return -ENODEV; + if ((chip->hardware & WSS_HW_CS4236B_MASK) == 0) { + snd_printd("chip is not CS4236+, hardware=0x%x\n", + chip->hardware); + *rchip = chip; + return 0; } #if 0 { @@ -308,9 +309,16 @@ int snd_cs4236_create(struct snd_card *card, idx, snd_cs4236_ctrl_in(chip, idx)); } #endif + if (cport < 0x100 || cport == SNDRV_AUTO_PORT) { + snd_printk(KERN_ERR "please, specify control port " + "for CS4236+ chips\n"); + snd_device_free(card, chip); + return -ENODEV; + } ver1 = snd_cs4236_ctrl_in(chip, 1); ver2 = snd_cs4236_ext_in(chip, CS4236_VERSION); - snd_printdd("CS4236: [0x%lx] C1 (version) = 0x%x, ext = 0x%x\n", cport, ver1, ver2); + snd_printdd("CS4236: [0x%lx] C1 (version) = 0x%x, ext = 0x%x\n", + cport, ver1, ver2); if (ver1 != ver2) { snd_printk(KERN_ERR "CS4236+ chip detected, but " "control port 0x%lx is not valid\n", cport); @@ -321,13 +329,17 @@ int snd_cs4236_create(struct snd_card *card, snd_cs4236_ctrl_out(chip, 2, 0xff); snd_cs4236_ctrl_out(chip, 3, 0x00); snd_cs4236_ctrl_out(chip, 4, 0x80); - snd_cs4236_ctrl_out(chip, 5, ((IEC958_AES1_CON_PCM_CODER & 3) << 6) | IEC958_AES0_CON_EMPHASIS_NONE); + reg = ((IEC958_AES1_CON_PCM_CODER & 3) << 6) | + IEC958_AES0_CON_EMPHASIS_NONE; + snd_cs4236_ctrl_out(chip, 5, reg); snd_cs4236_ctrl_out(chip, 6, IEC958_AES1_CON_PCM_CODER >> 2); snd_cs4236_ctrl_out(chip, 7, 0x00); - /* 0x8c for C8 is valid for Turtle Beach Malibu - the IEC-958 output */ - /* is working with this setup, other hardware should have */ - /* different signal paths and this value should be selectable */ - /* in the future */ + /* + * 0x8c for C8 is valid for Turtle Beach Malibu - the IEC-958 + * output is working with this setup, other hardware should + * have different signal paths and this value should be + * selectable in the future + */ snd_cs4236_ctrl_out(chip, 8, 0x8c); chip->rate_constraint = snd_cs4236_xrate; chip->set_playback_format = snd_cs4236_playback_format; @@ -339,9 +351,10 @@ int snd_cs4236_create(struct snd_card *card, /* initialize extended registers */ for (reg = 0; reg < sizeof(snd_cs4236_ext_map); reg++) - snd_cs4236_ext_out(chip, CS4236_I23VAL(reg), snd_cs4236_ext_map[reg]); + snd_cs4236_ext_out(chip, CS4236_I23VAL(reg), + snd_cs4236_ext_map[reg]); - /* initialize compatible but more featured registers */ + /* initialize compatible but more featured registers */ snd_wss_out(chip, CS4231_LEFT_INPUT, 0x40); snd_wss_out(chip, CS4231_RIGHT_INPUT, 0x40); snd_wss_out(chip, CS4231_AUX1_LEFT_INPUT, 0xff); @@ -387,6 +400,14 @@ int snd_cs4236_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm) .get = snd_cs4236_get_single, .put = snd_cs4236_put_single, \ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } +#define CS4236_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = snd_cs4236_info_single, \ + .get = snd_cs4236_get_single, .put = snd_cs4236_put_single, \ + .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24), \ + .tlv = { .p = (xtlv) } } + static int snd_cs4236_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { int mask = (kcontrol->private_value >> 16) & 0xff; @@ -490,6 +511,16 @@ static int snd_cs4236_put_singlec(struct snd_kcontrol *kcontrol, struct snd_ctl_ .get = snd_cs4236_get_double, .put = snd_cs4236_put_double, \ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } +#define CS4236_DOUBLE_TLV(xname, xindex, left_reg, right_reg, shift_left, \ + shift_right, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = snd_cs4236_info_double, \ + .get = snd_cs4236_get_double, .put = snd_cs4236_put_double, \ + .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \ + (shift_right << 19) | (mask << 24) | (invert << 22), \ + .tlv = { .p = (xtlv) } } + static int snd_cs4236_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { int mask = (kcontrol->private_value >> 24) & 0xff; @@ -560,12 +591,23 @@ static int snd_cs4236_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_e return change; } -#define CS4236_DOUBLE1(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ +#define CS4236_DOUBLE1(xname, xindex, left_reg, right_reg, shift_left, \ + shift_right, mask, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_cs4236_info_double, \ .get = snd_cs4236_get_double1, .put = snd_cs4236_put_double1, \ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } +#define CS4236_DOUBLE1_TLV(xname, xindex, left_reg, right_reg, shift_left, \ + shift_right, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = snd_cs4236_info_double, \ + .get = snd_cs4236_get_double1, .put = snd_cs4236_put_double1, \ + .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \ + (shift_right << 19) | (mask << 24) | (invert << 22), \ + .tlv = { .p = (xtlv) } } + static int snd_cs4236_get_double1(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_wss *chip = snd_kcontrol_chip(kcontrol); @@ -619,16 +661,18 @@ static int snd_cs4236_put_double1(struct snd_kcontrol *kcontrol, struct snd_ctl_ return change; } -#define CS4236_MASTER_DIGITAL(xname, xindex) \ +#define CS4236_MASTER_DIGITAL(xname, xindex, xtlv) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ .info = snd_cs4236_info_double, \ .get = snd_cs4236_get_master_digital, .put = snd_cs4236_put_master_digital, \ - .private_value = 71 << 24 } + .private_value = 71 << 24, \ + .tlv = { .p = (xtlv) } } static inline int snd_cs4236_mixer_master_digital_invert_volume(int vol) { return (vol < 64) ? 63 - vol : 64 + (71 - vol); -} +} static int snd_cs4236_get_master_digital(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -661,11 +705,13 @@ static int snd_cs4236_put_master_digital(struct snd_kcontrol *kcontrol, struct s return change; } -#define CS4235_OUTPUT_ACCU(xname, xindex) \ +#define CS4235_OUTPUT_ACCU(xname, xindex, xtlv) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ .info = snd_cs4236_info_double, \ .get = snd_cs4235_get_output_accu, .put = snd_cs4235_put_output_accu, \ - .private_value = 3 << 24 } + .private_value = 3 << 24, \ + .tlv = { .p = (xtlv) } } static inline int snd_cs4235_mixer_output_accu_get_volume(int vol) { @@ -720,41 +766,56 @@ static int snd_cs4235_put_output_accu(struct snd_kcontrol *kcontrol, struct snd_ return change; } +static const DECLARE_TLV_DB_SCALE(db_scale_7bit, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_6bit_12db_max, -8250, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_5bit_22db_max, -2400, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_4bit, -4500, 300, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_2bit, -1800, 600, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0); + static struct snd_kcontrol_new snd_cs4236_controls[] = { CS4236_DOUBLE("Master Digital Playback Switch", 0, CS4236_LEFT_MASTER, CS4236_RIGHT_MASTER, 7, 7, 1, 1), CS4236_DOUBLE("Master Digital Capture Switch", 0, CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), -CS4236_MASTER_DIGITAL("Master Digital Volume", 0), +CS4236_MASTER_DIGITAL("Master Digital Volume", 0, db_scale_7bit), -CS4236_DOUBLE("Capture Boost Volume", 0, - CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1), +CS4236_DOUBLE_TLV("Capture Boost Volume", 0, + CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1, + db_scale_2bit), WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -WSS_DOUBLE("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +WSS_DOUBLE_TLV("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, + db_scale_6bit), CS4236_DOUBLE("DSP Playback Switch", 0, CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 7, 7, 1, 1), -CS4236_DOUBLE("DSP Playback Volume", 0, - CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 0, 0, 63, 1), +CS4236_DOUBLE_TLV("DSP Playback Volume", 0, + CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 0, 0, 63, 1, + db_scale_6bit), CS4236_DOUBLE("FM Playback Switch", 0, CS4236_LEFT_FM, CS4236_RIGHT_FM, 7, 7, 1, 1), -CS4236_DOUBLE("FM Playback Volume", 0, - CS4236_LEFT_FM, CS4236_RIGHT_FM, 0, 0, 63, 1), +CS4236_DOUBLE_TLV("FM Playback Volume", 0, + CS4236_LEFT_FM, CS4236_RIGHT_FM, 0, 0, 63, 1, + db_scale_6bit), CS4236_DOUBLE("Wavetable Playback Switch", 0, CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 7, 7, 1, 1), -CS4236_DOUBLE("Wavetable Playback Volume", 0, - CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 0, 0, 63, 1), +CS4236_DOUBLE_TLV("Wavetable Playback Volume", 0, + CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 0, 0, 63, 1, + db_scale_6bit_12db_max), WSS_DOUBLE("Synth Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Synth Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Synth Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("Synth Capture Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 6, 6, 1, 1), WSS_DOUBLE("Synth Capture Bypass", 0, @@ -764,14 +825,16 @@ CS4236_DOUBLE("Mic Playback Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 6, 6, 1, 1), CS4236_DOUBLE("Mic Capture Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 7, 7, 1, 1), -CS4236_DOUBLE("Mic Volume", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 0, 0, 31, 1), -CS4236_DOUBLE("Mic Playback Boost", 0, +CS4236_DOUBLE_TLV("Mic Volume", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, + 0, 0, 31, 1, db_scale_5bit_22db_max), +CS4236_DOUBLE("Mic Playback Boost (+20dB)", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 5, 5, 1, 0), WSS_DOUBLE("Line Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Line Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Line Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("Line Capture Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 6, 6, 1, 1), WSS_DOUBLE("Line Capture Bypass", 0, @@ -779,57 +842,63 @@ WSS_DOUBLE("Line Capture Bypass", 0, WSS_DOUBLE("CD Playback Switch", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("CD Volume", 0, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("CD Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("CD Capture Switch", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 6, 6, 1, 1), CS4236_DOUBLE1("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1), -CS4236_DOUBLE1("Mono Playback Switch", 0, +CS4236_DOUBLE1("Beep Playback Switch", 0, CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), -WSS_SINGLE("Mono Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), -WSS_SINGLE("Mono Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0), +WSS_SINGLE_TLV("Beep Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1, + db_scale_4bit), +WSS_SINGLE("Beep Bypass Playback Switch", 0, CS4231_MONO_CTRL, 5, 1, 0), -WSS_DOUBLE("Capture Volume", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, + 0, 0, 15, 0, db_scale_rec_gain), WSS_DOUBLE("Analog Loopback Capture Switch", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 7, 7, 1, 0), -WSS_SINGLE("Digital Loopback Playback Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -CS4236_DOUBLE1("Digital Loopback Playback Volume", 0, - CS4231_LOOPBACK, CS4236_RIGHT_LOOPBACK, 2, 0, 63, 1) +WSS_SINGLE("Loopback Digital Playback Switch", 0, CS4231_LOOPBACK, 0, 1, 0), +CS4236_DOUBLE1_TLV("Loopback Digital Playback Volume", 0, + CS4231_LOOPBACK, CS4236_RIGHT_LOOPBACK, 2, 0, 63, 1, + db_scale_6bit), }; +static const DECLARE_TLV_DB_SCALE(db_scale_5bit_6db_max, -5600, 200, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_2bit_16db_max, -2400, 800, 0); + static struct snd_kcontrol_new snd_cs4235_controls[] = { -WSS_DOUBLE("Master Switch", 0, +WSS_DOUBLE("Master Playback Switch", 0, CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 7, 7, 1, 1), -WSS_DOUBLE("Master Volume", 0, - CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 0, 0, 31, 1), - -CS4235_OUTPUT_ACCU("Playback Volume", 0), +WSS_DOUBLE_TLV("Master Playback Volume", 0, + CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 0, 0, 31, 1, + db_scale_5bit_6db_max), -CS4236_DOUBLE("Master Digital Playback Switch", 0, - CS4236_LEFT_MASTER, CS4236_RIGHT_MASTER, 7, 7, 1, 1), -CS4236_DOUBLE("Master Digital Capture Switch", 0, - CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), -CS4236_MASTER_DIGITAL("Master Digital Volume", 0), +CS4235_OUTPUT_ACCU("Playback Volume", 0, db_scale_2bit_16db_max), -WSS_DOUBLE("Master Digital Playback Switch", 1, +WSS_DOUBLE("Synth Playback Switch", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Master Digital Capture Switch", 1, +WSS_DOUBLE("Synth Capture Switch", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 6, 6, 1, 1), -WSS_DOUBLE("Master Digital Volume", 1, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Synth Volume", 1, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, + db_scale_5bit_12db_max), -CS4236_DOUBLE("Capture Volume", 0, - CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1), +CS4236_DOUBLE_TLV("Capture Volume", 0, + CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1, + db_scale_2bit), -WSS_DOUBLE("PCM Switch", 0, +WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -WSS_DOUBLE("PCM Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +WSS_DOUBLE("PCM Capture Switch", 0, + CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), +WSS_DOUBLE_TLV("PCM Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, + db_scale_6bit), CS4236_DOUBLE("DSP Switch", 0, CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 7, 7, 1, 1), @@ -842,29 +911,29 @@ CS4236_DOUBLE("Mic Capture Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 7, 7, 1, 1), CS4236_DOUBLE("Mic Playback Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 6, 6, 1, 1), -CS4236_SINGLE("Mic Volume", 0, CS4236_LEFT_MIC, 0, 31, 1), -CS4236_SINGLE("Mic Playback Boost", 0, CS4236_LEFT_MIC, 5, 1, 0), +CS4236_SINGLE_TLV("Mic Volume", 0, CS4236_LEFT_MIC, 0, 31, 1, + db_scale_5bit_22db_max), +CS4236_SINGLE("Mic Boost (+20dB)", 0, CS4236_LEFT_MIC, 5, 1, 0), -WSS_DOUBLE("Aux Playback Switch", 0, +WSS_DOUBLE("Line Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Capture Switch", 0, +WSS_DOUBLE("Line Capture Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 6, 6, 1, 1), -WSS_DOUBLE("Aux Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Line Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), -WSS_DOUBLE("Aux Playback Switch", 1, +WSS_DOUBLE("CD Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Capture Switch", 1, +WSS_DOUBLE("CD Capture Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 6, 6, 1, 1), -WSS_DOUBLE("Aux Volume", 1, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), - -CS4236_DOUBLE1("Master Mono Switch", 0, - CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1), +WSS_DOUBLE_TLV("CD Volume", 1, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), -CS4236_DOUBLE1("Mono Switch", 0, +CS4236_DOUBLE1("Beep Playback Switch", 0, CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), -WSS_SINGLE("Mono Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), +WSS_SINGLE("Beep Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), WSS_DOUBLE("Analog Loopback Switch", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 7, 7, 1, 0), diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index e5bf335..9a43baa 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -102,8 +102,6 @@ struct snd_es18xx { unsigned long port; /* port of ESS chip */ - unsigned long mpu_port; /* MPU-401 port of ESS chip */ - unsigned long fm_port; /* FM port */ unsigned long ctrl_port; /* Control port of ESS chip */ struct resource *res_port; struct resource *res_mpu_port; @@ -116,8 +114,6 @@ struct snd_es18xx { unsigned short audio2_vol; /* volume level of audio2 */ unsigned short active; /* active channel mask */ - unsigned int dma1_size; - unsigned int dma2_size; unsigned int dma1_shift; unsigned int dma2_shift; @@ -135,7 +131,6 @@ struct snd_es18xx { spinlock_t reg_lock; spinlock_t mixer_lock; - spinlock_t ctrl_lock; #ifdef CONFIG_PM unsigned char pm_reg; #endif @@ -354,7 +349,7 @@ static inline int snd_es18xx_mixer_writable(struct snd_es18xx *chip, unsigned ch } -static int snd_es18xx_reset(struct snd_es18xx *chip) +static int __devinit snd_es18xx_reset(struct snd_es18xx *chip) { int i; outb(0x03, chip->port + 0x06); @@ -490,8 +485,6 @@ static int snd_es18xx_playback1_prepare(struct snd_es18xx *chip, unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); - chip->dma2_size = size; - snd_es18xx_rate_set(chip, substream, DAC2); /* Transfer Count Reload */ @@ -591,8 +584,6 @@ static int snd_es18xx_capture_prepare(struct snd_pcm_substream *substream) unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); - chip->dma1_size = size; - snd_es18xx_reset_fifo(chip); /* Set stereo/mono */ @@ -659,8 +650,6 @@ static int snd_es18xx_playback2_prepare(struct snd_es18xx *chip, unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); - chip->dma1_size = size; - snd_es18xx_reset_fifo(chip); /* Set stereo/mono */ @@ -821,17 +810,18 @@ static irqreturn_t snd_es18xx_interrupt(int irq, void *dev_id) static snd_pcm_uframes_t snd_es18xx_playback_pointer(struct snd_pcm_substream *substream) { struct snd_es18xx *chip = snd_pcm_substream_chip(substream); + unsigned int size = snd_pcm_lib_buffer_bytes(substream); int pos; if (substream->number == 0 && (chip->caps & ES18XX_PCM2)) { if (!(chip->active & DAC2)) return 0; - pos = snd_dma_pointer(chip->dma2, chip->dma2_size); + pos = snd_dma_pointer(chip->dma2, size); return pos >> chip->dma2_shift; } else { if (!(chip->active & DAC1)) return 0; - pos = snd_dma_pointer(chip->dma1, chip->dma1_size); + pos = snd_dma_pointer(chip->dma1, size); return pos >> chip->dma1_shift; } } @@ -839,11 +829,12 @@ static snd_pcm_uframes_t snd_es18xx_playback_pointer(struct snd_pcm_substream *s static snd_pcm_uframes_t snd_es18xx_capture_pointer(struct snd_pcm_substream *substream) { struct snd_es18xx *chip = snd_pcm_substream_chip(substream); + unsigned int size = snd_pcm_lib_buffer_bytes(substream); int pos; if (!(chip->active & ADC1)) return 0; - pos = snd_dma_pointer(chip->dma1, chip->dma1_size); + pos = snd_dma_pointer(chip->dma1, size); return pos >> chip->dma1_shift; } @@ -974,9 +965,6 @@ static int snd_es18xx_capture_close(struct snd_pcm_substream *substream) static int snd_es18xx_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts4Source[4] = { - "Mic", "CD", "Line", "Master" - }; static char *texts5Source[5] = { "Mic", "CD", "Line", "Master", "Mix" }; @@ -994,7 +982,8 @@ static int snd_es18xx_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_ele uinfo->value.enumerated.items = 4; if (uinfo->value.enumerated.item > 3) uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, texts4Source[uinfo->value.enumerated.item]); + strcpy(uinfo->value.enumerated.name, + texts5Source[uinfo->value.enumerated.item]); break; case 0x1887: case 0x1888: @@ -1378,11 +1367,9 @@ ES18XX_SINGLE("Hardware Master Volume Split", 0, 0x64, 7, 1, 0), static int __devinit snd_es18xx_config_read(struct snd_es18xx *chip, unsigned char reg) { int data; - unsigned long flags; - spin_lock_irqsave(&chip->ctrl_lock, flags); + outb(reg, chip->ctrl_port); data = inb(chip->ctrl_port + 1); - spin_unlock_irqrestore(&chip->ctrl_lock, flags); return data; } @@ -1398,7 +1385,9 @@ static void __devinit snd_es18xx_config_write(struct snd_es18xx *chip, #endif } -static int __devinit snd_es18xx_initialize(struct snd_es18xx *chip) +static int __devinit snd_es18xx_initialize(struct snd_es18xx *chip, + unsigned long mpu_port, + unsigned long fm_port) { int mask = 0; @@ -1412,15 +1401,15 @@ static int __devinit snd_es18xx_initialize(struct snd_es18xx *chip) if (chip->caps & ES18XX_CONTROL) { /* Hardware volume IRQ */ snd_es18xx_config_write(chip, 0x27, chip->irq); - if (chip->fm_port > 0 && chip->fm_port != SNDRV_AUTO_PORT) { + if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) { /* FM I/O */ - snd_es18xx_config_write(chip, 0x62, chip->fm_port >> 8); - snd_es18xx_config_write(chip, 0x63, chip->fm_port & 0xff); + snd_es18xx_config_write(chip, 0x62, fm_port >> 8); + snd_es18xx_config_write(chip, 0x63, fm_port & 0xff); } - if (chip->mpu_port > 0 && chip->mpu_port != SNDRV_AUTO_PORT) { + if (mpu_port > 0 && mpu_port != SNDRV_AUTO_PORT) { /* MPU-401 I/O */ - snd_es18xx_config_write(chip, 0x64, chip->mpu_port >> 8); - snd_es18xx_config_write(chip, 0x65, chip->mpu_port & 0xff); + snd_es18xx_config_write(chip, 0x64, mpu_port >> 8); + snd_es18xx_config_write(chip, 0x65, mpu_port & 0xff); /* MPU-401 IRQ */ snd_es18xx_config_write(chip, 0x28, chip->irq); } @@ -1507,11 +1496,12 @@ static int __devinit snd_es18xx_initialize(struct snd_es18xx *chip) snd_es18xx_mixer_write(chip, 0x7A, 0x68); /* Enable and set hardware volume interrupt */ snd_es18xx_mixer_write(chip, 0x64, 0x06); - if (chip->mpu_port > 0 && chip->mpu_port != SNDRV_AUTO_PORT) { + if (mpu_port > 0 && mpu_port != SNDRV_AUTO_PORT) { /* MPU401 share irq with audio Joystick enabled FM enabled */ - snd_es18xx_mixer_write(chip, 0x40, 0x43 | (chip->mpu_port & 0xf0) >> 1); + snd_es18xx_mixer_write(chip, 0x40, + 0x43 | (mpu_port & 0xf0) >> 1); } snd_es18xx_mixer_write(chip, 0x7f, ((irqmask + 1) << 1) | 0x01); } @@ -1629,7 +1619,9 @@ static int __devinit snd_es18xx_identify(struct snd_es18xx *chip) return 0; } -static int __devinit snd_es18xx_probe(struct snd_es18xx *chip) +static int __devinit snd_es18xx_probe(struct snd_es18xx *chip, + unsigned long mpu_port, + unsigned long fm_port) { if (snd_es18xx_identify(chip) < 0) { snd_printk(KERN_ERR PFX "[0x%lx] ESS chip not found\n", chip->port); @@ -1650,8 +1642,6 @@ static int __devinit snd_es18xx_probe(struct snd_es18xx *chip) chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_I2S | ES18XX_CONTROL | ES18XX_HWV; break; case 0x1887: - chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME; - break; case 0x1888: chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME; break; @@ -1666,7 +1656,7 @@ static int __devinit snd_es18xx_probe(struct snd_es18xx *chip) if (chip->dma1 == chip->dma2) chip->caps &= ~(ES18XX_PCM2 | ES18XX_DUPLEX_SAME); - return snd_es18xx_initialize(chip); + return snd_es18xx_initialize(chip, mpu_port, fm_port); } static struct snd_pcm_ops snd_es18xx_playback_ops = { @@ -1802,10 +1792,7 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card, spin_lock_init(&chip->reg_lock); spin_lock_init(&chip->mixer_lock); - spin_lock_init(&chip->ctrl_lock); chip->port = port; - chip->mpu_port = mpu_port; - chip->fm_port = fm_port; chip->irq = -1; chip->dma1 = -1; chip->dma2 = -1; @@ -1841,11 +1828,11 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card, } chip->dma2 = dma2; - if (snd_es18xx_probe(chip) < 0) { + if (snd_es18xx_probe(chip, mpu_port, fm_port) < 0) { snd_es18xx_free(card); - return -ENODEV; - } - err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, NULL, &ops); + return -ENODEV; + } + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { snd_es18xx_free(card); return err; @@ -1980,7 +1967,7 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ #ifdef CONFIG_PNP -static int isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; +static int isapnp[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; #endif static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260,0x280 */ #ifndef CONFIG_PNP @@ -2160,19 +2147,23 @@ static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev) return err; if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) { - if (snd_opl3_create(card, chip->fm_port, chip->fm_port + 2, OPL3_HW_OPL3, 0, &opl3) < 0) { - snd_printk(KERN_WARNING PFX "opl3 not detected at 0x%lx\n", chip->fm_port); + if (snd_opl3_create(card, fm_port[dev], fm_port[dev] + 2, + OPL3_HW_OPL3, 0, &opl3) < 0) { + snd_printk(KERN_WARNING PFX + "opl3 not detected at 0x%lx\n", + fm_port[dev]); } else { - if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) return err; } } if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { - if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ES18XX, - chip->mpu_port, 0, - irq[dev], 0, - &chip->rmidi)) < 0) + err = snd_mpu401_uart_new(card, 0, MPU401_HW_ES18XX, + mpu_port[dev], 0, + irq[dev], 0, &chip->rmidi); + if (err < 0) return err; } diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 02e30d7..b865e45 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -25,6 +25,7 @@ #include <linux/init.h> #include <linux/err.h> #include <linux/isa.h> +#include <linux/pnp.h> #include <linux/delay.h> #include <linux/slab.h> #include <linux/ioport.h> @@ -40,7 +41,7 @@ #define SNDRV_LEGACY_FIND_FREE_IRQ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -#include "miro.h" +#include <sound/aci.h> MODULE_AUTHOR("Martin Langer <martin-langer@gmx.de>"); MODULE_LICENSE("GPL"); @@ -60,6 +61,9 @@ static int dma1 = SNDRV_DEFAULT_DMA1; /* 0,1,3 */ static int dma2 = SNDRV_DEFAULT_DMA1; /* 0,1,3 */ static int wss; static int ide; +#ifdef CONFIG_PNP +static int isapnp = 1; /* Enable ISA PnP detection */ +#endif module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for miro soundcard."); @@ -83,6 +87,10 @@ module_param(wss, int, 0444); MODULE_PARM_DESC(wss, "wss mode"); module_param(ide, int, 0444); MODULE_PARM_DESC(ide, "enable ide port"); +#ifdef CONFIG_PNP +module_param(isapnp, bool, 0444); +MODULE_PARM_DESC(isapnp, "Enable ISA PnP detection for specified soundcard."); +#endif #define OPTi9XX_HW_DETECT 0 #define OPTi9XX_HW_82C928 1 @@ -96,7 +104,6 @@ MODULE_PARM_DESC(ide, "enable ide port"); #define OPTi9XX_MC_REG(n) n - struct snd_miro { unsigned short hardware; unsigned char password; @@ -110,7 +117,6 @@ struct snd_miro { unsigned long pwd_reg; spinlock_t lock; - struct snd_card *card; struct snd_pcm *pcm; long wss_base; @@ -118,42 +124,48 @@ struct snd_miro { int dma1; int dma2; - long fm_port; - long mpu_port; int mpu_irq; - unsigned long aci_port; - int aci_vendor; - int aci_product; - int aci_version; - int aci_amp; - int aci_preamp; - int aci_solomode; - - struct mutex aci_mutex; + struct snd_miro_aci *aci; }; -static void snd_miro_proc_init(struct snd_miro * miro); +static struct snd_miro_aci aci_device; static char * snd_opti9xx_names[] = { - "unkown", + "unknown", "82C928", "82C929", "82C924", "82C925", "82C930", "82C931", "82C933" }; +static int snd_miro_pnp_is_probed; + +#ifdef CONFIG_PNP + +static struct pnp_card_device_id snd_miro_pnpids[] = { + /* PCM20 and PCM12 in PnP mode */ + { .id = "MIR0924", + .devs = { { "MIR0000" }, { "MIR0002" }, { "MIR0005" } }, }, + { .id = "" } +}; + +MODULE_DEVICE_TABLE(pnp_card, snd_miro_pnpids); + +#endif /* CONFIG_PNP */ + /* * ACI control */ -static int aci_busy_wait(struct snd_miro * miro) +static int aci_busy_wait(struct snd_miro_aci *aci) { long timeout; unsigned char byte; - for (timeout = 1; timeout <= ACI_MINTIME+30; timeout++) { - if (((byte=inb(miro->aci_port + ACI_REG_BUSY)) & 1) == 0) { + for (timeout = 1; timeout <= ACI_MINTIME + 30; timeout++) { + byte = inb(aci->aci_port + ACI_REG_BUSY); + if ((byte & 1) == 0) { if (timeout >= ACI_MINTIME) snd_printd("aci ready in round %ld.\n", timeout-ACI_MINTIME); @@ -179,10 +191,10 @@ static int aci_busy_wait(struct snd_miro * miro) return -EBUSY; } -static inline int aci_write(struct snd_miro * miro, unsigned char byte) +static inline int aci_write(struct snd_miro_aci *aci, unsigned char byte) { - if (aci_busy_wait(miro) >= 0) { - outb(byte, miro->aci_port + ACI_REG_COMMAND); + if (aci_busy_wait(aci) >= 0) { + outb(byte, aci->aci_port + ACI_REG_COMMAND); return 0; } else { snd_printk(KERN_ERR "aci busy, aci_write(0x%x) stopped.\n", byte); @@ -190,12 +202,12 @@ static inline int aci_write(struct snd_miro * miro, unsigned char byte) } } -static inline int aci_read(struct snd_miro * miro) +static inline int aci_read(struct snd_miro_aci *aci) { unsigned char byte; - if (aci_busy_wait(miro) >= 0) { - byte=inb(miro->aci_port + ACI_REG_STATUS); + if (aci_busy_wait(aci) >= 0) { + byte = inb(aci->aci_port + ACI_REG_STATUS); return byte; } else { snd_printk(KERN_ERR "aci busy, aci_read() stopped.\n"); @@ -203,39 +215,49 @@ static inline int aci_read(struct snd_miro * miro) } } -static int aci_cmd(struct snd_miro * miro, int write1, int write2, int write3) +int snd_aci_cmd(struct snd_miro_aci *aci, int write1, int write2, int write3) { int write[] = {write1, write2, write3}; int value, i; - if (mutex_lock_interruptible(&miro->aci_mutex)) + if (mutex_lock_interruptible(&aci->aci_mutex)) return -EINTR; for (i=0; i<3; i++) { if (write[i]< 0 || write[i] > 255) break; else { - value = aci_write(miro, write[i]); + value = aci_write(aci, write[i]); if (value < 0) goto out; } } - value = aci_read(miro); + value = aci_read(aci); -out: mutex_unlock(&miro->aci_mutex); +out: mutex_unlock(&aci->aci_mutex); return value; } +EXPORT_SYMBOL(snd_aci_cmd); + +static int aci_getvalue(struct snd_miro_aci *aci, unsigned char index) +{ + return snd_aci_cmd(aci, ACI_STATUS, index, -1); +} -static int aci_getvalue(struct snd_miro * miro, unsigned char index) +static int aci_setvalue(struct snd_miro_aci *aci, unsigned char index, + int value) { - return aci_cmd(miro, ACI_STATUS, index, -1); + return snd_aci_cmd(aci, index, value, -1); } -static int aci_setvalue(struct snd_miro * miro, unsigned char index, int value) +struct snd_miro_aci *snd_aci_get_aci(void) { - return aci_cmd(miro, index, value, -1); + if (aci_device.aci_port == 0) + return NULL; + return &aci_device; } +EXPORT_SYMBOL(snd_aci_get_aci); /* * MIXER part @@ -249,8 +271,10 @@ static int snd_miro_get_capture(struct snd_kcontrol *kcontrol, struct snd_miro *miro = snd_kcontrol_chip(kcontrol); int value; - if ((value = aci_getvalue(miro, ACI_S_GENERAL)) < 0) { - snd_printk(KERN_ERR "snd_miro_get_capture() failed: %d\n", value); + value = aci_getvalue(miro->aci, ACI_S_GENERAL); + if (value < 0) { + snd_printk(KERN_ERR "snd_miro_get_capture() failed: %d\n", + value); return value; } @@ -267,13 +291,15 @@ static int snd_miro_put_capture(struct snd_kcontrol *kcontrol, value = !(ucontrol->value.integer.value[0]); - if ((error = aci_setvalue(miro, ACI_SET_SOLOMODE, value)) < 0) { - snd_printk(KERN_ERR "snd_miro_put_capture() failed: %d\n", error); + error = aci_setvalue(miro->aci, ACI_SET_SOLOMODE, value); + if (error < 0) { + snd_printk(KERN_ERR "snd_miro_put_capture() failed: %d\n", + error); return error; } - change = (value != miro->aci_solomode); - miro->aci_solomode = value; + change = (value != miro->aci->aci_solomode); + miro->aci->aci_solomode = value; return change; } @@ -295,7 +321,7 @@ static int snd_miro_get_preamp(struct snd_kcontrol *kcontrol, struct snd_miro *miro = snd_kcontrol_chip(kcontrol); int value; - if (miro->aci_version <= 176) { + if (miro->aci->aci_version <= 176) { /* OSS says it's not readable with versions < 176. @@ -303,12 +329,14 @@ static int snd_miro_get_preamp(struct snd_kcontrol *kcontrol, which is a PCM12 with aci_version = 176. */ - ucontrol->value.integer.value[0] = miro->aci_preamp; + ucontrol->value.integer.value[0] = miro->aci->aci_preamp; return 0; } - if ((value = aci_getvalue(miro, ACI_GET_PREAMP)) < 0) { - snd_printk(KERN_ERR "snd_miro_get_preamp() failed: %d\n", value); + value = aci_getvalue(miro->aci, ACI_GET_PREAMP); + if (value < 0) { + snd_printk(KERN_ERR "snd_miro_get_preamp() failed: %d\n", + value); return value; } @@ -325,13 +353,15 @@ static int snd_miro_put_preamp(struct snd_kcontrol *kcontrol, value = ucontrol->value.integer.value[0]; - if ((error = aci_setvalue(miro, ACI_SET_PREAMP, value)) < 0) { - snd_printk(KERN_ERR "snd_miro_put_preamp() failed: %d\n", error); + error = aci_setvalue(miro->aci, ACI_SET_PREAMP, value); + if (error < 0) { + snd_printk(KERN_ERR "snd_miro_put_preamp() failed: %d\n", + error); return error; } - change = (value != miro->aci_preamp); - miro->aci_preamp = value; + change = (value != miro->aci->aci_preamp); + miro->aci->aci_preamp = value; return change; } @@ -342,7 +372,7 @@ static int snd_miro_get_amp(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_miro *miro = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = miro->aci_amp; + ucontrol->value.integer.value[0] = miro->aci->aci_amp; return 0; } @@ -355,13 +385,14 @@ static int snd_miro_put_amp(struct snd_kcontrol *kcontrol, value = ucontrol->value.integer.value[0]; - if ((error = aci_setvalue(miro, ACI_SET_POWERAMP, value)) < 0) { + error = aci_setvalue(miro->aci, ACI_SET_POWERAMP, value); + if (error < 0) { snd_printk(KERN_ERR "snd_miro_put_amp() to %d failed: %d\n", value, error); return error; } - change = (value != miro->aci_amp); - miro->aci_amp = value; + change = (value != miro->aci->aci_amp); + miro->aci->aci_amp = value; return change; } @@ -410,12 +441,14 @@ static int snd_miro_get_double(struct snd_kcontrol *kcontrol, int right_reg = kcontrol->private_value & 0xff; int left_reg = right_reg + 1; - if ((right_val = aci_getvalue(miro, right_reg)) < 0) { + right_val = aci_getvalue(miro->aci, right_reg); + if (right_val < 0) { snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", right_reg, right_val); return right_val; } - if ((left_val = aci_getvalue(miro, left_reg)) < 0) { + left_val = aci_getvalue(miro->aci, left_reg); + if (left_val < 0) { snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", left_reg, left_val); return left_val; } @@ -451,6 +484,7 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_miro *miro = snd_kcontrol_chip(kcontrol); + struct snd_miro_aci *aci = miro->aci; int left, right, left_old, right_old; int setreg_left, setreg_right, getreg_left, getreg_right; int change, error; @@ -459,21 +493,21 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, right = ucontrol->value.integer.value[1]; setreg_right = (kcontrol->private_value >> 8) & 0xff; - if (setreg_right == ACI_SET_MASTER) { - setreg_left = setreg_right + 1; - } else { - setreg_left = setreg_right + 8; - } + setreg_left = setreg_right + 8; + if (setreg_right == ACI_SET_MASTER) + setreg_left -= 7; getreg_right = kcontrol->private_value & 0xff; getreg_left = getreg_right + 1; - if ((left_old = aci_getvalue(miro, getreg_left)) < 0) { + left_old = aci_getvalue(aci, getreg_left); + if (left_old < 0) { snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", getreg_left, left_old); return left_old; } - if ((right_old = aci_getvalue(miro, getreg_right)) < 0) { + right_old = aci_getvalue(aci, getreg_right); + if (right_old < 0) { snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", getreg_right, right_old); return right_old; } @@ -492,13 +526,15 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, right_old = 0x80 - right_old; if (left >= 0) { - if ((error = aci_setvalue(miro, setreg_left, left)) < 0) { + error = aci_setvalue(aci, setreg_left, left); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", left, error); return error; } } else { - if ((error = aci_setvalue(miro, setreg_left, 0x80 - left)) < 0) { + error = aci_setvalue(aci, setreg_left, 0x80 - left); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", 0x80 - left, error); return error; @@ -506,13 +542,15 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, } if (right >= 0) { - if ((error = aci_setvalue(miro, setreg_right, right)) < 0) { + error = aci_setvalue(aci, setreg_right, right); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", right, error); return error; } } else { - if ((error = aci_setvalue(miro, setreg_right, 0x80 - right)) < 0) { + error = aci_setvalue(aci, setreg_right, 0x80 - right); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", 0x80 - right, error); return error; @@ -530,12 +568,14 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, left_old = 0x20 - left_old; right_old = 0x20 - right_old; - if ((error = aci_setvalue(miro, setreg_left, 0x20 - left)) < 0) { + error = aci_setvalue(aci, setreg_left, 0x20 - left); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", 0x20 - left, error); return error; } - if ((error = aci_setvalue(miro, setreg_right, 0x20 - right)) < 0) { + error = aci_setvalue(aci, setreg_right, 0x20 - right); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", 0x20 - right, error); return error; @@ -633,11 +673,13 @@ static unsigned char aci_init_values[][2] __devinitdata = { static int __devinit snd_set_aci_init_values(struct snd_miro *miro) { int idx, error; + struct snd_miro_aci *aci = miro->aci; /* enable WSS on PCM1 */ - if ((miro->aci_product == 'A') && wss) { - if ((error = aci_setvalue(miro, ACI_SET_WSS, wss)) < 0) { + if ((aci->aci_product == 'A') && wss) { + error = aci_setvalue(aci, ACI_SET_WSS, wss); + if (error < 0) { snd_printk(KERN_ERR "enabling WSS mode failed\n"); return error; } @@ -646,7 +688,8 @@ static int __devinit snd_set_aci_init_values(struct snd_miro *miro) /* enable IDE port */ if (ide) { - if ((error = aci_setvalue(miro, ACI_SET_IDE, ide)) < 0) { + error = aci_setvalue(aci, ACI_SET_IDE, ide); + if (error < 0) { snd_printk(KERN_ERR "enabling IDE port failed\n"); return error; } @@ -654,32 +697,31 @@ static int __devinit snd_set_aci_init_values(struct snd_miro *miro) /* set common aci values */ - for (idx = 0; idx < ARRAY_SIZE(aci_init_values); idx++) - if ((error = aci_setvalue(miro, aci_init_values[idx][0], - aci_init_values[idx][1])) < 0) { + for (idx = 0; idx < ARRAY_SIZE(aci_init_values); idx++) { + error = aci_setvalue(aci, aci_init_values[idx][0], + aci_init_values[idx][1]); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", aci_init_values[idx][0], error); return error; } - - miro->aci_amp = 0; - miro->aci_preamp = 0; - miro->aci_solomode = 1; + } + aci->aci_amp = 0; + aci->aci_preamp = 0; + aci->aci_solomode = 1; return 0; } -static int __devinit snd_miro_mixer(struct snd_miro *miro) +static int __devinit snd_miro_mixer(struct snd_card *card, + struct snd_miro *miro) { - struct snd_card *card; unsigned int idx; int err; - if (snd_BUG_ON(!miro || !miro->card)) + if (snd_BUG_ON(!miro || !card)) return -EINVAL; - card = miro->card; - switch (miro->hardware) { case OPTi9XX_HW_82C924: strcpy(card->mixername, "ACI & OPTi924"); @@ -697,7 +739,8 @@ static int __devinit snd_miro_mixer(struct snd_miro *miro) return err; } - if ((miro->aci_product == 'A') || (miro->aci_product == 'B')) { + if ((miro->aci->aci_product == 'A') || + (miro->aci->aci_product == 'B')) { /* PCM1/PCM12 with power-amp and Line 2 */ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_line_control[0], miro))) < 0) return err; @@ -705,16 +748,17 @@ static int __devinit snd_miro_mixer(struct snd_miro *miro) return err; } - if ((miro->aci_product == 'B') || (miro->aci_product == 'C')) { + if ((miro->aci->aci_product == 'B') || + (miro->aci->aci_product == 'C')) { /* PCM12/PCM20 with mic-preamp */ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_preamp_control[0], miro))) < 0) return err; - if (miro->aci_version >= 176) + if (miro->aci->aci_version >= 176) if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_capture_control[0], miro))) < 0) return err; } - if (miro->aci_product == 'C') { + if (miro->aci->aci_product == 'C') { /* PCM20 with radio and 7 band equalizer */ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_radio_control[0], miro))) < 0) return err; @@ -757,21 +801,26 @@ static int __devinit snd_miro_init(struct snd_miro *chip, chip->irq = -1; chip->dma1 = -1; chip->dma2 = -1; - chip->fm_port = -1; chip->mpu_port = -1; chip->mpu_irq = -1; + chip->pwd_reg = 3; + +#ifdef CONFIG_PNP + if (isapnp && chip->mc_base) + /* PnP resource gives the least 10 bits */ + chip->mc_base |= 0xc00; + else +#endif + chip->mc_base = 0xf8c; + switch (hardware) { case OPTi9XX_HW_82C929: - chip->mc_base = 0xf8c; chip->password = 0xe3; - chip->pwd_reg = 3; break; case OPTi9XX_HW_82C924: - chip->mc_base = 0xf8c; chip->password = 0xe5; - chip->pwd_reg = 3; break; default: @@ -853,14 +902,15 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, struct snd_info_buffer *buffer) { struct snd_miro *miro = (struct snd_miro *) entry->private_data; + struct snd_miro_aci *aci = miro->aci; char* model = "unknown"; /* miroSOUND PCM1 pro, early PCM12 */ if ((miro->hardware == OPTi9XX_HW_82C929) && - (miro->aci_vendor == 'm') && - (miro->aci_product == 'A')) { - switch(miro->aci_version) { + (aci->aci_vendor == 'm') && + (aci->aci_product == 'A')) { + switch (aci->aci_version) { case 3: model = "miroSOUND PCM1 pro"; break; @@ -873,9 +923,9 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, /* miroSOUND PCM12, PCM12 (Rev. E), PCM12 pnp */ if ((miro->hardware == OPTi9XX_HW_82C924) && - (miro->aci_vendor == 'm') && - (miro->aci_product == 'B')) { - switch(miro->aci_version) { + (aci->aci_vendor == 'm') && + (aci->aci_product == 'B')) { + switch (aci->aci_version) { case 4: model = "miroSOUND PCM12"; break; @@ -891,9 +941,9 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, /* miroSOUND PCM20 radio */ if ((miro->hardware == OPTi9XX_HW_82C924) && - (miro->aci_vendor == 'm') && - (miro->aci_product == 'C')) { - switch(miro->aci_version) { + (aci->aci_vendor == 'm') && + (aci->aci_product == 'C')) { + switch (aci->aci_version) { case 7: model = "miroSOUND PCM20 radio (Rev. E)"; break; @@ -917,17 +967,17 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, snd_iprintf(buffer, "ACI information:\n"); snd_iprintf(buffer, " vendor : "); - switch(miro->aci_vendor) { + switch (aci->aci_vendor) { case 'm': snd_iprintf(buffer, "Miro\n"); break; default: - snd_iprintf(buffer, "unknown (0x%x)\n", miro->aci_vendor); + snd_iprintf(buffer, "unknown (0x%x)\n", aci->aci_vendor); break; } snd_iprintf(buffer, " product : "); - switch(miro->aci_product) { + switch (aci->aci_product) { case 'A': snd_iprintf(buffer, "miroSOUND PCM1 pro / (early) PCM12\n"); break; @@ -938,26 +988,27 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, snd_iprintf(buffer, "miroSOUND PCM20 radio\n"); break; default: - snd_iprintf(buffer, "unknown (0x%x)\n", miro->aci_product); + snd_iprintf(buffer, "unknown (0x%x)\n", aci->aci_product); break; } snd_iprintf(buffer, " firmware: %d (0x%x)\n", - miro->aci_version, miro->aci_version); + aci->aci_version, aci->aci_version); snd_iprintf(buffer, " port : 0x%lx-0x%lx\n", - miro->aci_port, miro->aci_port+2); + aci->aci_port, aci->aci_port+2); snd_iprintf(buffer, " wss : 0x%x\n", wss); snd_iprintf(buffer, " ide : 0x%x\n", ide); - snd_iprintf(buffer, " solomode: 0x%x\n", miro->aci_solomode); - snd_iprintf(buffer, " amp : 0x%x\n", miro->aci_amp); - snd_iprintf(buffer, " preamp : 0x%x\n", miro->aci_preamp); + snd_iprintf(buffer, " solomode: 0x%x\n", aci->aci_solomode); + snd_iprintf(buffer, " amp : 0x%x\n", aci->aci_amp); + snd_iprintf(buffer, " preamp : 0x%x\n", aci->aci_preamp); } -static void __devinit snd_miro_proc_init(struct snd_miro * miro) +static void __devinit snd_miro_proc_init(struct snd_card *card, + struct snd_miro *miro) { struct snd_info_entry *entry; - if (! snd_card_proc_new(miro->card, "miro", &entry)) + if (!snd_card_proc_new(card, "miro", &entry)) snd_info_set_text_ops(entry, miro, snd_miro_proc_read); } @@ -974,37 +1025,40 @@ static int __devinit snd_miro_configure(struct snd_miro *chip) unsigned char mpu_irq_bits; unsigned long flags; + snd_miro_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80); + snd_miro_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); /* OPL4 */ + snd_miro_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02); + switch (chip->hardware) { case OPTi9XX_HW_82C924: snd_miro_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); /* OPL4 */ snd_miro_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02); break; case OPTi9XX_HW_82C929: /* untested init commands for OPTi929 */ - snd_miro_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); /* OPL4 */ snd_miro_write_mask(chip, OPTi9XX_MC_REG(4), 0x00, 0x0c); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02); break; default: snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware); return -EINVAL; } - switch (chip->wss_base) { - case 0x530: + /* PnP resource says it decodes only 10 bits of address */ + switch (chip->wss_base & 0x3ff) { + case 0x130: + chip->wss_base = 0x530; wss_base_bits = 0x00; break; - case 0x604: + case 0x204: + chip->wss_base = 0x604; wss_base_bits = 0x03; break; - case 0xe80: + case 0x280: + chip->wss_base = 0xe80; wss_base_bits = 0x01; break; - case 0xf40: + case 0x340: + chip->wss_base = 0xf40; wss_base_bits = 0x02; break; default: @@ -1122,75 +1176,92 @@ __skip_mpu: return 0; } +static int __devinit snd_miro_opti_check(struct snd_miro *chip) +{ + unsigned char value; + + chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, + "OPTi9xx MC"); + if (chip->res_mc_base == NULL) + return -ENOMEM; + + value = snd_miro_read(chip, OPTi9XX_MC_REG(1)); + if (value != 0xff && value != inb(chip->mc_base + OPTi9XX_MC_REG(1))) + if (value == snd_miro_read(chip, OPTi9XX_MC_REG(1))) + return 0; + + release_and_free_resource(chip->res_mc_base); + chip->res_mc_base = NULL; + + return -ENODEV; +} + static int __devinit snd_card_miro_detect(struct snd_card *card, struct snd_miro *chip) { int i, err; - unsigned char value; for (i = OPTi9XX_HW_82C929; i <= OPTi9XX_HW_82C924; i++) { if ((err = snd_miro_init(chip, i)) < 0) return err; - if ((chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, "OPTi9xx MC")) == NULL) - continue; - - value = snd_miro_read(chip, OPTi9XX_MC_REG(1)); - if ((value != 0xff) && (value != inb(chip->mc_base + 1))) - if (value == snd_miro_read(chip, OPTi9XX_MC_REG(1))) - return 1; - - release_and_free_resource(chip->res_mc_base); - chip->res_mc_base = NULL; - + err = snd_miro_opti_check(chip); + if (err == 0) + return 1; } return -ENODEV; } static int __devinit snd_card_miro_aci_detect(struct snd_card *card, - struct snd_miro * miro) + struct snd_miro *miro) { unsigned char regval; int i; + struct snd_miro_aci *aci = &aci_device; + + miro->aci = aci; - mutex_init(&miro->aci_mutex); + mutex_init(&aci->aci_mutex); /* get ACI port from OPTi9xx MC 4 */ - miro->mc_base = 0xf8c; regval=inb(miro->mc_base + 4); - miro->aci_port = (regval & 0x10) ? 0x344: 0x354; + aci->aci_port = (regval & 0x10) ? 0x344 : 0x354; - if ((miro->res_aci_port = request_region(miro->aci_port, 3, "miro aci")) == NULL) { + miro->res_aci_port = request_region(aci->aci_port, 3, "miro aci"); + if (miro->res_aci_port == NULL) { snd_printk(KERN_ERR "aci i/o area 0x%lx-0x%lx already used.\n", - miro->aci_port, miro->aci_port+2); + aci->aci_port, aci->aci_port+2); return -ENOMEM; } /* force ACI into a known state */ for (i = 0; i < 3; i++) - if (aci_cmd(miro, ACI_ERROR_OP, -1, -1) < 0) { + if (snd_aci_cmd(aci, ACI_ERROR_OP, -1, -1) < 0) { snd_printk(KERN_ERR "can't force aci into known state.\n"); return -ENXIO; } - if ((miro->aci_vendor=aci_cmd(miro, ACI_READ_IDCODE, -1, -1)) < 0 || - (miro->aci_product=aci_cmd(miro, ACI_READ_IDCODE, -1, -1)) < 0) { - snd_printk(KERN_ERR "can't read aci id on 0x%lx.\n", miro->aci_port); + aci->aci_vendor = snd_aci_cmd(aci, ACI_READ_IDCODE, -1, -1); + aci->aci_product = snd_aci_cmd(aci, ACI_READ_IDCODE, -1, -1); + if (aci->aci_vendor < 0 || aci->aci_product < 0) { + snd_printk(KERN_ERR "can't read aci id on 0x%lx.\n", + aci->aci_port); return -ENXIO; } - if ((miro->aci_version=aci_cmd(miro, ACI_READ_VERSION, -1, -1)) < 0) { + aci->aci_version = snd_aci_cmd(aci, ACI_READ_VERSION, -1, -1); + if (aci->aci_version < 0) { snd_printk(KERN_ERR "can't read aci version on 0x%lx.\n", - miro->aci_port); + aci->aci_port); return -ENXIO; } - if (aci_cmd(miro, ACI_INIT, -1, -1) < 0 || - aci_cmd(miro, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0 || - aci_cmd(miro, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0) { + if (snd_aci_cmd(aci, ACI_INIT, -1, -1) < 0 || + snd_aci_cmd(aci, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0 || + snd_aci_cmd(aci, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0) { snd_printk(KERN_ERR "can't initialize aci.\n"); return -ENXIO; } @@ -1201,157 +1272,80 @@ static int __devinit snd_card_miro_aci_detect(struct snd_card *card, static void snd_card_miro_free(struct snd_card *card) { struct snd_miro *miro = card->private_data; - + release_and_free_resource(miro->res_aci_port); + if (miro->aci) + miro->aci->aci_port = 0; release_and_free_resource(miro->res_mc_base); } -static int __devinit snd_miro_match(struct device *devptr, unsigned int n) -{ - return 1; -} - -static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) +static int __devinit snd_miro_probe(struct snd_card *card) { - static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; - static long possible_mpu_ports[] = {0x330, 0x300, 0x310, 0x320, -1}; - static int possible_irqs[] = {11, 9, 10, 7, -1}; - static int possible_mpu_irqs[] = {10, 5, 9, 7, -1}; - static int possible_dma1s[] = {3, 1, 0, -1}; - static int possible_dma2s[][2] = {{1,-1}, {0,-1}, {-1,-1}, {0,-1}}; - int error; - struct snd_miro *miro; + struct snd_miro *miro = card->private_data; struct snd_wss *codec; struct snd_timer *timer; - struct snd_card *card; struct snd_pcm *pcm; struct snd_rawmidi *rmidi; - error = snd_card_create(index, id, THIS_MODULE, - sizeof(struct snd_miro), &card); - if (error < 0) - return error; - - card->private_free = snd_card_miro_free; - miro = card->private_data; - miro->card = card; - - if ((error = snd_card_miro_aci_detect(card, miro)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to detect aci chip\n"); - return -ENODEV; + if (!miro->res_mc_base) { + miro->res_mc_base = request_region(miro->mc_base, + miro->mc_base_size, + "miro (OPTi9xx MC)"); + if (miro->res_mc_base == NULL) { + snd_printk(KERN_ERR "request for OPTI9xx MC failed\n"); + return -ENOMEM; + } } - /* init proc interface */ - snd_miro_proc_init(miro); - - if ((error = snd_card_miro_detect(card, miro)) < 0) { + error = snd_card_miro_aci_detect(card, miro); + if (error < 0) { snd_card_free(card); - snd_printk(KERN_ERR "unable to detect OPTi9xx chip\n"); + snd_printk(KERN_ERR "unable to detect aci chip\n"); return -ENODEV; } - if (! miro->res_mc_base && - (miro->res_mc_base = request_region(miro->mc_base, miro->mc_base_size, - "miro (OPTi9xx MC)")) == NULL) { - snd_card_free(card); - snd_printk(KERN_ERR "request for OPTI9xx MC failed\n"); - return -ENOMEM; - } - miro->wss_base = port; - miro->fm_port = fm_port; miro->mpu_port = mpu_port; miro->irq = irq; miro->mpu_irq = mpu_irq; miro->dma1 = dma1; miro->dma2 = dma2; - if (miro->wss_base == SNDRV_AUTO_PORT) { - if ((miro->wss_base = snd_legacy_find_free_ioport(possible_ports, 4)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free WSS port\n"); - return -EBUSY; - } - } - - if (miro->mpu_port == SNDRV_AUTO_PORT) { - if ((miro->mpu_port = snd_legacy_find_free_ioport(possible_mpu_ports, 2)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free MPU401 port\n"); - return -EBUSY; - } - } - if (miro->irq == SNDRV_AUTO_IRQ) { - if ((miro->irq = snd_legacy_find_free_irq(possible_irqs)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free IRQ\n"); - return -EBUSY; - } - } - if (miro->mpu_irq == SNDRV_AUTO_IRQ) { - if ((miro->mpu_irq = snd_legacy_find_free_irq(possible_mpu_irqs)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free MPU401 IRQ\n"); - return -EBUSY; - } - } - if (miro->dma1 == SNDRV_AUTO_DMA) { - if ((miro->dma1 = snd_legacy_find_free_dma(possible_dma1s)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free DMA1\n"); - return -EBUSY; - } - } - if (miro->dma2 == SNDRV_AUTO_DMA) { - if ((miro->dma2 = snd_legacy_find_free_dma(possible_dma2s[miro->dma1 % 4])) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free DMA2\n"); - return -EBUSY; - } - } + /* init proc interface */ + snd_miro_proc_init(card, miro); error = snd_miro_configure(miro); - if (error) { - snd_card_free(card); + if (error) return error; - } error = snd_wss_create(card, miro->wss_base + 4, -1, - miro->irq, miro->dma1, miro->dma2, - WSS_HW_AD1845, 0, &codec); - if (error < 0) { - snd_card_free(card); + miro->irq, miro->dma1, miro->dma2, + WSS_HW_DETECT, 0, &codec); + if (error < 0) return error; - } error = snd_wss_pcm(codec, 0, &pcm); - if (error < 0) { - snd_card_free(card); + if (error < 0) return error; - } + error = snd_wss_mixer(codec); - if (error < 0) { - snd_card_free(card); + if (error < 0) return error; - } + error = snd_wss_timer(codec, 0, &timer); - if (error < 0) { - snd_card_free(card); + if (error < 0) return error; - } miro->pcm = pcm; - if ((error = snd_miro_mixer(miro)) < 0) { - snd_card_free(card); + error = snd_miro_mixer(card, miro); + if (error < 0) return error; - } - if (miro->aci_vendor == 'm') { + if (miro->aci->aci_vendor == 'm') { /* It looks like a miro sound card. */ - switch (miro->aci_product) { + switch (miro->aci->aci_product) { case 'A': sprintf(card->shortname, "miroSOUND PCM1 pro / PCM12"); @@ -1380,30 +1374,131 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) card->shortname, miro->name, pcm->name, miro->wss_base + 4, miro->irq, miro->dma1, miro->dma2); - if (miro->mpu_port <= 0 || miro->mpu_port == SNDRV_AUTO_PORT) + if (mpu_port <= 0 || mpu_port == SNDRV_AUTO_PORT) rmidi = NULL; - else - if ((error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - miro->mpu_port, 0, miro->mpu_irq, IRQF_DISABLED, - &rmidi))) - snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", miro->mpu_port); + else { + error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpu_port, 0, miro->mpu_irq, IRQF_DISABLED, + &rmidi); + if (error < 0) + snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", + mpu_port); + } - if (miro->fm_port > 0 && miro->fm_port != SNDRV_AUTO_PORT) { + if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) { struct snd_opl3 *opl3 = NULL; struct snd_opl4 *opl4; - if (snd_opl4_create(card, miro->fm_port, miro->fm_port - 8, + + if (snd_opl4_create(card, fm_port, fm_port - 8, 2, &opl3, &opl4) < 0) - snd_printk(KERN_WARNING "no OPL4 device at 0x%lx\n", miro->fm_port); + snd_printk(KERN_WARNING "no OPL4 device at 0x%lx\n", + fm_port); } - if ((error = snd_set_aci_init_values(miro)) < 0) { - snd_card_free(card); + error = snd_set_aci_init_values(miro); + if (error < 0) return error; + + return snd_card_register(card); +} + +static int __devinit snd_miro_isa_match(struct device *devptr, unsigned int n) +{ +#ifdef CONFIG_PNP + if (snd_miro_pnp_is_probed) + return 0; + if (isapnp) + return 0; +#endif + return 1; +} + +static int __devinit snd_miro_isa_probe(struct device *devptr, unsigned int n) +{ + static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; + static long possible_mpu_ports[] = {0x330, 0x300, 0x310, 0x320, -1}; + static int possible_irqs[] = {11, 9, 10, 7, -1}; + static int possible_mpu_irqs[] = {10, 5, 9, 7, -1}; + static int possible_dma1s[] = {3, 1, 0, -1}; + static int possible_dma2s[][2] = { {1, -1}, {0, -1}, {-1, -1}, + {0, -1} }; + + int error; + struct snd_miro *miro; + struct snd_card *card; + + error = snd_card_create(index, id, THIS_MODULE, + sizeof(struct snd_miro), &card); + if (error < 0) + return error; + + card->private_free = snd_card_miro_free; + miro = card->private_data; + + error = snd_card_miro_detect(card, miro); + if (error < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to detect OPTi9xx chip\n"); + return -ENODEV; + } + + if (port == SNDRV_AUTO_PORT) { + port = snd_legacy_find_free_ioport(possible_ports, 4); + if (port < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to find a free WSS port\n"); + return -EBUSY; + } + } + + if (mpu_port == SNDRV_AUTO_PORT) { + mpu_port = snd_legacy_find_free_ioport(possible_mpu_ports, 2); + if (mpu_port < 0) { + snd_card_free(card); + snd_printk(KERN_ERR + "unable to find a free MPU401 port\n"); + return -EBUSY; + } + } + + if (irq == SNDRV_AUTO_IRQ) { + irq = snd_legacy_find_free_irq(possible_irqs); + if (irq < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to find a free IRQ\n"); + return -EBUSY; + } + } + if (mpu_irq == SNDRV_AUTO_IRQ) { + mpu_irq = snd_legacy_find_free_irq(possible_mpu_irqs); + if (mpu_irq < 0) { + snd_card_free(card); + snd_printk(KERN_ERR + "unable to find a free MPU401 IRQ\n"); + return -EBUSY; + } + } + if (dma1 == SNDRV_AUTO_DMA) { + dma1 = snd_legacy_find_free_dma(possible_dma1s); + if (dma1 < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to find a free DMA1\n"); + return -EBUSY; + } + } + if (dma2 == SNDRV_AUTO_DMA) { + dma2 = snd_legacy_find_free_dma(possible_dma2s[dma1 % 4]); + if (dma2 < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to find a free DMA2\n"); + return -EBUSY; + } } snd_card_set_dev(card, devptr); - if ((error = snd_card_register(card))) { + error = snd_miro_probe(card); + if (error < 0) { snd_card_free(card); return error; } @@ -1412,7 +1507,8 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) return 0; } -static int __devexit snd_miro_remove(struct device *devptr, unsigned int dev) +static int __devexit snd_miro_isa_remove(struct device *devptr, + unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); dev_set_drvdata(devptr, NULL); @@ -1422,23 +1518,164 @@ static int __devexit snd_miro_remove(struct device *devptr, unsigned int dev) #define DEV_NAME "miro" static struct isa_driver snd_miro_driver = { - .match = snd_miro_match, - .probe = snd_miro_probe, - .remove = __devexit_p(snd_miro_remove), + .match = snd_miro_isa_match, + .probe = snd_miro_isa_probe, + .remove = __devexit_p(snd_miro_isa_remove), /* FIXME: suspend/resume */ .driver = { .name = DEV_NAME }, }; +#ifdef CONFIG_PNP + +static int __devinit snd_card_miro_pnp(struct snd_miro *chip, + struct pnp_card_link *card, + const struct pnp_card_device_id *pid) +{ + struct pnp_dev *pdev; + int err; + struct pnp_dev *devmpu; + struct pnp_dev *devmc; + + pdev = pnp_request_card_device(card, pid->devs[0].id, NULL); + if (pdev == NULL) + return -EBUSY; + + devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); + if (devmpu == NULL) + return -EBUSY; + + devmc = pnp_request_card_device(card, pid->devs[2].id, NULL); + if (devmc == NULL) + return -EBUSY; + + err = pnp_activate_dev(pdev); + if (err < 0) { + snd_printk(KERN_ERR "AUDIO pnp configure failure: %d\n", err); + return err; + } + + err = pnp_activate_dev(devmc); + if (err < 0) { + snd_printk(KERN_ERR "OPL syntg pnp configure failure: %d\n", + err); + return err; + } + + port = pnp_port_start(pdev, 1); + fm_port = pnp_port_start(pdev, 2) + 8; + + /* + * The MC(0) is never accessed and the miroSOUND PCM20 card does not + * include it in the PnP resource range. OPTI93x include it. + */ + chip->mc_base = pnp_port_start(devmc, 0) - 1; + chip->mc_base_size = pnp_port_len(devmc, 0) + 1; + + irq = pnp_irq(pdev, 0); + dma1 = pnp_dma(pdev, 0); + dma2 = pnp_dma(pdev, 1); + + if (mpu_port > 0) { + err = pnp_activate_dev(devmpu); + if (err < 0) { + snd_printk(KERN_ERR "MPU401 pnp configure failure\n"); + mpu_port = -1; + return err; + } + mpu_port = pnp_port_start(devmpu, 0); + mpu_irq = pnp_irq(devmpu, 0); + } + return 0; +} + +static int __devinit snd_miro_pnp_probe(struct pnp_card_link *pcard, + const struct pnp_card_device_id *pid) +{ + struct snd_card *card; + int err; + struct snd_miro *miro; + + if (snd_miro_pnp_is_probed) + return -EBUSY; + if (!isapnp) + return -ENODEV; + err = snd_card_create(index, id, THIS_MODULE, + sizeof(struct snd_miro), &card); + if (err < 0) + return err; + + card->private_free = snd_card_miro_free; + miro = card->private_data; + + err = snd_card_miro_pnp(miro, pcard, pid); + if (err) { + snd_card_free(card); + return err; + } + + /* only miroSOUND PCM20 and PCM12 == OPTi924 */ + err = snd_miro_init(miro, OPTi9XX_HW_82C924); + if (err) { + snd_card_free(card); + return err; + } + + err = snd_miro_opti_check(miro); + if (err) { + snd_printk(KERN_ERR "OPTI chip not found\n"); + snd_card_free(card); + return err; + } + + snd_card_set_dev(card, &pcard->card->dev); + err = snd_miro_probe(card); + if (err < 0) { + snd_card_free(card); + return err; + } + pnp_set_card_drvdata(pcard, card); + snd_miro_pnp_is_probed = 1; + return 0; +} + +static void __devexit snd_miro_pnp_remove(struct pnp_card_link * pcard) +{ + snd_card_free(pnp_get_card_drvdata(pcard)); + pnp_set_card_drvdata(pcard, NULL); + snd_miro_pnp_is_probed = 0; +} + +static struct pnp_card_driver miro_pnpc_driver = { + .flags = PNP_DRIVER_RES_DISABLE, + .name = "miro", + .id_table = snd_miro_pnpids, + .probe = snd_miro_pnp_probe, + .remove = __devexit_p(snd_miro_pnp_remove), +}; +#endif + static int __init alsa_card_miro_init(void) { +#ifdef CONFIG_PNP + pnp_register_card_driver(&miro_pnpc_driver); + if (snd_miro_pnp_is_probed) + return 0; + pnp_unregister_card_driver(&miro_pnpc_driver); +#endif return isa_register_driver(&snd_miro_driver, 1); } static void __exit alsa_card_miro_exit(void) { - isa_unregister_driver(&snd_miro_driver); + if (!snd_miro_pnp_is_probed) { + isa_unregister_driver(&snd_miro_driver); + return; + } +#ifdef CONFIG_PNP + pnp_unregister_card_driver(&miro_pnpc_driver); +#endif } module_init(alsa_card_miro_init) diff --git a/sound/isa/opti9xx/miro.h b/sound/isa/opti9xx/miro.h deleted file mode 100644 index 6e1385b..0000000 --- a/sound/isa/opti9xx/miro.h +++ /dev/null @@ -1,73 +0,0 @@ -#ifndef _MIRO_H_ -#define _MIRO_H_ - -#define ACI_REG_COMMAND 0 /* write register offset */ -#define ACI_REG_STATUS 1 /* read register offset */ -#define ACI_REG_BUSY 2 /* busy register offset */ -#define ACI_REG_RDS 2 /* PCM20: RDS register offset */ -#define ACI_MINTIME 500 /* ACI time out limit */ - -#define ACI_SET_MUTE 0x0d -#define ACI_SET_POWERAMP 0x0f -#define ACI_SET_TUNERMUTE 0xa3 -#define ACI_SET_TUNERMONO 0xa4 -#define ACI_SET_IDE 0xd0 -#define ACI_SET_WSS 0xd1 -#define ACI_SET_SOLOMODE 0xd2 -#define ACI_SET_PREAMP 0x03 -#define ACI_GET_PREAMP 0x21 -#define ACI_WRITE_TUNE 0xa7 -#define ACI_READ_TUNERSTEREO 0xa8 -#define ACI_READ_TUNERSTATION 0xa9 -#define ACI_READ_VERSION 0xf1 -#define ACI_READ_IDCODE 0xf2 -#define ACI_INIT 0xff -#define ACI_STATUS 0xf0 -#define ACI_S_GENERAL 0x00 -#define ACI_ERROR_OP 0xdf - -/* ACI Mixer */ - -/* These are the values for the right channel GET registers. - Add an offset of 0x01 for the left channel register. - (left=right+0x01) */ - -#define ACI_GET_MASTER 0x03 -#define ACI_GET_MIC 0x05 -#define ACI_GET_LINE 0x07 -#define ACI_GET_CD 0x09 -#define ACI_GET_SYNTH 0x0b -#define ACI_GET_PCM 0x0d -#define ACI_GET_LINE1 0x10 /* Radio on PCM20 */ -#define ACI_GET_LINE2 0x12 - -#define ACI_GET_EQ1 0x22 /* from Bass ... */ -#define ACI_GET_EQ2 0x24 -#define ACI_GET_EQ3 0x26 -#define ACI_GET_EQ4 0x28 -#define ACI_GET_EQ5 0x2a -#define ACI_GET_EQ6 0x2c -#define ACI_GET_EQ7 0x2e /* ... to Treble */ - -/* And these are the values for the right channel SET registers. - For left channel access you have to add an offset of 0x08. - MASTER is an exception, which needs an offset of 0x01 */ - -#define ACI_SET_MASTER 0x00 -#define ACI_SET_MIC 0x30 -#define ACI_SET_LINE 0x31 -#define ACI_SET_CD 0x34 -#define ACI_SET_SYNTH 0x33 -#define ACI_SET_PCM 0x32 -#define ACI_SET_LINE1 0x35 /* Radio on PCM20 */ -#define ACI_SET_LINE2 0x36 - -#define ACI_SET_EQ1 0x40 /* from Bass ... */ -#define ACI_SET_EQ2 0x41 -#define ACI_SET_EQ3 0x42 -#define ACI_SET_EQ4 0x43 -#define ACI_SET_EQ5 0x44 -#define ACI_SET_EQ6 0x45 -#define ACI_SET_EQ7 0x46 /* ... to Treble */ - -#endif /* _MIRO_H_ */ diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 5cd5553..c8a8da0 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -135,21 +135,15 @@ struct snd_opti9xx { unsigned long mc_base_size; #ifdef OPTi93X unsigned long mc_indir_index; + unsigned long mc_indir_size; + struct resource *res_mc_indir; struct snd_wss *codec; #endif /* OPTi93X */ unsigned long pwd_reg; spinlock_t lock; - long wss_base; int irq; - int dma1; - int dma2; - - long fm_port; - - long mpu_port; - int mpu_irq; #ifdef CONFIG_PNP struct pnp_dev *dev; @@ -185,7 +179,7 @@ MODULE_DEVICE_TABLE(pnp_card, snd_opti9xx_pnpids); #endif static char * snd_opti9xx_names[] = { - "unkown", + "unknown", "82C928", "82C929", "82C924", "82C925", "82C930", "82C931", "82C933" @@ -216,13 +210,7 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, spin_lock_init(&chip->lock); - chip->wss_base = -1; chip->irq = -1; - chip->dma1 = -1; - chip->dma2 = -1; - chip->fm_port = -1; - chip->mpu_port = -1; - chip->mpu_irq = -1; switch (hardware) { #ifndef OPTi93X @@ -245,7 +233,10 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, case OPTi9XX_HW_82C931: case OPTi9XX_HW_82C933: chip->mc_base = (hardware == OPTi9XX_HW_82C930) ? 0xf8f : 0xf8d; - chip->mc_indir_index = 0xe0e; + if (!chip->mc_indir_index) { + chip->mc_indir_index = 0xe0e; + chip->mc_indir_size = 2; + } chip->password = 0xe4; chip->pwd_reg = 0; break; @@ -348,7 +339,10 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask))) -static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) +static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, + long wss_base, + int irq, int dma1, int dma2, + long mpu_port, int mpu_irq) { unsigned char wss_base_bits; unsigned char irq_bits; @@ -416,7 +410,7 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) return -EINVAL; } - switch (chip->wss_base) { + switch (wss_base) { case 0x530: wss_base_bits = 0x00; break; @@ -430,14 +424,13 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) wss_base_bits = 0x02; break; default: - snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", - chip->wss_base); + snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", wss_base); goto __skip_base; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), wss_base_bits << 4, 0x30); __skip_base: - switch (chip->irq) { + switch (irq) { //#ifdef OPTi93X case 5: irq_bits = 0x05; @@ -456,11 +449,11 @@ __skip_base: irq_bits = 0x04; break; default: - snd_printk(KERN_WARNING "WSS irq # %d not valid\n", chip->irq); + snd_printk(KERN_WARNING "WSS irq # %d not valid\n", irq); goto __skip_resources; } - switch (chip->dma1) { + switch (dma1) { case 0: dma_bits = 0x01; break; @@ -471,38 +464,36 @@ __skip_base: dma_bits = 0x03; break; default: - snd_printk(KERN_WARNING "WSS dma1 # %d not valid\n", - chip->dma1); + snd_printk(KERN_WARNING "WSS dma1 # %d not valid\n", dma1); goto __skip_resources; } #if defined(CS4231) || defined(OPTi93X) - if (chip->dma1 == chip->dma2) { + if (dma1 == dma2) { snd_printk(KERN_ERR "don't want to share dmas\n"); return -EBUSY; } - switch (chip->dma2) { + switch (dma2) { case 0: case 1: break; default: - snd_printk(KERN_WARNING "WSS dma2 # %d not valid\n", - chip->dma2); + snd_printk(KERN_WARNING "WSS dma2 # %d not valid\n", dma2); goto __skip_resources; } dma_bits |= 0x04; #endif /* CS4231 || OPTi93X */ #ifndef OPTi93X - outb(irq_bits << 3 | dma_bits, chip->wss_base); + outb(irq_bits << 3 | dma_bits, wss_base); #else /* OPTi93X */ snd_opti9xx_write(chip, OPTi9XX_MC_REG(3), (irq_bits << 3 | dma_bits)); #endif /* OPTi93X */ __skip_resources: if (chip->hardware > OPTi9XX_HW_82C928) { - switch (chip->mpu_port) { + switch (mpu_port) { case 0: case -1: break; @@ -520,12 +511,11 @@ __skip_resources: break; default: snd_printk(KERN_WARNING - "MPU-401 port 0x%lx not valid\n", - chip->mpu_port); + "MPU-401 port 0x%lx not valid\n", mpu_port); goto __skip_mpu; } - switch (chip->mpu_irq) { + switch (mpu_irq) { case 5: mpu_irq_bits = 0x02; break; @@ -540,12 +530,12 @@ __skip_resources: break; default: snd_printk(KERN_WARNING "MPU-401 irq # %d not valid\n", - chip->mpu_irq); + mpu_irq); goto __skip_mpu; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), - (chip->mpu_port <= 0) ? 0x00 : + (mpu_port <= 0) ? 0x00 : 0x80 | mpu_port_bits << 5 | mpu_irq_bits << 3, 0xf8); } @@ -558,10 +548,13 @@ __skip_mpu: static irqreturn_t snd_opti93x_interrupt(int irq, void *dev_id) { - struct snd_wss *codec = dev_id; - struct snd_opti9xx *chip = codec->card->private_data; + struct snd_opti9xx *chip = dev_id; + struct snd_wss *codec = chip->codec; unsigned char status; + if (!codec) + return IRQ_HANDLED; + status = snd_opti9xx_read(chip, OPTi9XX_MC_REG(11)); if ((status & OPTi93X_IRQ_PLAYBACK) && codec->playback_substream) snd_pcm_period_elapsed(codec->playback_substream); @@ -575,57 +568,69 @@ static irqreturn_t snd_opti93x_interrupt(int irq, void *dev_id) #endif /* OPTi93X */ -static int __devinit snd_card_opti9xx_detect(struct snd_card *card, - struct snd_opti9xx *chip) +static int __devinit snd_opti9xx_read_check(struct snd_opti9xx *chip) { - int i, err; + unsigned char value; +#ifdef OPTi93X + unsigned long flags; +#endif + chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, + "OPTi9xx MC"); + if (chip->res_mc_base == NULL) + return -EBUSY; #ifndef OPTi93X - for (i = OPTi9XX_HW_82C928; i < OPTi9XX_HW_82C930; i++) { - unsigned char value; + value = snd_opti9xx_read(chip, OPTi9XX_MC_REG(1)); + if (value != 0xff && value != inb(chip->mc_base + OPTi9XX_MC_REG(1))) + if (value == snd_opti9xx_read(chip, OPTi9XX_MC_REG(1))) + return 0; +#else /* OPTi93X */ + chip->res_mc_indir = request_region(chip->mc_indir_index, + chip->mc_indir_size, + "OPTi93x MC"); + if (chip->res_mc_indir == NULL) + return -EBUSY; - if ((err = snd_opti9xx_init(chip, i)) < 0) - return err; + spin_lock_irqsave(&chip->lock, flags); + outb(chip->password, chip->mc_base + chip->pwd_reg); + outb(((chip->mc_indir_index & 0x1f0) >> 4), chip->mc_base); + spin_unlock_irqrestore(&chip->lock, flags); - if ((chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, "OPTi9xx MC")) == NULL) - continue; + value = snd_opti9xx_read(chip, OPTi9XX_MC_REG(7)); + snd_opti9xx_write(chip, OPTi9XX_MC_REG(7), 0xff - value); + if (snd_opti9xx_read(chip, OPTi9XX_MC_REG(7)) == 0xff - value) + return 0; - value = snd_opti9xx_read(chip, OPTi9XX_MC_REG(1)); - if ((value != 0xff) && (value != inb(chip->mc_base + 1))) - if (value == snd_opti9xx_read(chip, OPTi9XX_MC_REG(1))) - return 1; + release_and_free_resource(chip->res_mc_indir); + chip->res_mc_indir = NULL; +#endif /* OPTi93X */ + release_and_free_resource(chip->res_mc_base); + chip->res_mc_base = NULL; - release_and_free_resource(chip->res_mc_base); - chip->res_mc_base = NULL; + return -ENODEV; +} - } -#else /* OPTi93X */ - for (i = OPTi9XX_HW_82C931; i >= OPTi9XX_HW_82C930; i--) { - unsigned long flags; - unsigned char value; +static int __devinit snd_card_opti9xx_detect(struct snd_card *card, + struct snd_opti9xx *chip) +{ + int i, err; - if ((err = snd_opti9xx_init(chip, i)) < 0) +#ifndef OPTi93X + for (i = OPTi9XX_HW_82C928; i < OPTi9XX_HW_82C930; i++) { +#else + for (i = OPTi9XX_HW_82C931; i >= OPTi9XX_HW_82C930; i--) { +#endif + err = snd_opti9xx_init(chip, i); + if (err < 0) return err; - if ((chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, "OPTi9xx MC")) == NULL) - continue; - - spin_lock_irqsave(&chip->lock, flags); - outb(chip->password, chip->mc_base + chip->pwd_reg); - outb(((chip->mc_indir_index & (1 << 8)) >> 4) | - ((chip->mc_indir_index & 0xf0) >> 4), chip->mc_base); - spin_unlock_irqrestore(&chip->lock, flags); - - value = snd_opti9xx_read(chip, OPTi9XX_MC_REG(7)); - snd_opti9xx_write(chip, OPTi9XX_MC_REG(7), 0xff - value); - if (snd_opti9xx_read(chip, OPTi9XX_MC_REG(7)) == 0xff - value) + err = snd_opti9xx_read_check(chip); + if (err == 0) return 1; - - release_and_free_resource(chip->res_mc_base); - chip->res_mc_base = NULL; +#ifdef OPTi93X + chip->mc_indir_index = 0; +#endif } -#endif /* OPTi93X */ - return -ENODEV; } @@ -654,6 +659,8 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, #ifdef OPTi93X port = pnp_port_start(pdev, 0) - 4; fm_port = pnp_port_start(pdev, 1) + 8; + chip->mc_indir_index = pnp_port_start(pdev, 3) + 2; + chip->mc_indir_size = pnp_port_len(pdev, 3) - 2; #else if (pid->driver_data != 0x0924) port = pnp_port_start(pdev, 1); @@ -684,14 +691,14 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, static void snd_card_opti9xx_free(struct snd_card *card) { struct snd_opti9xx *chip = card->private_data; - + if (chip) { #ifdef OPTi93X - struct snd_wss *codec = chip->codec; - if (codec && codec->irq > 0) { - disable_irq(codec->irq); - free_irq(codec->irq, codec); + if (chip->irq > 0) { + disable_irq(chip->irq); + free_irq(chip->irq, chip); } + release_and_free_resource(chip->res_mc_indir); #endif release_and_free_resource(chip->res_mc_base); } @@ -701,6 +708,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) { static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; int error; + int xdma2; struct snd_opti9xx *chip = card->private_data; struct snd_wss *codec; #ifdef CS4231 @@ -710,36 +718,25 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) struct snd_rawmidi *rmidi; struct snd_hwdep *synth; - if (! chip->res_mc_base && - (chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, - "OPTi9xx MC")) == NULL) - return -ENOMEM; - - chip->wss_base = port; - chip->fm_port = fm_port; - chip->mpu_port = mpu_port; - chip->irq = irq; - chip->mpu_irq = mpu_irq; - chip->dma1 = dma1; #if defined(CS4231) || defined(OPTi93X) - chip->dma2 = dma2; + xdma2 = dma2; #else - chip->dma2 = -1; + xdma2 = -1; #endif - if (chip->wss_base == SNDRV_AUTO_PORT) { - chip->wss_base = snd_legacy_find_free_ioport(possible_ports, 4); - if (chip->wss_base < 0) { + if (port == SNDRV_AUTO_PORT) { + port = snd_legacy_find_free_ioport(possible_ports, 4); + if (port < 0) { snd_printk(KERN_ERR "unable to find a free WSS port\n"); return -EBUSY; } } - error = snd_opti9xx_configure(chip); + error = snd_opti9xx_configure(chip, port, irq, dma1, xdma2, + mpu_port, mpu_irq); if (error) return error; - error = snd_wss_create(card, chip->wss_base + 4, -1, - chip->irq, chip->dma1, chip->dma2, + error = snd_wss_create(card, port + 4, -1, irq, dma1, xdma2, #ifdef OPTi93X WSS_HW_OPTI93X, WSS_HWSHARE_IRQ, #else @@ -763,35 +760,35 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) return error; #endif #ifdef OPTi93X - error = request_irq(chip->irq, snd_opti93x_interrupt, - IRQF_DISABLED, DEV_NAME" - WSS", codec); + error = request_irq(irq, snd_opti93x_interrupt, + IRQF_DISABLED, DEV_NAME" - WSS", chip); if (error < 0) { - snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); + snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", irq); return error; } #endif + chip->irq = irq; strcpy(card->driver, chip->name); sprintf(card->shortname, "OPTi %s", card->driver); #if defined(CS4231) || defined(OPTi93X) sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", - card->shortname, pcm->name, chip->wss_base + 4, - chip->irq, chip->dma1, chip->dma2); + card->shortname, pcm->name, port + 4, irq, dma1, xdma2); #else sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", - card->shortname, pcm->name, chip->wss_base + 4, - chip->irq, chip->dma1); + card->shortname, pcm->name, port + 4, irq, dma1); #endif /* CS4231 || OPTi93X */ - if (chip->mpu_port <= 0 || chip->mpu_port == SNDRV_AUTO_PORT) + if (mpu_port <= 0 || mpu_port == SNDRV_AUTO_PORT) rmidi = NULL; - else - if ((error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - chip->mpu_port, 0, chip->mpu_irq, IRQF_DISABLED, - &rmidi))) + else { + error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpu_port, 0, mpu_irq, IRQF_DISABLED, &rmidi); + if (error) snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", - chip->mpu_port); + mpu_port); + } - if (chip->fm_port > 0 && chip->fm_port != SNDRV_AUTO_PORT) { + if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) { struct snd_opl3 *opl3 = NULL; #ifndef OPTi93X if (chip->hardware == OPTi9XX_HW_82C928 || @@ -801,9 +798,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) /* assume we have an OPL4 */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); - if (snd_opl4_create(card, - chip->fm_port, - chip->fm_port - 8, + if (snd_opl4_create(card, fm_port, fm_port - 8, 2, &opl3, &opl4) < 0) { /* no luck, use OPL3 instead */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2), @@ -811,12 +806,10 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) } } #endif /* !OPTi93X */ - if (!opl3 && snd_opl3_create(card, - chip->fm_port, - chip->fm_port + 2, + if (!opl3 && snd_opl3_create(card, fm_port, fm_port + 2, OPL3_HW_AUTO, 0, &opl3) < 0) { snd_printk(KERN_WARNING "no OPL device at 0x%lx-0x%lx\n", - chip->fm_port, chip->fm_port + 4 - 1); + fm_port, fm_port + 4 - 1); } if (opl3) { error = snd_opl3_hwdep_new(opl3, 0, 1, &synth); @@ -978,6 +971,13 @@ static int __devinit snd_opti9xx_pnp_probe(struct pnp_card_link *pcard, } if (hw <= OPTi9XX_HW_82C930) chip->mc_base -= 0x80; + + error = snd_opti9xx_read_check(chip); + if (error) { + snd_printk(KERN_ERR "OPTI chip not found\n"); + snd_card_free(card); + return error; + } snd_card_set_dev(card, &pcard->card->dev); if ((error = snd_opti9xx_probe(card)) < 0) { snd_card_free(card); diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 2ba1897..5b9d6c1 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1682,7 +1682,7 @@ static void snd_wss_resume(struct snd_wss *chip) } #endif /* CONFIG_PM */ -int snd_wss_free(struct snd_wss *chip) +static int snd_wss_free(struct snd_wss *chip) { release_and_free_resource(chip->res_port); release_and_free_resource(chip->res_cport); @@ -1705,7 +1705,6 @@ int snd_wss_free(struct snd_wss *chip) kfree(chip); return 0; } -EXPORT_SYMBOL(snd_wss_free); static int snd_wss_dev_free(struct snd_device *device) { @@ -2225,7 +2224,7 @@ WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, .get = snd_wss_get_mux, .put = snd_wss_put_mux, }, -WSS_DOUBLE("Mic Boost", 0, +WSS_DOUBLE("Mic Boost (+20dB)", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), @@ -2236,14 +2235,14 @@ WSS_DOUBLE("Line Playback Switch", 0, WSS_DOUBLE_TLV("Line Playback Volume", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, db_scale_5bit_12db_max), -WSS_SINGLE("Mono Playback Switch", 0, +WSS_SINGLE("Beep Playback Switch", 0, CS4231_MONO_CTRL, 7, 1, 1), -WSS_SINGLE_TLV("Mono Playback Volume", 0, +WSS_SINGLE_TLV("Beep Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1, db_scale_4bit), WSS_SINGLE("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, 6, 1, 1), -WSS_SINGLE("Mono Output Playback Bypass", 0, +WSS_SINGLE("Beep Bypass Playback Switch", 0, CS4231_MONO_CTRL, 5, 1, 0), }; diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index 135a2b7..a513651 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -1,5 +1,3 @@ -# drivers/sound/Config.in -# # 18 Apr 1998, Michael Elizabeth Chastain, <mailto:mec@shout.net> # More hacking for modularisation. # diff --git a/sound/oss/audio.c b/sound/oss/audio.c index b69c05b..7df48a2 100644 --- a/sound/oss/audio.c +++ b/sound/oss/audio.c @@ -838,7 +838,7 @@ static int dma_ioctl(int dev, unsigned int cmd, void __user *arg) if ((err = audio_devs[dev]->d->prepare_for_input(dev, dmap_in->fragment_size, dmap_in->nbufs)) < 0) { spin_unlock_irqrestore(&dmap_in->lock,flags); - return -err; + return err; } dmap_in->dma_mode = DMODE_INPUT; audio_devs[dev]->enable_bits |= PCM_ENABLE_INPUT; diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index 793b7f4..3f3c3f7 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -219,7 +219,9 @@ static int shared_resources_initialised; * Mid level stuff */ -struct sound_settings dmasound = { .lock = SPIN_LOCK_UNLOCKED }; +struct sound_settings dmasound = { + .lock = __SPIN_LOCK_UNLOCKED(dmasound.lock) +}; static inline void sound_silence(void) { diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c index 06e9e88..bb14e4c 100644 --- a/sound/oss/dmasound/dmasound_paula.c +++ b/sound/oss/dmasound/dmasound_paula.c @@ -657,7 +657,7 @@ static int AmiStateInfo(char *buffer, size_t space) len += sprintf(buffer+len, "\tsound.volume_right = %d [0...64]\n", dmasound.volume_right); if (len >= space) { - printk(KERN_ERR "dmasound_paula: overlowed state buffer alloc.\n") ; + printk(KERN_ERR "dmasound_paula: overflowed state buffer alloc.\n") ; len = space ; } return len; diff --git a/sound/oss/hex2hex.c b/sound/oss/hex2hex.c index 5460faa..041ef5c 100644 --- a/sound/oss/hex2hex.c +++ b/sound/oss/hex2hex.c @@ -12,7 +12,7 @@ #define MAX_SIZE (256*1024) unsigned char buf[MAX_SIZE]; -int loadhex(FILE *inf, unsigned char *buf) +static int loadhex(FILE *inf, unsigned char *buf) { int l=0, c, i; diff --git a/sound/oss/midi_synth.c b/sound/oss/midi_synth.c index 9e45098..3bc7104 100644 --- a/sound/oss/midi_synth.c +++ b/sound/oss/midi_synth.c @@ -426,7 +426,7 @@ midi_synth_open(int dev, int mode) int err; struct midi_input_info *inc; - if (orig_dev < 0 || orig_dev > num_midis || midi_devs[orig_dev] == NULL) + if (orig_dev < 0 || orig_dev >= num_midis || midi_devs[orig_dev] == NULL) return -ENXIO; midi2synth[orig_dev] = dev; diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c index 734b8f9..0af9d24 100644 --- a/sound/oss/mpu401.c +++ b/sound/oss/mpu401.c @@ -770,7 +770,7 @@ static int mpu_synth_ioctl(int dev, unsigned int cmd, void __user *arg) midi_dev = synth_devs[dev]->midi_dev; - if (midi_dev < 0 || midi_dev > num_midis || midi_devs[midi_dev] == NULL) + if (midi_dev < 0 || midi_dev >= num_midis || midi_devs[midi_dev] == NULL) return -ENXIO; devc = &dev_conf[midi_dev]; diff --git a/sound/oss/sb_common.c b/sound/oss/sb_common.c index 77d0e5e..ce4db49 100644 --- a/sound/oss/sb_common.c +++ b/sound/oss/sb_common.c @@ -157,7 +157,7 @@ static void sb_intr (sb_devc *devc) break; default: - /* printk(KERN_WARN "Sound Blaster: Unexpected interrupt\n"); */ + /* printk(KERN_WARNING "Sound Blaster: Unexpected interrupt\n"); */ ; } } @@ -177,7 +177,7 @@ static void sb_intr (sb_devc *devc) break; default: - /* printk(KERN_WARN "Sound Blaster: Unexpected interrupt\n"); */ + /* printk(KERN_WARNING "Sound Blaster: Unexpected interrupt\n"); */ ; } } diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index 180e95c..51a3d38 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -782,7 +782,7 @@ printk(KERN_INFO "FKS: ess_handle_channel %s irq_mode=%d\n", channel, irq_mode); break; default:; - /* printk(KERN_WARN "ESS: Unexpected interrupt\n"); */ + /* printk(KERN_WARNING "ESS: Unexpected interrupt\n"); */ } } diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 75c602b..351654c 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -570,6 +570,7 @@ config SND_ICE1712 tristate "ICEnsemble ICE1712 (Envy24)" select SND_MPU401_UART select SND_AC97_CODEC + select BITREVERSE help Say Y here to include support for soundcards based on the ICE1712 (Envy24) chip. diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 20cb60a..c119206 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2122,7 +2122,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, } /* nothing should be in powerdown mode */ snd_ac97_write_cache(ac97, AC97_GENERAL_PURPOSE, 0); - end_time = jiffies + msecs_to_jiffies(120); + end_time = jiffies + msecs_to_jiffies(5000); do { if ((snd_ac97_read(ac97, AC97_POWERDOWN) & 0x0f) == 0x0f) goto __ready_ok; diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index 15523e6..0470461 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -233,7 +233,7 @@ static void snd_ca0106_proc_dump_iec958( struct snd_info_buffer *buffer, u32 val snd_iprintf(buffer, "user-defined\n"); break; default: - snd_iprintf(buffer, "unkown\n"); + snd_iprintf(buffer, "unknown\n"); break; } snd_iprintf(buffer, "Sample Bits: "); diff --git a/sound/pci/cs46xx/imgs/cwcdma.asp b/sound/pci/cs46xx/imgs/cwcdma.asp index 09d24c7..a65e119 100644 --- a/sound/pci/cs46xx/imgs/cwcdma.asp +++ b/sound/pci/cs46xx/imgs/cwcdma.asp @@ -26,10 +26,11 @@ // // // The purpose of this code is very simple: make it possible to tranfser -// the samples 'as they are' with no alteration from a PCMreader SCB (DMA from host) -// to any other SCB. This is useful for AC3 throug SPDIF. SRC (source rate converters) -// task always alters the samples in some how, however it's from 48khz -> 48khz. The -// alterations are not audible, but AC3 wont work. +// the samples 'as they are' with no alteration from a PCMreader +// SCB (DMA from host) to any other SCB. This is useful for AC3 through SPDIF. +// SRC (source rate converters) task always alters the samples in somehow, +// however it's from 48khz -> 48khz. +// The alterations are not audible, but AC3 wont work. // // ... // | diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 6b8ae7b..1d369ff 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -184,7 +184,7 @@ MODULE_PARM_DESC(enable, "Enable the EMU10K1X soundcard."); * The hardware has 3 channels for playback and 1 for capture. * - channel 0 is the front channel * - channel 1 is the rear channel - * - channel 2 is the center/lfe chanel + * - channel 2 is the center/lfe channel * Volume is controlled by the AC97 for the front and rear channels by * the PCM Playback Volume, Sigmatel Surround Playback Volume and * Surround Playback Volume. The Sigmatel 4-Speaker Stereo switch affects diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 60cdb9e..83508b3 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -55,7 +55,7 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card * * 1 = MediaForte 256-PCS * 2 = MediaForte 256-PCPR * 3 = MediaForte 64-PCR - * 16 = setup tuner only (this is additional bit), i.e. SF-64-PCR FM card + * 16 = setup tuner only (this is additional bit), i.e. SF64-PCR FM card * High 16-bits are video (radio) device number + 1 */ static int tea575x_tuner[SNDRV_CARDS]; @@ -67,7 +67,10 @@ MODULE_PARM_DESC(id, "ID string for the FM801 soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable FM801 soundcard."); module_param_array(tea575x_tuner, int, NULL, 0444); -MODULE_PARM_DESC(tea575x_tuner, "Enable TEA575x tuner."); +MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (1 = SF256-PCS, 2=SF256-PCPR, 3=SF64-PCR, +16=tuner-only)."); + +#define TUNER_ONLY (1<<4) +#define TUNER_TYPE_MASK (~TUNER_ONLY & 0xFFFF) /* * Direct registers @@ -160,7 +163,7 @@ struct fm801 { unsigned int multichannel: 1, /* multichannel support */ secondary: 1; /* secondary codec */ unsigned char secondary_addr; /* address of the secondary codec */ - unsigned int tea575x_tuner; /* tuner flags */ + unsigned int tea575x_tuner; /* tuner access method & flags */ unsigned short ply_ctrl; /* playback control */ unsigned short cap_ctrl; /* capture control */ @@ -1287,7 +1290,7 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume) { unsigned short cmdw; - if (chip->tea575x_tuner & 0x0010) + if (chip->tea575x_tuner & TUNER_ONLY) goto __ac97_ok; /* codec cold reset + AC'97 warm reset */ @@ -1296,11 +1299,13 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume) udelay(100); outw(0, FM801_REG(chip, CODEC_CTRL)); - if (wait_for_codec(chip, 0, AC97_RESET, msecs_to_jiffies(750)) < 0) { - snd_printk(KERN_ERR "Primary AC'97 codec not found\n"); - if (! resume) - return -EIO; - } + if (wait_for_codec(chip, 0, AC97_RESET, msecs_to_jiffies(750)) < 0) + if (!resume) { + snd_printk(KERN_INFO "Primary AC'97 codec not found, " + "assume SF64-PCR (tuner-only)\n"); + chip->tea575x_tuner = 3 | TUNER_ONLY; + goto __ac97_ok; + } if (chip->multichannel) { if (chip->secondary_addr) { @@ -1414,7 +1419,7 @@ static int __devinit snd_fm801_create(struct snd_card *card, return err; } chip->port = pci_resource_start(pci, 0); - if ((tea575x_tuner & 0x0010) == 0) { + if ((tea575x_tuner & TUNER_ONLY) == 0) { if (request_irq(pci->irq, snd_fm801_interrupt, IRQF_SHARED, "FM801", chip)) { snd_printk(KERN_ERR "unable to grab IRQ %d\n", chip->irq); @@ -1429,6 +1434,14 @@ static int __devinit snd_fm801_create(struct snd_card *card, chip->multichannel = 1; snd_fm801_chip_init(chip, 0); + /* init might set tuner access method */ + tea575x_tuner = chip->tea575x_tuner; + + if (chip->irq >= 0 && (tea575x_tuner & TUNER_ONLY)) { + pci_clear_master(pci); + free_irq(chip->irq, chip); + chip->irq = -1; + } if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { snd_fm801_free(chip); @@ -1438,12 +1451,13 @@ static int __devinit snd_fm801_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); #ifdef TEA575X_RADIO - if (tea575x_tuner > 0 && (tea575x_tuner & 0x000f) < 4) { + if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 && + (tea575x_tuner & TUNER_TYPE_MASK) < 4) { chip->tea.dev_nr = tea575x_tuner >> 16; chip->tea.card = card; chip->tea.freq_fixup = 10700; chip->tea.private_data = chip; - chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & 0x000f) - 1]; + chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & TUNER_TYPE_MASK) - 1]; snd_tea575x_init(&chip->tea); } #endif @@ -1483,7 +1497,7 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci, sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, chip->port, chip->irq); - if (tea575x_tuner[dev] & 0x0010) + if (chip->tea575x_tuner & TUNER_ONLY) goto __fm801_tuner_only; if ((err = snd_fm801_pcm(chip, 0, NULL)) < 0) { diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 2439e84..4b200da 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -938,7 +938,7 @@ static void init_input(struct hda_codec *codec) coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */ if (is_active_pin(codec, CS_DMIC1_PIN_NID)) coef |= 0x1800; /* DMIC1 enable 2 channels, disable GPIO0 - * No effect if SPDIF_OUT2 is slected in + * No effect if SPDIF_OUT2 is selected in * IDX_SPDIF_CTL. */ cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 85c81fe..a45c116 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -66,7 +66,7 @@ struct cmi_spec { struct hda_pcm pcm_rec[2]; /* PCM information */ - /* pin deafault configuration */ + /* pin default configuration */ hda_nid_t pin_nid[NUM_PINS]; unsigned int def_conf[NUM_PINS]; unsigned int pin_def_confs; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b3abe9c..0877bae 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6673,7 +6673,7 @@ static struct hda_input_mux alc889A_mb31_capture_source = { /* Front Mic (0x01) unused */ { "Line", 0x2 }, /* Line 2 (0x03) unused */ - /* CD (0x04) unsused? */ + /* CD (0x04) unused? */ }, }; @@ -9287,8 +9287,6 @@ static struct alc_config_preset alc882_presets[] = { .dac_nids = alc883_dac_nids, .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), .adc_nids = alc889_adc_nids, - .capsrc_nids = alc889_capsrc_nids, - .capsrc_nids = alc889_capsrc_nids, .dig_out_nid = ALC883_DIGOUT_NID, .dig_in_nid = ALC883_DIGIN_NID, .slave_dig_outs = alc883_slave_dig_outs, diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 110d16e..765d7bd 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -689,32 +689,14 @@ static int aureon_ac97_mmute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e return change; } -static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1); +static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -10000, 100, 1); static const DECLARE_TLV_DB_SCALE(db_scale_wm_pcm, -6400, 50, 1); static const DECLARE_TLV_DB_SCALE(db_scale_wm_adc, -1200, 100, 0); static const DECLARE_TLV_DB_SCALE(db_scale_ac97_master, -4650, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_ac97_gain, -3450, 150, 0); -/* - * Logarithmic volume values for WM8770 - * Computed as 20 * Log10(255 / x) - */ -static const unsigned char wm_vol[256] = { - 127, 48, 42, 39, 36, 34, 33, 31, 30, 29, 28, 27, 27, 26, 25, 25, 24, 24, 23, - 23, 22, 22, 21, 21, 21, 20, 20, 20, 19, 19, 19, 18, 18, 18, 18, 17, 17, 17, - 17, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 14, 14, 14, 13, 13, 13, - 13, 13, 13, 13, 12, 12, 12, 12, 12, 12, 12, 11, 11, 11, 11, 11, 11, 11, 11, - 11, 10, 10, 10, 10, 10, 10, 10, 10, 10, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 8, 8, - 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 6, 6, 6, - 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, - 5, 5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 3, 3, 3, 3, 3, - 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, - 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, - 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0 -}; - -#define WM_VOL_MAX (sizeof(wm_vol) - 1) +#define WM_VOL_MAX 100 +#define WM_VOL_CNT 101 /* 0dB .. -100dB */ #define WM_VOL_MUTE 0x8000 static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned short vol, unsigned short master) @@ -724,7 +706,8 @@ static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned sho if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE)) nvol = 0; else - nvol = 127 - wm_vol[(((vol & ~WM_VOL_MUTE) * (master & ~WM_VOL_MUTE)) / 127) & WM_VOL_MAX]; + nvol = ((vol % WM_VOL_CNT) * (master % WM_VOL_CNT)) / + WM_VOL_MAX; wm_put(ice, index, nvol); wm_put_nocache(ice, index, 0x180 | nvol); @@ -820,7 +803,7 @@ static int wm_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info * uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = voices; uinfo->value.integer.min = 0; /* mute (-101dB) */ - uinfo->value.integer.max = 0x7F; /* 0dB */ + uinfo->value.integer.max = WM_VOL_MAX; /* 0dB */ return 0; } @@ -850,7 +833,7 @@ static int wm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value * snd_ice1712_save_gpio_status(ice); for (i = 0; i < voices; i++) { unsigned int vol = ucontrol->value.integer.value[i]; - if (vol > 0x7f) + if (vol > WM_VOL_MAX) continue; vol |= spec->vol[ofs+i]; if (vol != spec->vol[ofs+i]) { diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 4615bca..0da778a 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -387,8 +387,8 @@ struct snd_ice1712 { #ifdef CONFIG_PM int (*pm_suspend)(struct snd_ice1712 *); int (*pm_resume)(struct snd_ice1712 *); - int pm_suspend_enabled:1; - int pm_saved_is_spdif_master:1; + unsigned int pm_suspend_enabled:1; + unsigned int pm_saved_is_spdif_master:1; unsigned int pm_saved_spdif_ctrl; unsigned char pm_saved_spdif_cfg; unsigned int pm_saved_route; diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index 4bed963..98bc3b7 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -380,7 +380,7 @@ static struct snd_kcontrol_new juli_mute_controls[] __devinitdata = { * inputs) are fed from Xilinx. * * I even checked traces on board and coded a support in driver for - * an alternative possiblity - the unused I2S ICE output channels + * an alternative possibility - the unused I2S ICE output channels * switched to HW-IN/SPDIF-IN and providing the monitoring signal to * the DAC - to no avail. The I2S outputs seem to be unconnected. * @@ -483,6 +483,31 @@ static int __devinit juli_add_controls(struct snd_ice1712 *ice) } /* + * suspend/resume + * */ + +#ifdef CONFIG_PM +static int juli_resume(struct snd_ice1712 *ice) +{ + struct snd_akm4xxx *ak = ice->akm; + struct juli_spec *spec = ice->spec; + /* akm4358 un-reset, un-mute */ + snd_akm4xxx_reset(ak, 0); + /* reinit ak4114 */ + snd_ak4114_reinit(spec->ak4114); + return 0; +} + +static int juli_suspend(struct snd_ice1712 *ice) +{ + struct snd_akm4xxx *ak = ice->akm; + /* akm4358 reset and soft-mute */ + snd_akm4xxx_reset(ak, 1); + return 0; +} +#endif + +/* * initialize the chip */ @@ -626,6 +651,13 @@ static int __devinit juli_init(struct snd_ice1712 *ice) ice->set_spdif_clock = juli_set_spdif_clock; ice->spdif.ops.open = juli_spdif_in_open; + +#ifdef CONFIG_PM + ice->pm_resume = juli_resume; + ice->pm_suspend = juli_suspend; + ice->pm_suspend_enabled = 1; +#endif + return 0; } diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index c75515f..6a9fee3 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -1100,7 +1100,7 @@ static void ak4396_init(struct snd_ice1712 *ice) } #ifdef CONFIG_PM -static int __devinit prodigy_hd2_resume(struct snd_ice1712 *ice) +static int prodigy_hd2_resume(struct snd_ice1712 *ice) { /* initialize ak4396 codec and restore previous mixer volumes */ struct prodigy_hifi_spec *spec = ice->spec; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 754867e..b990143 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1950,6 +1950,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { }, { .subvendor = 0x104d, + .subdevice = 0x8144, + .name = "Sony", + .type = AC97_TUNE_INV_EAPD + }, + { + .subvendor = 0x104d, .subdevice = 0x8197, .name = "Sony S1XP", .type = AC97_TUNE_INV_EAPD @@ -2057,6 +2063,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { .type = AC97_TUNE_HP_ONLY }, { + .subvendor = 0x161f, + .subdevice = 0x203a, + .name = "Gateway 4525GZ", /* AD1981B */ + .type = AC97_TUNE_INV_EAPD + }, + { .subvendor = 0x1734, .subdevice = 0x0088, .name = "Fujitsu-Siemens D1522", /* AD1981 */ diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 0dce331..a1b10d1 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -3017,7 +3017,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry, insel = "Coaxial"; break; default: - insel = "Unkown"; + insel = "Unknown"; } switch (hdspm->control_register & HDSPM_SyncRefMask) { @@ -3028,7 +3028,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry, syncref = "MADI"; break; default: - syncref = "Unkown"; + syncref = "Unknown"; } snd_iprintf(buffer, "Inputsel = %s, SyncRef = %s\n", insel, syncref); diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 7dea74b..7717e01 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -131,7 +131,7 @@ static int snd_pdacf_probe(struct pcmcia_device *link) return err; } - snd_card_set_dev(card, &handle_to_dev(link)); + snd_card_set_dev(card, &link->dev); pdacf->index = i; card_list[i] = card; @@ -142,12 +142,10 @@ static int snd_pdacf_probe(struct pcmcia_device *link) link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; link->io.NumPorts1 = 16; - link->irq.Attributes = IRQ_TYPE_EXCLUSIVE | IRQ_HANDLE_PRESENT | IRQ_FORCED_PULSE; + link->irq.Attributes = IRQ_TYPE_EXCLUSIVE | IRQ_FORCED_PULSE; // link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING|IRQ_FIRST_SHARED; - link->irq.IRQInfo1 = 0 /* | IRQ_LEVEL_ID */; link->irq.Handler = pdacf_interrupt; - link->irq.Instance = pdacf; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; link->conf.ConfigIndex = 1; @@ -217,20 +215,25 @@ static void snd_pdacf_detach(struct pcmcia_device *link) * configuration callback */ -#define CS_CHECK(fn, ret) \ -do { last_fn = (fn); if ((last_ret = (ret)) != 0) goto cs_failed; } while (0) - static int pdacf_config(struct pcmcia_device *link) { struct snd_pdacf *pdacf = link->priv; - int last_fn, last_ret; + int ret; snd_printdd(KERN_DEBUG "pdacf_config called\n"); link->conf.ConfigIndex = 0x5; - CS_CHECK(RequestIO, pcmcia_request_io(link, &link->io)); - CS_CHECK(RequestIRQ, pcmcia_request_irq(link, &link->irq)); - CS_CHECK(RequestConfiguration, pcmcia_request_configuration(link, &link->conf)); + ret = pcmcia_request_io(link, &link->io); + if (ret) + goto failed; + + ret = pcmcia_request_irq(link, &link->irq); + if (ret) + goto failed; + + ret = pcmcia_request_configuration(link, &link->conf); + if (ret) + goto failed; if (snd_pdacf_assign_resources(pdacf, link->io.BasePort1, link->irq.AssignedIRQ) < 0) goto failed; @@ -238,8 +241,6 @@ static int pdacf_config(struct pcmcia_device *link) link->dev_node = &pdacf->node; return 0; -cs_failed: - cs_error(link, last_fn, last_ret); failed: pcmcia_disable_device(link); return -ENODEV; diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 7445cc8..7be3b33 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -161,11 +161,9 @@ static int snd_vxpocket_new(struct snd_card *card, int ibl, link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; link->io.NumPorts1 = 16; - link->irq.Attributes = IRQ_TYPE_EXCLUSIVE | IRQ_HANDLE_PRESENT; + link->irq.Attributes = IRQ_TYPE_EXCLUSIVE; - link->irq.IRQInfo1 = IRQ_LEVEL_ID; link->irq.Handler = &snd_vx_irq_handler; - link->irq.Instance = chip; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; @@ -213,14 +211,11 @@ static int snd_vxpocket_assign_resources(struct vx_core *chip, int port, int irq * configuration callback */ -#define CS_CHECK(fn, ret) \ -do { last_fn = (fn); if ((last_ret = (ret)) != 0) goto cs_failed; } while (0) - static int vxpocket_config(struct pcmcia_device *link) { struct vx_core *chip = link->priv; struct snd_vxpocket *vxp = (struct snd_vxpocket *)chip; - int last_fn, last_ret; + int ret; snd_printdd(KERN_DEBUG "vxpocket_config called\n"); @@ -235,11 +230,19 @@ static int vxpocket_config(struct pcmcia_device *link) strcpy(chip->card->driver, vxp440_hw.name); } - CS_CHECK(RequestIO, pcmcia_request_io(link, &link->io)); - CS_CHECK(RequestIRQ, pcmcia_request_irq(link, &link->irq)); - CS_CHECK(RequestConfiguration, pcmcia_request_configuration(link, &link->conf)); + ret = pcmcia_request_io(link, &link->io); + if (ret) + goto failed; + + ret = pcmcia_request_irq(link, &link->irq); + if (ret) + goto failed; + + ret = pcmcia_request_configuration(link, &link->conf); + if (ret) + goto failed; - chip->dev = &handle_to_dev(link); + chip->dev = &link->dev; snd_card_set_dev(chip->card, chip->dev); if (snd_vxpocket_assign_resources(chip, link->io.BasePort1, link->irq.AssignedIRQ) < 0) @@ -248,8 +251,6 @@ static int vxpocket_config(struct pcmcia_device *link) link->dev_node = &vxp->node; return 0; -cs_failed: - cs_error(link, last_fn, last_ret); failed: pcmcia_disable_device(link); return -ENODEV; diff --git a/sound/ppc/Kconfig b/sound/ppc/Kconfig index bd2338a..0519c60 100644 --- a/sound/ppc/Kconfig +++ b/sound/ppc/Kconfig @@ -2,7 +2,7 @@ menuconfig SND_PPC bool "PowerPC sound devices" - depends on PPC64 || PPC32 + depends on PPC default y help Support for sound devices specific to PowerPC architectures. diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 583a369..a0df401 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -49,6 +49,7 @@ MODULE_AUTHOR("Adrian McMenamin <adrian@mcmen.demon.co.uk>"); MODULE_DESCRIPTION("Dreamcast AICA sound (pcm) driver"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Yamaha/SEGA, AICA}}"); +MODULE_FIRMWARE("aica_firmware.bin"); /* module parameters */ #define CARD_NAME "AICA" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 0c5eac0..1470141 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ -snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o +snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 9eb610c..9df4c68 100644 --- a/sound/soc/atmel/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -268,7 +268,7 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, #endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - ret = snd_soc_dai_set_pll(codec_dai, 0, + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, clk_get_rate(CODEC_CLK), pll_out); if (ret < 0) { pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 885ba01..e028744 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -207,7 +207,7 @@ static int __init at91sam9g20ek_init(void) struct clk *pllb; int ret; - if (!machine_is_at91sam9g20ek()) + if (!(machine_is_at91sam9g20ek() || machine_is_at91sam9g20ek_2mmc())) return -ENODEV; /* diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 594c6c5..19e4d37 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -2,7 +2,7 @@ * Au12x0/Au1550 PSC ALSA ASoC audio support. * * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano@roarinelk.homelinux.net> + * Manuel Lauss <manuel.lauss@gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -333,6 +333,30 @@ static int au1xpsc_pcm_new(struct snd_card *card, static int au1xpsc_pcm_probe(struct platform_device *pdev) { + if (!au1xpsc_audio_pcmdma[PCM_TX] || !au1xpsc_audio_pcmdma[PCM_RX]) + return -ENODEV; + + return 0; +} + +static int au1xpsc_pcm_remove(struct platform_device *pdev) +{ + return 0; +} + +/* au1xpsc audio platform */ +struct snd_soc_platform au1xpsc_soc_platform = { + .name = "au1xpsc-pcm-dbdma", + .probe = au1xpsc_pcm_probe, + .remove = au1xpsc_pcm_remove, + .pcm_ops = &au1xpsc_pcm_ops, + .pcm_new = au1xpsc_pcm_new, + .pcm_free = au1xpsc_pcm_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(au1xpsc_soc_platform); + +static int __devinit au1xpsc_pcm_drvprobe(struct platform_device *pdev) +{ struct resource *r; int ret; @@ -365,7 +389,9 @@ static int au1xpsc_pcm_probe(struct platform_device *pdev) } (au1xpsc_audio_pcmdma[PCM_RX])->ddma_id = r->start; - return 0; + ret = snd_soc_register_platform(&au1xpsc_soc_platform); + if (!ret) + return ret; out2: kfree(au1xpsc_audio_pcmdma[PCM_RX]); @@ -376,10 +402,12 @@ out1: return ret; } -static int au1xpsc_pcm_remove(struct platform_device *pdev) +static int __devexit au1xpsc_pcm_drvremove(struct platform_device *pdev) { int i; + snd_soc_unregister_platform(&au1xpsc_soc_platform); + for (i = 0; i < 2; i++) { if (au1xpsc_audio_pcmdma[i]) { au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[i]); @@ -391,32 +419,81 @@ static int au1xpsc_pcm_remove(struct platform_device *pdev) return 0; } -/* au1xpsc audio platform */ -struct snd_soc_platform au1xpsc_soc_platform = { - .name = "au1xpsc-pcm-dbdma", - .probe = au1xpsc_pcm_probe, - .remove = au1xpsc_pcm_remove, - .pcm_ops = &au1xpsc_pcm_ops, - .pcm_new = au1xpsc_pcm_new, - .pcm_free = au1xpsc_pcm_free_dma_buffers, +static struct platform_driver au1xpsc_pcm_driver = { + .driver = { + .name = "au1xpsc-pcm", + .owner = THIS_MODULE, + }, + .probe = au1xpsc_pcm_drvprobe, + .remove = __devexit_p(au1xpsc_pcm_drvremove), }; -EXPORT_SYMBOL_GPL(au1xpsc_soc_platform); -static int __init au1xpsc_audio_dbdma_init(void) +static int __init au1xpsc_audio_dbdma_load(void) { au1xpsc_audio_pcmdma[PCM_TX] = NULL; au1xpsc_audio_pcmdma[PCM_RX] = NULL; - return snd_soc_register_platform(&au1xpsc_soc_platform); + return platform_driver_register(&au1xpsc_pcm_driver); } -static void __exit au1xpsc_audio_dbdma_exit(void) +static void __exit au1xpsc_audio_dbdma_unload(void) { - snd_soc_unregister_platform(&au1xpsc_soc_platform); + platform_driver_unregister(&au1xpsc_pcm_driver); } -module_init(au1xpsc_audio_dbdma_init); -module_exit(au1xpsc_audio_dbdma_exit); +module_init(au1xpsc_audio_dbdma_load); +module_exit(au1xpsc_audio_dbdma_unload); + + +struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev) +{ + struct resource *res, *r; + struct platform_device *pd; + int id[2]; + int ret; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r) + return NULL; + id[0] = r->start; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r) + return NULL; + id[1] = r->start; + + res = kzalloc(sizeof(struct resource) * 2, GFP_KERNEL); + if (!res) + return NULL; + + res[0].start = res[0].end = id[0]; + res[1].start = res[1].end = id[1]; + res[0].flags = res[1].flags = IORESOURCE_DMA; + + pd = platform_device_alloc("au1xpsc-pcm", -1); + if (!pd) + goto out; + + pd->resource = res; + pd->num_resources = 2; + + ret = platform_device_add(pd); + if (!ret) + return pd; + + platform_device_put(pd); +out: + kfree(res); + return NULL; +} +EXPORT_SYMBOL_GPL(au1xpsc_pcm_add); + +void au1xpsc_pcm_destroy(struct platform_device *dmapd) +{ + if (dmapd) + platform_device_unregister(dmapd); +} +EXPORT_SYMBOL_GPL(au1xpsc_pcm_destroy); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver"); -MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index a521aa9..340311d 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -61,7 +61,8 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned short data, retry, tmo; + unsigned short retry, tmo; + unsigned long data; au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); @@ -74,20 +75,26 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, AC97_CDC(pscdata)); au_sync(); - tmo = 2000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) - && --tmo) - udelay(2); + tmo = 20; + do { + udelay(21); + if (au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD) + break; + } while (--tmo); - data = au_readl(AC97_CDC(pscdata)) & 0xffff; + data = au_readl(AC97_CDC(pscdata)); au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); mutex_unlock(&pscdata->lock); + + if (reg != ((data >> 16) & 0x7f)) + tmo = 1; /* wrong register, try again */ + } while (--retry && !tmo); - return retry ? data : 0xffff; + return retry ? data & 0xffff : 0xffff; } /* AC97 controller writes to codec register */ @@ -109,10 +116,12 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, AC97_CDC(pscdata)); au_sync(); - tmo = 2000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) - && --tmo) - udelay(2); + tmo = 20; + do { + udelay(21); + if (au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD) + break; + } while (--tmo); au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); @@ -195,7 +204,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; unsigned long r, ro, stat; - int chans, stype = SUBSTREAM_TYPE(substream); + int chans, t, stype = SUBSTREAM_TYPE(substream); chans = params_channels(params); @@ -237,8 +246,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, au_sync(); /* ...wait for it... */ - while (au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) - asm volatile ("nop"); + t = 100; + while ((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) && --t) + msleep(1); + + if (!t) + printk(KERN_ERR "PSC-AC97: can't disable!\n"); /* ...write config... */ au_writel(r, AC97_CFG(pscdata)); @@ -249,8 +262,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, au_sync(); /* ...and wait for ready bit */ - while (!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) - asm volatile ("nop"); + t = 100; + while ((!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && --t) + msleep(1); + + if (!t) + printk(KERN_ERR "PSC-AC97: can't enable!\n"); mutex_unlock(&pscdata->lock); @@ -300,19 +317,55 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, static int au1xpsc_ac97_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { + return au1xpsc_ac97_workdata ? 0 : -ENODEV; +} + +static void au1xpsc_ac97_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ +} + +static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { + .trigger = au1xpsc_ac97_trigger, + .hw_params = au1xpsc_ac97_hw_params, +}; + +struct snd_soc_dai au1xpsc_ac97_dai = { + .name = "au1xpsc_ac97", + .ac97_control = 1, + .probe = au1xpsc_ac97_probe, + .remove = au1xpsc_ac97_remove, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = &au1xpsc_ac97_dai_ops, +}; +EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); + +static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) +{ int ret; struct resource *r; unsigned long sel; + struct au1xpsc_audio_data *wd; if (au1xpsc_ac97_workdata) return -EBUSY; - au1xpsc_ac97_workdata = - kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); - if (!au1xpsc_ac97_workdata) + wd = kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); + if (!wd) return -ENOMEM; - mutex_init(&au1xpsc_ac97_workdata->lock); + mutex_init(&wd->lock); r = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!r) { @@ -321,81 +374,95 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, } ret = -EBUSY; - au1xpsc_ac97_workdata->ioarea = - request_mem_region(r->start, r->end - r->start + 1, + wd->ioarea = request_mem_region(r->start, r->end - r->start + 1, "au1xpsc_ac97"); - if (!au1xpsc_ac97_workdata->ioarea) + if (!wd->ioarea) goto out0; - au1xpsc_ac97_workdata->mmio = ioremap(r->start, 0xffff); - if (!au1xpsc_ac97_workdata->mmio) + wd->mmio = ioremap(r->start, 0xffff); + if (!wd->mmio) goto out1; /* configuration: max dma trigger threshold, enable ac97 */ - au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | - PSC_AC97CFG_TT_FIFO8 | - PSC_AC97CFG_DE_ENABLE; + wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 | + PSC_AC97CFG_DE_ENABLE; - /* preserve PSC clock source set up by platform (dev.platform_data - * is already occupied by soc layer) - */ - sel = au_readl(PSC_SEL(au1xpsc_ac97_workdata)) & PSC_SEL_CLK_MASK; - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + /* preserve PSC clock source set up by platform */ + sel = au_readl(PSC_SEL(wd)) & PSC_SEL_CLK_MASK; + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - au_writel(0, PSC_SEL(au1xpsc_ac97_workdata)); + au_writel(0, PSC_SEL(wd)); au_sync(); - au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(au1xpsc_ac97_workdata)); + au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(wd)); au_sync(); - /* next up: cold reset. Dont check for PSC-ready now since - * there may not be any codec clock yet. - */ - return 0; + ret = snd_soc_register_dai(&au1xpsc_ac97_dai); + if (ret) + goto out1; + wd->dmapd = au1xpsc_pcm_add(pdev); + if (wd->dmapd) { + platform_set_drvdata(pdev, wd); + au1xpsc_ac97_workdata = wd; /* MDEV */ + return 0; + } + + snd_soc_unregister_dai(&au1xpsc_ac97_dai); out1: - release_resource(au1xpsc_ac97_workdata->ioarea); - kfree(au1xpsc_ac97_workdata->ioarea); + release_resource(wd->ioarea); + kfree(wd->ioarea); out0: - kfree(au1xpsc_ac97_workdata); - au1xpsc_ac97_workdata = NULL; + kfree(wd); return ret; } -static void au1xpsc_ac97_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev) { + struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); + + if (wd->dmapd) + au1xpsc_pcm_destroy(wd->dmapd); + + snd_soc_unregister_dai(&au1xpsc_ac97_dai); + /* disable PSC completely */ - au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); + au_writel(0, AC97_CFG(wd)); au_sync(); - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - iounmap(au1xpsc_ac97_workdata->mmio); - release_resource(au1xpsc_ac97_workdata->ioarea); - kfree(au1xpsc_ac97_workdata->ioarea); - kfree(au1xpsc_ac97_workdata); - au1xpsc_ac97_workdata = NULL; + iounmap(wd->mmio); + release_resource(wd->ioarea); + kfree(wd->ioarea); + kfree(wd); + + au1xpsc_ac97_workdata = NULL; /* MDEV */ + + return 0; } -static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai) +#ifdef CONFIG_PM +static int au1xpsc_ac97_drvsuspend(struct device *dev) { + struct au1xpsc_audio_data *wd = dev_get_drvdata(dev); + /* save interesting registers and disable PSC */ - au1xpsc_ac97_workdata->pm[0] = - au_readl(PSC_SEL(au1xpsc_ac97_workdata)); + wd->pm[0] = au_readl(PSC_SEL(wd)); - au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); + au_writel(0, AC97_CFG(wd)); au_sync(); - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); return 0; } -static int au1xpsc_ac97_resume(struct snd_soc_dai *dai) +static int au1xpsc_ac97_drvresume(struct device *dev) { + struct au1xpsc_audio_data *wd = dev_get_drvdata(dev); + /* restore PSC clock config */ - au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE, - PSC_SEL(au1xpsc_ac97_workdata)); + au_writel(wd->pm[0] | PSC_SEL_PS_AC97MODE, PSC_SEL(wd)); au_sync(); /* after this point the ac97 core will cold-reset the codec. @@ -405,48 +472,44 @@ static int au1xpsc_ac97_resume(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { - .trigger = au1xpsc_ac97_trigger, - .hw_params = au1xpsc_ac97_hw_params, +static struct dev_pm_ops au1xpscac97_pmops = { + .suspend = au1xpsc_ac97_drvsuspend, + .resume = au1xpsc_ac97_drvresume, }; -struct snd_soc_dai au1xpsc_ac97_dai = { - .name = "au1xpsc_ac97", - .ac97_control = 1, - .probe = au1xpsc_ac97_probe, - .remove = au1xpsc_ac97_remove, - .suspend = au1xpsc_ac97_suspend, - .resume = au1xpsc_ac97_resume, - .playback = { - .rates = AC97_RATES, - .formats = AC97_FMTS, - .channels_min = 2, - .channels_max = 2, - }, - .capture = { - .rates = AC97_RATES, - .formats = AC97_FMTS, - .channels_min = 2, - .channels_max = 2, +#define AU1XPSCAC97_PMOPS &au1xpscac97_pmops + +#else + +#define AU1XPSCAC97_PMOPS NULL + +#endif + +static struct platform_driver au1xpsc_ac97_driver = { + .driver = { + .name = "au1xpsc_ac97", + .owner = THIS_MODULE, + .pm = AU1XPSCAC97_PMOPS, }, - .ops = &au1xpsc_ac97_dai_ops, + .probe = au1xpsc_ac97_drvprobe, + .remove = __devexit_p(au1xpsc_ac97_drvremove), }; -EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); -static int __init au1xpsc_ac97_init(void) +static int __init au1xpsc_ac97_load(void) { au1xpsc_ac97_workdata = NULL; - return snd_soc_register_dai(&au1xpsc_ac97_dai); + return platform_driver_register(&au1xpsc_ac97_driver); } -static void __exit au1xpsc_ac97_exit(void) +static void __exit au1xpsc_ac97_unload(void) { - snd_soc_unregister_dai(&au1xpsc_ac97_dai); + platform_driver_unregister(&au1xpsc_ac97_driver); } -module_init(au1xpsc_ac97_init); -module_exit(au1xpsc_ac97_exit); +module_init(au1xpsc_ac97_load); +module_exit(au1xpsc_ac97_unload); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <manuel.lauss@gmail.com>"); +MODULE_AUTHOR("Manuel Lauss"); + diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index bb58932..0cf2ca6 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -2,7 +2,7 @@ * Au12x0/Au1550 PSC ALSA ASoC audio support. * * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano@roarinelk.homelinux.net> + * Manuel Lauss <manuel.lauss@gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -265,16 +265,52 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, static int au1xpsc_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { + return au1xpsc_i2s_workdata ? 0 : -ENODEV; +} + +static void au1xpsc_i2s_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ +} + +static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { + .trigger = au1xpsc_i2s_trigger, + .hw_params = au1xpsc_i2s_hw_params, + .set_fmt = au1xpsc_i2s_set_fmt, +}; + +struct snd_soc_dai au1xpsc_i2s_dai = { + .name = "au1xpsc_i2s", + .probe = au1xpsc_i2s_probe, + .remove = au1xpsc_i2s_remove, + .playback = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .capture = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .ops = &au1xpsc_i2s_dai_ops, +}; +EXPORT_SYMBOL(au1xpsc_i2s_dai); + +static int __init au1xpsc_i2s_drvprobe(struct platform_device *pdev) +{ struct resource *r; unsigned long sel; int ret; + struct au1xpsc_audio_data *wd; if (au1xpsc_i2s_workdata) return -EBUSY; - au1xpsc_i2s_workdata = - kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); - if (!au1xpsc_i2s_workdata) + wd = kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); + if (!wd) return -ENOMEM; r = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -284,131 +320,146 @@ static int au1xpsc_i2s_probe(struct platform_device *pdev, } ret = -EBUSY; - au1xpsc_i2s_workdata->ioarea = - request_mem_region(r->start, r->end - r->start + 1, + wd->ioarea = request_mem_region(r->start, r->end - r->start + 1, "au1xpsc_i2s"); - if (!au1xpsc_i2s_workdata->ioarea) + if (!wd->ioarea) goto out0; - au1xpsc_i2s_workdata->mmio = ioremap(r->start, 0xffff); - if (!au1xpsc_i2s_workdata->mmio) + wd->mmio = ioremap(r->start, 0xffff); + if (!wd->mmio) goto out1; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) */ - sel = au_readl(PSC_SEL(au1xpsc_i2s_workdata)) & PSC_SEL_CLK_MASK; - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + sel = au_readl(PSC_SEL(wd)) & PSC_SEL_CLK_MASK; + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - au_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(au1xpsc_i2s_workdata)); - au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + au_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(wd)); + au_writel(0, I2S_CFG(wd)); au_sync(); /* preconfigure: set max rx/tx fifo depths */ - au1xpsc_i2s_workdata->cfg |= - PSC_I2SCFG_RT_FIFO8 | PSC_I2SCFG_TT_FIFO8; + wd->cfg |= PSC_I2SCFG_RT_FIFO8 | PSC_I2SCFG_TT_FIFO8; /* don't wait for I2S core to become ready now; clocks may not * be running yet; depending on clock input for PSC a wait might * time out. */ - return 0; + ret = snd_soc_register_dai(&au1xpsc_i2s_dai); + if (ret) + goto out1; + /* finally add the DMA device for this PSC */ + wd->dmapd = au1xpsc_pcm_add(pdev); + if (wd->dmapd) { + platform_set_drvdata(pdev, wd); + au1xpsc_i2s_workdata = wd; + return 0; + } + + snd_soc_unregister_dai(&au1xpsc_i2s_dai); out1: - release_resource(au1xpsc_i2s_workdata->ioarea); - kfree(au1xpsc_i2s_workdata->ioarea); + release_resource(wd->ioarea); + kfree(wd->ioarea); out0: - kfree(au1xpsc_i2s_workdata); - au1xpsc_i2s_workdata = NULL; + kfree(wd); return ret; } -static void au1xpsc_i2s_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev) { - au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); + + if (wd->dmapd) + au1xpsc_pcm_destroy(wd->dmapd); + + snd_soc_unregister_dai(&au1xpsc_i2s_dai); + + au_writel(0, I2S_CFG(wd)); au_sync(); - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - iounmap(au1xpsc_i2s_workdata->mmio); - release_resource(au1xpsc_i2s_workdata->ioarea); - kfree(au1xpsc_i2s_workdata->ioarea); - kfree(au1xpsc_i2s_workdata); - au1xpsc_i2s_workdata = NULL; + iounmap(wd->mmio); + release_resource(wd->ioarea); + kfree(wd->ioarea); + kfree(wd); + + au1xpsc_i2s_workdata = NULL; /* MDEV */ + + return 0; } -static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai) +#ifdef CONFIG_PM +static int au1xpsc_i2s_drvsuspend(struct device *dev) { + struct au1xpsc_audio_data *wd = dev_get_drvdata(dev); + /* save interesting register and disable PSC */ - au1xpsc_i2s_workdata->pm[0] = - au_readl(PSC_SEL(au1xpsc_i2s_workdata)); + wd->pm[0] = au_readl(PSC_SEL(wd)); - au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + au_writel(0, I2S_CFG(wd)); au_sync(); - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); return 0; } -static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai) +static int au1xpsc_i2s_drvresume(struct device *dev) { + struct au1xpsc_audio_data *wd = dev_get_drvdata(dev); + /* select I2S mode and PSC clock */ - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - au_writel(0, PSC_SEL(au1xpsc_i2s_workdata)); + au_writel(0, PSC_SEL(wd)); au_sync(); - au_writel(au1xpsc_i2s_workdata->pm[0], - PSC_SEL(au1xpsc_i2s_workdata)); + au_writel(wd->pm[0], PSC_SEL(wd)); au_sync(); return 0; } -static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { - .trigger = au1xpsc_i2s_trigger, - .hw_params = au1xpsc_i2s_hw_params, - .set_fmt = au1xpsc_i2s_set_fmt, +static struct dev_pm_ops au1xpsci2s_pmops = { + .suspend = au1xpsc_i2s_drvsuspend, + .resume = au1xpsc_i2s_drvresume, }; -struct snd_soc_dai au1xpsc_i2s_dai = { - .name = "au1xpsc_i2s", - .probe = au1xpsc_i2s_probe, - .remove = au1xpsc_i2s_remove, - .suspend = au1xpsc_i2s_suspend, - .resume = au1xpsc_i2s_resume, - .playback = { - .rates = AU1XPSC_I2S_RATES, - .formats = AU1XPSC_I2S_FMTS, - .channels_min = 2, - .channels_max = 8, /* 2 without external help */ - }, - .capture = { - .rates = AU1XPSC_I2S_RATES, - .formats = AU1XPSC_I2S_FMTS, - .channels_min = 2, - .channels_max = 8, /* 2 without external help */ +#define AU1XPSCI2S_PMOPS &au1xpsci2s_pmops + +#else + +#define AU1XPSCI2S_PMOPS NULL + +#endif + +static struct platform_driver au1xpsc_i2s_driver = { + .driver = { + .name = "au1xpsc_i2s", + .owner = THIS_MODULE, + .pm = AU1XPSCI2S_PMOPS, }, - .ops = &au1xpsc_i2s_dai_ops, + .probe = au1xpsc_i2s_drvprobe, + .remove = __devexit_p(au1xpsc_i2s_drvremove), }; -EXPORT_SYMBOL(au1xpsc_i2s_dai); -static int __init au1xpsc_i2s_init(void) +static int __init au1xpsc_i2s_load(void) { au1xpsc_i2s_workdata = NULL; - return snd_soc_register_dai(&au1xpsc_i2s_dai); + return platform_driver_register(&au1xpsc_i2s_driver); } -static void __exit au1xpsc_i2s_exit(void) +static void __exit au1xpsc_i2s_unload(void) { - snd_soc_unregister_dai(&au1xpsc_i2s_dai); + platform_driver_unregister(&au1xpsc_i2s_driver); } -module_init(au1xpsc_i2s_init); -module_exit(au1xpsc_i2s_exit); +module_init(au1xpsc_i2s_load); +module_exit(au1xpsc_i2s_unload); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index 3f474e8..32d3807 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -2,7 +2,7 @@ * Au12x0/Au1550 PSC ALSA ASoC audio support. * * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano@roarinelk.homelinux.net> + * Manuel Lauss <manuel.lauss@gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -21,6 +21,10 @@ extern struct snd_soc_dai au1xpsc_i2s_dai; extern struct snd_soc_platform au1xpsc_soc_platform; extern struct snd_ac97_bus_ops soc_ac97_ops; +/* DBDMA helpers */ +extern struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev); +extern void au1xpsc_pcm_destroy(struct platform_device *dmapd); + struct au1xpsc_audio_data { void __iomem *mmio; @@ -30,6 +34,7 @@ struct au1xpsc_audio_data { unsigned long pm[2]; struct resource *ioarea; struct mutex lock; + struct platform_device *dmapd; }; #define PCM_TX 0 diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index cd361e3..0f45a3f 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -52,6 +52,7 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int channel_map[] = {0, 4, 1, 5, 2, 6, 3, 7}; int ret = 0; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | @@ -65,6 +66,12 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + /* set cpu DAI channel mapping */ + ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), + channel_map, ARRAY_SIZE(channel_map), channel_map); + if (ret < 0) + return ret; + return 0; } diff --git a/sound/soc/blackfin/bf5xx-ad1938.c b/sound/soc/blackfin/bf5xx-ad1938.c index 08269e9..2ef1e50 100644 --- a/sound/soc/blackfin/bf5xx-ad1938.c +++ b/sound/soc/blackfin/bf5xx-ad1938.c @@ -61,6 +61,7 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int channel_map[] = {0, 1, 2, 3, 4, 5, 6, 7}; int ret = 0; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | @@ -75,7 +76,13 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, return ret; /* set codec DAI slots, 8 channels, all channels are enabled */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 8); + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 0xFF, 8, 32); + if (ret < 0) + return ret; + + /* set cpu DAI channel mapping */ + ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), + channel_map, ARRAY_SIZE(channel_map), channel_map); if (ret < 0) return ret; diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 084b688..3e6ada0 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -49,7 +49,6 @@ struct bf5xx_i2s_port { u16 rcr1; u16 tcr2; u16 rcr2; - int counter; int configured; }; @@ -133,16 +132,6 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return ret; } -static int bf5xx_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - pr_debug("%s enter\n", __func__); - - /*this counter is used for counting how many pcm streams are opened*/ - bf5xx_i2s.counter++; - return 0; -} - static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -201,9 +190,8 @@ static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { pr_debug("%s enter\n", __func__); - bf5xx_i2s.counter--; /* No active stream, SPORT is allowed to be configured again. */ - if (!bf5xx_i2s.counter) + if (!dai->active) bf5xx_i2s.configured = 0; } @@ -284,7 +272,6 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai) SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = { - .startup = bf5xx_i2s_startup, .shutdown = bf5xx_i2s_shutdown, .hw_params = bf5xx_i2s_hw_params, .set_fmt = bf5xx_i2s_set_dai_fmt, diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index ccb5e82..a8c73cb 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -43,7 +43,7 @@ #include "bf5xx-tdm.h" #include "bf5xx-sport.h" -#define PCM_BUFFER_MAX 0x10000 +#define PCM_BUFFER_MAX 0x8000 #define FRAGMENT_SIZE_MIN (4*1024) #define FRAGMENTS_MIN 2 #define FRAGMENTS_MAX 32 @@ -177,6 +177,9 @@ out: static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count) { + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + struct bf5xx_tdm_port *tdm_port = sport->private_data; unsigned int *src; unsigned int *dst; int i; @@ -188,7 +191,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, dst += pos * 8; while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *(dst + i) = *src++; + *(dst + tdm_port->tx_map[i]) = *src++; dst += 8; } } else { @@ -198,7 +201,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, src += pos * 8; while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *dst++ = *(src+i); + *dst++ = *(src + tdm_port->rx_map[i]); src += 8; } } diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index ff546e9..4b36012 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -46,14 +46,6 @@ #include "bf5xx-sport.h" #include "bf5xx-tdm.h" -struct bf5xx_tdm_port { - u16 tcr1; - u16 rcr1; - u16 tcr2; - u16 rcr2; - int configured; -}; - static struct bf5xx_tdm_port bf5xx_tdm; static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; @@ -181,6 +173,40 @@ static void bf5xx_tdm_shutdown(struct snd_pcm_substream *substream, bf5xx_tdm.configured = 0; } +static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + int i; + unsigned int slot; + unsigned int tx_mapped = 0, rx_mapped = 0; + + if ((tx_num > BFIN_TDM_DAI_MAX_SLOTS) || + (rx_num > BFIN_TDM_DAI_MAX_SLOTS)) + return -EINVAL; + + for (i = 0; i < tx_num; i++) { + slot = tx_slot[i]; + if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && + (!(tx_mapped & (1 << slot)))) { + bf5xx_tdm.tx_map[i] = slot; + tx_mapped |= 1 << slot; + } else + return -EINVAL; + } + for (i = 0; i < rx_num; i++) { + slot = rx_slot[i]; + if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && + (!(rx_mapped & (1 << slot)))) { + bf5xx_tdm.rx_map[i] = slot; + rx_mapped |= 1 << slot; + } else + return -EINVAL; + } + + return 0; +} + #ifdef CONFIG_PM static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) { @@ -235,6 +261,7 @@ static struct snd_soc_dai_ops bf5xx_tdm_dai_ops = { .hw_params = bf5xx_tdm_hw_params, .set_fmt = bf5xx_tdm_set_dai_fmt, .shutdown = bf5xx_tdm_shutdown, + .set_channel_map = bf5xx_tdm_set_channel_map, }; struct snd_soc_dai bf5xx_tdm_dai = { @@ -300,6 +327,8 @@ static int __devinit bfin_tdm_probe(struct platform_device *pdev) pr_err("Failed to register DAI: %d\n", ret); goto sport_config_err; } + + sport_handle->private_data = &bf5xx_tdm; return 0; sport_config_err: diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h index 618ec3d..04189a1 100644 --- a/sound/soc/blackfin/bf5xx-tdm.h +++ b/sound/soc/blackfin/bf5xx-tdm.h @@ -9,6 +9,17 @@ #ifndef _BF5XX_TDM_H #define _BF5XX_TDM_H +#define BFIN_TDM_DAI_MAX_SLOTS 8 +struct bf5xx_tdm_port { + u16 tcr1; + u16 rcr1; + u16 tcr2; + u16 rcr2; + unsigned int tx_map[BFIN_TDM_DAI_MAX_SLOTS]; + unsigned int rx_map[BFIN_TDM_DAI_MAX_SLOTS]; + int configured; +}; + extern struct snd_soc_dai bf5xx_tdm_dai; #endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0edca93..52b005f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -15,10 +15,12 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD1836 if SPI_MASTER select SND_SOC_AD1938 if SPI_MASTER select SND_SOC_AD1980 if SND_SOC_AC97_BUS + select SND_SOC_ADS117X select SND_SOC_AD73311 if I2C select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C + select SND_SOC_AK4671 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -28,6 +30,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C + select SND_SOC_TPA6130A2 if I2C + select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C @@ -36,6 +40,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8523 if I2C select SND_SOC_WM8580 if I2C + select SND_SOC_WM8711 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8727 select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI @@ -86,6 +92,9 @@ config SND_SOC_AD1980 config SND_SOC_AD73311 tristate + +config SND_SOC_ADS117X + tristate config SND_SOC_AK4104 tristate @@ -96,6 +105,9 @@ config SND_SOC_AK4535 config SND_SOC_AK4642 tristate +config SND_SOC_AK4671 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate @@ -136,7 +148,11 @@ config SND_SOC_TLV320AIC26 config SND_SOC_TLV320AIC3X tristate +config SND_SOC_TLV320DAC33 + tristate + config SND_SOC_TWL4030 + select TWL4030_CODEC tristate config SND_SOC_UDA134X @@ -160,6 +176,12 @@ config SND_SOC_WM8523 config SND_SOC_WM8580 tristate +config SND_SOC_WM8711 + tristate + +config SND_SOC_WM8727 + tristate + config SND_SOC_WM8728 tristate @@ -220,3 +242,6 @@ config SND_SOC_WM9713 # Amp config SND_SOC_MAX9877 tristate + +config SND_SOC_TPA6130A2 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fb4af28..dbaecb1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,9 +3,11 @@ snd-soc-ad1836-objs := ad1836.o snd-soc-ad1938-objs := ad1938.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o +snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o +snd-soc-ak4671-objs := ak4671.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -16,6 +18,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-tlv320dac33-objs := tlv320dac33.o snd-soc-twl4030-objs := twl4030.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o @@ -24,6 +27,8 @@ snd-soc-wm8400-objs := wm8400.o snd-soc-wm8510-objs := wm8510.o snd-soc-wm8523-objs := wm8523.o snd-soc-wm8580-objs := wm8580.o +snd-soc-wm8711-objs := wm8711.o +snd-soc-wm8727-objs := wm8727.o snd-soc-wm8728-objs := wm8728.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o @@ -47,15 +52,18 @@ snd-soc-wm-hubs-objs := wm_hubs.o # Amp snd-soc-max9877-objs := max9877.o +snd-soc-tpa6130a2-objs := tpa6130a2.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD1938) += snd-soc-ad1938.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o +obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o +obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o @@ -66,6 +74,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.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_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o @@ -74,6 +83,8 @@ obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8523) += snd-soc-wm8523.o obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o +obj-$(CONFIG_SND_SOC_WM8711) += snd-soc-wm8711.o +obj-$(CONFIG_SND_SOC_WM8727) += snd-soc-wm8727.o obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o @@ -97,3 +108,4 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o +obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 932299b..69bd0ac 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -117,9 +117,6 @@ static int ac97_soc_probe(struct platform_device *pdev) if (ret < 0) goto bus_err; - ret = snd_soc_init_card(socdev); - if (ret < 0) - goto bus_err; return 0; bus_err: diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index c48485f..2c18e3d 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -385,19 +385,7 @@ static int ad1836_probe(struct platform_device *pdev) snd_soc_dapm_new_controls(codec, ad1836_dapm_widgets, ARRAY_SIZE(ad1836_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); - snd_soc_dapm_new_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } - - return ret; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index 34b30ef..5d48918 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -592,21 +592,9 @@ static int ad1938_probe(struct platform_device *pdev) snd_soc_dapm_new_controls(codec, ad1938_dapm_widgets, ARRAY_SIZE(ad1938_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); - snd_soc_dapm_new_widgets(codec); ad1938_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } - - return ret; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index d7440a9..39c0f758 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -257,11 +257,6 @@ static int ad1980_soc_probe(struct platform_device *pdev) snd_soc_add_controls(codec, ad1980_snd_ac97_controls, ARRAY_SIZE(ad1980_snd_ac97_controls)); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "ad1980: failed to register card\n"); - goto reset_err; - } return 0; diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index e61dac5..d2fcc60 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -64,16 +64,8 @@ static int ad73311_soc_probe(struct platform_device *pdev) goto pcm_err; } - ret = snd_soc_init_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->card->codec); socdev->card->codec = NULL; diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c new file mode 100644 index 0000000..cc96411 --- /dev/null +++ b/sound/soc/codecs/ads117x.c @@ -0,0 +1,123 @@ +/* + * ads117x.c -- Driver for ads1174/8 ADC chips + * + * Copyright 2009 ShotSpotter Inc. + * Author: Graeme Gregory <gg@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 + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "ads117x.h" + +#define ADS117X_RATES (SNDRV_PCM_RATE_8000_48000) + +#define ADS117X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) + +struct snd_soc_dai ads117x_dai = { +/* ADC */ + .name = "ADS117X ADC", + .id = 1, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 32, + .rates = ADS117X_RATES, + .formats = ADS117X_FORMATS,}, +}; +EXPORT_SYMBOL_GPL(ads117x_dai); + +static int ads117x_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->card->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->name = "ADS117X"; + codec->owner = THIS_MODULE; + codec->dai = &ads117x_dai; + codec->num_dai = 1; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ads117x: failed to create pcms\n"); + kfree(codec); + return ret; + } + + return 0; +} + +static int ads117x_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_free_pcms(socdev); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ads117x = { + .probe = ads117x_probe, + .remove = ads117x_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ads117x); + +static __devinit int ads117x_platform_probe(struct platform_device *pdev) +{ + ads117x_dai.dev = &pdev->dev; + return snd_soc_register_dai(&ads117x_dai); +} + +static int __devexit ads117x_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&ads117x_dai); + return 0; +} + +static struct platform_driver ads117x_codec_driver = { + .driver = { + .name = "ads117x", + .owner = THIS_MODULE, + }, + + .probe = ads117x_platform_probe, + .remove = __devexit_p(ads117x_platform_remove), +}; + +static int __init ads117x_init(void) +{ + return platform_driver_register(&ads117x_codec_driver); +} +module_init(ads117x_init); + +static void __exit ads117x_exit(void) +{ + platform_driver_unregister(&ads117x_codec_driver); +} +module_exit(ads117x_exit); + +MODULE_DESCRIPTION("ASoC ads117x driver"); +MODULE_AUTHOR("Graeme Gregory"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ads117x.h b/sound/soc/codecs/ads117x.h new file mode 100644 index 0000000..dbcf50e --- /dev/null +++ b/sound/soc/codecs/ads117x.h @@ -0,0 +1,13 @@ +/* + * ads117x.h -- Driver for ads1174/8 ADC chips + * + * Copyright 2009 ShotSpotter Inc. + * Author: Graeme Gregory <gg@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 + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +extern struct snd_soc_dai ads117x_dai; +extern struct snd_soc_codec_device soc_codec_dev_ads117x; diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 4d47bc4..3a14c6f 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -313,14 +313,6 @@ static int ak4104_probe(struct platform_device *pdev) return ret; } - /* Register the socdev */ - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card\n"); - snd_soc_free_pcms(socdev); - return ret; - } - return 0; } diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 0abec0d..ff96656 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -294,7 +294,6 @@ static int ak4535_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -485,17 +484,9 @@ static int ak4535_init(struct snd_soc_device *socdev) snd_soc_add_controls(codec, ak4535_snd_controls, ARRAY_SIZE(ak4535_snd_controls)); ak4535_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "ak4535: 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); diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index e057c7b..b69861d 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -442,18 +442,9 @@ static int ak4642_probe(struct platform_device *pdev) goto pcm_err; } - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "ak4642: failed to register card\n"); - goto card_err; - } - dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c new file mode 100644 index 0000000..82fca28 --- /dev/null +++ b/sound/soc/codecs/ak4671.c @@ -0,0 +1,815 @@ +/* + * ak4671.c -- audio driver for AK4671 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim@samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/delay.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "ak4671.h" + +static struct snd_soc_codec *ak4671_codec; + +/* codec private data */ +struct ak4671_priv { + struct snd_soc_codec codec; + u8 reg_cache[AK4671_CACHEREGNUM]; +}; + +/* ak4671 register cache & default register settings */ +static const u8 ak4671_reg[AK4671_CACHEREGNUM] = { + 0x00, /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) */ + 0xf6, /* AK4671_PLL_MODE_SELECT0 (0x01) */ + 0x00, /* AK4671_PLL_MODE_SELECT1 (0x02) */ + 0x02, /* AK4671_FORMAT_SELECT (0x03) */ + 0x00, /* AK4671_MIC_SIGNAL_SELECT (0x04) */ + 0x55, /* AK4671_MIC_AMP_GAIN (0x05) */ + 0x00, /* AK4671_MIXING_POWER_MANAGEMENT0 (0x06) */ + 0x00, /* AK4671_MIXING_POWER_MANAGEMENT1 (0x07) */ + 0xb5, /* AK4671_OUTPUT_VOLUME_CONTROL (0x08) */ + 0x00, /* AK4671_LOUT1_SIGNAL_SELECT (0x09) */ + 0x00, /* AK4671_ROUT1_SIGNAL_SELECT (0x0a) */ + 0x00, /* AK4671_LOUT2_SIGNAL_SELECT (0x0b) */ + 0x00, /* AK4671_ROUT2_SIGNAL_SELECT (0x0c) */ + 0x00, /* AK4671_LOUT3_SIGNAL_SELECT (0x0d) */ + 0x00, /* AK4671_ROUT3_SIGNAL_SELECT (0x0e) */ + 0x00, /* AK4671_LOUT1_POWER_MANAGERMENT (0x0f) */ + 0x00, /* AK4671_LOUT2_POWER_MANAGERMENT (0x10) */ + 0x80, /* AK4671_LOUT3_POWER_MANAGERMENT (0x11) */ + 0x91, /* AK4671_LCH_INPUT_VOLUME_CONTROL (0x12) */ + 0x91, /* AK4671_RCH_INPUT_VOLUME_CONTROL (0x13) */ + 0xe1, /* AK4671_ALC_REFERENCE_SELECT (0x14) */ + 0x00, /* AK4671_DIGITAL_MIXING_CONTROL (0x15) */ + 0x00, /* AK4671_ALC_TIMER_SELECT (0x16) */ + 0x00, /* AK4671_ALC_MODE_CONTROL (0x17) */ + 0x02, /* AK4671_MODE_CONTROL1 (0x18) */ + 0x01, /* AK4671_MODE_CONTROL2 (0x19) */ + 0x18, /* AK4671_LCH_OUTPUT_VOLUME_CONTROL (0x1a) */ + 0x18, /* AK4671_RCH_OUTPUT_VOLUME_CONTROL (0x1b) */ + 0x00, /* AK4671_SIDETONE_A_CONTROL (0x1c) */ + 0x02, /* AK4671_DIGITAL_FILTER_SELECT (0x1d) */ + 0x00, /* AK4671_FIL3_COEFFICIENT0 (0x1e) */ + 0x00, /* AK4671_FIL3_COEFFICIENT1 (0x1f) */ + 0x00, /* AK4671_FIL3_COEFFICIENT2 (0x20) */ + 0x00, /* AK4671_FIL3_COEFFICIENT3 (0x21) */ + 0x00, /* AK4671_EQ_COEFFICIENT0 (0x22) */ + 0x00, /* AK4671_EQ_COEFFICIENT1 (0x23) */ + 0x00, /* AK4671_EQ_COEFFICIENT2 (0x24) */ + 0x00, /* AK4671_EQ_COEFFICIENT3 (0x25) */ + 0x00, /* AK4671_EQ_COEFFICIENT4 (0x26) */ + 0x00, /* AK4671_EQ_COEFFICIENT5 (0x27) */ + 0xa9, /* AK4671_FIL1_COEFFICIENT0 (0x28) */ + 0x1f, /* AK4671_FIL1_COEFFICIENT1 (0x29) */ + 0xad, /* AK4671_FIL1_COEFFICIENT2 (0x2a) */ + 0x20, /* AK4671_FIL1_COEFFICIENT3 (0x2b) */ + 0x00, /* AK4671_FIL2_COEFFICIENT0 (0x2c) */ + 0x00, /* AK4671_FIL2_COEFFICIENT1 (0x2d) */ + 0x00, /* AK4671_FIL2_COEFFICIENT2 (0x2e) */ + 0x00, /* AK4671_FIL2_COEFFICIENT3 (0x2f) */ + 0x00, /* AK4671_DIGITAL_FILTER_SELECT2 (0x30) */ + 0x00, /* this register not used */ + 0x00, /* AK4671_E1_COEFFICIENT0 (0x32) */ + 0x00, /* AK4671_E1_COEFFICIENT1 (0x33) */ + 0x00, /* AK4671_E1_COEFFICIENT2 (0x34) */ + 0x00, /* AK4671_E1_COEFFICIENT3 (0x35) */ + 0x00, /* AK4671_E1_COEFFICIENT4 (0x36) */ + 0x00, /* AK4671_E1_COEFFICIENT5 (0x37) */ + 0x00, /* AK4671_E2_COEFFICIENT0 (0x38) */ + 0x00, /* AK4671_E2_COEFFICIENT1 (0x39) */ + 0x00, /* AK4671_E2_COEFFICIENT2 (0x3a) */ + 0x00, /* AK4671_E2_COEFFICIENT3 (0x3b) */ + 0x00, /* AK4671_E2_COEFFICIENT4 (0x3c) */ + 0x00, /* AK4671_E2_COEFFICIENT5 (0x3d) */ + 0x00, /* AK4671_E3_COEFFICIENT0 (0x3e) */ + 0x00, /* AK4671_E3_COEFFICIENT1 (0x3f) */ + 0x00, /* AK4671_E3_COEFFICIENT2 (0x40) */ + 0x00, /* AK4671_E3_COEFFICIENT3 (0x41) */ + 0x00, /* AK4671_E3_COEFFICIENT4 (0x42) */ + 0x00, /* AK4671_E3_COEFFICIENT5 (0x43) */ + 0x00, /* AK4671_E4_COEFFICIENT0 (0x44) */ + 0x00, /* AK4671_E4_COEFFICIENT1 (0x45) */ + 0x00, /* AK4671_E4_COEFFICIENT2 (0x46) */ + 0x00, /* AK4671_E4_COEFFICIENT3 (0x47) */ + 0x00, /* AK4671_E4_COEFFICIENT4 (0x48) */ + 0x00, /* AK4671_E4_COEFFICIENT5 (0x49) */ + 0x00, /* AK4671_E5_COEFFICIENT0 (0x4a) */ + 0x00, /* AK4671_E5_COEFFICIENT1 (0x4b) */ + 0x00, /* AK4671_E5_COEFFICIENT2 (0x4c) */ + 0x00, /* AK4671_E5_COEFFICIENT3 (0x4d) */ + 0x00, /* AK4671_E5_COEFFICIENT4 (0x4e) */ + 0x00, /* AK4671_E5_COEFFICIENT5 (0x4f) */ + 0x88, /* AK4671_EQ_CONTROL_250HZ_100HZ (0x50) */ + 0x88, /* AK4671_EQ_CONTROL_3500HZ_1KHZ (0x51) */ + 0x08, /* AK4671_EQ_CONTRO_10KHZ (0x52) */ + 0x00, /* AK4671_PCM_IF_CONTROL0 (0x53) */ + 0x00, /* AK4671_PCM_IF_CONTROL1 (0x54) */ + 0x00, /* AK4671_PCM_IF_CONTROL2 (0x55) */ + 0x18, /* AK4671_DIGITAL_VOLUME_B_CONTROL (0x56) */ + 0x18, /* AK4671_DIGITAL_VOLUME_C_CONTROL (0x57) */ + 0x00, /* AK4671_SIDETONE_VOLUME_CONTROL (0x58) */ + 0x00, /* AK4671_DIGITAL_MIXING_CONTROL2 (0x59) */ + 0x00, /* AK4671_SAR_ADC_CONTROL (0x5a) */ +}; + +/* + * LOUT1/ROUT1 output volume control: + * from -24 to 6 dB in 6 dB steps (mute instead of -30 dB) + */ +static DECLARE_TLV_DB_SCALE(out1_tlv, -3000, 600, 1); + +/* + * LOUT2/ROUT2 output volume control: + * from -33 to 6 dB in 3 dB steps (mute instead of -33 dB) + */ +static DECLARE_TLV_DB_SCALE(out2_tlv, -3300, 300, 1); + +/* + * LOUT3/ROUT3 output volume control: + * from -6 to 3 dB in 3 dB steps + */ +static DECLARE_TLV_DB_SCALE(out3_tlv, -600, 300, 0); + +/* + * Mic amp gain control: + * from -15 to 30 dB in 3 dB steps + * REVISIT: The actual min value(0x01) is -12 dB and the reg value 0x00 is not + * available + */ +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -1500, 300, 0); + +static const struct snd_kcontrol_new ak4671_snd_controls[] = { + /* Common playback gain controls */ + SOC_SINGLE_TLV("Line Output1 Playback Volume", + AK4671_OUTPUT_VOLUME_CONTROL, 0, 0x6, 0, out1_tlv), + SOC_SINGLE_TLV("Headphone Output2 Playback Volume", + AK4671_OUTPUT_VOLUME_CONTROL, 4, 0xd, 0, out2_tlv), + SOC_SINGLE_TLV("Line Output3 Playback Volume", + AK4671_LOUT3_POWER_MANAGERMENT, 6, 0x3, 0, out3_tlv), + + /* Common capture gain controls */ + SOC_DOUBLE_TLV("Mic Amp Capture Volume", + AK4671_MIC_AMP_GAIN, 0, 4, 0xf, 0, mic_amp_tlv), +}; + +/* event handlers */ +static int ak4671_out2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u8 reg; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); + reg |= AK4671_MUTEN; + snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + break; + case SND_SOC_DAPM_PRE_PMD: + reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); + reg &= ~AK4671_MUTEN; + snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + break; + } + + return 0; +} + +/* Output Mixers */ +static const struct snd_kcontrol_new ak4671_lout1_mixer_controls[] = { + SOC_DAPM_SINGLE("DACL", AK4671_LOUT1_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINL1", AK4671_LOUT1_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINL2", AK4671_LOUT1_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINL3", AK4671_LOUT1_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINL4", AK4671_LOUT1_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPL", AK4671_LOUT1_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout1_mixer_controls[] = { + SOC_DAPM_SINGLE("DACR", AK4671_ROUT1_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINR1", AK4671_ROUT1_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINR2", AK4671_ROUT1_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINR3", AK4671_ROUT1_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINR4", AK4671_ROUT1_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPR", AK4671_ROUT1_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_lout2_mixer_controls[] = { + SOC_DAPM_SINGLE("DACHL", AK4671_LOUT2_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINH1", AK4671_LOUT2_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINH2", AK4671_LOUT2_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINH3", AK4671_LOUT2_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINH4", AK4671_LOUT2_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPHL", AK4671_LOUT2_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout2_mixer_controls[] = { + SOC_DAPM_SINGLE("DACHR", AK4671_ROUT2_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINH1", AK4671_ROUT2_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINH2", AK4671_ROUT2_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINH3", AK4671_ROUT2_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINH4", AK4671_ROUT2_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPHR", AK4671_ROUT2_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_lout3_mixer_controls[] = { + SOC_DAPM_SINGLE("DACSL", AK4671_LOUT3_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINS1", AK4671_LOUT3_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINS2", AK4671_LOUT3_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINS3", AK4671_LOUT3_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINS4", AK4671_LOUT3_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPSL", AK4671_LOUT3_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout3_mixer_controls[] = { + SOC_DAPM_SINGLE("DACSR", AK4671_ROUT3_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINS1", AK4671_ROUT3_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINS2", AK4671_ROUT3_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINS3", AK4671_ROUT3_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINS4", AK4671_ROUT3_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPSR", AK4671_ROUT3_SIGNAL_SELECT, 5, 1, 0), +}; + +/* Input MUXs */ +static const char *ak4671_lin_mux_texts[] = + {"LIN1", "LIN2", "LIN3", "LIN4"}; +static const struct soc_enum ak4671_lin_mux_enum = + SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 0, + ARRAY_SIZE(ak4671_lin_mux_texts), + ak4671_lin_mux_texts); +static const struct snd_kcontrol_new ak4671_lin_mux_control = + SOC_DAPM_ENUM("Route", ak4671_lin_mux_enum); + +static const char *ak4671_rin_mux_texts[] = + {"RIN1", "RIN2", "RIN3", "RIN4"}; +static const struct soc_enum ak4671_rin_mux_enum = + SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 2, + ARRAY_SIZE(ak4671_rin_mux_texts), + ak4671_rin_mux_texts); +static const struct snd_kcontrol_new ak4671_rin_mux_control = + SOC_DAPM_ENUM("Route", ak4671_rin_mux_enum); + +static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("LIN1"), + SND_SOC_DAPM_INPUT("RIN1"), + SND_SOC_DAPM_INPUT("LIN2"), + SND_SOC_DAPM_INPUT("RIN2"), + SND_SOC_DAPM_INPUT("LIN3"), + SND_SOC_DAPM_INPUT("RIN3"), + SND_SOC_DAPM_INPUT("LIN4"), + SND_SOC_DAPM_INPUT("RIN4"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + SND_SOC_DAPM_OUTPUT("LOUT3"), + SND_SOC_DAPM_OUTPUT("ROUT3"), + + /* DAC */ + SND_SOC_DAPM_DAC("DAC Left", "Left HiFi Playback", + AK4671_AD_DA_POWER_MANAGEMENT, 6, 0), + SND_SOC_DAPM_DAC("DAC Right", "Right HiFi Playback", + AK4671_AD_DA_POWER_MANAGEMENT, 7, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC Left", "Left HiFi Capture", + AK4671_AD_DA_POWER_MANAGEMENT, 4, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right HiFi Capture", + AK4671_AD_DA_POWER_MANAGEMENT, 5, 0), + + /* PGA */ + SND_SOC_DAPM_PGA("LOUT2 Mix Amp", + AK4671_LOUT2_POWER_MANAGERMENT, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("ROUT2 Mix Amp", + AK4671_LOUT2_POWER_MANAGERMENT, 6, 0, NULL, 0), + + SND_SOC_DAPM_PGA("LIN1 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN1 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN2 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN2 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN3 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN3 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN4 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN4 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 7, 0, NULL, 0), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("LOUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 0, 0, + &ak4671_lout1_mixer_controls[0], + ARRAY_SIZE(ak4671_lout1_mixer_controls)), + SND_SOC_DAPM_MIXER("ROUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 1, 0, + &ak4671_rout1_mixer_controls[0], + ARRAY_SIZE(ak4671_rout1_mixer_controls)), + SND_SOC_DAPM_MIXER_E("LOUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT, + 0, 0, &ak4671_lout2_mixer_controls[0], + ARRAY_SIZE(ak4671_lout2_mixer_controls), + ak4671_out2_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MIXER_E("ROUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT, + 1, 0, &ak4671_rout2_mixer_controls[0], + ARRAY_SIZE(ak4671_rout2_mixer_controls), + ak4671_out2_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MIXER("LOUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 0, 0, + &ak4671_lout3_mixer_controls[0], + ARRAY_SIZE(ak4671_lout3_mixer_controls)), + SND_SOC_DAPM_MIXER("ROUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 1, 0, + &ak4671_rout3_mixer_controls[0], + ARRAY_SIZE(ak4671_rout3_mixer_controls)), + + /* Input MUXs */ + SND_SOC_DAPM_MUX("LIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 2, 0, + &ak4671_lin_mux_control), + SND_SOC_DAPM_MUX("RIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 3, 0, + &ak4671_rin_mux_control), + + /* Mic Power */ + SND_SOC_DAPM_MICBIAS("Mic Bias", AK4671_AD_DA_POWER_MANAGEMENT, 1, 0), + + /* Supply */ + SND_SOC_DAPM_SUPPLY("PMPLL", AK4671_PLL_MODE_SELECT1, 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"DAC Left", "NULL", "PMPLL"}, + {"DAC Right", "NULL", "PMPLL"}, + {"ADC Left", "NULL", "PMPLL"}, + {"ADC Right", "NULL", "PMPLL"}, + + /* Outputs */ + {"LOUT1", "NULL", "LOUT1 Mixer"}, + {"ROUT1", "NULL", "ROUT1 Mixer"}, + {"LOUT2", "NULL", "LOUT2 Mix Amp"}, + {"ROUT2", "NULL", "ROUT2 Mix Amp"}, + {"LOUT3", "NULL", "LOUT3 Mixer"}, + {"ROUT3", "NULL", "ROUT3 Mixer"}, + + {"LOUT1 Mixer", "DACL", "DAC Left"}, + {"ROUT1 Mixer", "DACR", "DAC Right"}, + {"LOUT2 Mixer", "DACHL", "DAC Left"}, + {"ROUT2 Mixer", "DACHR", "DAC Right"}, + {"LOUT2 Mix Amp", "NULL", "LOUT2 Mixer"}, + {"ROUT2 Mix Amp", "NULL", "ROUT2 Mixer"}, + {"LOUT3 Mixer", "DACSL", "DAC Left"}, + {"ROUT3 Mixer", "DACSR", "DAC Right"}, + + /* Inputs */ + {"LIN MUX", "LIN1", "LIN1"}, + {"LIN MUX", "LIN2", "LIN2"}, + {"LIN MUX", "LIN3", "LIN3"}, + {"LIN MUX", "LIN4", "LIN4"}, + + {"RIN MUX", "RIN1", "RIN1"}, + {"RIN MUX", "RIN2", "RIN2"}, + {"RIN MUX", "RIN3", "RIN3"}, + {"RIN MUX", "RIN4", "RIN4"}, + + {"LIN1", NULL, "Mic Bias"}, + {"RIN1", NULL, "Mic Bias"}, + {"LIN2", NULL, "Mic Bias"}, + {"RIN2", NULL, "Mic Bias"}, + + {"ADC Left", "NULL", "LIN MUX"}, + {"ADC Right", "NULL", "RIN MUX"}, + + /* Analog Loops */ + {"LIN1 Mixing Circuit", "NULL", "LIN1"}, + {"RIN1 Mixing Circuit", "NULL", "RIN1"}, + {"LIN2 Mixing Circuit", "NULL", "LIN2"}, + {"RIN2 Mixing Circuit", "NULL", "RIN2"}, + {"LIN3 Mixing Circuit", "NULL", "LIN3"}, + {"RIN3 Mixing Circuit", "NULL", "RIN3"}, + {"LIN4 Mixing Circuit", "NULL", "LIN4"}, + {"RIN4 Mixing Circuit", "NULL", "RIN4"}, + + {"LOUT1 Mixer", "LINL1", "LIN1 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR1", "RIN1 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH1", "LIN1 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH1", "RIN1 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS1", "LIN1 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS1", "RIN1 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL2", "LIN2 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR2", "RIN2 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH2", "LIN2 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH2", "RIN2 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS2", "LIN2 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS2", "RIN2 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL3", "LIN3 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR3", "RIN3 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH3", "LIN3 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH3", "RIN3 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS3", "LIN3 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS3", "RIN3 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL4", "LIN4 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR4", "RIN4 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH4", "LIN4 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH4", "RIN4 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS4", "LIN4 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS4", "RIN4 Mixing Circuit"}, +}; + +static int ak4671_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, ak4671_dapm_widgets, + ARRAY_SIZE(ak4671_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + return 0; +} + +static int ak4671_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + u8 fs; + + fs = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0); + fs &= ~AK4671_FS; + + switch (params_rate(params)) { + case 8000: + fs |= AK4671_FS_8KHZ; + break; + case 12000: + fs |= AK4671_FS_12KHZ; + break; + case 16000: + fs |= AK4671_FS_16KHZ; + break; + case 24000: + fs |= AK4671_FS_24KHZ; + break; + case 11025: + fs |= AK4671_FS_11_025KHZ; + break; + case 22050: + fs |= AK4671_FS_22_05KHZ; + break; + case 32000: + fs |= AK4671_FS_32KHZ; + break; + case 44100: + fs |= AK4671_FS_44_1KHZ; + break; + case 48000: + fs |= AK4671_FS_48KHZ; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, fs); + + return 0; +} + +static int ak4671_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + u8 pll; + + pll = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0); + pll &= ~AK4671_PLL; + + switch (freq) { + case 11289600: + pll |= AK4671_PLL_11_2896MHZ; + break; + case 12000000: + pll |= AK4671_PLL_12MHZ; + break; + case 12288000: + pll |= AK4671_PLL_12_288MHZ; + break; + case 13000000: + pll |= AK4671_PLL_13MHZ; + break; + case 13500000: + pll |= AK4671_PLL_13_5MHZ; + break; + case 19200000: + pll |= AK4671_PLL_19_2MHZ; + break; + case 24000000: + pll |= AK4671_PLL_24MHZ; + break; + case 26000000: + pll |= AK4671_PLL_26MHZ; + break; + case 27000000: + pll |= AK4671_PLL_27MHZ; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, pll); + + return 0; +} + +static int ak4671_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mode; + u8 format; + + /* set master/slave audio interface */ + mode = snd_soc_read(codec, AK4671_PLL_MODE_SELECT1); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mode |= AK4671_M_S; + break; + case SND_SOC_DAIFMT_CBM_CFS: + mode &= ~(AK4671_M_S); + break; + default: + return -EINVAL; + } + + /* interface format */ + format = snd_soc_read(codec, AK4671_FORMAT_SELECT); + format &= ~AK4671_DIF; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + format |= AK4671_DIF_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + format |= AK4671_DIF_MSB_MODE; + break; + case SND_SOC_DAIFMT_DSP_A: + format |= AK4671_DIF_DSP_MODE; + format |= AK4671_BCKP; + format |= AK4671_MSBS; + break; + default: + return -EINVAL; + } + + /* set mode and format */ + snd_soc_write(codec, AK4671_PLL_MODE_SELECT1, mode); + snd_soc_write(codec, AK4671_FORMAT_SELECT, format); + + return 0; +} + +static int ak4671_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u8 reg; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + reg = snd_soc_read(codec, AK4671_AD_DA_POWER_MANAGEMENT); + snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, + reg | AK4671_PMVCM); + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00); + break; + } + codec->bias_level = level; + return 0; +} + +#define AK4671_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) + +#define AK4671_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +static struct snd_soc_dai_ops ak4671_dai_ops = { + .hw_params = ak4671_hw_params, + .set_sysclk = ak4671_set_dai_sysclk, + .set_fmt = ak4671_set_dai_fmt, +}; + +struct snd_soc_dai ak4671_dai = { + .name = "AK4671", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AK4671_RATES, + .formats = AK4671_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AK4671_RATES, + .formats = AK4671_FORMATS,}, + .ops = &ak4671_dai_ops, +}; +EXPORT_SYMBOL_GPL(ak4671_dai); + +static int ak4671_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (ak4671_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ak4671_codec; + codec = ak4671_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, ak4671_snd_controls, + ARRAY_SIZE(ak4671_snd_controls)); + ak4671_add_widgets(codec); + + ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return ret; + +pcm_err: + return ret; +} + +static int ak4671_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak4671 = { + .probe = ak4671_probe, + .remove = ak4671_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4671); + +static int ak4671_register(struct ak4671_priv *ak4671, + enum snd_soc_control_type control) +{ + int ret; + struct snd_soc_codec *codec = &ak4671->codec; + + if (ak4671_codec) { + dev_err(codec->dev, "Another AK4671 is registered\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = ak4671; + codec->name = "AK4671"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = ak4671_set_bias_level; + codec->dai = &ak4671_dai; + codec->num_dai = 1; + codec->reg_cache_size = AK4671_CACHEREGNUM; + codec->reg_cache = &ak4671->reg_cache; + + memcpy(codec->reg_cache, ak4671_reg, sizeof(ak4671_reg)); + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, control); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + ak4671_dai.dev = codec->dev; + ak4671_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&ak4671_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(ak4671); + return ret; +} + +static void ak4671_unregister(struct ak4671_priv *ak4671) +{ + ak4671_set_bias_level(&ak4671->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&ak4671_dai); + snd_soc_unregister_codec(&ak4671->codec); + kfree(ak4671); + ak4671_codec = NULL; +} + +static int __devinit ak4671_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct ak4671_priv *ak4671; + struct snd_soc_codec *codec; + + ak4671 = kzalloc(sizeof(struct ak4671_priv), GFP_KERNEL); + if (ak4671 == NULL) + return -ENOMEM; + + codec = &ak4671->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(client, ak4671); + codec->control_data = client; + + codec->dev = &client->dev; + + return ak4671_register(ak4671, SND_SOC_I2C); +} + +static __devexit int ak4671_i2c_remove(struct i2c_client *client) +{ + struct ak4671_priv *ak4671 = i2c_get_clientdata(client); + + ak4671_unregister(ak4671); + + return 0; +} + +static const struct i2c_device_id ak4671_i2c_id[] = { + { "ak4671", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4671_i2c_id); + +static struct i2c_driver ak4671_i2c_driver = { + .driver = { + .name = "ak4671", + .owner = THIS_MODULE, + }, + .probe = ak4671_i2c_probe, + .remove = __devexit_p(ak4671_i2c_remove), + .id_table = ak4671_i2c_id, +}; + +static int __init ak4671_modinit(void) +{ + return i2c_add_driver(&ak4671_i2c_driver); +} +module_init(ak4671_modinit); + +static void __exit ak4671_exit(void) +{ + i2c_del_driver(&ak4671_i2c_driver); +} +module_exit(ak4671_exit); + +MODULE_DESCRIPTION("ASoC AK4671 codec driver"); +MODULE_AUTHOR("Joonyoung Shim <jy0922.shim@samsung.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4671.h b/sound/soc/codecs/ak4671.h new file mode 100644 index 0000000..e2fad96 --- /dev/null +++ b/sound/soc/codecs/ak4671.h @@ -0,0 +1,156 @@ +/* + * ak4671.h -- audio driver for AK4671 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim@samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef _AK4671_H +#define _AK4671_H + +#define AK4671_AD_DA_POWER_MANAGEMENT 0x00 +#define AK4671_PLL_MODE_SELECT0 0x01 +#define AK4671_PLL_MODE_SELECT1 0x02 +#define AK4671_FORMAT_SELECT 0x03 +#define AK4671_MIC_SIGNAL_SELECT 0x04 +#define AK4671_MIC_AMP_GAIN 0x05 +#define AK4671_MIXING_POWER_MANAGEMENT0 0x06 +#define AK4671_MIXING_POWER_MANAGEMENT1 0x07 +#define AK4671_OUTPUT_VOLUME_CONTROL 0x08 +#define AK4671_LOUT1_SIGNAL_SELECT 0x09 +#define AK4671_ROUT1_SIGNAL_SELECT 0x0a +#define AK4671_LOUT2_SIGNAL_SELECT 0x0b +#define AK4671_ROUT2_SIGNAL_SELECT 0x0c +#define AK4671_LOUT3_SIGNAL_SELECT 0x0d +#define AK4671_ROUT3_SIGNAL_SELECT 0x0e +#define AK4671_LOUT1_POWER_MANAGERMENT 0x0f +#define AK4671_LOUT2_POWER_MANAGERMENT 0x10 +#define AK4671_LOUT3_POWER_MANAGERMENT 0x11 +#define AK4671_LCH_INPUT_VOLUME_CONTROL 0x12 +#define AK4671_RCH_INPUT_VOLUME_CONTROL 0x13 +#define AK4671_ALC_REFERENCE_SELECT 0x14 +#define AK4671_DIGITAL_MIXING_CONTROL 0x15 +#define AK4671_ALC_TIMER_SELECT 0x16 +#define AK4671_ALC_MODE_CONTROL 0x17 +#define AK4671_MODE_CONTROL1 0x18 +#define AK4671_MODE_CONTROL2 0x19 +#define AK4671_LCH_OUTPUT_VOLUME_CONTROL 0x1a +#define AK4671_RCH_OUTPUT_VOLUME_CONTROL 0x1b +#define AK4671_SIDETONE_A_CONTROL 0x1c +#define AK4671_DIGITAL_FILTER_SELECT 0x1d +#define AK4671_FIL3_COEFFICIENT0 0x1e +#define AK4671_FIL3_COEFFICIENT1 0x1f +#define AK4671_FIL3_COEFFICIENT2 0x20 +#define AK4671_FIL3_COEFFICIENT3 0x21 +#define AK4671_EQ_COEFFICIENT0 0x22 +#define AK4671_EQ_COEFFICIENT1 0x23 +#define AK4671_EQ_COEFFICIENT2 0x24 +#define AK4671_EQ_COEFFICIENT3 0x25 +#define AK4671_EQ_COEFFICIENT4 0x26 +#define AK4671_EQ_COEFFICIENT5 0x27 +#define AK4671_FIL1_COEFFICIENT0 0x28 +#define AK4671_FIL1_COEFFICIENT1 0x29 +#define AK4671_FIL1_COEFFICIENT2 0x2a +#define AK4671_FIL1_COEFFICIENT3 0x2b +#define AK4671_FIL2_COEFFICIENT0 0x2c +#define AK4671_FIL2_COEFFICIENT1 0x2d +#define AK4671_FIL2_COEFFICIENT2 0x2e +#define AK4671_FIL2_COEFFICIENT3 0x2f +#define AK4671_DIGITAL_FILTER_SELECT2 0x30 +#define AK4671_E1_COEFFICIENT0 0x32 +#define AK4671_E1_COEFFICIENT1 0x33 +#define AK4671_E1_COEFFICIENT2 0x34 +#define AK4671_E1_COEFFICIENT3 0x35 +#define AK4671_E1_COEFFICIENT4 0x36 +#define AK4671_E1_COEFFICIENT5 0x37 +#define AK4671_E2_COEFFICIENT0 0x38 +#define AK4671_E2_COEFFICIENT1 0x39 +#define AK4671_E2_COEFFICIENT2 0x3a +#define AK4671_E2_COEFFICIENT3 0x3b +#define AK4671_E2_COEFFICIENT4 0x3c +#define AK4671_E2_COEFFICIENT5 0x3d +#define AK4671_E3_COEFFICIENT0 0x3e +#define AK4671_E3_COEFFICIENT1 0x3f +#define AK4671_E3_COEFFICIENT2 0x40 +#define AK4671_E3_COEFFICIENT3 0x41 +#define AK4671_E3_COEFFICIENT4 0x42 +#define AK4671_E3_COEFFICIENT5 0x43 +#define AK4671_E4_COEFFICIENT0 0x44 +#define AK4671_E4_COEFFICIENT1 0x45 +#define AK4671_E4_COEFFICIENT2 0x46 +#define AK4671_E4_COEFFICIENT3 0x47 +#define AK4671_E4_COEFFICIENT4 0x48 +#define AK4671_E4_COEFFICIENT5 0x49 +#define AK4671_E5_COEFFICIENT0 0x4a +#define AK4671_E5_COEFFICIENT1 0x4b +#define AK4671_E5_COEFFICIENT2 0x4c +#define AK4671_E5_COEFFICIENT3 0x4d +#define AK4671_E5_COEFFICIENT4 0x4e +#define AK4671_E5_COEFFICIENT5 0x4f +#define AK4671_EQ_CONTROL_250HZ_100HZ 0x50 +#define AK4671_EQ_CONTROL_3500HZ_1KHZ 0x51 +#define AK4671_EQ_CONTRO_10KHZ 0x52 +#define AK4671_PCM_IF_CONTROL0 0x53 +#define AK4671_PCM_IF_CONTROL1 0x54 +#define AK4671_PCM_IF_CONTROL2 0x55 +#define AK4671_DIGITAL_VOLUME_B_CONTROL 0x56 +#define AK4671_DIGITAL_VOLUME_C_CONTROL 0x57 +#define AK4671_SIDETONE_VOLUME_CONTROL 0x58 +#define AK4671_DIGITAL_MIXING_CONTROL2 0x59 +#define AK4671_SAR_ADC_CONTROL 0x5a + +#define AK4671_CACHEREGNUM (AK4671_SAR_ADC_CONTROL + 1) + +/* Bitfield Definitions */ + +/* AK4671_AD_DA_POWER_MANAGEMENT (0x00) Fields */ +#define AK4671_PMVCM 0x01 + +/* AK4671_PLL_MODE_SELECT0 (0x01) Fields */ +#define AK4671_PLL 0x0f +#define AK4671_PLL_11_2896MHZ (4 << 0) +#define AK4671_PLL_12_288MHZ (5 << 0) +#define AK4671_PLL_12MHZ (6 << 0) +#define AK4671_PLL_24MHZ (7 << 0) +#define AK4671_PLL_19_2MHZ (8 << 0) +#define AK4671_PLL_13_5MHZ (12 << 0) +#define AK4671_PLL_27MHZ (13 << 0) +#define AK4671_PLL_13MHZ (14 << 0) +#define AK4671_PLL_26MHZ (15 << 0) +#define AK4671_FS 0xf0 +#define AK4671_FS_8KHZ (0 << 4) +#define AK4671_FS_12KHZ (1 << 4) +#define AK4671_FS_16KHZ (2 << 4) +#define AK4671_FS_24KHZ (3 << 4) +#define AK4671_FS_11_025KHZ (5 << 4) +#define AK4671_FS_22_05KHZ (7 << 4) +#define AK4671_FS_32KHZ (10 << 4) +#define AK4671_FS_48KHZ (11 << 4) +#define AK4671_FS_44_1KHZ (15 << 4) + +/* AK4671_PLL_MODE_SELECT1 (0x02) Fields */ +#define AK4671_PMPLL 0x01 +#define AK4671_M_S 0x02 + +/* AK4671_FORMAT_SELECT (0x03) Fields */ +#define AK4671_DIF 0x03 +#define AK4671_DIF_DSP_MODE (0 << 0) +#define AK4671_DIF_MSB_MODE (2 << 0) +#define AK4671_DIF_I2S_MODE (3 << 0) +#define AK4671_BCKP 0x04 +#define AK4671_MSBS 0x08 +#define AK4671_SDOD 0x10 + +/* AK4671_LOUT2_POWER_MANAGEMENT (0x10) Fields */ +#define AK4671_MUTEN 0x04 + +extern struct snd_soc_dai ak4671_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4671; + +#endif diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index ca1e24a..ffe122d 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -520,6 +520,7 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { SOC_SINGLE("Digital Sidetone Switch", CS4270_FORMAT, 5, 1, 0), SOC_SINGLE("Soft Ramp Switch", CS4270_TRANS, 6, 1, 0), SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0), + SOC_SINGLE("De-emphasis filter", CS4270_TRANS, 0, 1, 0), SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1), SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0), SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1), @@ -598,13 +599,6 @@ static int cs4270_probe(struct platform_device *pdev) goto error_free_pcms; } - /* And finally, register the socdev */ - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card\n"); - goto error_free_pcms; - } - return 0; error_free_pcms: @@ -802,22 +796,6 @@ MODULE_DEVICE_TABLE(i2c, cs4270_id); * and all registers are written back to the hardware when resuming. */ -static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg) -{ - struct cs4270_private *cs4270 = i2c_get_clientdata(client); - struct snd_soc_codec *codec = &cs4270->codec; - - return snd_soc_suspend_device(codec->dev); -} - -static int cs4270_i2c_resume(struct i2c_client *client) -{ - struct cs4270_private *cs4270 = i2c_get_clientdata(client); - struct snd_soc_codec *codec = &cs4270->codec; - - return snd_soc_resume_device(codec->dev); -} - static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg) { struct snd_soc_codec *codec = cs4270_codec; @@ -853,8 +831,6 @@ static int cs4270_soc_resume(struct platform_device *pdev) return snd_soc_write(codec, CS4270_PWRCTL, reg); } #else -#define cs4270_i2c_suspend NULL -#define cs4270_i2c_resume NULL #define cs4270_soc_suspend NULL #define cs4270_soc_resume NULL #endif /* CONFIG_PM */ @@ -873,8 +849,6 @@ static struct i2c_driver cs4270_i2c_driver = { .id_table = cs4270_id, .probe = cs4270_i2c_probe, .remove = cs4270_i2c_remove, - .suspend = cs4270_i2c_suspend, - .resume = cs4270_i2c_resume, }; /* diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index 38eac9c..e000cdf 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -93,7 +93,6 @@ static int cx20442_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, cx20442_audio_map, ARRAY_SIZE(cx20442_audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -355,17 +354,6 @@ static int cx20442_codec_probe(struct platform_device *pdev) cx20442_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&pdev->dev, "failed to register card\n"); - goto card_err; - } - - return ret; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 5cda9e6..2afcd0a 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -90,13 +90,6 @@ static int pcm3008_soc_probe(struct platform_device *pdev) goto pcm_err; } - /* Register Card. */ - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "pcm3008: failed to register card\n"); - goto card_err; - } - /* DEM1 DEM0 DE-EMPHASIS_MODE * Low Low De-emphasis 44.1 kHz ON * Low High De-emphasis OFF @@ -136,8 +129,6 @@ static int pcm3008_soc_probe(struct platform_device *pdev) gpio_err: pcm3008_gpio_free(setup); -card_err: - snd_soc_free_pcms(socdev); pcm_err: kfree(socdev->card->codec); diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index c550750..d2ff1cd 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -210,7 +210,6 @@ static int ssm2602_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_conn, ARRAY_SIZE(audio_conn)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -613,17 +612,9 @@ static int ssm2602_init(struct snd_soc_device *socdev) snd_soc_add_controls(codec, ssm2602_snd_controls, ARRAY_SIZE(ssm2602_snd_controls)); ssm2602_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - pr_err("ssm2602: 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; diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index befc648..bbc72c2 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -418,9 +418,6 @@ static int stac9766_codec_probe(struct platform_device *pdev) snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE(stac9766_snd_ac97_controls)); - ret = snd_soc_init_card(socdev); - if (ret < 0) - goto reset_err; return 0; reset_err: diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 35606ae..a9dc5fb 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -265,8 +265,8 @@ static const int bosr_usb_divisor_table[] = { #define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15)) static const unsigned short sr_valid_mask[] = { LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/ - LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/ LOWER_GROUP, /* Usb, bosr - 0*/ + LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/ UPPER_GROUP, /* Usb, bosr - 1*/ }; /* @@ -395,7 +395,6 @@ static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) /* set up audio path interconnects */ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -625,11 +624,10 @@ static int tlv320aic23_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - int i; u16 reg; /* Sync reg_cache with the hardware */ - for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) { + for (reg = 0; reg < TLV320AIC23_RESET; reg++) { u16 val = tlv320aic23_read_reg_cache(codec, reg); tlv320aic23_write(codec, reg, val); } @@ -707,17 +705,9 @@ static int tlv320aic23_init(struct snd_soc_device *socdev) snd_soc_add_controls(codec, tlv320aic23_snd_controls, ARRAY_SIZE(tlv320aic23_snd_controls)); tlv320aic23_add_widgets(codec); - ret = snd_soc_init_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; diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 3387d9e..357b609 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -356,18 +356,7 @@ static int aic26_probe(struct platform_device *pdev) ARRAY_SIZE(aic26_snd_controls)); WARN_ON(err < 0); - /* CODEC is setup, we can register the card now */ - dev_dbg(&pdev->dev, "Registering card\n"); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&pdev->dev, "aic26: failed to register card\n"); - goto card_err; - } return 0; - - card_err: - snd_soc_free_pcms(socdev); - return ret; } static int aic26_remove(struct platform_device *pdev) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 3395cf9..2b4dc2b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -753,7 +753,6 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) /* set up audio path interconnects */ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -1405,18 +1404,8 @@ static int aic3x_probe(struct platform_device *pdev) aic3x_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "aic3x: 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; diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c new file mode 100644 index 0000000..9c8903d --- /dev/null +++ b/sound/soc/codecs/tlv320dac33.c @@ -0,0 +1,1229 @@ +/* + * ALSA SoC Texas Instruments TLV320DAC33 codec driver + * + * Author: Peter Ujfalusi <peter.ujfalusi@nokia.com> + * + * Copyright: (C) 2009 Nokia Corporation + * + * 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 <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/interrupt.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include <sound/tlv320dac33-plat.h> +#include "tlv320dac33.h" + +#define DAC33_BUFFER_SIZE_BYTES 24576 /* bytes, 12288 16 bit words, + * 6144 stereo */ +#define DAC33_BUFFER_SIZE_SAMPLES 6144 + +#define NSAMPLE_MAX 5700 + +#define LATENCY_TIME_MS 20 + +static struct snd_soc_codec *tlv320dac33_codec; + +enum dac33_state { + DAC33_IDLE = 0, + DAC33_PREFILL, + DAC33_PLAYBACK, + DAC33_FLUSH, +}; + +struct tlv320dac33_priv { + struct mutex mutex; + struct workqueue_struct *dac33_wq; + struct work_struct work; + struct snd_soc_codec codec; + int power_gpio; + int chip_power; + int irq; + unsigned int refclk; + + unsigned int alarm_threshold; /* set to be half of LATENCY_TIME_MS */ + unsigned int nsample_min; /* nsample should not be lower than + * this */ + unsigned int nsample_max; /* nsample should not be higher than + * this */ + unsigned int nsample_switch; /* Use FIFO or bypass FIFO switch */ + unsigned int nsample; /* burst read amount from host */ + + enum dac33_state state; +}; + +static const u8 dac33_reg[DAC33_CACHEREGNUM] = { +0x00, 0x00, 0x00, 0x00, /* 0x00 - 0x03 */ +0x00, 0x00, 0x00, 0x00, /* 0x04 - 0x07 */ +0x00, 0x00, 0x00, 0x00, /* 0x08 - 0x0b */ +0x00, 0x00, 0x00, 0x00, /* 0x0c - 0x0f */ +0x00, 0x00, 0x00, 0x00, /* 0x10 - 0x13 */ +0x00, 0x00, 0x00, 0x00, /* 0x14 - 0x17 */ +0x00, 0x00, 0x00, 0x00, /* 0x18 - 0x1b */ +0x00, 0x00, 0x00, 0x00, /* 0x1c - 0x1f */ +0x00, 0x00, 0x00, 0x00, /* 0x20 - 0x23 */ +0x00, 0x00, 0x00, 0x00, /* 0x24 - 0x27 */ +0x00, 0x00, 0x00, 0x00, /* 0x28 - 0x2b */ +0x00, 0x00, 0x00, 0x80, /* 0x2c - 0x2f */ +0x80, 0x00, 0x00, 0x00, /* 0x30 - 0x33 */ +0x00, 0x00, 0x00, 0x00, /* 0x34 - 0x37 */ +0x00, 0x00, /* 0x38 - 0x39 */ +/* Registers 0x3a - 0x3f are reserved */ + 0x00, 0x00, /* 0x3a - 0x3b */ +0x00, 0x00, 0x00, 0x00, /* 0x3c - 0x3f */ + +0x00, 0x00, 0x00, 0x00, /* 0x40 - 0x43 */ +0x00, 0x80, /* 0x44 - 0x45 */ +/* Registers 0x46 - 0x47 are reserved */ + 0x80, 0x80, /* 0x46 - 0x47 */ + +0x80, 0x00, 0x00, /* 0x48 - 0x4a */ +/* Registers 0x4b - 0x7c are reserved */ + 0x00, /* 0x4b */ +0x00, 0x00, 0x00, 0x00, /* 0x4c - 0x4f */ +0x00, 0x00, 0x00, 0x00, /* 0x50 - 0x53 */ +0x00, 0x00, 0x00, 0x00, /* 0x54 - 0x57 */ +0x00, 0x00, 0x00, 0x00, /* 0x58 - 0x5b */ +0x00, 0x00, 0x00, 0x00, /* 0x5c - 0x5f */ +0x00, 0x00, 0x00, 0x00, /* 0x60 - 0x63 */ +0x00, 0x00, 0x00, 0x00, /* 0x64 - 0x67 */ +0x00, 0x00, 0x00, 0x00, /* 0x68 - 0x6b */ +0x00, 0x00, 0x00, 0x00, /* 0x6c - 0x6f */ +0x00, 0x00, 0x00, 0x00, /* 0x70 - 0x73 */ +0x00, 0x00, 0x00, 0x00, /* 0x74 - 0x77 */ +0x00, 0x00, 0x00, 0x00, /* 0x78 - 0x7b */ +0x00, /* 0x7c */ + + 0xda, 0x33, 0x03, /* 0x7d - 0x7f */ +}; + +/* Register read and write */ +static inline unsigned int dac33_read_reg_cache(struct snd_soc_codec *codec, + unsigned reg) +{ + u8 *cache = codec->reg_cache; + if (reg >= DAC33_CACHEREGNUM) + return 0; + + return cache[reg]; +} + +static inline void dac33_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + if (reg >= DAC33_CACHEREGNUM) + return; + + cache[reg] = value; +} + +static int dac33_read(struct snd_soc_codec *codec, unsigned int reg, + u8 *value) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + int val; + + *value = reg & 0xff; + + /* If powered off, return the cached value */ + if (dac33->chip_power) { + val = i2c_smbus_read_byte_data(codec->control_data, value[0]); + if (val < 0) { + dev_err(codec->dev, "Read failed (%d)\n", val); + value[0] = dac33_read_reg_cache(codec, reg); + } else { + value[0] = val; + dac33_write_reg_cache(codec, reg, val); + } + } else { + value[0] = dac33_read_reg_cache(codec, reg); + } + + return 0; +} + +static int dac33_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + u8 data[2]; + int ret = 0; + + /* + * data is + * D15..D8 dac33 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + dac33_write_reg_cache(codec, data[0], data[1]); + if (dac33->chip_power) { + ret = codec->hw_write(codec->control_data, data, 2); + if (ret != 2) + dev_err(codec->dev, "Write failed (%d)\n", ret); + else + ret = 0; + } + + return ret; +} + +static int dac33_write_locked(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + int ret; + + mutex_lock(&dac33->mutex); + ret = dac33_write(codec, reg, value); + mutex_unlock(&dac33->mutex); + + return ret; +} + +#define DAC33_I2C_ADDR_AUTOINC 0x80 +static int dac33_write16(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + u8 data[3]; + int ret = 0; + + /* + * data is + * D23..D16 dac33 register offset + * D15..D8 register data MSB + * D7...D0 register data LSB + */ + data[0] = reg & 0xff; + data[1] = (value >> 8) & 0xff; + data[2] = value & 0xff; + + dac33_write_reg_cache(codec, data[0], data[1]); + dac33_write_reg_cache(codec, data[0] + 1, data[2]); + + if (dac33->chip_power) { + /* We need to set autoincrement mode for 16 bit writes */ + data[0] |= DAC33_I2C_ADDR_AUTOINC; + ret = codec->hw_write(codec->control_data, data, 3); + if (ret != 3) + dev_err(codec->dev, "Write failed (%d)\n", ret); + else + ret = 0; + } + + return ret; +} + +static void dac33_restore_regs(struct snd_soc_codec *codec) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + u8 *cache = codec->reg_cache; + u8 data[2]; + int i, ret; + + if (!dac33->chip_power) + return; + + for (i = DAC33_PWR_CTRL; i <= DAC33_INTP_CTRL_B; i++) { + data[0] = i; + data[1] = cache[i]; + /* Skip the read only registers */ + if ((i >= DAC33_INT_OSC_STATUS && + i <= DAC33_INT_OSC_FREQ_RAT_READ_B) || + (i >= DAC33_FIFO_WPTR_MSB && i <= DAC33_FIFO_IRQ_FLAG) || + i == DAC33_DAC_STATUS_FLAGS || + i == DAC33_SRC_EST_REF_CLK_RATIO_A || + i == DAC33_SRC_EST_REF_CLK_RATIO_B) + continue; + ret = codec->hw_write(codec->control_data, data, 2); + if (ret != 2) + dev_err(codec->dev, "Write failed (%d)\n", ret); + } + for (i = DAC33_LDAC_PWR_CTRL; i <= DAC33_LINEL_TO_LLO_VOL; i++) { + data[0] = i; + data[1] = cache[i]; + ret = codec->hw_write(codec->control_data, data, 2); + if (ret != 2) + dev_err(codec->dev, "Write failed (%d)\n", ret); + } + for (i = DAC33_LINER_TO_RLO_VOL; i <= DAC33_OSC_TRIM; i++) { + data[0] = i; + data[1] = cache[i]; + ret = codec->hw_write(codec->control_data, data, 2); + if (ret != 2) + dev_err(codec->dev, "Write failed (%d)\n", ret); + } +} + +static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) +{ + u8 reg; + + reg = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); + if (power) + reg |= DAC33_PDNALLB; + else + reg &= ~DAC33_PDNALLB; + dac33_write(codec, DAC33_PWR_CTRL, reg); +} + +static void dac33_hard_power(struct snd_soc_codec *codec, int power) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + + mutex_lock(&dac33->mutex); + if (power) { + if (dac33->power_gpio >= 0) { + gpio_set_value(dac33->power_gpio, 1); + dac33->chip_power = 1; + /* Restore registers */ + dac33_restore_regs(codec); + } + dac33_soft_power(codec, 1); + } else { + dac33_soft_power(codec, 0); + if (dac33->power_gpio >= 0) { + gpio_set_value(dac33->power_gpio, 0); + dac33->chip_power = 0; + } + } + mutex_unlock(&dac33->mutex); + +} + +static int dac33_get_nsample(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + + ucontrol->value.integer.value[0] = dac33->nsample; + + return 0; +} + +static int dac33_set_nsample(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + int ret = 0; + + if (dac33->nsample == ucontrol->value.integer.value[0]) + return 0; + + if (ucontrol->value.integer.value[0] < dac33->nsample_min || + ucontrol->value.integer.value[0] > dac33->nsample_max) + ret = -EINVAL; + else + dac33->nsample = ucontrol->value.integer.value[0]; + + return ret; +} + +static int dac33_get_nsample_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + + ucontrol->value.integer.value[0] = dac33->nsample_switch; + + return 0; +} + +static int dac33_set_nsample_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + int ret = 0; + + if (dac33->nsample_switch == ucontrol->value.integer.value[0]) + return 0; + /* Do not allow changes while stream is running*/ + if (codec->active) + return -EPERM; + + if (ucontrol->value.integer.value[0] < 0 || + ucontrol->value.integer.value[0] > 1) + ret = -EINVAL; + else + dac33->nsample_switch = ucontrol->value.integer.value[0]; + + return ret; +} + +/* + * DACL/R digital volume control: + * from 0 dB to -63.5 in 0.5 dB steps + * Need to be inverted later on: + * 0x00 == 0 dB + * 0x7f == -63.5 dB + */ +static DECLARE_TLV_DB_SCALE(dac_digivol_tlv, -6350, 50, 0); + +static const struct snd_kcontrol_new dac33_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC Digital Playback Volume", + DAC33_LDAC_DIG_VOL_CTRL, DAC33_RDAC_DIG_VOL_CTRL, + 0, 0x7f, 1, dac_digivol_tlv), + SOC_DOUBLE_R("DAC Digital Playback Switch", + DAC33_LDAC_DIG_VOL_CTRL, DAC33_RDAC_DIG_VOL_CTRL, 7, 1, 1), + SOC_DOUBLE_R("Line to Line Out Volume", + DAC33_LINEL_TO_LLO_VOL, DAC33_LINER_TO_RLO_VOL, 0, 127, 1), +}; + +static const struct snd_kcontrol_new dac33_nsample_snd_controls[] = { + SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0, + dac33_get_nsample, dac33_set_nsample), + SOC_SINGLE_EXT("nSample Switch", 0, 0, 1, 0, + dac33_get_nsample_switch, dac33_set_nsample_switch), +}; + +/* Analog bypass */ +static const struct snd_kcontrol_new dac33_dapm_abypassl_control = + SOC_DAPM_SINGLE("Switch", DAC33_LINEL_TO_LLO_VOL, 7, 1, 1); + +static const struct snd_kcontrol_new dac33_dapm_abypassr_control = + SOC_DAPM_SINGLE("Switch", DAC33_LINER_TO_RLO_VOL, 7, 1, 1); + +static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("LEFT_LO"), + SND_SOC_DAPM_OUTPUT("RIGHT_LO"), + + SND_SOC_DAPM_INPUT("LINEL"), + SND_SOC_DAPM_INPUT("LINER"), + + SND_SOC_DAPM_DAC("DACL", "Left Playback", DAC33_LDAC_PWR_CTRL, 2, 0), + SND_SOC_DAPM_DAC("DACR", "Right Playback", DAC33_RDAC_PWR_CTRL, 2, 0), + + /* Analog bypass */ + SND_SOC_DAPM_SWITCH("Analog Left Bypass", SND_SOC_NOPM, 0, 0, + &dac33_dapm_abypassl_control), + SND_SOC_DAPM_SWITCH("Analog Right Bypass", SND_SOC_NOPM, 0, 0, + &dac33_dapm_abypassr_control), + + SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Left Amp Power", + DAC33_OUT_AMP_PWR_CTRL, 6, 3, 3, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Right Amp Power", + DAC33_OUT_AMP_PWR_CTRL, 4, 3, 3, 0), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Analog bypass */ + {"Analog Left Bypass", "Switch", "LINEL"}, + {"Analog Right Bypass", "Switch", "LINER"}, + + {"Output Left Amp Power", NULL, "DACL"}, + {"Output Right Amp Power", NULL, "DACR"}, + + {"Output Left Amp Power", NULL, "Analog Left Bypass"}, + {"Output Right Amp Power", NULL, "Analog Right Bypass"}, + + /* output */ + {"LEFT_LO", NULL, "Output Left Amp Power"}, + {"RIGHT_LO", NULL, "Output Right Amp Power"}, +}; + +static int dac33_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dac33_dapm_widgets, + ARRAY_SIZE(dac33_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + return 0; +} + +static int dac33_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + dac33_soft_power(codec, 1); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) + dac33_hard_power(codec, 1); + dac33_soft_power(codec, 0); + break; + case SND_SOC_BIAS_OFF: + dac33_hard_power(codec, 0); + break; + } + codec->bias_level = level; + + return 0; +} + +static void dac33_work(struct work_struct *work) +{ + struct snd_soc_codec *codec; + struct tlv320dac33_priv *dac33; + u8 reg; + + dac33 = container_of(work, struct tlv320dac33_priv, work); + codec = &dac33->codec; + + mutex_lock(&dac33->mutex); + switch (dac33->state) { + case DAC33_PREFILL: + dac33->state = DAC33_PLAYBACK; + dac33_write16(codec, DAC33_NSAMPLE_MSB, + DAC33_THRREG(dac33->nsample)); + dac33_write16(codec, DAC33_PREFILL_MSB, + DAC33_THRREG(dac33->alarm_threshold)); + break; + case DAC33_PLAYBACK: + dac33_write16(codec, DAC33_NSAMPLE_MSB, + DAC33_THRREG(dac33->nsample)); + break; + case DAC33_IDLE: + break; + case DAC33_FLUSH: + dac33->state = DAC33_IDLE; + /* Mask all interrupts from dac33 */ + dac33_write(codec, DAC33_FIFO_IRQ_MASK, 0); + + /* flush fifo */ + reg = dac33_read_reg_cache(codec, DAC33_FIFO_CTRL_A); + reg |= DAC33_FIFOFLUSH; + dac33_write(codec, DAC33_FIFO_CTRL_A, reg); + break; + } + mutex_unlock(&dac33->mutex); +} + +static irqreturn_t dac33_interrupt_handler(int irq, void *dev) +{ + struct snd_soc_codec *codec = dev; + struct tlv320dac33_priv *dac33 = codec->private_data; + + queue_work(dac33->dac33_wq, &dac33->work); + + return IRQ_HANDLED; +} + +static void dac33_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + unsigned int pwr_ctrl; + + /* Stop pending workqueue */ + if (dac33->nsample_switch) + cancel_work_sync(&dac33->work); + + mutex_lock(&dac33->mutex); + pwr_ctrl = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); + pwr_ctrl &= ~(DAC33_OSCPDNB | DAC33_DACRPDNB | DAC33_DACLPDNB); + dac33_write(codec, DAC33_PWR_CTRL, pwr_ctrl); + mutex_unlock(&dac33->mutex); +} + +static void dac33_oscwait(struct snd_soc_codec *codec) +{ + int timeout = 20; + u8 reg; + + do { + msleep(1); + dac33_read(codec, DAC33_INT_OSC_STATUS, ®); + } while (((reg & 0x03) != DAC33_OSCSTATUS_NORMAL) && timeout--); + if ((reg & 0x03) != DAC33_OSCSTATUS_NORMAL) + dev_err(codec->dev, + "internal oscillator calibration failed\n"); +} + +static int dac33_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + + /* Check parameters for validity */ + switch (params_rate(params)) { + case 44100: + case 48000: + break; + default: + dev_err(codec->dev, "unsupported rate %d\n", + params_rate(params)); + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + default: + dev_err(codec->dev, "unsupported format %d\n", + params_format(params)); + return -EINVAL; + } + + return 0; +} + +#define CALC_OSCSET(rate, refclk) ( \ + ((((rate * 10000) / refclk) * 4096) + 5000) / 10000) +#define CALC_RATIOSET(rate, refclk) ( \ + ((((refclk * 100000) / rate) * 16384) + 50000) / 100000) + +/* + * tlv320dac33 is strict on the sequence of the register writes, if the register + * writes happens in different order, than dac33 might end up in unknown state. + * Use the known, working sequence of register writes to initialize the dac33. + */ +static int dac33_prepare_chip(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->card->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + unsigned int oscset, ratioset, pwr_ctrl, reg_tmp; + u8 aictrl_a, fifoctrl_a; + + switch (substream->runtime->rate) { + case 44100: + case 48000: + oscset = CALC_OSCSET(substream->runtime->rate, dac33->refclk); + ratioset = CALC_RATIOSET(substream->runtime->rate, + dac33->refclk); + break; + default: + dev_err(codec->dev, "unsupported rate %d\n", + substream->runtime->rate); + return -EINVAL; + } + + + aictrl_a = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A); + aictrl_a &= ~(DAC33_NCYCL_MASK | DAC33_WLEN_MASK); + fifoctrl_a = dac33_read_reg_cache(codec, DAC33_FIFO_CTRL_A); + fifoctrl_a &= ~DAC33_WIDTH; + switch (substream->runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + aictrl_a |= (DAC33_NCYCL_16 | DAC33_WLEN_16); + fifoctrl_a |= DAC33_WIDTH; + break; + default: + dev_err(codec->dev, "unsupported format %d\n", + substream->runtime->format); + return -EINVAL; + } + + mutex_lock(&dac33->mutex); + dac33_soft_power(codec, 1); + + reg_tmp = dac33_read_reg_cache(codec, DAC33_INT_OSC_CTRL); + dac33_write(codec, DAC33_INT_OSC_CTRL, reg_tmp); + + /* Write registers 0x08 and 0x09 (MSB, LSB) */ + dac33_write16(codec, DAC33_INT_OSC_FREQ_RAT_A, oscset); + + /* calib time: 128 is a nice number ;) */ + dac33_write(codec, DAC33_CALIB_TIME, 128); + + /* adjustment treshold & step */ + dac33_write(codec, DAC33_INT_OSC_CTRL_B, DAC33_ADJTHRSHLD(2) | + DAC33_ADJSTEP(1)); + + /* div=4 / gain=1 / div */ + dac33_write(codec, DAC33_INT_OSC_CTRL_C, DAC33_REFDIV(4)); + + pwr_ctrl = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); + pwr_ctrl |= DAC33_OSCPDNB | DAC33_DACRPDNB | DAC33_DACLPDNB; + dac33_write(codec, DAC33_PWR_CTRL, pwr_ctrl); + + dac33_oscwait(codec); + + if (dac33->nsample_switch) { + /* 50-51 : ASRC Control registers */ + dac33_write(codec, DAC33_ASRC_CTRL_A, (1 << 4)); /* div=2 */ + dac33_write(codec, DAC33_ASRC_CTRL_B, 1); /* ??? */ + + /* Write registers 0x34 and 0x35 (MSB, LSB) */ + dac33_write16(codec, DAC33_SRC_REF_CLK_RATIO_A, ratioset); + + /* Set interrupts to high active */ + dac33_write(codec, DAC33_INTP_CTRL_A, DAC33_INTPM_AHIGH); + + dac33_write(codec, DAC33_FIFO_IRQ_MODE_B, + DAC33_ATM(DAC33_FIFO_IRQ_MODE_LEVEL)); + dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MAT); + } else { + /* 50-51 : ASRC Control registers */ + dac33_write(codec, DAC33_ASRC_CTRL_A, DAC33_SRCBYP); + dac33_write(codec, DAC33_ASRC_CTRL_B, 0); /* ??? */ + } + + if (dac33->nsample_switch) + fifoctrl_a &= ~DAC33_FBYPAS; + else + fifoctrl_a |= DAC33_FBYPAS; + dac33_write(codec, DAC33_FIFO_CTRL_A, fifoctrl_a); + + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a); + reg_tmp = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); + if (dac33->nsample_switch) + reg_tmp &= ~DAC33_BCLKON; + else + reg_tmp |= DAC33_BCLKON; + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, reg_tmp); + + if (dac33->nsample_switch) { + /* 20: BCLK divide ratio */ + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 3); + + dac33_write16(codec, DAC33_ATHR_MSB, + DAC33_THRREG(dac33->alarm_threshold)); + } else { + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32); + } + + mutex_unlock(&dac33->mutex); + + return 0; +} + +static void dac33_calculate_times(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->card->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + unsigned int nsample_limit; + + /* Number of samples (16bit, stereo) in one period */ + dac33->nsample_min = snd_pcm_lib_period_bytes(substream) / 4; + + /* Number of samples (16bit, stereo) in ALSA buffer */ + dac33->nsample_max = snd_pcm_lib_buffer_bytes(substream) / 4; + /* Subtract one period from the total */ + dac33->nsample_max -= dac33->nsample_min; + + /* Number of samples for LATENCY_TIME_MS / 2 */ + dac33->alarm_threshold = substream->runtime->rate / + (1000 / (LATENCY_TIME_MS / 2)); + + /* Find and fix up the lowest nsmaple limit */ + nsample_limit = substream->runtime->rate / (1000 / LATENCY_TIME_MS); + + if (dac33->nsample_min < nsample_limit) + dac33->nsample_min = nsample_limit; + + if (dac33->nsample < dac33->nsample_min) + dac33->nsample = dac33->nsample_min; + + /* + * Find and fix up the highest nsmaple limit + * In order to not overflow the DAC33 buffer substract the + * alarm_threshold value from the size of the DAC33 buffer + */ + nsample_limit = DAC33_BUFFER_SIZE_SAMPLES - dac33->alarm_threshold; + + if (dac33->nsample_max > nsample_limit) + dac33->nsample_max = nsample_limit; + + if (dac33->nsample > dac33->nsample_max) + dac33->nsample = dac33->nsample_max; +} + +static int dac33_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + dac33_calculate_times(substream); + dac33_prepare_chip(substream); + + return 0; +} + +static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (dac33->nsample_switch) { + dac33->state = DAC33_PREFILL; + queue_work(dac33->dac33_wq, &dac33->work); + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (dac33->nsample_switch) { + dac33->state = DAC33_FLUSH; + queue_work(dac33->dac33_wq, &dac33->work); + } + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static int dac33_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; + struct tlv320dac33_priv *dac33 = codec->private_data; + u8 ioc_reg, asrcb_reg; + + ioc_reg = dac33_read_reg_cache(codec, DAC33_INT_OSC_CTRL); + asrcb_reg = dac33_read_reg_cache(codec, DAC33_ASRC_CTRL_B); + switch (clk_id) { + case TLV320DAC33_MCLK: + ioc_reg |= DAC33_REFSEL; + asrcb_reg |= DAC33_SRCREFSEL; + break; + case TLV320DAC33_SLEEPCLK: + ioc_reg &= ~DAC33_REFSEL; + asrcb_reg &= ~DAC33_SRCREFSEL; + break; + default: + dev_err(codec->dev, "Invalid clock ID (%d)\n", clk_id); + break; + } + dac33->refclk = freq; + + dac33_write_reg_cache(codec, DAC33_INT_OSC_CTRL, ioc_reg); + dac33_write_reg_cache(codec, DAC33_ASRC_CTRL_B, asrcb_reg); + + return 0; +} + +static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 aictrl_a, aictrl_b; + + aictrl_a = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A); + aictrl_b = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* Codec Master */ + aictrl_a |= (DAC33_MSBCLK | DAC33_MSWCLK); + break; + case SND_SOC_DAIFMT_CBS_CFS: + /* Codec Slave */ + aictrl_a &= ~(DAC33_MSBCLK | DAC33_MSWCLK); + break; + default: + return -EINVAL; + } + + aictrl_a &= ~DAC33_AFMT_MASK; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aictrl_a |= DAC33_AFMT_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + aictrl_a |= DAC33_AFMT_DSP; + aictrl_b &= ~DAC33_DATA_DELAY_MASK; + aictrl_b |= DAC33_DATA_DELAY(1); /* 1 bit delay */ + break; + case SND_SOC_DAIFMT_DSP_B: + aictrl_a |= DAC33_AFMT_DSP; + aictrl_b &= ~DAC33_DATA_DELAY_MASK; /* No delay */ + break; + case SND_SOC_DAIFMT_RIGHT_J: + aictrl_a |= DAC33_AFMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + aictrl_a |= DAC33_AFMT_LEFT_J; + break; + default: + dev_err(codec->dev, "Unsupported format (%u)\n", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + dac33_write_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a); + dac33_write_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B, aictrl_b); + + return 0; +} + +static void dac33_init_chip(struct snd_soc_codec *codec) +{ + /* 44-46: DAC Control Registers */ + /* A : DAC sample rate Fsref/1.5 */ + dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(1)); + /* B : DAC src=normal, not muted */ + dac33_write(codec, DAC33_DAC_CTRL_B, DAC33_DACSRCR_RIGHT | + DAC33_DACSRCL_LEFT); + /* C : (defaults) */ + dac33_write(codec, DAC33_DAC_CTRL_C, 0x00); + + /* 64-65 : L&R DAC power control + Line In -> OUT 1V/V Gain, DAC -> OUT 4V/V Gain*/ + dac33_write(codec, DAC33_LDAC_PWR_CTRL, DAC33_LROUT_GAIN(2)); + dac33_write(codec, DAC33_RDAC_PWR_CTRL, DAC33_LROUT_GAIN(2)); + + /* 73 : volume soft stepping control, + clock source = internal osc (?) */ + dac33_write(codec, DAC33_ANA_VOL_SOFT_STEP_CTRL, DAC33_VOLCLKEN); + + /* 66 : LOP/LOM Modes */ + dac33_write(codec, DAC33_OUT_AMP_CM_CTRL, 0xff); + + /* 68 : LOM inverted from LOP */ + dac33_write(codec, DAC33_OUT_AMP_CTRL, (3<<2)); + + dac33_write(codec, DAC33_PWR_CTRL, DAC33_PDNALLB); +} + +static int dac33_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct tlv320dac33_priv *dac33; + int ret = 0; + + BUG_ON(!tlv320dac33_codec); + + codec = tlv320dac33_codec; + socdev->card->codec = codec; + dac33 = codec->private_data; + + /* Power up the codec */ + dac33_hard_power(codec, 1); + /* Set default configuration */ + dac33_init_chip(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms\n"); + goto pcm_err; + } + + snd_soc_add_controls(codec, dac33_snd_controls, + ARRAY_SIZE(dac33_snd_controls)); + /* Only add the nSample controls, if we have valid IRQ number */ + if (dac33->irq >= 0) + snd_soc_add_controls(codec, dac33_nsample_snd_controls, + ARRAY_SIZE(dac33_nsample_snd_controls)); + + dac33_add_widgets(codec); + + /* power on device */ + dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; + +pcm_err: + dac33_hard_power(codec, 0); + return ret; +} + +static int dac33_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + dac33_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +static int dac33_soc_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + dac33_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int dac33_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + dac33_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_tlv320dac33 = { + .probe = dac33_soc_probe, + .remove = dac33_soc_remove, + .suspend = dac33_soc_suspend, + .resume = dac33_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320dac33); + +#define DAC33_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) +#define DAC33_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +static struct snd_soc_dai_ops dac33_dai_ops = { + .shutdown = dac33_shutdown, + .hw_params = dac33_hw_params, + .prepare = dac33_pcm_prepare, + .trigger = dac33_pcm_trigger, + .set_sysclk = dac33_set_dai_sysclk, + .set_fmt = dac33_set_dai_fmt, +}; + +struct snd_soc_dai dac33_dai = { + .name = "tlv320dac33", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = DAC33_RATES, + .formats = DAC33_FORMATS,}, + .ops = &dac33_dai_ops, +}; +EXPORT_SYMBOL_GPL(dac33_dai); + +static int dac33_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct tlv320dac33_platform_data *pdata; + struct tlv320dac33_priv *dac33; + struct snd_soc_codec *codec; + int ret = 0; + + if (client->dev.platform_data == NULL) { + dev_err(&client->dev, "Platform data not set\n"); + return -ENODEV; + } + pdata = client->dev.platform_data; + + dac33 = kzalloc(sizeof(struct tlv320dac33_priv), GFP_KERNEL); + if (dac33 == NULL) + return -ENOMEM; + + codec = &dac33->codec; + codec->private_data = dac33; + codec->control_data = client; + + mutex_init(&codec->mutex); + mutex_init(&dac33->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "tlv320dac33"; + codec->owner = THIS_MODULE; + codec->read = dac33_read_reg_cache; + codec->write = dac33_write_locked; + codec->hw_write = (hw_write_t) i2c_master_send; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = dac33_set_bias_level; + codec->dai = &dac33_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(dac33_reg); + codec->reg_cache = kmemdup(dac33_reg, ARRAY_SIZE(dac33_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto error_reg; + } + + i2c_set_clientdata(client, dac33); + + dac33->power_gpio = pdata->power_gpio; + dac33->irq = client->irq; + dac33->nsample = NSAMPLE_MAX; + /* Disable FIFO use by default */ + dac33->nsample_switch = 0; + + tlv320dac33_codec = codec; + + codec->dev = &client->dev; + dac33_dai.dev = codec->dev; + + /* Check if the reset GPIO number is valid and request it */ + if (dac33->power_gpio >= 0) { + ret = gpio_request(dac33->power_gpio, "tlv320dac33 reset"); + if (ret < 0) { + dev_err(codec->dev, + "Failed to request reset GPIO (%d)\n", + dac33->power_gpio); + snd_soc_unregister_dai(&dac33_dai); + snd_soc_unregister_codec(codec); + goto error_gpio; + } + gpio_direction_output(dac33->power_gpio, 0); + } else { + dac33->chip_power = 1; + } + + /* Check if the IRQ number is valid and request it */ + if (dac33->irq >= 0) { + ret = request_irq(dac33->irq, dac33_interrupt_handler, + IRQF_TRIGGER_RISING | IRQF_DISABLED, + codec->name, codec); + if (ret < 0) { + dev_err(codec->dev, "Could not request IRQ%d (%d)\n", + dac33->irq, ret); + dac33->irq = -1; + } + if (dac33->irq != -1) { + /* Setup work queue */ + dac33->dac33_wq = + create_singlethread_workqueue("tlv320dac33"); + if (dac33->dac33_wq == NULL) { + free_irq(dac33->irq, &dac33->codec); + ret = -ENOMEM; + goto error_wq; + } + + INIT_WORK(&dac33->work, dac33_work); + } + } + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto error_codec; + } + + ret = snd_soc_register_dai(&dac33_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + goto error_codec; + } + + /* Shut down the codec for now */ + dac33_hard_power(codec, 0); + + return ret; + +error_codec: + if (dac33->irq >= 0) { + free_irq(dac33->irq, &dac33->codec); + destroy_workqueue(dac33->dac33_wq); + } +error_wq: + if (dac33->power_gpio >= 0) + gpio_free(dac33->power_gpio); +error_gpio: + kfree(codec->reg_cache); +error_reg: + tlv320dac33_codec = NULL; + kfree(dac33); + + return ret; +} + +static int dac33_i2c_remove(struct i2c_client *client) +{ + struct tlv320dac33_priv *dac33; + + dac33 = i2c_get_clientdata(client); + dac33_hard_power(&dac33->codec, 0); + + if (dac33->power_gpio >= 0) + gpio_free(dac33->power_gpio); + if (dac33->irq >= 0) + free_irq(dac33->irq, &dac33->codec); + + destroy_workqueue(dac33->dac33_wq); + snd_soc_unregister_dai(&dac33_dai); + snd_soc_unregister_codec(&dac33->codec); + kfree(dac33->codec.reg_cache); + kfree(dac33); + tlv320dac33_codec = NULL; + + return 0; +} + +static const struct i2c_device_id tlv320dac33_i2c_id[] = { + { + .name = "tlv320dac33", + .driver_data = 0, + }, + { }, +}; + +static struct i2c_driver tlv320dac33_i2c_driver = { + .driver = { + .name = "tlv320dac33", + .owner = THIS_MODULE, + }, + .probe = dac33_i2c_probe, + .remove = __devexit_p(dac33_i2c_remove), + .id_table = tlv320dac33_i2c_id, +}; + +static int __init dac33_module_init(void) +{ + int r; + r = i2c_add_driver(&tlv320dac33_i2c_driver); + if (r < 0) { + printk(KERN_ERR "DAC33: driver registration failed\n"); + return r; + } + return 0; +} +module_init(dac33_module_init); + +static void __exit dac33_module_exit(void) +{ + i2c_del_driver(&tlv320dac33_i2c_driver); +} +module_exit(dac33_module_exit); + + +MODULE_DESCRIPTION("ASoC TLV320DAC33 codec driver"); +MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@nokia.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320dac33.h b/sound/soc/codecs/tlv320dac33.h new file mode 100644 index 0000000..eb8ae07 --- /dev/null +++ b/sound/soc/codecs/tlv320dac33.h @@ -0,0 +1,267 @@ +/* + * ALSA SoC Texas Instruments TLV320DAC33 codec driver + * + * Author: Peter Ujfalusi <peter.ujfalusi@nokia.com> + * + * Copyright: (C) 2009 Nokia Corporation + * + * 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 + * + */ + +#ifndef __TLV320DAC33_H +#define __TLV320DAC33_H + +#define DAC33_PAGE_SELECT 0x00 +#define DAC33_PWR_CTRL 0x01 +#define DAC33_PLL_CTRL_A 0x02 +#define DAC33_PLL_CTRL_B 0x03 +#define DAC33_PLL_CTRL_C 0x04 +#define DAC33_PLL_CTRL_D 0x05 +#define DAC33_PLL_CTRL_E 0x06 +#define DAC33_INT_OSC_CTRL 0x07 +#define DAC33_INT_OSC_FREQ_RAT_A 0x08 +#define DAC33_INT_OSC_FREQ_RAT_B 0x09 +#define DAC33_INT_OSC_DAC_RATIO_SET 0x0A +#define DAC33_CALIB_TIME 0x0B +#define DAC33_INT_OSC_CTRL_B 0x0C +#define DAC33_INT_OSC_CTRL_C 0x0D +#define DAC33_INT_OSC_STATUS 0x0E +#define DAC33_INT_OSC_DAC_RATIO_READ 0x0F +#define DAC33_INT_OSC_FREQ_RAT_READ_A 0x10 +#define DAC33_INT_OSC_FREQ_RAT_READ_B 0x11 +#define DAC33_SER_AUDIOIF_CTRL_A 0x12 +#define DAC33_SER_AUDIOIF_CTRL_B 0x13 +#define DAC33_SER_AUDIOIF_CTRL_C 0x14 +#define DAC33_FIFO_CTRL_A 0x15 +#define DAC33_UTHR_MSB 0x16 +#define DAC33_UTHR_LSB 0x17 +#define DAC33_ATHR_MSB 0x18 +#define DAC33_ATHR_LSB 0x19 +#define DAC33_LTHR_MSB 0x1A +#define DAC33_LTHR_LSB 0x1B +#define DAC33_PREFILL_MSB 0x1C +#define DAC33_PREFILL_LSB 0x1D +#define DAC33_NSAMPLE_MSB 0x1E +#define DAC33_NSAMPLE_LSB 0x1F +#define DAC33_FIFO_WPTR_MSB 0x20 +#define DAC33_FIFO_WPTR_LSB 0x21 +#define DAC33_FIFO_RPTR_MSB 0x22 +#define DAC33_FIFO_RPTR_LSB 0x23 +#define DAC33_FIFO_DEPTH_MSB 0x24 +#define DAC33_FIFO_DEPTH_LSB 0x25 +#define DAC33_SAMPLES_REMAINING_MSB 0x26 +#define DAC33_SAMPLES_REMAINING_LSB 0x27 +#define DAC33_FIFO_IRQ_FLAG 0x28 +#define DAC33_FIFO_IRQ_MASK 0x29 +#define DAC33_FIFO_IRQ_MODE_A 0x2A +#define DAC33_FIFO_IRQ_MODE_B 0x2B +#define DAC33_DAC_CTRL_A 0x2C +#define DAC33_DAC_CTRL_B 0x2D +#define DAC33_DAC_CTRL_C 0x2E +#define DAC33_LDAC_DIG_VOL_CTRL 0x2F +#define DAC33_RDAC_DIG_VOL_CTRL 0x30 +#define DAC33_DAC_STATUS_FLAGS 0x31 +#define DAC33_ASRC_CTRL_A 0x32 +#define DAC33_ASRC_CTRL_B 0x33 +#define DAC33_SRC_REF_CLK_RATIO_A 0x34 +#define DAC33_SRC_REF_CLK_RATIO_B 0x35 +#define DAC33_SRC_EST_REF_CLK_RATIO_A 0x36 +#define DAC33_SRC_EST_REF_CLK_RATIO_B 0x37 +#define DAC33_INTP_CTRL_A 0x38 +#define DAC33_INTP_CTRL_B 0x39 +/* Registers 0x3A - 0x3F Reserved */ +#define DAC33_LDAC_PWR_CTRL 0x40 +#define DAC33_RDAC_PWR_CTRL 0x41 +#define DAC33_OUT_AMP_CM_CTRL 0x42 +#define DAC33_OUT_AMP_PWR_CTRL 0x43 +#define DAC33_OUT_AMP_CTRL 0x44 +#define DAC33_LINEL_TO_LLO_VOL 0x45 +/* Registers 0x45 - 0x47 Reserved */ +#define DAC33_LINER_TO_RLO_VOL 0x48 +#define DAC33_ANA_VOL_SOFT_STEP_CTRL 0x49 +#define DAC33_OSC_TRIM 0x4A +/* Registers 0x4B - 0x7C Reserved */ +#define DAC33_DEVICE_ID_MSB 0x7D +#define DAC33_DEVICE_ID_LSB 0x7E +#define DAC33_DEVICE_REV_ID 0x7F + +#define DAC33_CACHEREGNUM 128 + +/* Bit definitions */ + +/* DAC33_PWR_CTRL (0x01) */ +#define DAC33_DACRPDNB (0x01 << 0) +#define DAC33_DACLPDNB (0x01 << 1) +#define DAC33_OSCPDNB (0x01 << 2) +#define DAC33_PLLPDNB (0x01 << 3) +#define DAC33_PDNALLB (0x01 << 4) +#define DAC33_SOFT_RESET (0x01 << 7) + +/* DAC33_INT_OSC_CTRL (0x07) */ +#define DAC33_REFSEL (0x01 << 1) + +/* DAC33_INT_OSC_CTRL_B (0x0C) */ +#define DAC33_ADJSTEP(x) (x << 0) +#define DAC33_ADJTHRSHLD(x) (x << 4) + +/* DAC33_INT_OSC_CTRL_C (0x0D) */ +#define DAC33_REFDIV(x) (x << 4) + +/* DAC33_INT_OSC_STATUS (0x0E) */ +#define DAC33_OSCSTATUS_IDLE_CALIB (0x00) +#define DAC33_OSCSTATUS_NORMAL (0x01) +#define DAC33_OSCSTATUS_ADJUSTMENT (0x03) +#define DAC33_OSCSTATUS_NOT_USED (0x02) + +/* DAC33_SER_AUDIOIF_CTRL_A (0x12) */ +#define DAC33_MSWCLK (0x01 << 0) +#define DAC33_MSBCLK (0x01 << 1) +#define DAC33_AFMT_MASK (0x03 << 2) +#define DAC33_AFMT_I2S (0x00 << 2) +#define DAC33_AFMT_DSP (0x01 << 2) +#define DAC33_AFMT_RIGHT_J (0x02 << 2) +#define DAC33_AFMT_LEFT_J (0x03 << 2) +#define DAC33_WLEN_MASK (0x03 << 4) +#define DAC33_WLEN_16 (0x00 << 4) +#define DAC33_WLEN_20 (0x01 << 4) +#define DAC33_WLEN_24 (0x02 << 4) +#define DAC33_WLEN_32 (0x03 << 4) +#define DAC33_NCYCL_MASK (0x03 << 6) +#define DAC33_NCYCL_16 (0x00 << 6) +#define DAC33_NCYCL_20 (0x01 << 6) +#define DAC33_NCYCL_24 (0x02 << 6) +#define DAC33_NCYCL_32 (0x03 << 6) + +/* DAC33_SER_AUDIOIF_CTRL_B (0x13) */ +#define DAC33_DATA_DELAY_MASK (0x03 << 2) +#define DAC33_DATA_DELAY(x) (x << 2) +#define DAC33_BCLKON (0x01 << 5) + +/* DAC33_FIFO_CTRL_A (0x15) */ +#define DAC33_WIDTH (0x01 << 0) +#define DAC33_FBYPAS (0x01 << 1) +#define DAC33_FAUTO (0x01 << 2) +#define DAC33_FIFOFLUSH (0x01 << 3) + +/* + * UTHR, ATHR, LTHR, PREFILL, NSAMPLE (0x16 - 0x1F) + * 13-bit values +*/ +#define DAC33_THRREG(x) (((x) & 0x1FFF) << 3) + +/* DAC33_FIFO_IRQ_MASK (0x29) */ +#define DAC33_MNS (0x01 << 0) +#define DAC33_MPS (0x01 << 1) +#define DAC33_MAT (0x01 << 2) +#define DAC33_MLT (0x01 << 3) +#define DAC33_MUT (0x01 << 4) +#define DAC33_MUF (0x01 << 5) +#define DAC33_MOF (0x01 << 6) + +#define DAC33_FIFO_IRQ_MODE_MASK (0x03) +#define DAC33_FIFO_IRQ_MODE_RISING (0x00) +#define DAC33_FIFO_IRQ_MODE_FALLING (0x01) +#define DAC33_FIFO_IRQ_MODE_LEVEL (0x02) +#define DAC33_FIFO_IRQ_MODE_EDGE (0x03) + +/* DAC33_FIFO_IRQ_MODE_A (0x2A) */ +#define DAC33_UTM(x) (x << 0) +#define DAC33_UFM(x) (x << 2) +#define DAC33_OFM(x) (x << 4) + +/* DAC33_FIFO_IRQ_MODE_B (0x2B) */ +#define DAC33_NSM(x) (x << 0) +#define DAC33_PSM(x) (x << 2) +#define DAC33_ATM(x) (x << 4) +#define DAC33_LTM(x) (x << 6) + +/* DAC33_DAC_CTRL_A (0x2C) */ +#define DAC33_DACRATE(x) (x << 0) +#define DAC33_DACDUAL (0x01 << 4) +#define DAC33_DACLKSEL_MASK (0x03 << 5) +#define DAC33_DACLKSEL_INTSOC (0x00 << 5) +#define DAC33_DACLKSEL_PLL (0x01 << 5) +#define DAC33_DACLKSEL_MCLK (0x02 << 5) +#define DAC33_DACLKSEL_BCLK (0x03 << 5) + +/* DAC33_DAC_CTRL_B (0x2D) */ +#define DAC33_DACSRCR_MASK (0x03 << 0) +#define DAC33_DACSRCR_MUTE (0x00 << 0) +#define DAC33_DACSRCR_RIGHT (0x01 << 0) +#define DAC33_DACSRCR_LEFT (0x02 << 0) +#define DAC33_DACSRCR_MONOMIX (0x03 << 0) +#define DAC33_DACSRCL_MASK (0x03 << 2) +#define DAC33_DACSRCL_MUTE (0x00 << 2) +#define DAC33_DACSRCL_LEFT (0x01 << 2) +#define DAC33_DACSRCL_RIGHT (0x02 << 2) +#define DAC33_DACSRCL_MONOMIX (0x03 << 2) +#define DAC33_DVOLSTEP_MASK (0x03 << 4) +#define DAC33_DVOLSTEP_SS_PERFS (0x00 << 4) +#define DAC33_DVOLSTEP_SS_PER2FS (0x01 << 4) +#define DAC33_DVOLSTEP_SS_DISABLED (0x02 << 4) +#define DAC33_DVOLCTRL_MASK (0x03 << 6) +#define DAC33_DVOLCTRL_LR_INDEPENDENT1 (0x00 << 6) +#define DAC33_DVOLCTRL_LR_RIGHT_CONTROL (0x01 << 6) +#define DAC33_DVOLCTRL_LR_LEFT_CONTROL (0x02 << 6) +#define DAC33_DVOLCTRL_LR_INDEPENDENT2 (0x03 << 6) + +/* DAC33_DAC_CTRL_C (0x2E) */ +#define DAC33_DEEMENR (0x01 << 0) +#define DAC33_EFFENR (0x01 << 1) +#define DAC33_DEEMENL (0x01 << 2) +#define DAC33_EFFENL (0x01 << 3) +#define DAC33_EN3D (0x01 << 4) +#define DAC33_RESYNMUTE (0x01 << 5) +#define DAC33_RESYNEN (0x01 << 6) + +/* DAC33_ASRC_CTRL_A (0x32) */ +#define DAC33_SRCBYP (0x01 << 0) +#define DAC33_SRCLKSEL_MASK (0x03 << 1) +#define DAC33_SRCLKSEL_INTSOC (0x00 << 1) +#define DAC33_SRCLKSEL_PLL (0x01 << 1) +#define DAC33_SRCLKSEL_MCLK (0x02 << 1) +#define DAC33_SRCLKSEL_BCLK (0x03 << 1) +#define DAC33_SRCLKDIV(x) (x << 3) + +/* DAC33_ASRC_CTRL_B (0x33) */ +#define DAC33_SRCSETUP(x) (x << 0) +#define DAC33_SRCREFSEL (0x01 << 4) +#define DAC33_SRCREFDIV(x) (x << 5) + +/* DAC33_INTP_CTRL_A (0x38) */ +#define DAC33_INTPSEL (0x01 << 0) +#define DAC33_INTPM_MASK (0x03 << 1) +#define DAC33_INTPM_ALOW_OPENDRAIN (0x00 << 1) +#define DAC33_INTPM_ALOW (0x01 << 1) +#define DAC33_INTPM_AHIGH (0x02 << 1) + +/* DAC33_LDAC_PWR_CTRL (0x40) */ +/* DAC33_RDAC_PWR_CTRL (0x41) */ +#define DAC33_DACLRNUM (0x01 << 2) +#define DAC33_LROUT_GAIN(x) (x << 0) + +/* DAC33_ANA_VOL_SOFT_STEP_CTRL (0x49) */ +#define DAC33_VOLCLKSEL (0x01 << 0) +#define DAC33_VOLCLKEN (0x01 << 1) +#define DAC33_VOLBYPASS (0x01 << 2) + +#define TLV320DAC33_MCLK 0 +#define TLV320DAC33_SLEEPCLK 1 + +extern struct snd_soc_dai dac33_dai; +extern struct snd_soc_codec_device soc_codec_dev_tlv320dac33; + +#endif /* __TLV320DAC33_H */ diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c new file mode 100644 index 0000000..6b650c1 --- /dev/null +++ b/sound/soc/codecs/tpa6130a2.c @@ -0,0 +1,463 @@ +/* + * ALSA SoC Texas Instruments TPA6130A2 headset stereo amplifier driver + * + * Copyright (C) Nokia Corporation + * + * Author: Peter Ujfalusi <peter.ujfalusi@nokia.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * 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 <linux/module.h> +#include <linux/errno.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <sound/tpa6130a2-plat.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "tpa6130a2.h" + +static struct i2c_client *tpa6130a2_client; + +/* This struct is used to save the context */ +struct tpa6130a2_data { + struct mutex mutex; + unsigned char regs[TPA6130A2_CACHEREGNUM]; + int power_gpio; + unsigned char power_state; +}; + +static int tpa6130a2_i2c_read(int reg) +{ + struct tpa6130a2_data *data; + int val; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + /* If powered off, return the cached value */ + if (data->power_state) { + val = i2c_smbus_read_byte_data(tpa6130a2_client, reg); + if (val < 0) + dev_err(&tpa6130a2_client->dev, "Read failed\n"); + else + data->regs[reg] = val; + } else { + val = data->regs[reg]; + } + + return val; +} + +static int tpa6130a2_i2c_write(int reg, u8 value) +{ + struct tpa6130a2_data *data; + int val = 0; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + if (data->power_state) { + val = i2c_smbus_write_byte_data(tpa6130a2_client, reg, value); + if (val < 0) + dev_err(&tpa6130a2_client->dev, "Write failed\n"); + } + + /* Either powered on or off, we save the context */ + data->regs[reg] = value; + + return val; +} + +static u8 tpa6130a2_read(int reg) +{ + struct tpa6130a2_data *data; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + return data->regs[reg]; +} + +static void tpa6130a2_initialize(void) +{ + struct tpa6130a2_data *data; + int i; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + for (i = 1; i < TPA6130A2_REG_VERSION; i++) + tpa6130a2_i2c_write(i, data->regs[i]); +} + +static void tpa6130a2_power(int power) +{ + struct tpa6130a2_data *data; + u8 val; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + mutex_lock(&data->mutex); + if (power) { + /* Power on */ + if (data->power_gpio >= 0) { + gpio_set_value(data->power_gpio, 1); + data->power_state = 1; + tpa6130a2_initialize(); + } + /* Clear SWS */ + val = tpa6130a2_read(TPA6130A2_REG_CONTROL); + val &= ~TPA6130A2_SWS; + tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); + } else { + /* set SWS */ + val = tpa6130a2_read(TPA6130A2_REG_CONTROL); + val |= TPA6130A2_SWS; + tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); + /* Power off */ + if (data->power_gpio >= 0) { + gpio_set_value(data->power_gpio, 0); + data->power_state = 0; + } + } + mutex_unlock(&data->mutex); +} + +static int tpa6130a2_get_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct tpa6130a2_data *data; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + unsigned int invert = mc->invert; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + mutex_lock(&data->mutex); + + ucontrol->value.integer.value[0] = + (tpa6130a2_read(reg) >> shift) & mask; + + if (invert) + ucontrol->value.integer.value[0] = + mask - ucontrol->value.integer.value[0]; + + mutex_unlock(&data->mutex); + return 0; +} + +static int tpa6130a2_set_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct tpa6130a2_data *data; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + unsigned int invert = mc->invert; + unsigned int val = (ucontrol->value.integer.value[0] & mask); + unsigned int val_reg; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + if (invert) + val = mask - val; + + mutex_lock(&data->mutex); + + val_reg = tpa6130a2_read(reg); + if (((val_reg >> shift) & mask) == val) { + mutex_unlock(&data->mutex); + return 0; + } + + val_reg &= ~(mask << shift); + val_reg |= val << shift; + tpa6130a2_i2c_write(reg, val_reg); + + mutex_unlock(&data->mutex); + + return 1; +} + +/* + * TPA6130 volume. From -59.5 to 4 dB with increasing step size when going + * down in gain. + */ +static const unsigned int tpa6130_tlv[] = { + TLV_DB_RANGE_HEAD(10), + 0, 1, TLV_DB_SCALE_ITEM(-5950, 600, 0), + 2, 3, TLV_DB_SCALE_ITEM(-5000, 250, 0), + 4, 5, TLV_DB_SCALE_ITEM(-4550, 160, 0), + 6, 7, TLV_DB_SCALE_ITEM(-4140, 190, 0), + 8, 9, TLV_DB_SCALE_ITEM(-3650, 120, 0), + 10, 11, TLV_DB_SCALE_ITEM(-3330, 160, 0), + 12, 13, TLV_DB_SCALE_ITEM(-3040, 180, 0), + 14, 20, TLV_DB_SCALE_ITEM(-2710, 110, 0), + 21, 37, TLV_DB_SCALE_ITEM(-1960, 74, 0), + 38, 63, TLV_DB_SCALE_ITEM(-720, 45, 0), +}; + +static const struct snd_kcontrol_new tpa6130a2_controls[] = { + SOC_SINGLE_EXT_TLV("TPA6130A2 Headphone Playback Volume", + TPA6130A2_REG_VOL_MUTE, 0, 0x3f, 0, + tpa6130a2_get_reg, tpa6130a2_set_reg, + tpa6130_tlv), +}; + +/* + * Enable or disable channel (left or right) + * The bit number for mute and amplifier are the same per channel: + * bit 6: Right channel + * bit 7: Left channel + * in both registers. + */ +static void tpa6130a2_channel_enable(u8 channel, int enable) +{ + struct tpa6130a2_data *data; + u8 val; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + if (enable) { + /* Enable channel */ + /* Enable amplifier */ + val = tpa6130a2_read(TPA6130A2_REG_CONTROL); + val |= channel; + tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); + + /* Unmute channel */ + val = tpa6130a2_read(TPA6130A2_REG_VOL_MUTE); + val &= ~channel; + tpa6130a2_i2c_write(TPA6130A2_REG_VOL_MUTE, val); + } else { + /* Disable channel */ + /* Mute channel */ + val = tpa6130a2_read(TPA6130A2_REG_VOL_MUTE); + val |= channel; + tpa6130a2_i2c_write(TPA6130A2_REG_VOL_MUTE, val); + + /* Disable amplifier */ + val = tpa6130a2_read(TPA6130A2_REG_CONTROL); + val &= ~channel; + tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); + } +} + +static int tpa6130a2_left_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + tpa6130a2_channel_enable(TPA6130A2_HP_EN_L, 1); + break; + case SND_SOC_DAPM_POST_PMD: + tpa6130a2_channel_enable(TPA6130A2_HP_EN_L, 0); + break; + } + return 0; +} + +static int tpa6130a2_right_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + tpa6130a2_channel_enable(TPA6130A2_HP_EN_R, 1); + break; + case SND_SOC_DAPM_POST_PMD: + tpa6130a2_channel_enable(TPA6130A2_HP_EN_R, 0); + break; + } + return 0; +} + +static int tpa6130a2_supply_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + tpa6130a2_power(1); + break; + case SND_SOC_DAPM_POST_PMD: + tpa6130a2_power(0); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget tpa6130a2_dapm_widgets[] = { + SND_SOC_DAPM_PGA_E("TPA6130A2 Left", SND_SOC_NOPM, + 0, 0, NULL, 0, tpa6130a2_left_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("TPA6130A2 Right", SND_SOC_NOPM, + 0, 0, NULL, 0, tpa6130a2_right_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("TPA6130A2 Enable", SND_SOC_NOPM, + 0, 0, tpa6130a2_supply_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + /* Outputs */ + SND_SOC_DAPM_HP("TPA6130A2 Headphone Left", NULL), + SND_SOC_DAPM_HP("TPA6130A2 Headphone Right", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"TPA6130A2 Headphone Left", NULL, "TPA6130A2 Left"}, + {"TPA6130A2 Headphone Right", NULL, "TPA6130A2 Right"}, + + {"TPA6130A2 Headphone Left", NULL, "TPA6130A2 Enable"}, + {"TPA6130A2 Headphone Right", NULL, "TPA6130A2 Enable"}, +}; + +int tpa6130a2_add_controls(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, tpa6130a2_dapm_widgets, + ARRAY_SIZE(tpa6130a2_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + return snd_soc_add_controls(codec, tpa6130a2_controls, + ARRAY_SIZE(tpa6130a2_controls)); + +} +EXPORT_SYMBOL_GPL(tpa6130a2_add_controls); + +static int tpa6130a2_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device *dev; + struct tpa6130a2_data *data; + struct tpa6130a2_platform_data *pdata; + int ret; + + dev = &client->dev; + + if (client->dev.platform_data == NULL) { + dev_err(dev, "Platform data not set\n"); + dump_stack(); + return -ENODEV; + } + + data = kzalloc(sizeof(*data), GFP_KERNEL); + if (data == NULL) { + dev_err(dev, "Can not allocate memory\n"); + return -ENOMEM; + } + + tpa6130a2_client = client; + + i2c_set_clientdata(tpa6130a2_client, data); + + pdata = client->dev.platform_data; + data->power_gpio = pdata->power_gpio; + + mutex_init(&data->mutex); + + /* Set default register values */ + data->regs[TPA6130A2_REG_CONTROL] = TPA6130A2_SWS; + data->regs[TPA6130A2_REG_VOL_MUTE] = TPA6130A2_MUTE_R | + TPA6130A2_MUTE_L; + + if (data->power_gpio >= 0) { + ret = gpio_request(data->power_gpio, "tpa6130a2 enable"); + if (ret < 0) { + dev_err(dev, "Failed to request power GPIO (%d)\n", + data->power_gpio); + goto fail; + } + gpio_direction_output(data->power_gpio, 0); + } else { + data->power_state = 1; + tpa6130a2_initialize(); + } + + tpa6130a2_power(1); + + /* Read version */ + ret = tpa6130a2_i2c_read(TPA6130A2_REG_VERSION) & + TPA6130A2_VERSION_MASK; + if ((ret != 1) && (ret != 2)) + dev_warn(dev, "UNTESTED version detected (%d)\n", ret); + + /* Disable the chip */ + tpa6130a2_power(0); + + return 0; +fail: + kfree(data); + i2c_set_clientdata(tpa6130a2_client, NULL); + tpa6130a2_client = NULL; + + return ret; +} + +static int tpa6130a2_remove(struct i2c_client *client) +{ + struct tpa6130a2_data *data = i2c_get_clientdata(client); + + tpa6130a2_power(0); + + if (data->power_gpio >= 0) + gpio_free(data->power_gpio); + kfree(data); + tpa6130a2_client = NULL; + + return 0; +} + +static const struct i2c_device_id tpa6130a2_id[] = { + { "tpa6130a2", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tpa6130a2_id); + +static struct i2c_driver tpa6130a2_i2c_driver = { + .driver = { + .name = "tpa6130a2", + .owner = THIS_MODULE, + }, + .probe = tpa6130a2_probe, + .remove = __devexit_p(tpa6130a2_remove), + .id_table = tpa6130a2_id, +}; + +static int __init tpa6130a2_init(void) +{ + return i2c_add_driver(&tpa6130a2_i2c_driver); +} + +static void __exit tpa6130a2_exit(void) +{ + i2c_del_driver(&tpa6130a2_i2c_driver); +} + +MODULE_AUTHOR("Peter Ujfalusi"); +MODULE_DESCRIPTION("TPA6130A2 Headphone amplifier driver"); +MODULE_LICENSE("GPL"); + +module_init(tpa6130a2_init); +module_exit(tpa6130a2_exit); diff --git a/sound/soc/codecs/tpa6130a2.h b/sound/soc/codecs/tpa6130a2.h new file mode 100644 index 0000000..57e867f --- /dev/null +++ b/sound/soc/codecs/tpa6130a2.h @@ -0,0 +1,61 @@ +/* + * ALSA SoC TPA6130A2 amplifier driver + * + * Copyright (C) Nokia Corporation + * + * Author: Peter Ujfalusi <peter.ujfalusi@nokia.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * 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 + * + */ + +#ifndef __TPA6130A2_H__ +#define __TPA6130A2_H__ + +/* Register addresses */ +#define TPA6130A2_REG_CONTROL 0x01 +#define TPA6130A2_REG_VOL_MUTE 0x02 +#define TPA6130A2_REG_OUT_IMPEDANCE 0x03 +#define TPA6130A2_REG_VERSION 0x04 + +#define TPA6130A2_CACHEREGNUM (TPA6130A2_REG_VERSION + 1) + +/* Register bits */ +/* TPA6130A2_REG_CONTROL (0x01) */ +#define TPA6130A2_SWS (0x01 << 0) +#define TPA6130A2_TERMAL (0x01 << 1) +#define TPA6130A2_MODE(x) (x << 4) +#define TPA6130A2_MODE_STEREO (0x00) +#define TPA6130A2_MODE_DUAL_MONO (0x01) +#define TPA6130A2_MODE_BRIDGE (0x02) +#define TPA6130A2_MODE_MASK (0x03) +#define TPA6130A2_HP_EN_R (0x01 << 6) +#define TPA6130A2_HP_EN_L (0x01 << 7) + +/* TPA6130A2_REG_VOL_MUTE (0x02) */ +#define TPA6130A2_VOLUME(x) ((x & 0x3f) << 0) +#define TPA6130A2_MUTE_R (0x01 << 6) +#define TPA6130A2_MUTE_L (0x01 << 7) + +/* TPA6130A2_REG_OUT_IMPEDANCE (0x03) */ +#define TPA6130A2_HIZ_R (0x01 << 0) +#define TPA6130A2_HIZ_L (0x01 << 1) + +/* TPA6130A2_REG_VERSION (0x04) */ +#define TPA6130A2_VERSION_MASK (0x0f) + +extern int tpa6130a2_add_controls(struct snd_soc_codec *codec); + +#endif /* __TPA6130A2_H__ */ diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 4df7c6c..5f1681f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -120,9 +120,10 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { /* codec private data */ struct twl4030_priv { - unsigned int bypass_state; + struct snd_soc_codec codec; + unsigned int codec_powered; - unsigned int codec_muted; + unsigned int apll_enabled; struct snd_pcm_substream *master_substream; struct snd_pcm_substream *slave_substream; @@ -183,19 +184,20 @@ static int twl4030_write(struct snd_soc_codec *codec, static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) { struct twl4030_priv *twl4030 = codec->private_data; - u8 mode; + int mode; if (enable == twl4030->codec_powered) return; - mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); if (enable) - mode |= TWL4030_CODECPDZ; + mode = twl4030_codec_enable_resource(TWL4030_CODEC_RES_POWER); else - mode &= ~TWL4030_CODECPDZ; + mode = twl4030_codec_disable_resource(TWL4030_CODEC_RES_POWER); - twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); - twl4030->codec_powered = enable; + if (mode >= 0) { + twl4030_write_reg_cache(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030->codec_powered = enable; + } /* REVISIT: this delay is present in TI sample drivers */ /* but there seems to be no TRM requirement for it */ @@ -212,31 +214,30 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) /* set all audio section registers to reasonable defaults */ for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++) - twl4030_write(codec, i, cache[i]); + if (i != TWL4030_REG_APLL_CTL) + twl4030_write(codec, i, cache[i]); } -static void twl4030_codec_mute(struct snd_soc_codec *codec, int mute) +static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable) { struct twl4030_priv *twl4030 = codec->private_data; - u8 reg_val; + int status; - if (mute == twl4030->codec_muted) + if (enable == twl4030->apll_enabled) return; - if (mute) { - /* Disable PLL */ - reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL); - reg_val &= ~TWL4030_APLL_EN; - twl4030_write(codec, TWL4030_REG_APLL_CTL, reg_val); - } else { + if (enable) /* Enable PLL */ - reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL); - reg_val |= TWL4030_APLL_EN; - twl4030_write(codec, TWL4030_REG_APLL_CTL, reg_val); - } + status = twl4030_codec_enable_resource(TWL4030_CODEC_RES_APLL); + else + /* Disable PLL */ + status = twl4030_codec_disable_resource(TWL4030_CODEC_RES_APLL); + + if (status >= 0) + twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status); - twl4030->codec_muted = mute; + twl4030->apll_enabled = enable; } static void twl4030_power_up(struct snd_soc_codec *codec) @@ -613,6 +614,27 @@ static int handsfreerpga_event(struct snd_soc_dapm_widget *w, return 0; } +static int vibramux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + twl4030_write(w->codec, TWL4030_REG_VIBRA_SET, 0xff); + return 0; +} + +static int apll_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + twl4030_apll_enable(w->codec, 1); + break; + case SND_SOC_DAPM_POST_PMD: + twl4030_apll_enable(w->codec, 0); + break; + } + return 0; +} + static void headset_ramp(struct snd_soc_codec *codec, int ramp) { struct snd_soc_device *socdev = codec->socdev; @@ -724,67 +746,6 @@ static int headsetrpga_event(struct snd_soc_dapm_widget *w, return 0; } -static int bypass_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct soc_mixer_control *m = - (struct soc_mixer_control *)w->kcontrols->private_value; - struct twl4030_priv *twl4030 = w->codec->private_data; - unsigned char reg, misc; - - reg = twl4030_read_reg_cache(w->codec, m->reg); - - /* - * bypass_state[0:3] - analog HiFi bypass - * bypass_state[4] - analog voice bypass - * bypass_state[5] - digital voice bypass - * bypass_state[6:7] - digital HiFi bypass - */ - if (m->reg == TWL4030_REG_VSTPGA) { - /* Voice digital bypass */ - if (reg) - twl4030->bypass_state |= (1 << 5); - else - twl4030->bypass_state &= ~(1 << 5); - } else if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { - /* Analog bypass */ - if (reg & (1 << m->shift)) - twl4030->bypass_state |= - (1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL)); - else - twl4030->bypass_state &= - ~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL)); - } else if (m->reg == TWL4030_REG_VDL_APGA_CTL) { - /* Analog voice bypass */ - if (reg & (1 << m->shift)) - twl4030->bypass_state |= (1 << 4); - else - twl4030->bypass_state &= ~(1 << 4); - } else { - /* Digital bypass */ - if (reg & (0x7 << m->shift)) - twl4030->bypass_state |= (1 << (m->shift ? 7 : 6)); - else - twl4030->bypass_state &= ~(1 << (m->shift ? 7 : 6)); - } - - /* Enable master analog loopback mode if any analog switch is enabled*/ - misc = twl4030_read_reg_cache(w->codec, TWL4030_REG_MISC_SET_1); - if (twl4030->bypass_state & 0x1F) - misc |= TWL4030_FMLOOP_EN; - else - misc &= ~TWL4030_FMLOOP_EN; - twl4030_write(w->codec, TWL4030_REG_MISC_SET_1, misc); - - if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) { - if (twl4030->bypass_state) - twl4030_codec_mute(w->codec, 0); - else - twl4030_codec_mute(w->codec, 1); - } - return 0; -} - /* * Some of the gain controls in TWL (mostly those which are associated with * the outputs) are implemented in an interesting way: @@ -1192,32 +1153,28 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_NOPM, 0, 0), /* Analog bypasses */ - SND_SOC_DAPM_SWITCH_E("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_abypassr1_control, bypass_event, - SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_SWITCH_E("Left1 Analog Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_abypassl1_control, - bypass_event, SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_SWITCH_E("Right2 Analog Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_abypassr2_control, - bypass_event, SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_SWITCH_E("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_abypassl2_control, - bypass_event, SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_SWITCH_E("Voice Analog Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_abypassv_control, - bypass_event, SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassr1_control), + SND_SOC_DAPM_SWITCH("Left1 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassl1_control), + SND_SOC_DAPM_SWITCH("Right2 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassr2_control), + SND_SOC_DAPM_SWITCH("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassl2_control), + SND_SOC_DAPM_SWITCH("Voice Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassv_control), + + /* Master analog loopback switch */ + SND_SOC_DAPM_SUPPLY("FM Loop Enable", TWL4030_REG_MISC_SET_1, 5, 0, + NULL, 0), /* Digital bypasses */ - SND_SOC_DAPM_SWITCH_E("Left Digital Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_dbypassl_control, bypass_event, - SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_SWITCH_E("Right Digital Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_dbypassr_control, bypass_event, - SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_SWITCH_E("Voice Digital Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_dbypassv_control, bypass_event, - SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH("Left Digital Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_dbypassl_control), + SND_SOC_DAPM_SWITCH("Right Digital Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_dbypassr_control), + SND_SOC_DAPM_SWITCH("Voice Digital Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_dbypassv_control), /* Digital mixers, power control for the physical DACs */ SND_SOC_DAPM_MIXER("Digital R1 Playback Mixer", @@ -1243,6 +1200,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer", TWL4030_REG_VDL_APGA_CTL, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("APLL Enable", SND_SOC_NOPM, 0, 0, apll_event, + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), + /* Output MIXER controls */ /* Earpiece */ SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, @@ -1308,8 +1268,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { 0, 0, NULL, 0, handsfreerpga_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), /* Vibra */ - SND_SOC_DAPM_MUX("Vibra Mux", TWL4030_REG_VIBRA_CTL, 0, 0, - &twl4030_dapm_vibra_control), + SND_SOC_DAPM_MUX_E("Vibra Mux", TWL4030_REG_VIBRA_CTL, 0, 0, + &twl4030_dapm_vibra_control, vibramux_event, + SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_MUX("Vibra Route", SND_SOC_NOPM, 0, 0, &twl4030_dapm_vibrapath_control), @@ -1369,6 +1330,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Digital R2 Playback Mixer", NULL, "DAC Right2"}, {"Digital Voice Playback Mixer", NULL, "DAC Voice"}, + /* Supply for the digital part (APLL) */ + {"Digital R1 Playback Mixer", NULL, "APLL Enable"}, + {"Digital L1 Playback Mixer", NULL, "APLL Enable"}, + {"Digital R2 Playback Mixer", NULL, "APLL Enable"}, + {"Digital L2 Playback Mixer", NULL, "APLL Enable"}, + {"Digital Voice Playback Mixer", NULL, "APLL Enable"}, + {"Analog L1 Playback Mixer", NULL, "Digital L1 Playback Mixer"}, {"Analog R1 Playback Mixer", NULL, "Digital R1 Playback Mixer"}, {"Analog L2 Playback Mixer", NULL, "Digital L2 Playback Mixer"}, @@ -1482,6 +1450,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Virtual Left2", NULL, "TX2 Capture Route"}, {"ADC Virtual Right2", NULL, "TX2 Capture Route"}, + {"ADC Virtual Left1", NULL, "APLL Enable"}, + {"ADC Virtual Right1", NULL, "APLL Enable"}, + {"ADC Virtual Left2", NULL, "APLL Enable"}, + {"ADC Virtual Right2", NULL, "APLL Enable"}, + /* Analog bypass routes */ {"Right1 Analog Loopback", "Switch", "Analog Right"}, {"Left1 Analog Loopback", "Switch", "Analog Left"}, @@ -1489,6 +1462,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left2 Analog Loopback", "Switch", "Analog Left"}, {"Voice Analog Loopback", "Switch", "Analog Left"}, + /* Supply for the Analog loopbacks */ + {"Right1 Analog Loopback", NULL, "FM Loop Enable"}, + {"Left1 Analog Loopback", NULL, "FM Loop Enable"}, + {"Right2 Analog Loopback", NULL, "FM Loop Enable"}, + {"Left2 Analog Loopback", NULL, "FM Loop Enable"}, + {"Voice Analog Loopback", NULL, "FM Loop Enable"}, + {"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"}, {"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"}, {"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"}, @@ -1513,32 +1493,20 @@ static int twl4030_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); return 0; } static int twl4030_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - struct twl4030_priv *twl4030 = codec->private_data; - switch (level) { case SND_SOC_BIAS_ON: - twl4030_codec_mute(codec, 0); break; case SND_SOC_BIAS_PREPARE: - twl4030_power_up(codec); - if (twl4030->bypass_state) - twl4030_codec_mute(codec, 0); - else - twl4030_codec_mute(codec, 1); break; case SND_SOC_BIAS_STANDBY: - twl4030_power_up(codec); - if (twl4030->bypass_state) - twl4030_codec_mute(codec, 0); - else - twl4030_codec_mute(codec, 1); + if (codec->bias_level == SND_SOC_BIAS_OFF) + twl4030_power_up(codec); break; case SND_SOC_BIAS_OFF: twl4030_power_down(codec); @@ -1785,29 +1753,23 @@ static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct twl4030_priv *twl4030 = codec->private_data; - u8 infreq; switch (freq) { case 19200000: - infreq = TWL4030_APLL_INFREQ_19200KHZ; - twl4030->sysclk = 19200; - break; case 26000000: - infreq = TWL4030_APLL_INFREQ_26000KHZ; - twl4030->sysclk = 26000; - break; case 38400000: - infreq = TWL4030_APLL_INFREQ_38400KHZ; - twl4030->sysclk = 38400; break; default: - printk(KERN_ERR "TWL4030 set sysclk: unknown rate %d\n", - freq); + dev_err(codec->dev, "Unsupported APLL mclk: %u\n", freq); return -EINVAL; } - infreq |= TWL4030_APLL_EN; - twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq); + if ((freq / 1000) != twl4030->sysclk) { + dev_err(codec->dev, + "Mismatch in APLL mclk: %u (configured: %u)\n", + freq, twl4030->sysclk * 1000); + return -EINVAL; + } return 0; } @@ -1905,18 +1867,16 @@ static int twl4030_voice_startup(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->card->codec; - u8 infreq; + struct twl4030_priv *twl4030 = codec->private_data; u8 mode; /* If the system master clock is not 26MHz, the voice PCM interface is * not avilable. */ - infreq = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL) - & TWL4030_APLL_INFREQ; - - if (infreq != TWL4030_APLL_INFREQ_26000KHZ) { - printk(KERN_ERR "TWL4030 voice startup: " - "MCLK is not 26MHz, call set_sysclk() on init\n"); + if (twl4030->sysclk != 26000) { + dev_err(codec->dev, "The board is configured for %u Hz, while" + "the Voice interface needs 26MHz APLL mclk\n", + twl4030->sysclk * 1000); return -EINVAL; } @@ -1989,21 +1949,19 @@ static int twl4030_voice_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; - u8 infreq; + struct twl4030_priv *twl4030 = codec->private_data; - switch (freq) { - case 26000000: - infreq = TWL4030_APLL_INFREQ_26000KHZ; - break; - default: - printk(KERN_ERR "TWL4030 voice set sysclk: unknown rate %d\n", - freq); + if (freq != 26000000) { + dev_err(codec->dev, "Unsupported APLL mclk: %u, the Voice" + "interface needs 26MHz APLL mclk\n", freq); + return -EINVAL; + } + if ((freq / 1000) != twl4030->sysclk) { + dev_err(codec->dev, + "Mismatch in APLL mclk: %u (configured: %u)\n", + freq, twl4030->sysclk * 1000); return -EINVAL; } - - infreq |= TWL4030_APLL_EN; - twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq); - return 0; } @@ -2121,7 +2079,7 @@ struct snd_soc_dai twl4030_dai[] = { }; EXPORT_SYMBOL_GPL(twl4030_dai); -static int twl4030_suspend(struct platform_device *pdev, pm_message_t state) +static int twl4030_soc_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; @@ -2131,7 +2089,7 @@ static int twl4030_suspend(struct platform_device *pdev, pm_message_t state) return 0; } -static int twl4030_resume(struct platform_device *pdev) +static int twl4030_soc_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; @@ -2141,147 +2099,181 @@ static int twl4030_resume(struct platform_device *pdev) return 0; } -/* - * initialize the driver - * register the mixer and dsp interfaces with the kernel - */ +static struct snd_soc_codec *twl4030_codec; -static int twl4030_init(struct snd_soc_device *socdev) +static int twl4030_soc_probe(struct platform_device *pdev) { - struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct twl4030_setup_data *setup = socdev->codec_data; - struct twl4030_priv *twl4030 = codec->private_data; - int ret = 0; + struct snd_soc_codec *codec; + struct twl4030_priv *twl4030; + int ret; - printk(KERN_INFO "TWL4030 Audio Codec init \n"); + BUG_ON(!twl4030_codec); - codec->name = "twl4030"; - codec->owner = THIS_MODULE; - codec->read = twl4030_read_reg_cache; - codec->write = twl4030_write; - codec->set_bias_level = twl4030_set_bias_level; - codec->dai = twl4030_dai; - codec->num_dai = ARRAY_SIZE(twl4030_dai), - codec->reg_cache_size = sizeof(twl4030_reg); - codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), - GFP_KERNEL); - if (codec->reg_cache == NULL) - return -ENOMEM; + codec = twl4030_codec; + twl4030 = codec->private_data; + socdev->card->codec = codec; /* Configuration for headset ramp delay from setup data */ if (setup) { unsigned char hs_pop; - if (setup->sysclk) - twl4030->sysclk = setup->sysclk; - else - twl4030->sysclk = 26000; + if (setup->sysclk != twl4030->sysclk) + dev_warn(&pdev->dev, + "Mismatch in APLL mclk: %u (configured: %u)\n", + setup->sysclk, twl4030->sysclk); hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); hs_pop &= ~TWL4030_RAMP_DELAY; hs_pop |= (setup->ramp_delay_value << 2); twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, hs_pop); - } else { - twl4030->sysclk = 26000; } /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { - printk(KERN_ERR "twl4030: failed to create pcms\n"); - goto pcm_err; + dev_err(&pdev->dev, "failed to create pcms\n"); + return ret; } - twl4030_init_chip(codec); - - /* power on device */ - twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, twl4030_snd_controls, ARRAY_SIZE(twl4030_snd_controls)); twl4030_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "twl4030: failed to register card\n"); - goto card_err; - } + return 0; +} - return ret; +static int twl4030_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; -card_err: + twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -pcm_err: - kfree(codec->reg_cache); - return ret; -} + kfree(codec->private_data); + kfree(codec); -static struct snd_soc_device *twl4030_socdev; + return 0; +} -static int twl4030_probe(struct platform_device *pdev) +static int __devinit twl4030_codec_probe(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct twl4030_codec_audio_data *pdata = pdev->dev.platform_data; struct snd_soc_codec *codec; struct twl4030_priv *twl4030; + int ret; - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; + if (!pdata) { + dev_err(&pdev->dev, "platform_data is missing\n"); + return -EINVAL; + } twl4030 = kzalloc(sizeof(struct twl4030_priv), GFP_KERNEL); if (twl4030 == NULL) { - kfree(codec); + dev_err(&pdev->dev, "Can not allocate memroy\n"); return -ENOMEM; } + codec = &twl4030->codec; codec->private_data = twl4030; - socdev->card->codec = codec; + codec->dev = &pdev->dev; + twl4030_dai[0].dev = &pdev->dev; + twl4030_dai[1].dev = &pdev->dev; + mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); - twl4030_socdev = socdev; - twl4030_init(socdev); + codec->name = "twl4030"; + codec->owner = THIS_MODULE; + codec->read = twl4030_read_reg_cache; + codec->write = twl4030_write; + codec->set_bias_level = twl4030_set_bias_level; + codec->dai = twl4030_dai; + codec->num_dai = ARRAY_SIZE(twl4030_dai), + codec->reg_cache_size = sizeof(twl4030_reg); + codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto error_cache; + } + + platform_set_drvdata(pdev, twl4030); + twl4030_codec = codec; + + /* Set the defaults, and power up the codec */ + twl4030->sysclk = twl4030_codec_get_mclk() / 1000; + twl4030_init_chip(codec); + codec->bias_level = SND_SOC_BIAS_OFF; + twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto error_codec; + } + + ret = snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", ret); + snd_soc_unregister_codec(codec); + goto error_codec; + } return 0; + +error_codec: + twl4030_power_down(codec); + kfree(codec->reg_cache); +error_cache: + kfree(twl4030); + return ret; } -static int twl4030_remove(struct platform_device *pdev) +static int __devexit twl4030_codec_remove(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; + struct twl4030_priv *twl4030 = platform_get_drvdata(pdev); - printk(KERN_INFO "TWL4030 Audio Codec remove\n"); - twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); - kfree(codec->private_data); - kfree(codec); + kfree(twl4030); + twl4030_codec = NULL; return 0; } -struct snd_soc_codec_device soc_codec_dev_twl4030 = { - .probe = twl4030_probe, - .remove = twl4030_remove, - .suspend = twl4030_suspend, - .resume = twl4030_resume, +MODULE_ALIAS("platform:twl4030_codec_audio"); + +static struct platform_driver twl4030_codec_driver = { + .probe = twl4030_codec_probe, + .remove = __devexit_p(twl4030_codec_remove), + .driver = { + .name = "twl4030_codec_audio", + .owner = THIS_MODULE, + }, }; -EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); static int __init twl4030_modinit(void) { - return snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); + return platform_driver_register(&twl4030_codec_driver); } module_init(twl4030_modinit); static void __exit twl4030_exit(void) { - snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); + platform_driver_unregister(&twl4030_codec_driver); } module_exit(twl4030_exit); +struct snd_soc_codec_device soc_codec_dev_twl4030 = { + .probe = twl4030_soc_probe, + .remove = twl4030_soc_remove, + .suspend = twl4030_soc_suspend, + .resume = twl4030_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); + MODULE_DESCRIPTION("ASoC TWL4030 codec driver"); MODULE_AUTHOR("Steve Sakoman"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 2b4bfa2..dd6396e 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -22,245 +22,13 @@ #ifndef __TWL4030_AUDIO_H__ #define __TWL4030_AUDIO_H__ -#define TWL4030_REG_CODEC_MODE 0x1 -#define TWL4030_REG_OPTION 0x2 -#define TWL4030_REG_UNKNOWN 0x3 -#define TWL4030_REG_MICBIAS_CTL 0x4 -#define TWL4030_REG_ANAMICL 0x5 -#define TWL4030_REG_ANAMICR 0x6 -#define TWL4030_REG_AVADC_CTL 0x7 -#define TWL4030_REG_ADCMICSEL 0x8 -#define TWL4030_REG_DIGMIXING 0x9 -#define TWL4030_REG_ATXL1PGA 0xA -#define TWL4030_REG_ATXR1PGA 0xB -#define TWL4030_REG_AVTXL2PGA 0xC -#define TWL4030_REG_AVTXR2PGA 0xD -#define TWL4030_REG_AUDIO_IF 0xE -#define TWL4030_REG_VOICE_IF 0xF -#define TWL4030_REG_ARXR1PGA 0x10 -#define TWL4030_REG_ARXL1PGA 0x11 -#define TWL4030_REG_ARXR2PGA 0x12 -#define TWL4030_REG_ARXL2PGA 0x13 -#define TWL4030_REG_VRXPGA 0x14 -#define TWL4030_REG_VSTPGA 0x15 -#define TWL4030_REG_VRX2ARXPGA 0x16 -#define TWL4030_REG_AVDAC_CTL 0x17 -#define TWL4030_REG_ARX2VTXPGA 0x18 -#define TWL4030_REG_ARXL1_APGA_CTL 0x19 -#define TWL4030_REG_ARXR1_APGA_CTL 0x1A -#define TWL4030_REG_ARXL2_APGA_CTL 0x1B -#define TWL4030_REG_ARXR2_APGA_CTL 0x1C -#define TWL4030_REG_ATX2ARXPGA 0x1D -#define TWL4030_REG_BT_IF 0x1E -#define TWL4030_REG_BTPGA 0x1F -#define TWL4030_REG_BTSTPGA 0x20 -#define TWL4030_REG_EAR_CTL 0x21 -#define TWL4030_REG_HS_SEL 0x22 -#define TWL4030_REG_HS_GAIN_SET 0x23 -#define TWL4030_REG_HS_POPN_SET 0x24 -#define TWL4030_REG_PREDL_CTL 0x25 -#define TWL4030_REG_PREDR_CTL 0x26 -#define TWL4030_REG_PRECKL_CTL 0x27 -#define TWL4030_REG_PRECKR_CTL 0x28 -#define TWL4030_REG_HFL_CTL 0x29 -#define TWL4030_REG_HFR_CTL 0x2A -#define TWL4030_REG_ALC_CTL 0x2B -#define TWL4030_REG_ALC_SET1 0x2C -#define TWL4030_REG_ALC_SET2 0x2D -#define TWL4030_REG_BOOST_CTL 0x2E -#define TWL4030_REG_SOFTVOL_CTL 0x2F -#define TWL4030_REG_DTMF_FREQSEL 0x30 -#define TWL4030_REG_DTMF_TONEXT1H 0x31 -#define TWL4030_REG_DTMF_TONEXT1L 0x32 -#define TWL4030_REG_DTMF_TONEXT2H 0x33 -#define TWL4030_REG_DTMF_TONEXT2L 0x34 -#define TWL4030_REG_DTMF_TONOFF 0x35 -#define TWL4030_REG_DTMF_WANONOFF 0x36 -#define TWL4030_REG_I2S_RX_SCRAMBLE_H 0x37 -#define TWL4030_REG_I2S_RX_SCRAMBLE_M 0x38 -#define TWL4030_REG_I2S_RX_SCRAMBLE_L 0x39 -#define TWL4030_REG_APLL_CTL 0x3A -#define TWL4030_REG_DTMF_CTL 0x3B -#define TWL4030_REG_DTMF_PGA_CTL2 0x3C -#define TWL4030_REG_DTMF_PGA_CTL1 0x3D -#define TWL4030_REG_MISC_SET_1 0x3E -#define TWL4030_REG_PCMBTMUX 0x3F -#define TWL4030_REG_RX_PATH_SEL 0x43 -#define TWL4030_REG_VDL_APGA_CTL 0x44 -#define TWL4030_REG_VIBRA_CTL 0x45 -#define TWL4030_REG_VIBRA_SET 0x46 -#define TWL4030_REG_VIBRA_PWM_SET 0x47 -#define TWL4030_REG_ANAMIC_GAIN 0x48 -#define TWL4030_REG_MISC_SET_2 0x49 -#define TWL4030_REG_SW_SHADOW 0x4A +/* Register descriptions are here */ +#include <linux/mfd/twl4030-codec.h> +/* Sgadow register used by the audio driver */ +#define TWL4030_REG_SW_SHADOW 0x4A #define TWL4030_CACHEREGNUM (TWL4030_REG_SW_SHADOW + 1) -/* Bitfield Definitions */ - -/* TWL4030_CODEC_MODE (0x01) Fields */ - -#define TWL4030_APLL_RATE 0xF0 -#define TWL4030_APLL_RATE_8000 0x00 -#define TWL4030_APLL_RATE_11025 0x10 -#define TWL4030_APLL_RATE_12000 0x20 -#define TWL4030_APLL_RATE_16000 0x40 -#define TWL4030_APLL_RATE_22050 0x50 -#define TWL4030_APLL_RATE_24000 0x60 -#define TWL4030_APLL_RATE_32000 0x80 -#define TWL4030_APLL_RATE_44100 0x90 -#define TWL4030_APLL_RATE_48000 0xA0 -#define TWL4030_APLL_RATE_96000 0xE0 -#define TWL4030_SEL_16K 0x08 -#define TWL4030_CODECPDZ 0x02 -#define TWL4030_OPT_MODE 0x01 -#define TWL4030_OPTION_1 (1 << 0) -#define TWL4030_OPTION_2 (0 << 0) - -/* TWL4030_OPTION (0x02) Fields */ - -#define TWL4030_ATXL1_EN (1 << 0) -#define TWL4030_ATXR1_EN (1 << 1) -#define TWL4030_ATXL2_VTXL_EN (1 << 2) -#define TWL4030_ATXR2_VTXR_EN (1 << 3) -#define TWL4030_ARXL1_VRX_EN (1 << 4) -#define TWL4030_ARXR1_EN (1 << 5) -#define TWL4030_ARXL2_EN (1 << 6) -#define TWL4030_ARXR2_EN (1 << 7) - -/* TWL4030_REG_MICBIAS_CTL (0x04) Fields */ - -#define TWL4030_MICBIAS2_CTL 0x40 -#define TWL4030_MICBIAS1_CTL 0x20 -#define TWL4030_HSMICBIAS_EN 0x04 -#define TWL4030_MICBIAS2_EN 0x02 -#define TWL4030_MICBIAS1_EN 0x01 - -/* ANAMICL (0x05) Fields */ - -#define TWL4030_CNCL_OFFSET_START 0x80 -#define TWL4030_OFFSET_CNCL_SEL 0x60 -#define TWL4030_OFFSET_CNCL_SEL_ARX1 0x00 -#define TWL4030_OFFSET_CNCL_SEL_ARX2 0x20 -#define TWL4030_OFFSET_CNCL_SEL_VRX 0x40 -#define TWL4030_OFFSET_CNCL_SEL_ALL 0x60 -#define TWL4030_MICAMPL_EN 0x10 -#define TWL4030_CKMIC_EN 0x08 -#define TWL4030_AUXL_EN 0x04 -#define TWL4030_HSMIC_EN 0x02 -#define TWL4030_MAINMIC_EN 0x01 - -/* ANAMICR (0x06) Fields */ - -#define TWL4030_MICAMPR_EN 0x10 -#define TWL4030_AUXR_EN 0x04 -#define TWL4030_SUBMIC_EN 0x01 - -/* AVADC_CTL (0x07) Fields */ - -#define TWL4030_ADCL_EN 0x08 -#define TWL4030_AVADC_CLK_PRIORITY 0x04 -#define TWL4030_ADCR_EN 0x02 - -/* TWL4030_REG_ADCMICSEL (0x08) Fields */ - -#define TWL4030_DIGMIC1_EN 0x08 -#define TWL4030_TX2IN_SEL 0x04 -#define TWL4030_DIGMIC0_EN 0x02 -#define TWL4030_TX1IN_SEL 0x01 - -/* AUDIO_IF (0x0E) Fields */ - -#define TWL4030_AIF_SLAVE_EN 0x80 -#define TWL4030_DATA_WIDTH 0x60 -#define TWL4030_DATA_WIDTH_16S_16W 0x00 -#define TWL4030_DATA_WIDTH_32S_16W 0x40 -#define TWL4030_DATA_WIDTH_32S_24W 0x60 -#define TWL4030_AIF_FORMAT 0x18 -#define TWL4030_AIF_FORMAT_CODEC 0x00 -#define TWL4030_AIF_FORMAT_LEFT 0x08 -#define TWL4030_AIF_FORMAT_RIGHT 0x10 -#define TWL4030_AIF_FORMAT_TDM 0x18 -#define TWL4030_AIF_TRI_EN 0x04 -#define TWL4030_CLK256FS_EN 0x02 -#define TWL4030_AIF_EN 0x01 - -/* VOICE_IF (0x0F) Fields */ - -#define TWL4030_VIF_SLAVE_EN 0x80 -#define TWL4030_VIF_DIN_EN 0x40 -#define TWL4030_VIF_DOUT_EN 0x20 -#define TWL4030_VIF_SWAP 0x10 -#define TWL4030_VIF_FORMAT 0x08 -#define TWL4030_VIF_TRI_EN 0x04 -#define TWL4030_VIF_SUB_EN 0x02 -#define TWL4030_VIF_EN 0x01 - -/* EAR_CTL (0x21) */ -#define TWL4030_EAR_GAIN 0x30 - -/* HS_GAIN_SET (0x23) Fields */ - -#define TWL4030_HSR_GAIN 0x0C -#define TWL4030_HSR_GAIN_PWR_DOWN 0x00 -#define TWL4030_HSR_GAIN_PLUS_6DB 0x04 -#define TWL4030_HSR_GAIN_0DB 0x08 -#define TWL4030_HSR_GAIN_MINUS_6DB 0x0C -#define TWL4030_HSL_GAIN 0x03 -#define TWL4030_HSL_GAIN_PWR_DOWN 0x00 -#define TWL4030_HSL_GAIN_PLUS_6DB 0x01 -#define TWL4030_HSL_GAIN_0DB 0x02 -#define TWL4030_HSL_GAIN_MINUS_6DB 0x03 - -/* HS_POPN_SET (0x24) Fields */ - -#define TWL4030_VMID_EN 0x40 -#define TWL4030_EXTMUTE 0x20 -#define TWL4030_RAMP_DELAY 0x1C -#define TWL4030_RAMP_DELAY_20MS 0x00 -#define TWL4030_RAMP_DELAY_40MS 0x04 -#define TWL4030_RAMP_DELAY_81MS 0x08 -#define TWL4030_RAMP_DELAY_161MS 0x0C -#define TWL4030_RAMP_DELAY_323MS 0x10 -#define TWL4030_RAMP_DELAY_645MS 0x14 -#define TWL4030_RAMP_DELAY_1291MS 0x18 -#define TWL4030_RAMP_DELAY_2581MS 0x1C -#define TWL4030_RAMP_EN 0x02 - -/* PREDL_CTL (0x25) */ -#define TWL4030_PREDL_GAIN 0x30 - -/* PREDR_CTL (0x26) */ -#define TWL4030_PREDR_GAIN 0x30 - -/* PRECKL_CTL (0x27) */ -#define TWL4030_PRECKL_GAIN 0x30 - -/* PRECKR_CTL (0x28) */ -#define TWL4030_PRECKR_GAIN 0x30 - -/* HFL_CTL (0x29, 0x2A) Fields */ -#define TWL4030_HF_CTL_HB_EN 0x04 -#define TWL4030_HF_CTL_LOOP_EN 0x08 -#define TWL4030_HF_CTL_RAMP_EN 0x10 -#define TWL4030_HF_CTL_REF_EN 0x20 - -/* APLL_CTL (0x3A) Fields */ - -#define TWL4030_APLL_EN 0x10 -#define TWL4030_APLL_INFREQ 0x0F -#define TWL4030_APLL_INFREQ_19200KHZ 0x05 -#define TWL4030_APLL_INFREQ_26000KHZ 0x06 -#define TWL4030_APLL_INFREQ_38400KHZ 0x0F - -/* REG_MISC_SET_1 (0x3E) Fields */ - -#define TWL4030_CLK64_EN 0x80 -#define TWL4030_SCRAMBLE_EN 0x40 -#define TWL4030_FMLOOP_EN 0x20 -#define TWL4030_SMOOTH_ANAVOL_EN 0x02 -#define TWL4030_DIGMIC_LR_SWAP_EN 0x01 - /* TWL4030_REG_SW_SHADOW (0x4A) Fields */ #define TWL4030_HFL_EN 0x01 #define TWL4030_HFR_EN 0x02 @@ -279,3 +47,5 @@ struct twl4030_setup_data { }; #endif /* End of __TWL4030_AUDIO_H__ */ + + diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index c33b92e..3e99fe5 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -101,7 +101,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg, pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value); if (reg >= UDA134X_REGS_NUM) { - printk(KERN_ERR "%s unkown register: reg: %u", + printk(KERN_ERR "%s unknown register: reg: %u", __func__, reg); return -EINVAL; } @@ -552,7 +552,7 @@ static int uda134x_soc_probe(struct platform_device *pdev) ARRAY_SIZE(uda1341_snd_controls)); break; default: - printk(KERN_ERR "%s unkown codec type: %d", + printk(KERN_ERR "%s unknown codec type: %d", __func__, pd->model); return -EINVAL; } @@ -562,17 +562,8 @@ static int uda134x_soc_probe(struct platform_device *pdev) goto pcm_err; } - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "UDA134X: failed to register card\n"); - goto card_err; - } - return 0; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: kfree(codec->reg_cache); reg_err: diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 92ec034..a2763c2 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -378,7 +378,6 @@ static int uda1380_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -713,17 +712,9 @@ static int uda1380_probe(struct platform_device *pdev) snd_soc_add_controls(codec, uda1380_snd_controls, ARRAY_SIZE(uda1380_snd_controls)); uda1380_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 593d5b9..f82125d 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -800,7 +800,7 @@ static int wm8350_add_widgets(struct snd_soc_codec *codec) return ret; } - return snd_soc_dapm_new_widgets(codec); + return 0; } static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai, @@ -1101,7 +1101,7 @@ static inline int fll_factors(struct _fll_div *fll_div, unsigned int input, } static int wm8350_set_fll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, + int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1501,18 +1501,7 @@ static int wm8350_probe(struct platform_device *pdev) wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&pdev->dev, "failed to register card\n"); - goto card_err; - } - return 0; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); - return ret; } static int wm8350_remove(struct platform_device *pdev) @@ -1680,21 +1669,6 @@ static int __devexit wm8350_codec_remove(struct platform_device *pdev) return 0; } -#ifdef CONFIG_PM -static int wm8350_codec_suspend(struct platform_device *pdev, pm_message_t m) -{ - return snd_soc_suspend_device(&pdev->dev); -} - -static int wm8350_codec_resume(struct platform_device *pdev) -{ - return snd_soc_resume_device(&pdev->dev); -} -#else -#define wm8350_codec_suspend NULL -#define wm8350_codec_resume NULL -#endif - static struct platform_driver wm8350_codec_driver = { .driver = { .name = "wm8350-codec", @@ -1702,8 +1676,6 @@ static struct platform_driver wm8350_codec_driver = { }, .probe = wm8350_codec_probe, .remove = __devexit_p(wm8350_codec_remove), - .suspend = wm8350_codec_suspend, - .resume = wm8350_codec_resume, }; static __init int wm8350_init(void) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index b9ef4d9..b432f4d 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -915,7 +915,6 @@ static int wm8400_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -1011,7 +1010,8 @@ static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors, } static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, - unsigned int freq_in, unsigned int freq_out) + int source, unsigned int freq_in, + unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; struct wm8400_priv *wm8400 = codec->private_data; @@ -1399,17 +1399,6 @@ static int wm8400_probe(struct platform_device *pdev) wm8400_add_controls(codec); wm8400_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&pdev->dev, "failed to register card\n"); - goto card_err; - } - - return ret; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } @@ -1558,21 +1547,6 @@ static int __exit wm8400_codec_remove(struct platform_device *dev) return 0; } -#ifdef CONFIG_PM -static int wm8400_pdev_suspend(struct platform_device *pdev, pm_message_t msg) -{ - return snd_soc_suspend_device(&pdev->dev); -} - -static int wm8400_pdev_resume(struct platform_device *pdev) -{ - return snd_soc_resume_device(&pdev->dev); -} -#else -#define wm8400_pdev_suspend NULL -#define wm8400_pdev_resume NULL -#endif - static struct platform_driver wm8400_codec_driver = { .driver = { .name = "wm8400-codec", @@ -1580,8 +1554,6 @@ static struct platform_driver wm8400_codec_driver = { }, .probe = wm8400_codec_probe, .remove = __exit_p(wm8400_codec_remove), - .suspend = wm8400_pdev_suspend, - .resume = wm8400_pdev_resume, }; static int __init wm8400_codec_init(void) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 060d5d0..265e68c 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -219,7 +219,6 @@ static int wm8510_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -271,8 +270,8 @@ static void pll_factors(unsigned int target, unsigned int source) pll_div.k = K; } -static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; u16 reg; @@ -604,16 +603,9 @@ static int wm8510_init(struct snd_soc_device *socdev, snd_soc_add_controls(codec, wm8510_snd_controls, ARRAY_SIZE(wm8510_snd_controls)); wm8510_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8510: failed to register card\n"); - goto card_err; - } + return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 25870a4..d3a61d7 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -117,7 +117,6 @@ static int wm8523_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -448,17 +447,9 @@ static int wm8523_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm8523_snd_controls, ARRAY_SIZE(wm8523_snd_controls)); wm8523_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } @@ -638,21 +629,6 @@ static __devexit int wm8523_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8523_i2c_suspend(struct i2c_client *i2c, pm_message_t msg) -{ - return snd_soc_suspend_device(&i2c->dev); -} - -static int wm8523_i2c_resume(struct i2c_client *i2c) -{ - return snd_soc_resume_device(&i2c->dev); -} -#else -#define wm8523_i2c_suspend NULL -#define wm8523_i2c_resume NULL -#endif - static const struct i2c_device_id wm8523_i2c_id[] = { { "wm8523", 0 }, { } @@ -666,8 +642,6 @@ static struct i2c_driver wm8523_i2c_driver = { }, .probe = wm8523_i2c_probe, .remove = __devexit_p(wm8523_i2c_remove), - .suspend = wm8523_i2c_suspend, - .resume = wm8523_i2c_resume, .id_table = wm8523_i2c_id, }; #endif diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 6bded8c..d077df6 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -315,7 +315,6 @@ static int wm8580_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -407,8 +406,8 @@ static int pll_factors(struct _pll_div *pll_div, unsigned int target, return 0; } -static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { int offset; struct snd_soc_codec *codec = codec_dai->codec; @@ -800,17 +799,9 @@ static int wm8580_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm8580_snd_controls, ARRAY_SIZE(wm8580_snd_controls)); wm8580_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } @@ -961,21 +952,6 @@ static int wm8580_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8580_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8580_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8580_i2c_suspend NULL -#define wm8580_i2c_resume NULL -#endif - static const struct i2c_device_id wm8580_i2c_id[] = { { "wm8580", 0 }, { } @@ -989,8 +965,6 @@ static struct i2c_driver wm8580_i2c_driver = { }, .probe = wm8580_i2c_probe, .remove = wm8580_i2c_remove, - .suspend = wm8580_i2c_suspend, - .resume = wm8580_i2c_resume, .id_table = wm8580_i2c_id, }; #endif diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c new file mode 100644 index 0000000..24a3560 --- /dev/null +++ b/sound/soc/codecs/wm8711.c @@ -0,0 +1,633 @@ +/* + * wm8711.c -- WM8711 ALSA SoC Audio driver + * + * Copyright 2006 Wolfson Microelectronics + * + * Author: Mike Arthur <linux@wolfsonmicro.com> + * + * Based on 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. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/initval.h> + +#include "wm8711.h" + +static struct snd_soc_codec *wm8711_codec; + +/* codec private data */ +struct wm8711_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8711_CACHEREGNUM]; + unsigned int sysclk; +}; + +/* + * wm8711 register cache + * We can't read the WM8711 register space when we are + * using 2 wire for device control, so we cache them instead. + * There is no point in caching the reset register + */ +static const u16 wm8711_reg[WM8711_CACHEREGNUM] = { + 0x0079, 0x0079, 0x000a, 0x0008, + 0x009f, 0x000a, 0x0000, 0x0000 +}; + +#define wm8711_reset(c) snd_soc_write(c, WM8711_RESET, 0) + +static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); + +static const struct snd_kcontrol_new wm8711_snd_controls[] = { + +SOC_DOUBLE_R_TLV("Master Playback Volume", WM8711_LOUT1V, WM8711_ROUT1V, + 0, 127, 0, out_tlv), +SOC_DOUBLE_R("Master Playback ZC Switch", WM8711_LOUT1V, WM8711_ROUT1V, + 7, 1, 0), + +}; + +/* Output Mixer */ +static const struct snd_kcontrol_new wm8711_output_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8711_APANA, 3, 1, 0), +SOC_DAPM_SINGLE("HiFi Playback Switch", WM8711_APANA, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8711_dapm_widgets[] = { +SND_SOC_DAPM_MIXER("Output Mixer", WM8711_PWR, 4, 1, + &wm8711_output_mixer_controls[0], + ARRAY_SIZE(wm8711_output_mixer_controls)), +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8711_PWR, 3, 1), +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("LHPOUT"), +SND_SOC_DAPM_OUTPUT("ROUT"), +SND_SOC_DAPM_OUTPUT("RHPOUT"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* output mixer */ + {"Output Mixer", "Line Bypass Switch", "Line Input"}, + {"Output Mixer", "HiFi Playback Switch", "DAC"}, + + /* outputs */ + {"RHPOUT", NULL, "Output Mixer"}, + {"ROUT", NULL, "Output Mixer"}, + {"LHPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, +}; + +static int wm8711_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8711_dapm_widgets, + ARRAY_SIZE(wm8711_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + return 0; +} + +struct _coeff_div { + u32 mclk; + u32 rate; + u16 fs; + u8 sr:4; + u8 bosr:1; + u8 usb:1; +}; + +/* codec mclk clock divider coefficients */ +static const struct _coeff_div coeff_div[] = { + /* 48k */ + {12288000, 48000, 256, 0x0, 0x0, 0x0}, + {18432000, 48000, 384, 0x0, 0x1, 0x0}, + {12000000, 48000, 250, 0x0, 0x0, 0x1}, + + /* 32k */ + {12288000, 32000, 384, 0x6, 0x0, 0x0}, + {18432000, 32000, 576, 0x6, 0x1, 0x0}, + {12000000, 32000, 375, 0x6, 0x0, 0x1}, + + /* 8k */ + {12288000, 8000, 1536, 0x3, 0x0, 0x0}, + {18432000, 8000, 2304, 0x3, 0x1, 0x0}, + {11289600, 8000, 1408, 0xb, 0x0, 0x0}, + {16934400, 8000, 2112, 0xb, 0x1, 0x0}, + {12000000, 8000, 1500, 0x3, 0x0, 0x1}, + + /* 96k */ + {12288000, 96000, 128, 0x7, 0x0, 0x0}, + {18432000, 96000, 192, 0x7, 0x1, 0x0}, + {12000000, 96000, 125, 0x7, 0x0, 0x1}, + + /* 44.1k */ + {11289600, 44100, 256, 0x8, 0x0, 0x0}, + {16934400, 44100, 384, 0x8, 0x1, 0x0}, + {12000000, 44100, 272, 0x8, 0x1, 0x1}, + + /* 88.2k */ + {11289600, 88200, 128, 0xf, 0x0, 0x0}, + {16934400, 88200, 192, 0xf, 0x1, 0x0}, + {12000000, 88200, 136, 0xf, 0x1, 0x1}, +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return 0; +} + +static int wm8711_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8711_priv *wm8711 = codec->private_data; + u16 iface = snd_soc_read(codec, WM8711_IFACE) & 0xfffc; + int i = get_coeff(wm8711->sysclk, params_rate(params)); + u16 srate = (coeff_div[i].sr << 2) | + (coeff_div[i].bosr << 1) | coeff_div[i].usb; + + snd_soc_write(codec, WM8711_SRATE, srate); + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0008; + break; + } + + snd_soc_write(codec, WM8711_IFACE, iface); + return 0; +} + +static int wm8711_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + /* set active */ + snd_soc_write(codec, WM8711_ACTIVE, 0x0001); + + return 0; +} + +static void wm8711_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + /* deactivate */ + if (!codec->active) { + udelay(50); + snd_soc_write(codec, WM8711_ACTIVE, 0x0); + } +} + +static int wm8711_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = snd_soc_read(codec, WM8711_APDIGI) & 0xfff7; + + if (mute) + snd_soc_write(codec, WM8711_APDIGI, mute_reg | 0x8); + else + snd_soc_write(codec, WM8711_APDIGI, mute_reg); + + return 0; +} + +static int wm8711_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; + struct wm8711_priv *wm8711 = codec->private_data; + + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 16934400: + case 18432000: + wm8711->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int wm8711_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface |= 0x0040; + 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 |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0013; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0090; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0080; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0010; + break; + default: + return -EINVAL; + } + + /* set iface */ + snd_soc_write(codec, WM8711_IFACE, iface); + return 0; +} + + +static int wm8711_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg = snd_soc_read(codec, WM8711_PWR) & 0xff7f; + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_write(codec, WM8711_PWR, reg); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, WM8711_PWR, reg | 0x0040); + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, WM8711_ACTIVE, 0x0); + snd_soc_write(codec, WM8711_PWR, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8711_RATES SNDRV_PCM_RATE_8000_96000 + +#define WM8711_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops wm8711_ops = { + .prepare = wm8711_pcm_prepare, + .hw_params = wm8711_hw_params, + .shutdown = wm8711_shutdown, + .digital_mute = wm8711_mute, + .set_sysclk = wm8711_set_dai_sysclk, + .set_fmt = wm8711_set_dai_fmt, +}; + +struct snd_soc_dai wm8711_dai = { + .name = "WM8711", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8711_RATES, + .formats = WM8711_FORMATS, + }, + .ops = &wm8711_ops, +}; +EXPORT_SYMBOL_GPL(wm8711_dai); + +static int wm8711_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_write(codec, WM8711_ACTIVE, 0x0); + wm8711_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8711_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8711_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8711_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +static int wm8711_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8711_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8711_codec; + codec = wm8711_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8711_snd_controls, + ARRAY_SIZE(wm8711_snd_controls)); + wm8711_add_widgets(codec); + + return ret; + +pcm_err: + return ret; +} + +/* power down chip */ +static int wm8711_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8711 = { + .probe = wm8711_probe, + .remove = wm8711_remove, + .suspend = wm8711_suspend, + .resume = wm8711_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8711); + +static int wm8711_register(struct wm8711_priv *wm8711, + enum snd_soc_control_type control) +{ + int ret; + struct snd_soc_codec *codec = &wm8711->codec; + u16 reg; + + if (wm8711_codec) { + dev_err(codec->dev, "Another WM8711 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8711; + codec->name = "WM8711"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8711_set_bias_level; + codec->dai = &wm8711_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8711_CACHEREGNUM; + codec->reg_cache = &wm8711->reg_cache; + + memcpy(codec->reg_cache, wm8711_reg, sizeof(wm8711_reg)); + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + ret = wm8711_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err; + } + + wm8711_dai.dev = codec->dev; + + wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Latch the update bits */ + reg = snd_soc_read(codec, WM8711_LOUT1V); + snd_soc_write(codec, WM8711_LOUT1V, reg | 0x0100); + reg = snd_soc_read(codec, WM8711_ROUT1V); + snd_soc_write(codec, WM8711_ROUT1V, reg | 0x0100); + + wm8711_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&wm8711_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(wm8711); + return ret; +} + +static void wm8711_unregister(struct wm8711_priv *wm8711) +{ + wm8711_set_bias_level(&wm8711->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8711_dai); + snd_soc_unregister_codec(&wm8711->codec); + kfree(wm8711); + wm8711_codec = NULL; +} + +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8711_spi_probe(struct spi_device *spi) +{ + struct snd_soc_codec *codec; + struct wm8711_priv *wm8711; + + wm8711 = kzalloc(sizeof(struct wm8711_priv), GFP_KERNEL); + if (wm8711 == NULL) + return -ENOMEM; + + codec = &wm8711->codec; + codec->control_data = spi; + codec->dev = &spi->dev; + + dev_set_drvdata(&spi->dev, wm8711); + + return wm8711_register(wm8711, SND_SOC_SPI); +} + +static int __devexit wm8711_spi_remove(struct spi_device *spi) +{ + struct wm8711_priv *wm8711 = dev_get_drvdata(&spi->dev); + + wm8711_unregister(wm8711); + + return 0; +} + +static struct spi_driver wm8711_spi_driver = { + .driver = { + .name = "wm8711", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8711_spi_probe, + .remove = __devexit_p(wm8711_spi_remove), +}; +#endif /* CONFIG_SPI_MASTER */ + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static __devinit int wm8711_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8711_priv *wm8711; + struct snd_soc_codec *codec; + + wm8711 = kzalloc(sizeof(struct wm8711_priv), GFP_KERNEL); + if (wm8711 == NULL) + return -ENOMEM; + + codec = &wm8711->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8711); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8711_register(wm8711, SND_SOC_I2C); +} + +static __devexit int wm8711_i2c_remove(struct i2c_client *client) +{ + struct wm8711_priv *wm8711 = i2c_get_clientdata(client); + wm8711_unregister(wm8711); + return 0; +} + +static const struct i2c_device_id wm8711_i2c_id[] = { + { "wm8711", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8711_i2c_id); + +static struct i2c_driver wm8711_i2c_driver = { + .driver = { + .name = "WM8711 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = wm8711_i2c_probe, + .remove = __devexit_p(wm8711_i2c_remove), + .id_table = wm8711_i2c_id, +}; +#endif + +static int __init wm8711_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8711_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8711 I2C driver: %d\n", + ret); + } +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&wm8711_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8711 SPI driver: %d\n", + ret); + } +#endif + return 0; +} +module_init(wm8711_modinit); + +static void __exit wm8711_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8711_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8711_spi_driver); +#endif +} +module_exit(wm8711_exit); + +MODULE_DESCRIPTION("ASoC WM8711 driver"); +MODULE_AUTHOR("Mike Arthur"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8711.h b/sound/soc/codecs/wm8711.h new file mode 100644 index 0000000..381e84a --- /dev/null +++ b/sound/soc/codecs/wm8711.h @@ -0,0 +1,42 @@ +/* + * wm8711.h -- WM8711 Soc Audio driver + * + * Copyright 2006 Wolfson Microelectronics + * + * Author: Mike Arthur <linux@wolfsonmicro.com> + * + * Based on wm8731.h + * + * 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 _WM8711_H +#define _WM8711_H + +/* WM8711 register space */ + +#define WM8711_LOUT1V 0x02 +#define WM8711_ROUT1V 0x03 +#define WM8711_APANA 0x04 +#define WM8711_APDIGI 0x05 +#define WM8711_PWR 0x06 +#define WM8711_IFACE 0x07 +#define WM8711_SRATE 0x08 +#define WM8711_ACTIVE 0x09 +#define WM8711_RESET 0x0f + +#define WM8711_CACHEREGNUM 8 + +#define WM8711_SYSCLK 0 +#define WM8711_DAI 0 + +struct wm8711_setup_data { + unsigned short i2c_address; +}; + +extern struct snd_soc_dai wm8711_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8711; + +#endif diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c new file mode 100644 index 0000000..d8ffbd6 --- /dev/null +++ b/sound/soc/codecs/wm8727.c @@ -0,0 +1,135 @@ +/* + * wm8727.c + * + * Created on: 15-Oct-2009 + * Author: neil.jones@imgtec.com + * + * Copyright (C) 2009 Imagination Technologies Ltd. + * + * 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 <linux/init.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "wm8727.h" +/* + * Note this is a simple chip with no configuration interface, sample rate is + * determined automatically by examining the Master clock and Bit clock ratios + */ +#define WM8727_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_192000) + + +struct snd_soc_dai wm8727_dai = { + .name = "WM8727", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8727_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + }, +}; +EXPORT_SYMBOL_GPL(wm8727_dai); + +static int wm8727_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 = "WM8727"; + codec->owner = THIS_MODULE; + codec->dai = &wm8727_dai; + codec->num_dai = 1; + socdev->card->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 "wm8727: failed to create pcms\n"); + goto pcm_err; + } + + return ret; + +pcm_err: + kfree(socdev->card->codec); + socdev->card->codec = NULL; + return ret; +} + +static int wm8727_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec == NULL) + return 0; + snd_soc_free_pcms(socdev); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8727 = { + .probe = wm8727_soc_probe, + .remove = wm8727_soc_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8727); + + +static __devinit int wm8727_platform_probe(struct platform_device *pdev) +{ + wm8727_dai.dev = &pdev->dev; + return snd_soc_register_dai(&wm8727_dai); +} + +static int __devexit wm8727_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&wm8727_dai); + return 0; +} + +static struct platform_driver wm8727_codec_driver = { + .driver = { + .name = "wm8727-codec", + .owner = THIS_MODULE, + }, + + .probe = wm8727_platform_probe, + .remove = __devexit_p(wm8727_platform_remove), +}; + +static int __init wm8727_init(void) +{ + return platform_driver_register(&wm8727_codec_driver); +} +module_init(wm8727_init); + +static void __exit wm8727_exit(void) +{ + platform_driver_unregister(&wm8727_codec_driver); +} +module_exit(wm8727_exit); + +MODULE_DESCRIPTION("ASoC wm8727 driver"); +MODULE_AUTHOR("Neil Jones"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8727.h b/sound/soc/codecs/wm8727.h new file mode 100644 index 0000000..ee19aa7 --- /dev/null +++ b/sound/soc/codecs/wm8727.h @@ -0,0 +1,21 @@ +/* + * wm8727.h + * + * Created on: 15-Oct-2009 + * Author: neil.jones@imgtec.com + * + * Copyright (C) 2009 Imagination Technologies Ltd. + * + * 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. + */ + +#ifndef WM8727_H_ +#define WM8727_H_ + +extern struct snd_soc_dai wm8727_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8727; + +#endif /* WM8727_H_ */ diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 16e969a..3fb653b 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -74,8 +74,6 @@ static int wm8728_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); - return 0; } @@ -287,17 +285,9 @@ static int wm8728_init(struct snd_soc_device *socdev, snd_soc_add_controls(codec, wm8728_snd_controls, ARRAY_SIZE(wm8728_snd_controls)); wm8728_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8728: failed to register card\n"); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index d3fd4f2..3a49781 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -19,6 +19,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/regulator/consumer.h> #include <linux/spi/spi.h> #include <sound/core.h> #include <sound/pcm.h> @@ -33,9 +34,18 @@ static struct snd_soc_codec *wm8731_codec; struct snd_soc_codec_device soc_codec_dev_wm8731; +#define WM8731_NUM_SUPPLIES 4 +static const char *wm8731_supply_names[WM8731_NUM_SUPPLIES] = { + "AVDD", + "HPVDD", + "DCVDD", + "DBVDD", +}; + /* codec private data */ struct wm8731_priv { struct snd_soc_codec codec; + struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES]; u16 reg_cache[WM8731_CACHEREGNUM]; unsigned int sysclk; }; @@ -149,7 +159,6 @@ static int wm8731_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -422,9 +431,12 @@ static int wm8731_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; + struct wm8731_priv *wm8731 = codec->private_data; snd_soc_write(codec, WM8731_ACTIVE, 0x0); wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); + regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), + wm8731->supplies); return 0; } @@ -432,10 +444,16 @@ static int wm8731_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - int i; + struct wm8731_priv *wm8731 = codec->private_data; + int i, ret; u8 data[2]; u16 *cache = codec->reg_cache; + ret = regulator_bulk_enable(ARRAY_SIZE(wm8731->supplies), + wm8731->supplies); + if (ret != 0) + return ret; + /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8731_reg); i++) { data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); @@ -444,6 +462,7 @@ static int wm8731_resume(struct platform_device *pdev) } wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); wm8731_set_bias_level(codec, codec->suspend_bias_level); + return 0; } #else @@ -475,17 +494,9 @@ static int wm8731_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm8731_snd_controls, ARRAY_SIZE(wm8731_snd_controls)); wm8731_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } @@ -512,7 +523,7 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731); static int wm8731_register(struct wm8731_priv *wm8731, enum snd_soc_control_type control) { - int ret; + int ret, i; struct snd_soc_codec *codec = &wm8731->codec; if (wm8731_codec) { @@ -543,10 +554,27 @@ static int wm8731_register(struct wm8731_priv *wm8731, goto err; } + for (i = 0; i < ARRAY_SIZE(wm8731->supplies); i++) + wm8731->supplies[i].supply = wm8731_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8731->supplies), + wm8731->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8731->supplies), + wm8731->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_regulator_get; + } + ret = wm8731_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); - goto err; + goto err_regulator_enable; } wm8731_dai.dev = codec->dev; @@ -567,7 +595,7 @@ static int wm8731_register(struct wm8731_priv *wm8731, ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - goto err; + goto err_regulator_enable; } ret = snd_soc_register_dai(&wm8731_dai); @@ -581,6 +609,10 @@ static int wm8731_register(struct wm8731_priv *wm8731, err_codec: snd_soc_unregister_codec(codec); +err_regulator_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); +err_regulator_get: + regulator_bulk_free(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); err: kfree(wm8731); return ret; @@ -591,6 +623,8 @@ static void wm8731_unregister(struct wm8731_priv *wm8731) wm8731_set_bias_level(&wm8731->codec, SND_SOC_BIAS_OFF); snd_soc_unregister_dai(&wm8731_dai); snd_soc_unregister_codec(&wm8731->codec); + regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); + regulator_bulk_free(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); kfree(wm8731); wm8731_codec = NULL; } @@ -623,21 +657,6 @@ static int __devexit wm8731_spi_remove(struct spi_device *spi) return 0; } -#ifdef CONFIG_PM -static int wm8731_spi_suspend(struct spi_device *spi, pm_message_t msg) -{ - return snd_soc_suspend_device(&spi->dev); -} - -static int wm8731_spi_resume(struct spi_device *spi) -{ - return snd_soc_resume_device(&spi->dev); -} -#else -#define wm8731_spi_suspend NULL -#define wm8731_spi_resume NULL -#endif - static struct spi_driver wm8731_spi_driver = { .driver = { .name = "wm8731", @@ -645,8 +664,6 @@ static struct spi_driver wm8731_spi_driver = { .owner = THIS_MODULE, }, .probe = wm8731_spi_probe, - .suspend = wm8731_spi_suspend, - .resume = wm8731_spi_resume, .remove = __devexit_p(wm8731_spi_remove), }; #endif /* CONFIG_SPI_MASTER */ @@ -679,21 +696,6 @@ static __devexit int wm8731_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8731_i2c_suspend(struct i2c_client *i2c, pm_message_t msg) -{ - return snd_soc_suspend_device(&i2c->dev); -} - -static int wm8731_i2c_resume(struct i2c_client *i2c) -{ - return snd_soc_resume_device(&i2c->dev); -} -#else -#define wm8731_i2c_suspend NULL -#define wm8731_i2c_resume NULL -#endif - static const struct i2c_device_id wm8731_i2c_id[] = { { "wm8731", 0 }, { } @@ -707,8 +709,6 @@ static struct i2c_driver wm8731_i2c_driver = { }, .probe = wm8731_i2c_probe, .remove = __devexit_p(wm8731_i2c_remove), - .suspend = wm8731_i2c_suspend, - .resume = wm8731_i2c_resume, .id_table = wm8731_i2c_id, }; #endif diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 4ba1e7e..475c67a 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -403,7 +403,6 @@ static int wm8750_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -772,16 +771,8 @@ static int wm8750_init(struct snd_soc_device *socdev, snd_soc_add_controls(codec, wm8750_snd_controls, ARRAY_SIZE(wm8750_snd_controls)); wm8750_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8750: failed to register card\n"); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 5ad677c..d6850dacd 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -673,7 +673,6 @@ static int wm8753_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -724,8 +723,8 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, pll_div->k = K; } -static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { u16 reg, enable; int offset; @@ -1583,18 +1582,9 @@ static int wm8753_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm8753_snd_controls, ARRAY_SIZE(wm8753_snd_controls)); wm8753_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8753: failed to register card\n"); - goto card_err; - } return 0; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); - pcm_err: return ret; } @@ -1767,21 +1757,6 @@ static int wm8753_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8753_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8753_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8753_i2c_suspend NULL -#define wm8753_i2c_resume NULL -#endif - static const struct i2c_device_id wm8753_i2c_id[] = { { "wm8753", 0 }, { } @@ -1795,8 +1770,6 @@ static struct i2c_driver wm8753_i2c_driver = { }, .probe = wm8753_i2c_probe, .remove = wm8753_i2c_remove, - .suspend = wm8753_i2c_suspend, - .resume = wm8753_i2c_resume, .id_table = wm8753_i2c_id, }; #endif @@ -1852,22 +1825,6 @@ static int __devexit wm8753_spi_remove(struct spi_device *spi) return 0; } -#ifdef CONFIG_PM -static int wm8753_spi_suspend(struct spi_device *spi, pm_message_t msg) -{ - return snd_soc_suspend_device(&spi->dev); -} - -static int wm8753_spi_resume(struct spi_device *spi) -{ - return snd_soc_resume_device(&spi->dev); -} - -#else -#define wm8753_spi_suspend NULL -#define wm8753_spi_resume NULL -#endif - static struct spi_driver wm8753_spi_driver = { .driver = { .name = "wm8753", @@ -1876,8 +1833,6 @@ static struct spi_driver wm8753_spi_driver = { }, .probe = wm8753_spi_probe, .remove = __devexit_p(wm8753_spi_remove), - .suspend = wm8753_spi_suspend, - .resume = wm8753_spi_resume, }; #endif diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index a9829aa..ab2c0da 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -447,17 +447,8 @@ static int wm8776_probe(struct platform_device *pdev) ARRAY_SIZE(wm8776_dapm_widgets)); snd_soc_dapm_add_routes(codec, routes, ARRAY_SIZE(routes)); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } - return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } @@ -616,21 +607,6 @@ static int __devexit wm8776_spi_remove(struct spi_device *spi) return 0; } -#ifdef CONFIG_PM -static int wm8776_spi_suspend(struct spi_device *spi, pm_message_t msg) -{ - return snd_soc_suspend_device(&spi->dev); -} - -static int wm8776_spi_resume(struct spi_device *spi) -{ - return snd_soc_resume_device(&spi->dev); -} -#else -#define wm8776_spi_suspend NULL -#define wm8776_spi_resume NULL -#endif - static struct spi_driver wm8776_spi_driver = { .driver = { .name = "wm8776", @@ -638,8 +614,6 @@ static struct spi_driver wm8776_spi_driver = { .owner = THIS_MODULE, }, .probe = wm8776_spi_probe, - .suspend = wm8776_spi_suspend, - .resume = wm8776_spi_resume, .remove = __devexit_p(wm8776_spi_remove), }; #endif /* CONFIG_SPI_MASTER */ @@ -673,21 +647,6 @@ static __devexit int wm8776_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8776_i2c_suspend(struct i2c_client *i2c, pm_message_t msg) -{ - return snd_soc_suspend_device(&i2c->dev); -} - -static int wm8776_i2c_resume(struct i2c_client *i2c) -{ - return snd_soc_resume_device(&i2c->dev); -} -#else -#define wm8776_i2c_suspend NULL -#define wm8776_i2c_resume NULL -#endif - static const struct i2c_device_id wm8776_i2c_id[] = { { "wm8776", 0 }, { } @@ -701,8 +660,6 @@ static struct i2c_driver wm8776_i2c_driver = { }, .probe = wm8776_i2c_probe, .remove = __devexit_p(wm8776_i2c_remove), - .suspend = wm8776_i2c_suspend, - .resume = wm8776_i2c_resume, .id_table = wm8776_i2c_id, }; #endif diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 5e9c855..c9438dd 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -618,8 +618,6 @@ static int wm8900_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); - return 0; } @@ -814,8 +812,8 @@ reenable: return 0; } -static int wm8900_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8900_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { return wm8900_set_fll(codec_dai->codec, pll_id, freq_in, freq_out); } @@ -1312,21 +1310,6 @@ static __devexit int wm8900_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8900_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8900_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8900_i2c_suspend NULL -#define wm8900_i2c_resume NULL -#endif - static const struct i2c_device_id wm8900_i2c_id[] = { { "wm8900", 0 }, { } @@ -1340,8 +1323,6 @@ static struct i2c_driver wm8900_i2c_driver = { }, .probe = wm8900_i2c_probe, .remove = __devexit_p(wm8900_i2c_remove), - .suspend = wm8900_i2c_suspend, - .resume = wm8900_i2c_resume, .id_table = wm8900_i2c_id, }; @@ -1370,17 +1351,6 @@ static int wm8900_probe(struct platform_device *pdev) ARRAY_SIZE(wm8900_snd_controls)); wm8900_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&pdev->dev, "Failed to register card\n"); - goto card_err; - } - - return ret; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index fe1307b..ce5515e 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -607,7 +607,7 @@ SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1, SOC_SINGLE("DRC Switch", WM8903_DRC_0, 15, 1, 0), SOC_ENUM("DRC Compressor Slope R0", drc_slope_r0), SOC_ENUM("DRC Compressor Slope R1", drc_slope_r1), -SOC_SINGLE_TLV("DRC Compressor Threashold Volume", WM8903_DRC_3, 5, 124, 1, +SOC_SINGLE_TLV("DRC Compressor Threshold Volume", WM8903_DRC_3, 5, 124, 1, drc_tlv_thresh), SOC_SINGLE_TLV("DRC Volume", WM8903_DRC_3, 0, 30, 1, drc_tlv_amp), SOC_SINGLE_TLV("DRC Minimum Gain Volume", WM8903_DRC_1, 2, 3, 1, drc_tlv_min), @@ -617,11 +617,11 @@ SOC_ENUM("DRC Decay Rate", drc_decay), SOC_ENUM("DRC FF Delay", drc_ff_delay), SOC_SINGLE("DRC Anticlip Switch", WM8903_DRC_0, 1, 1, 0), SOC_SINGLE("DRC QR Switch", WM8903_DRC_0, 2, 1, 0), -SOC_SINGLE_TLV("DRC QR Threashold Volume", WM8903_DRC_0, 6, 3, 0, drc_tlv_max), +SOC_SINGLE_TLV("DRC QR Threshold Volume", WM8903_DRC_0, 6, 3, 0, drc_tlv_max), SOC_ENUM("DRC QR Decay Rate", drc_qr_decay), SOC_SINGLE("DRC Smoothing Switch", WM8903_DRC_0, 3, 1, 0), SOC_SINGLE("DRC Smoothing Hysteresis Switch", WM8903_DRC_0, 0, 1, 0), -SOC_ENUM("DRC Smoothing Threashold", drc_smoothing), +SOC_ENUM("DRC Smoothing Threshold", drc_smoothing), SOC_SINGLE_TLV("DRC Startup Volume", WM8903_DRC_0, 6, 18, 0, drc_tlv_startup), SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8903_ADC_DIGITAL_VOLUME_LEFT, @@ -919,8 +919,6 @@ static int wm8903_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); - return 0; } @@ -1655,21 +1653,6 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8903_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8903_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8903_i2c_suspend NULL -#define wm8903_i2c_resume NULL -#endif - /* i2c codec control layer */ static const struct i2c_device_id wm8903_i2c_id[] = { { "wm8903", 0 }, @@ -1684,8 +1667,6 @@ static struct i2c_driver wm8903_i2c_driver = { }, .probe = wm8903_i2c_probe, .remove = __devexit_p(wm8903_i2c_remove), - .suspend = wm8903_i2c_suspend, - .resume = wm8903_i2c_resume, .id_table = wm8903_i2c_id, }; @@ -1712,17 +1693,8 @@ static int wm8903_probe(struct platform_device *pdev) ARRAY_SIZE(wm8903_snd_controls)); wm8903_add_widgets(socdev->card->codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&pdev->dev, "wm8903: failed to register card\n"); - goto card_err; - } - return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); err: return ret; } diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 1ef2454..3d850b9 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -298,7 +298,6 @@ static int wm8940_add_widgets(struct snd_soc_codec *codec) ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); if (ret) goto error_ret; - ret = snd_soc_dapm_new_widgets(codec); error_ret: return ret; @@ -536,8 +535,8 @@ static void pll_factors(unsigned int target, unsigned int source) } /* Untested at the moment */ -static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; u16 reg; @@ -731,12 +730,6 @@ static int wm8940_probe(struct platform_device *pdev) if (ret) goto error_free_pcms; - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto error_free_pcms; - } - return ret; error_free_pcms: @@ -877,21 +870,6 @@ static int __devexit wm8940_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8940_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8940_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8940_i2c_suspend NULL -#define wm8940_i2c_resume NULL -#endif - static const struct i2c_device_id wm8940_i2c_id[] = { { "wm8940", 0 }, { } @@ -905,8 +883,6 @@ static struct i2c_driver wm8940_i2c_driver = { }, .probe = wm8940_i2c_probe, .remove = __devexit_p(wm8940_i2c_remove), - .suspend = wm8940_i2c_suspend, - .resume = wm8940_i2c_resume, .id_table = wm8940_i2c_id, }; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index f59703b..d07bcc1 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -307,7 +307,6 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -540,8 +539,8 @@ static int pll_factors(unsigned int source, unsigned int target, return 0; } -static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; u16 reg; @@ -713,17 +712,9 @@ static int wm8960_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm8960_snd_controls, ARRAY_SIZE(wm8960_snd_controls)); wm8960_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } @@ -883,21 +874,6 @@ static __devexit int wm8960_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8960_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8960_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8960_i2c_suspend NULL -#define wm8960_i2c_resume NULL -#endif - static const struct i2c_device_id wm8960_i2c_id[] = { { "wm8960", 0 }, { } @@ -911,8 +887,6 @@ static struct i2c_driver wm8960_i2c_driver = { }, .probe = wm8960_i2c_probe, .remove = __devexit_p(wm8960_i2c_remove), - .suspend = wm8960_i2c_suspend, - .resume = wm8960_i2c_resume, .id_table = wm8960_i2c_id, }; diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 5030320..a8007d5 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -986,19 +986,9 @@ static int wm8961_probe(struct platform_device *pdev) snd_soc_dapm_new_controls(codec, wm8961_dapm_widgets, ARRAY_SIZE(wm8961_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); - snd_soc_dapm_new_widgets(codec); - - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } @@ -1206,21 +1196,6 @@ static __devexit int wm8961_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8961_i2c_suspend(struct i2c_client *client, pm_message_t state) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8961_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8961_i2c_suspend NULL -#define wm8961_i2c_resume NULL -#endif - static const struct i2c_device_id wm8961_i2c_id[] = { { "wm8961", 0 }, { } @@ -1234,8 +1209,6 @@ static struct i2c_driver wm8961_i2c_driver = { }, .probe = wm8961_i2c_probe, .remove = __devexit_p(wm8961_i2c_remove), - .suspend = wm8961_i2c_suspend, - .resume = wm8961_i2c_resume, .id_table = wm8961_i2c_id, }; diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index d66efb0..d9540d5 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -338,8 +338,6 @@ static int wm8971_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); - return 0; } @@ -703,16 +701,9 @@ static int wm8971_init(struct snd_soc_device *socdev, snd_soc_add_controls(codec, wm8971_snd_controls, ARRAY_SIZE(wm8971_snd_controls)); wm8971_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8971: failed to register card\n"); - goto card_err; - } + return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 98d663a..81c57b5 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -276,41 +276,42 @@ static int wm8974_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } struct pll_ { - unsigned int pre_div:4; /* prescale - 1 */ + unsigned int pre_div:1; unsigned int n:4; unsigned int k; }; -static struct pll_ pll_div; - /* The size in bits of the pll divide multiplied by 10 * to allow rounding later */ #define FIXED_PLL_SIZE ((1 << 24) * 10) -static void pll_factors(unsigned int target, unsigned int source) +static void pll_factors(struct pll_ *pll_div, + unsigned int target, unsigned int source) { unsigned long long Kpart; unsigned int K, Ndiv, Nmod; + /* There is a fixed divide by 4 in the output path */ + target *= 4; + Ndiv = target / source; if (Ndiv < 6) { - source >>= 1; - pll_div.pre_div = 1; + source /= 2; + pll_div->pre_div = 1; Ndiv = target / source; } else - pll_div.pre_div = 0; + pll_div->pre_div = 0; if ((Ndiv < 6) || (Ndiv > 12)) printk(KERN_WARNING "WM8974 N value %u outwith recommended range!\n", Ndiv); - pll_div.n = Ndiv; + pll_div->n = Ndiv; Nmod = target % source; Kpart = FIXED_PLL_SIZE * (long long)Nmod; @@ -325,13 +326,14 @@ static void pll_factors(unsigned int target, unsigned int source) /* Move down to proper range now rounding is done */ K /= 10; - pll_div.k = K; + pll_div->k = K; } -static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; + struct pll_ pll_div; u16 reg; if (freq_in == 0 || freq_out == 0) { @@ -345,7 +347,7 @@ static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai, return 0; } - pll_factors(freq_out*4, freq_in); + pll_factors(&pll_div, freq_out, freq_in); snd_soc_write(codec, WM8974_PLLN, (pll_div.pre_div << 4) | pll_div.n); snd_soc_write(codec, WM8974_PLLK1, pll_div.k >> 18); @@ -638,17 +640,9 @@ static int wm8974_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm8974_snd_controls, ARRAY_SIZE(wm8974_snd_controls)); wm8974_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 3f530f8..2862e4d 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -790,19 +790,9 @@ static int wm8988_probe(struct platform_device *pdev) snd_soc_dapm_new_controls(codec, wm8988_dapm_widgets, ARRAY_SIZE(wm8988_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); - - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } @@ -944,21 +934,6 @@ static int wm8988_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8988_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8988_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8988_i2c_suspend NULL -#define wm8988_i2c_resume NULL -#endif - static const struct i2c_device_id wm8988_i2c_id[] = { { "wm8988", 0 }, { } @@ -972,8 +947,6 @@ static struct i2c_driver wm8988_i2c_driver = { }, .probe = wm8988_i2c_probe, .remove = wm8988_i2c_remove, - .suspend = wm8988_i2c_suspend, - .resume = wm8988_i2c_resume, .id_table = wm8988_i2c_id, }; #endif @@ -1006,21 +979,6 @@ static int __devexit wm8988_spi_remove(struct spi_device *spi) return 0; } -#ifdef CONFIG_PM -static int wm8988_spi_suspend(struct spi_device *spi, pm_message_t msg) -{ - return snd_soc_suspend_device(&spi->dev); -} - -static int wm8988_spi_resume(struct spi_device *spi) -{ - return snd_soc_resume_device(&spi->dev); -} -#else -#define wm8988_spi_suspend NULL -#define wm8988_spi_resume NULL -#endif - static struct spi_driver wm8988_spi_driver = { .driver = { .name = "wm8988", @@ -1029,8 +987,6 @@ static struct spi_driver wm8988_spi_driver = { }, .probe = wm8988_spi_probe, .remove = __devexit_p(wm8988_spi_remove), - .suspend = wm8988_spi_suspend, - .resume = wm8988_spi_resume, }; #endif diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 2d702db..341481e 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -920,7 +920,6 @@ static int wm8990_add_widgets(struct snd_soc_codec *codec) /* set up the WM8990 audio map */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -972,8 +971,8 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, pll_div->k = K; } -static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { u16 reg; struct snd_soc_codec *codec = codec_dai->codec; @@ -1409,16 +1408,9 @@ static int wm8990_init(struct snd_soc_device *socdev) snd_soc_add_controls(codec, wm8990_snd_controls, ARRAY_SIZE(wm8990_snd_controls)); wm8990_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8990: 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; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index d998799..2981afa 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -422,7 +422,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, return 0; } -static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, +static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, int source, unsigned int Fref, unsigned int Fout) { struct snd_soc_codec *codec = dai->codec; @@ -689,7 +689,7 @@ SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8993_DIGITAL_SIDE_TONE, SOC_SINGLE("DRC Switch", WM8993_DRC_CONTROL_1, 15, 1, 0), SOC_ENUM("DRC Path", drc_path), -SOC_SINGLE_TLV("DRC Compressor Threashold Volume", WM8993_DRC_CONTROL_2, +SOC_SINGLE_TLV("DRC Compressor Threshold Volume", WM8993_DRC_CONTROL_2, 2, 60, 1, drc_comp_threash), SOC_SINGLE_TLV("DRC Compressor Amplitude Volume", WM8993_DRC_CONTROL_3, 11, 30, 1, drc_comp_amp), @@ -709,7 +709,7 @@ SOC_SINGLE_TLV("DRC Quick Release Volume", WM8993_DRC_CONTROL_3, 2, 3, 0, SOC_ENUM("DRC Quick Release Rate", drc_qr_rate), SOC_SINGLE("DRC Smoothing Switch", WM8993_DRC_CONTROL_1, 11, 1, 0), SOC_SINGLE("DRC Smoothing Hysteresis Switch", WM8993_DRC_CONTROL_1, 8, 1, 0), -SOC_ENUM("DRC Smoothing Hysteresis Threashold", drc_smooth), +SOC_ENUM("DRC Smoothing Hysteresis Threshold", drc_smooth), SOC_SINGLE_TLV("DRC Startup Volume", WM8993_DRC_CONTROL_4, 8, 18, 0, drc_startup_tlv), @@ -1464,19 +1464,8 @@ static int wm8993_probe(struct platform_device *pdev) wm_hubs_add_analogue_routes(codec, wm8993->pdata.lineout1_diff, wm8993->pdata.lineout2_diff); - snd_soc_dapm_new_widgets(codec); - - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card\n"); - goto card_err; - } - return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); err: return ret; } @@ -1572,33 +1561,15 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, /* Use automatic clock configuration */ snd_soc_update_bits(codec, WM8993_CLOCKING_4, WM8993_SR_MODE, 0); - if (!wm8993->pdata.lineout1_diff) - snd_soc_update_bits(codec, WM8993_LINE_MIXER1, - WM8993_LINEOUT1_MODE, - WM8993_LINEOUT1_MODE); - if (!wm8993->pdata.lineout2_diff) - snd_soc_update_bits(codec, WM8993_LINE_MIXER2, - WM8993_LINEOUT2_MODE, - WM8993_LINEOUT2_MODE); - - if (wm8993->pdata.lineout1fb) - snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, - WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB); - - if (wm8993->pdata.lineout2fb) - snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, - WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB); - - /* Apply the microphone bias/detection configuration - the - * platform data is directly applicable to the register. */ - snd_soc_update_bits(codec, WM8993_MICBIAS, - WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK | - WM8993_MICB1_LVL | WM8993_MICB2_LVL, - wm8993->pdata.jd_scthr << WM8993_JD_SCTHR_SHIFT | - wm8993->pdata.jd_thr << WM8993_JD_THR_SHIFT | - wm8993->pdata.micbias1_lvl | - wm8993->pdata.micbias1_lvl << 1); - + wm_hubs_handle_analogue_pdata(codec, wm8993->pdata.lineout1_diff, + wm8993->pdata.lineout2_diff, + wm8993->pdata.lineout1fb, + wm8993->pdata.lineout2fb, + wm8993->pdata.jd_scthr, + wm8993->pdata.jd_thr, + wm8993->pdata.micbias1_lvl, + wm8993->pdata.micbias2_lvl); + ret = wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (ret != 0) goto err; diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 686e5aa..c468497 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1262,19 +1262,9 @@ static int wm9081_probe(struct platform_device *pdev) snd_soc_dapm_new_controls(codec, wm9081_dapm_widgets, ARRAY_SIZE(wm9081_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); - snd_soc_dapm_new_widgets(codec); - - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } @@ -1452,21 +1442,6 @@ static __devexit int wm9081_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm9081_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm9081_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm9081_i2c_suspend NULL -#define wm9081_i2c_resume NULL -#endif - static const struct i2c_device_id wm9081_i2c_id[] = { { "wm9081", 0 }, { } @@ -1480,8 +1455,6 @@ static struct i2c_driver wm9081_i2c_driver = { }, .probe = wm9081_i2c_probe, .remove = __devexit_p(wm9081_i2c_remove), - .suspend = wm9081_i2c_suspend, - .resume = wm9081_i2c_resume, .id_table = wm9081_i2c_id, }; diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index e7d2840..ec54c6d 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -205,7 +205,6 @@ static int wm9705_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_new_controls(codec, wm9705_dapm_widgets, ARRAY_SIZE(wm9705_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -403,12 +402,6 @@ static int wm9705_soc_probe(struct platform_device *pdev) ARRAY_SIZE(wm9705_snd_ac97_controls)); wm9705_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm9705: failed to register card\n"); - goto reset_err; - } - return 0; reset_err: diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 1fd4e88..0ac1215 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -436,7 +436,6 @@ static int wm9712_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -695,17 +694,11 @@ static int wm9712_soc_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm9712_snd_ac97_controls, ARRAY_SIZE(wm9712_snd_ac97_controls)); wm9712_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm9712: failed to register card\n"); - goto reset_err; - } return 0; reset_err: snd_soc_free_pcms(socdev); - pcm_err: snd_soc_free_ac97_codec(codec); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 60e360b..c58aab3 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -625,7 +625,6 @@ static int wm9713_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -800,8 +799,8 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, return 0; } -static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; return wm9713_set_pll(codec, pll_id, freq_in, freq_out); @@ -1247,14 +1246,11 @@ static int wm9713_soc_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm9713_snd_ac97_controls, ARRAY_SIZE(wm9713_snd_ac97_controls)); wm9713_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) - goto reset_err; + return 0; reset_err: snd_soc_free_pcms(socdev); - pcm_err: snd_soc_free_ac97_codec(codec); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index e542027..d73c305 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -438,11 +438,11 @@ static const struct snd_soc_dapm_widget analogue_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN1LN"), SND_SOC_DAPM_INPUT("IN1LP"), SND_SOC_DAPM_INPUT("IN2LN"), -SND_SOC_DAPM_INPUT("IN2LP/VXRN"), +SND_SOC_DAPM_INPUT("IN2LP:VXRN"), SND_SOC_DAPM_INPUT("IN1RN"), SND_SOC_DAPM_INPUT("IN1RP"), SND_SOC_DAPM_INPUT("IN2RN"), -SND_SOC_DAPM_INPUT("IN2RP/VXRP"), +SND_SOC_DAPM_INPUT("IN2RP:VXRP"), SND_SOC_DAPM_MICBIAS("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0), SND_SOC_DAPM_MICBIAS("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0), @@ -537,14 +537,14 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "IN1R PGA", "IN1RP Switch", "IN1RP" }, { "IN1R PGA", "IN1RN Switch", "IN1RN" }, - { "IN2L PGA", "IN2LP Switch", "IN2LP/VXRN" }, + { "IN2L PGA", "IN2LP Switch", "IN2LP:VXRN" }, { "IN2L PGA", "IN2LN Switch", "IN2LN" }, - { "IN2R PGA", "IN2RP Switch", "IN2RP/VXRP" }, + { "IN2R PGA", "IN2RP Switch", "IN2RP:VXRP" }, { "IN2R PGA", "IN2RN Switch", "IN2RN" }, - { "Direct Voice", NULL, "IN2LP/VXRN" }, - { "Direct Voice", NULL, "IN2RP/VXRP" }, + { "Direct Voice", NULL, "IN2LP:VXRN" }, + { "Direct Voice", NULL, "IN2RP:VXRP" }, { "MIXINL", "IN1L Switch", "IN1L PGA" }, { "MIXINL", "IN2L Switch", "IN2L PGA" }, @@ -565,7 +565,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Left Output Mixer", "Right Input Switch", "MIXINR" }, { "Left Output Mixer", "IN2RN Switch", "IN2RN" }, { "Left Output Mixer", "IN2LN Switch", "IN2LN" }, - { "Left Output Mixer", "IN2LP Switch", "IN2LP/VXRN" }, + { "Left Output Mixer", "IN2LP Switch", "IN2LP:VXRN" }, { "Left Output Mixer", "IN1L Switch", "IN1L PGA" }, { "Left Output Mixer", "IN1R Switch", "IN1R PGA" }, @@ -573,7 +573,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Right Output Mixer", "Right Input Switch", "MIXINR" }, { "Right Output Mixer", "IN2LN Switch", "IN2LN" }, { "Right Output Mixer", "IN2RN Switch", "IN2RN" }, - { "Right Output Mixer", "IN2RP Switch", "IN2RP/VXRP" }, + { "Right Output Mixer", "IN2RP Switch", "IN2RP:VXRP" }, { "Right Output Mixer", "IN1L Switch", "IN1L PGA" }, { "Right Output Mixer", "IN1R Switch", "IN1R PGA" }, @@ -738,6 +738,41 @@ int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_routes); +int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, + int lineout1_diff, int lineout2_diff, + int lineout1fb, int lineout2fb, + int jd_scthr, int jd_thr, int micbias1_lvl, + int micbias2_lvl) +{ + if (!lineout1_diff) + snd_soc_update_bits(codec, WM8993_LINE_MIXER1, + WM8993_LINEOUT1_MODE, + WM8993_LINEOUT1_MODE); + if (!lineout2_diff) + snd_soc_update_bits(codec, WM8993_LINE_MIXER2, + WM8993_LINEOUT2_MODE, + WM8993_LINEOUT2_MODE); + + if (lineout1fb) + snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, + WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB); + + if (lineout2fb) + snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, + WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB); + + snd_soc_update_bits(codec, WM8993_MICBIAS, + WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK | + WM8993_MICB1_LVL | WM8993_MICB2_LVL, + jd_scthr << WM8993_JD_SCTHR_SHIFT | + jd_thr << WM8993_JD_THR_SHIFT | + micbias1_lvl | + micbias2_lvl << WM8993_MICB2_LVL_SHIFT); + + return 0; +} +EXPORT_SYMBOL_GPL(wm_hubs_handle_analogue_pdata); + MODULE_DESCRIPTION("Shared support for Wolfson hubs products"); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index ec09cb6..36d3fba 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -20,5 +20,10 @@ extern const unsigned int wm_hubs_spkmix_tlv[]; extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *); extern int wm_hubs_add_analogue_routes(struct snd_soc_codec *, int, int); +extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *, + int lineout1_diff, int lineout2_diff, + int lineout1fb, int lineout2fb, + int jd_scthr, int jd_thr, + int micbias1_lvl, int micbias2_lvl); #endif diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 4dfd4ad..047ee39 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -13,9 +13,9 @@ config SND_DAVINCI_SOC_MCASP tristate config SND_DAVINCI_SOC_EVM - tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM" + tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X help diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 67414f6..7ccbe66 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -45,7 +45,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() || + machine_is_davinci_dm365_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -176,7 +177,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .ops = &evm_ops, }; -/* davinci-evm audio machine driver */ +/* davinci dm6446, dm355 or dm365 evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", .platform = &davinci_soc_platform, @@ -243,7 +244,7 @@ static int __init evm_init(void) int index; int ret; - if (machine_is_davinci_evm()) { + if (machine_is_davinci_evm() || machine_is_davinci_dm365_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 0; } else if (machine_is_davinci_dm355_evm()) { diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 4ae7070..6362ca0 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -97,12 +97,24 @@ enum { DAVINCI_MCBSP_WORD_32, }; +static const unsigned char data_type[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = 1, + [SNDRV_PCM_FORMAT_S16_LE] = 2, + [SNDRV_PCM_FORMAT_S32_LE] = 4, +}; + +static const unsigned char asp_word_length[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = DAVINCI_MCBSP_WORD_8, + [SNDRV_PCM_FORMAT_S16_LE] = DAVINCI_MCBSP_WORD_16, + [SNDRV_PCM_FORMAT_S32_LE] = DAVINCI_MCBSP_WORD_32, +}; + +static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = SNDRV_PCM_FORMAT_S16_LE, + [SNDRV_PCM_FORMAT_S16_LE] = SNDRV_PCM_FORMAT_S32_LE, +}; + struct davinci_mcbsp_dev { - /* - * dma_params must be first because rtd->dai->cpu_dai->private_data - * is cast to a pointer of an array of struct davinci_pcm_dma_params in - * davinci_pcm_open. - */ struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; #define MOD_DSP_A 0 @@ -110,6 +122,27 @@ struct davinci_mcbsp_dev { int mode; u32 pcr; struct clk *clk; + /* + * Combining both channels into 1 element will at least double the + * amount of time between servicing the dma channel, increase + * effiency, and reduce the chance of overrun/underrun. But, + * it will result in the left & right channels being swapped. + * + * If relabeling the left and right channels is not possible, + * you may want to let the codec know to swap them back. + * + * It may allow x10 the amount of time to service dma requests, + * if the codec is master and is using an unnecessarily fast bit clock + * (ie. tlvaic23b), independent of the sample rate. So, having an + * entire frame at once means it can be serviced at the sample rate + * instead of the bit clock rate. + * + * In the now unlikely case that an underrun still + * occurs, both the left and right samples will be repeated + * so that no pops are heard, and the left and right channels + * won't end up being swapped because of the underrun. + */ + unsigned enable_channel_combine:1; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -349,6 +382,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, int mcbsp_word_length; unsigned int rcr, xcr, srgr; u32 spcr; + snd_pcm_format_t fmt; + unsigned element_cnt = 1; /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -378,27 +413,24 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1); } /* Determine xfer data type */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - dma_params->data_type = 1; - mcbsp_word_length = DAVINCI_MCBSP_WORD_8; - break; - case SNDRV_PCM_FORMAT_S16_LE: - dma_params->data_type = 2; - mcbsp_word_length = DAVINCI_MCBSP_WORD_16; - break; - case SNDRV_PCM_FORMAT_S32_LE: - dma_params->data_type = 4; - mcbsp_word_length = DAVINCI_MCBSP_WORD_32; - break; - default: + fmt = params_format(params); + if ((fmt > SNDRV_PCM_FORMAT_S32_LE) || !data_type[fmt]) { printk(KERN_WARNING "davinci-i2s: unsupported PCM format\n"); return -EINVAL; } - dma_params->acnt = dma_params->data_type; - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1); + if (params_channels(params) == 2) { + element_cnt = 2; + if (double_fmt[fmt] && dev->enable_channel_combine) { + element_cnt = 1; + fmt = double_fmt[fmt]; + } + } + dma_params->acnt = dma_params->data_type = data_type[fmt]; + dma_params->fifo_level = 0; + mcbsp_word_length = asp_word_length[fmt]; + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); @@ -513,7 +545,13 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENOMEM; goto err_release_region; } - + if (pdata) { + dev->enable_channel_combine = pdata->enable_channel_combine; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].sram_size = + pdata->sram_size_playback; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = + pdata->sram_size_capture; + } dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; @@ -547,6 +585,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; davinci_i2s_dai.private_data = dev; + davinci_i2s_dai.dma_data = dev->dma_params; ret = snd_soc_register_dai(&davinci_i2s_dai); if (ret != 0) goto err_free_mem; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 5d1f98a..0a302e1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -714,16 +714,13 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = &dev->dma_params[substream->stream]; int word_length; - u8 numevt; + u8 fifo_level; davinci_hw_common_param(dev, substream->stream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - numevt = dev->txnumevt; + fifo_level = dev->txnumevt; else - numevt = dev->rxnumevt; - - if (!numevt) - numevt = 1; + fifo_level = dev->rxnumevt; if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) davinci_hw_dit_param(dev); @@ -751,12 +748,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (dev->version == MCASP_VERSION_2) { - dma_params->data_type *= numevt; - dma_params->acnt = 4 * numevt; - } else + if (dev->version == MCASP_VERSION_2 && !fifo_level) + dma_params->acnt = 4; + else dma_params->acnt = dma_params->data_type; + dma_params->fifo_level = fifo_level; davinci_config_channel_size(dev, word_length); return 0; @@ -907,6 +904,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->channel = res->start; davinci_mcasp_dai[pdata->op_mode].private_data = dev; + davinci_mcasp_dai[pdata->op_mode].dma_data = dev->dma_params; davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev; ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]); diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 9d179cc..582c924 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -39,11 +39,6 @@ enum { }; struct davinci_audio_dev { - /* - * dma_params must be first because rtd->dai->cpu_dai->private_data - * is cast to a pointer of an array of struct davinci_pcm_dma_params in - * davinci_pcm_open. - */ struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; int sample_rate; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index c73a915..ad4d7f4 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -3,6 +3,7 @@ * * Author: Vladimir Barinov, <vbarinov@embeddedalley.com> * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com> + * added SRAM ping/pong (C) 2008 Troy Kisky <troy.kisky@boundarydevices.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -23,10 +24,29 @@ #include <asm/dma.h> #include <mach/edma.h> +#include <mach/sram.h> #include "davinci-pcm.h" -static struct snd_pcm_hardware davinci_pcm_hardware = { +#ifdef DEBUG +static void print_buf_info(int slot, char *name) +{ + struct edmacc_param p; + if (slot < 0) + return; + edma_read_slot(slot, &p); + printk(KERN_DEBUG "%s: 0x%x, opt=%x, src=%x, a_b_cnt=%x dst=%x\n", + name, slot, p.opt, p.src, p.a_b_cnt, p.dst); + printk(KERN_DEBUG " src_dst_bidx=%x link_bcntrld=%x src_dst_cidx=%x ccnt=%x\n", + p.src_dst_bidx, p.link_bcntrld, p.src_dst_cidx, p.ccnt); +} +#else +static void print_buf_info(int slot, char *name) +{ +} +#endif + +static struct snd_pcm_hardware pcm_hardware_playback = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE), @@ -48,102 +68,432 @@ static struct snd_pcm_hardware davinci_pcm_hardware = { .fifo_size = 0, }; +static struct snd_pcm_hardware pcm_hardware_capture = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE), + .formats = (SNDRV_PCM_FMTBIT_S16_LE), + .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_KNOT), + .rate_min = 8000, + .rate_max = 96000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 128 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 8 * 1024, + .periods_min = 16, + .periods_max = 255, + .fifo_size = 0, +}; + +/* + * How ping/pong works.... + * + * Playback: + * ram_params - copys 2*ping_size from start of SDRAM to iram, + * links to ram_link2 + * ram_link2 - copys rest of SDRAM to iram in ping_size units, + * links to ram_link + * ram_link - copys entire SDRAM to iram in ping_size uints, + * links to self + * + * asp_params - same as asp_link[0] + * asp_link[0] - copys from lower half of iram to asp port + * links to asp_link[1], triggers iram copy event on completion + * asp_link[1] - copys from upper half of iram to asp port + * links to asp_link[0], triggers iram copy event on completion + * triggers interrupt only needed to let upper SOC levels update position + * in stream on completion + * + * When playback is started: + * ram_params started + * asp_params started + * + * Capture: + * ram_params - same as ram_link, + * links to ram_link + * ram_link - same as playback + * links to self + * + * asp_params - same as playback + * asp_link[0] - same as playback + * asp_link[1] - same as playback + * + * When capture is started: + * asp_params started + */ struct davinci_runtime_data { spinlock_t lock; int period; /* current DMA period */ - int master_lch; /* Master DMA channel */ - int slave_lch; /* linked parameter RAM reload slot */ + int asp_channel; /* Master DMA channel */ + int asp_link[2]; /* asp parameter link channel, ping/pong */ struct davinci_pcm_dma_params *params; /* DMA params */ + int ram_channel; + int ram_link; + int ram_link2; + struct edmacc_param asp_params; + struct edmacc_param ram_params; }; +/* + * Not used with ping/pong + */ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - int lch = prtd->slave_lch; + int link = prtd->asp_link[0]; unsigned int period_size; unsigned int dma_offset; dma_addr_t dma_pos; dma_addr_t src, dst; unsigned short src_bidx, dst_bidx; + unsigned short src_cidx, dst_cidx; unsigned int data_type; unsigned short acnt; unsigned int count; + unsigned int fifo_level; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; dma_pos = runtime->dma_addr + dma_offset; + fifo_level = prtd->params->fifo_level; pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " - "dma_ptr = %x period_size=%x\n", lch, dma_pos, period_size); + "dma_ptr = %x period_size=%x\n", link, dma_pos, period_size); data_type = prtd->params->data_type; count = period_size / data_type; + if (fifo_level) + count /= fifo_level; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; dst_bidx = 0; + src_cidx = data_type * fifo_level; + dst_cidx = 0; } else { src = prtd->params->dma_addr; dst = dma_pos; src_bidx = 0; dst_bidx = data_type; + src_cidx = 0; + dst_cidx = data_type * fifo_level; } acnt = prtd->params->acnt; - edma_set_src(lch, src, INCR, W8BIT); - edma_set_dest(lch, dst, INCR, W8BIT); - edma_set_src_index(lch, src_bidx, 0); - edma_set_dest_index(lch, dst_bidx, 0); - edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC); + edma_set_src(link, src, INCR, W8BIT); + edma_set_dest(link, dst, INCR, W8BIT); + + edma_set_src_index(link, src_bidx, src_cidx); + edma_set_dest_index(link, dst_bidx, dst_cidx); + + if (!fifo_level) + edma_set_transfer_params(link, acnt, count, 1, 0, ASYNC); + else + edma_set_transfer_params(link, acnt, fifo_level, count, + fifo_level, ABSYNC); prtd->period++; if (unlikely(prtd->period >= runtime->periods)) prtd->period = 0; } -static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data) +static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) { struct snd_pcm_substream *substream = data; struct davinci_runtime_data *prtd = substream->runtime->private_data; - pr_debug("davinci_pcm: lch=%d, status=0x%x\n", lch, ch_status); + print_buf_info(prtd->ram_channel, "i ram_channel"); + pr_debug("davinci_pcm: link=%d, status=0x%x\n", link, ch_status); if (unlikely(ch_status != DMA_COMPLETE)) return; if (snd_pcm_running(substream)) { + if (prtd->ram_channel < 0) { + /* No ping/pong must fix up link dma data*/ + spin_lock(&prtd->lock); + davinci_pcm_enqueue_dma(substream); + spin_unlock(&prtd->lock); + } snd_pcm_period_elapsed(substream); + } +} + +static int allocate_sram(struct snd_pcm_substream *substream, unsigned size, + struct snd_pcm_hardware *ppcm) +{ + struct snd_dma_buffer *buf = &substream->dma_buffer; + struct snd_dma_buffer *iram_dma = NULL; + dma_addr_t iram_phys = 0; + void *iram_virt = NULL; + + if (buf->private_data || !size) + return 0; + + ppcm->period_bytes_max = size; + iram_virt = sram_alloc(size, &iram_phys); + if (!iram_virt) + goto exit1; + iram_dma = kzalloc(sizeof(*iram_dma), GFP_KERNEL); + if (!iram_dma) + goto exit2; + iram_dma->area = iram_virt; + iram_dma->addr = iram_phys; + memset(iram_dma->area, 0, size); + iram_dma->bytes = size; + buf->private_data = iram_dma; + return 0; +exit2: + if (iram_virt) + sram_free(iram_virt, size); +exit1: + return -ENOMEM; +} + +/* + * Only used with ping/pong. + * This is called after runtime->dma_addr, period_bytes and data_type are valid + */ +static int ping_pong_dma_setup(struct snd_pcm_substream *substream) +{ + unsigned short ram_src_cidx, ram_dst_cidx; + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd = runtime->private_data; + struct snd_dma_buffer *iram_dma = + (struct snd_dma_buffer *)substream->dma_buffer.private_data; + struct davinci_pcm_dma_params *params = prtd->params; + unsigned int data_type = params->data_type; + unsigned int acnt = params->acnt; + /* divide by 2 for ping/pong */ + unsigned int ping_size = snd_pcm_lib_period_bytes(substream) >> 1; + int link = prtd->asp_link[1]; + unsigned int fifo_level = prtd->params->fifo_level; + unsigned int count; + if ((data_type == 0) || (data_type > 4)) { + printk(KERN_ERR "%s: data_type=%i\n", __func__, data_type); + return -EINVAL; + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dma_addr_t asp_src_pong = iram_dma->addr + ping_size; + ram_src_cidx = ping_size; + ram_dst_cidx = -ping_size; + edma_set_src(link, asp_src_pong, INCR, W8BIT); + + link = prtd->asp_link[0]; + edma_set_src_index(link, data_type, data_type * fifo_level); + link = prtd->asp_link[1]; + edma_set_src_index(link, data_type, data_type * fifo_level); + + link = prtd->ram_link; + edma_set_src(link, runtime->dma_addr, INCR, W32BIT); + } else { + dma_addr_t asp_dst_pong = iram_dma->addr + ping_size; + ram_src_cidx = -ping_size; + ram_dst_cidx = ping_size; + edma_set_dest(link, asp_dst_pong, INCR, W8BIT); + + link = prtd->asp_link[0]; + edma_set_dest_index(link, data_type, data_type * fifo_level); + link = prtd->asp_link[1]; + edma_set_dest_index(link, data_type, data_type * fifo_level); + + link = prtd->ram_link; + edma_set_dest(link, runtime->dma_addr, INCR, W32BIT); + } - spin_lock(&prtd->lock); - davinci_pcm_enqueue_dma(substream); - spin_unlock(&prtd->lock); + if (!fifo_level) { + count = ping_size / data_type; + edma_set_transfer_params(prtd->asp_link[0], acnt, count, + 1, 0, ASYNC); + edma_set_transfer_params(prtd->asp_link[1], acnt, count, + 1, 0, ASYNC); + } else { + count = ping_size / (data_type * fifo_level); + edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, + count, fifo_level, ABSYNC); + edma_set_transfer_params(prtd->asp_link[1], acnt, fifo_level, + count, fifo_level, ABSYNC); } + + link = prtd->ram_link; + edma_set_src_index(link, ping_size, ram_src_cidx); + edma_set_dest_index(link, ping_size, ram_dst_cidx); + edma_set_transfer_params(link, ping_size, 2, + runtime->periods, 2, ASYNC); + + /* init master params */ + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); + edma_read_slot(prtd->ram_link, &prtd->ram_params); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + struct edmacc_param p_ram; + /* Copy entire iram buffer before playback started */ + prtd->ram_params.a_b_cnt = (1 << 16) | (ping_size << 1); + /* 0 dst_bidx */ + prtd->ram_params.src_dst_bidx = (ping_size << 1); + /* 0 dst_cidx */ + prtd->ram_params.src_dst_cidx = (ping_size << 1); + prtd->ram_params.ccnt = 1; + + /* Skip 1st period */ + edma_read_slot(prtd->ram_link, &p_ram); + p_ram.src += (ping_size << 1); + p_ram.ccnt -= 1; + edma_write_slot(prtd->ram_link2, &p_ram); + /* + * When 1st started, ram -> iram dma channel will fill the + * entire iram. Then, whenever a ping/pong asp buffer finishes, + * 1/2 iram will be filled. + */ + prtd->ram_params.link_bcntrld = + EDMA_CHAN_SLOT(prtd->ram_link2) << 5; + } + return 0; +} + +/* 1 asp tx or rx channel using 2 parameter channels + * 1 ram to/from iram channel using 1 parameter channel + * + * Playback + * ram copy channel kicks off first, + * 1st ram copy of entire iram buffer completion kicks off asp channel + * asp tcc always kicks off ram copy of 1/2 iram buffer + * + * Record + * asp channel starts, tcc kicks off ram copy + */ +static int request_ping_pong(struct snd_pcm_substream *substream, + struct davinci_runtime_data *prtd, + struct snd_dma_buffer *iram_dma) +{ + dma_addr_t asp_src_ping; + dma_addr_t asp_dst_ping; + int link; + struct davinci_pcm_dma_params *params = prtd->params; + + /* Request ram master channel */ + link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, + davinci_pcm_dma_irq, substream, + EVENTQ_1); + if (link < 0) + goto exit1; + + /* Request ram link channel */ + link = prtd->ram_link = edma_alloc_slot( + EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); + if (link < 0) + goto exit2; + + link = prtd->asp_link[1] = edma_alloc_slot( + EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); + if (link < 0) + goto exit3; + + prtd->ram_link2 = -1; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + link = prtd->ram_link2 = edma_alloc_slot( + EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); + if (link < 0) + goto exit4; + } + /* circle ping-pong buffers */ + edma_link(prtd->asp_link[0], prtd->asp_link[1]); + edma_link(prtd->asp_link[1], prtd->asp_link[0]); + /* circle ram buffers */ + edma_link(prtd->ram_link, prtd->ram_link); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + asp_src_ping = iram_dma->addr; + asp_dst_ping = params->dma_addr; /* fifo */ + } else { + asp_src_ping = params->dma_addr; /* fifo */ + asp_dst_ping = iram_dma->addr; + } + /* ping */ + link = prtd->asp_link[0]; + edma_set_src(link, asp_src_ping, INCR, W16BIT); + edma_set_dest(link, asp_dst_ping, INCR, W16BIT); + edma_set_src_index(link, 0, 0); + edma_set_dest_index(link, 0, 0); + + edma_read_slot(link, &prtd->asp_params); + prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN); + prtd->asp_params.opt |= TCCHEN | EDMA_TCC(prtd->ram_channel & 0x3f); + edma_write_slot(link, &prtd->asp_params); + + /* pong */ + link = prtd->asp_link[1]; + edma_set_src(link, asp_src_ping, INCR, W16BIT); + edma_set_dest(link, asp_dst_ping, INCR, W16BIT); + edma_set_src_index(link, 0, 0); + edma_set_dest_index(link, 0, 0); + + edma_read_slot(link, &prtd->asp_params); + prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f)); + /* interrupt after every pong completion */ + prtd->asp_params.opt |= TCINTEN | TCCHEN | + EDMA_TCC(EDMA_CHAN_SLOT(prtd->ram_channel)); + edma_write_slot(link, &prtd->asp_params); + + /* ram */ + link = prtd->ram_link; + edma_set_src(link, iram_dma->addr, INCR, W32BIT); + edma_set_dest(link, iram_dma->addr, INCR, W32BIT); + pr_debug("%s: audio dma channels/slots in use for ram:%u %u %u," + "for asp:%u %u %u\n", __func__, + prtd->ram_channel, prtd->ram_link, prtd->ram_link2, + prtd->asp_channel, prtd->asp_link[0], + prtd->asp_link[1]); + return 0; +exit4: + edma_free_channel(prtd->asp_link[1]); + prtd->asp_link[1] = -1; +exit3: + edma_free_channel(prtd->ram_link); + prtd->ram_link = -1; +exit2: + edma_free_channel(prtd->ram_channel); + prtd->ram_channel = -1; +exit1: + return link; } static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) { + struct snd_dma_buffer *iram_dma; struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct edmacc_param p_ram; - int ret; + struct davinci_pcm_dma_params *params = prtd->params; + int link; - /* Request master DMA channel */ - ret = edma_alloc_channel(prtd->params->channel, - davinci_pcm_dma_irq, substream, - EVENTQ_0); - if (ret < 0) - return ret; - prtd->master_lch = ret; + if (!params) + return -ENODEV; - /* Request parameter RAM reload slot */ - ret = edma_alloc_slot(EDMA_CTLR(prtd->master_lch), EDMA_SLOT_ANY); - if (ret < 0) { - edma_free_channel(prtd->master_lch); - return ret; + /* Request asp master DMA channel */ + link = prtd->asp_channel = edma_alloc_channel(params->channel, + davinci_pcm_dma_irq, substream, EVENTQ_0); + if (link < 0) + goto exit1; + + /* Request asp link channels */ + link = prtd->asp_link[0] = edma_alloc_slot( + EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); + if (link < 0) + goto exit2; + + iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data; + if (iram_dma) { + if (request_ping_pong(substream, prtd, iram_dma) == 0) + return 0; + printk(KERN_WARNING "%s: dma channel allocation failed," + "not using sram\n", __func__); } - prtd->slave_lch = ret; /* Issue transfer completion IRQ when the channel completes a * transfer, then always reload from the same slot (by a kind @@ -154,12 +504,17 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) * the buffer and its length (ccnt) ... use it as a template * so davinci_pcm_enqueue_dma() takes less time in IRQ. */ - edma_read_slot(prtd->slave_lch, &p_ram); - p_ram.opt |= TCINTEN | EDMA_TCC(EDMA_CHAN_SLOT(prtd->master_lch)); - p_ram.link_bcntrld = EDMA_CHAN_SLOT(prtd->slave_lch) << 5; - edma_write_slot(prtd->slave_lch, &p_ram); - + edma_read_slot(link, &prtd->asp_params); + prtd->asp_params.opt |= TCINTEN | + EDMA_TCC(EDMA_CHAN_SLOT(prtd->asp_channel)); + prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(link) << 5; + edma_write_slot(link, &prtd->asp_params); return 0; +exit2: + edma_free_channel(prtd->asp_channel); + prtd->asp_channel = -1; +exit1: + return link; } static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) @@ -173,12 +528,12 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - edma_start(prtd->master_lch); + edma_resume(prtd->asp_channel); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - edma_stop(prtd->master_lch); + edma_pause(prtd->asp_channel); break; default: ret = -EINVAL; @@ -193,15 +548,37 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) static int davinci_pcm_prepare(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct edmacc_param temp; + if (prtd->ram_channel >= 0) { + int ret = ping_pong_dma_setup(substream); + if (ret < 0) + return ret; + + edma_write_slot(prtd->ram_channel, &prtd->ram_params); + edma_write_slot(prtd->asp_channel, &prtd->asp_params); + + print_buf_info(prtd->ram_channel, "ram_channel"); + print_buf_info(prtd->ram_link, "ram_link"); + print_buf_info(prtd->ram_link2, "ram_link2"); + print_buf_info(prtd->asp_channel, "asp_channel"); + print_buf_info(prtd->asp_link[0], "asp_link[0]"); + print_buf_info(prtd->asp_link[1], "asp_link[1]"); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* copy 1st iram buffer */ + edma_start(prtd->ram_channel); + } + edma_start(prtd->asp_channel); + return 0; + } prtd->period = 0; davinci_pcm_enqueue_dma(substream); /* Copy self-linked parameter RAM entry into master channel */ - edma_read_slot(prtd->slave_lch, &temp); - edma_write_slot(prtd->master_lch, &temp); + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); + edma_write_slot(prtd->asp_channel, &prtd->asp_params); davinci_pcm_enqueue_dma(substream); + edma_start(prtd->asp_channel); return 0; } @@ -212,20 +589,53 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd = runtime->private_data; unsigned int offset; - dma_addr_t count; - dma_addr_t src, dst; + int asp_count; + dma_addr_t asp_src, asp_dst; spin_lock(&prtd->lock); - - edma_get_position(prtd->master_lch, &src, &dst); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - count = src - runtime->dma_addr; - else - count = dst - runtime->dma_addr; - + if (prtd->ram_channel >= 0) { + int ram_count; + int mod_ram; + dma_addr_t ram_src, ram_dst; + unsigned int period_size = snd_pcm_lib_period_bytes(substream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* reading ram before asp should be safe + * as long as the asp transfers less than a ping size + * of bytes between the 2 reads + */ + edma_get_position(prtd->ram_channel, + &ram_src, &ram_dst); + edma_get_position(prtd->asp_channel, + &asp_src, &asp_dst); + asp_count = asp_src - prtd->asp_params.src; + ram_count = ram_src - prtd->ram_params.src; + mod_ram = ram_count % period_size; + mod_ram -= asp_count; + if (mod_ram < 0) + mod_ram += period_size; + else if (mod_ram == 0) { + if (snd_pcm_running(substream)) + mod_ram += period_size; + } + ram_count -= mod_ram; + if (ram_count < 0) + ram_count += period_size * runtime->periods; + } else { + edma_get_position(prtd->ram_channel, + &ram_src, &ram_dst); + ram_count = ram_dst - prtd->ram_params.dst; + } + asp_count = ram_count; + } else { + edma_get_position(prtd->asp_channel, &asp_src, &asp_dst); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + asp_count = asp_src - runtime->dma_addr; + else + asp_count = asp_dst - runtime->dma_addr; + } spin_unlock(&prtd->lock); - offset = bytes_to_frames(runtime, count); + offset = bytes_to_frames(runtime, asp_count); if (offset >= runtime->buffer_size) offset = 0; @@ -236,14 +646,19 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd; + struct snd_pcm_hardware *ppcm; int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->private_data; - struct davinci_pcm_dma_params *params = &pa[substream->stream]; - if (!params) + struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params *params; + if (!pa) return -ENODEV; + params = &pa[substream->stream]; - snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware); + ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + &pcm_hardware_playback : &pcm_hardware_capture; + allocate_sram(substream, params->sram_size, ppcm); + snd_soc_set_runtime_hwparams(substream, ppcm); /* ensure that buffer size is a multiple of period size */ ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -256,6 +671,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) spin_lock_init(&prtd->lock); prtd->params = params; + prtd->asp_channel = -1; + prtd->asp_link[0] = prtd->asp_link[1] = -1; + prtd->ram_channel = -1; + prtd->ram_link = -1; + prtd->ram_link2 = -1; runtime->private_data = prtd; @@ -273,10 +693,29 @@ static int davinci_pcm_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd = runtime->private_data; - edma_unlink(prtd->slave_lch); - - edma_free_slot(prtd->slave_lch); - edma_free_channel(prtd->master_lch); + if (prtd->ram_channel >= 0) + edma_stop(prtd->ram_channel); + if (prtd->asp_channel >= 0) + edma_stop(prtd->asp_channel); + if (prtd->asp_link[0] >= 0) + edma_unlink(prtd->asp_link[0]); + if (prtd->asp_link[1] >= 0) + edma_unlink(prtd->asp_link[1]); + if (prtd->ram_link >= 0) + edma_unlink(prtd->ram_link); + + if (prtd->asp_link[0] >= 0) + edma_free_slot(prtd->asp_link[0]); + if (prtd->asp_link[1] >= 0) + edma_free_slot(prtd->asp_link[1]); + if (prtd->asp_channel >= 0) + edma_free_channel(prtd->asp_channel); + if (prtd->ram_link >= 0) + edma_free_slot(prtd->ram_link); + if (prtd->ram_link2 >= 0) + edma_free_slot(prtd->ram_link2); + if (prtd->ram_channel >= 0) + edma_free_channel(prtd->ram_channel); kfree(prtd); @@ -318,11 +757,11 @@ static struct snd_pcm_ops davinci_pcm_ops = { .mmap = davinci_pcm_mmap, }; -static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream, + size_t size) { struct snd_pcm_substream *substream = pcm->streams[stream].substream; struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = davinci_pcm_hardware.buffer_bytes_max; buf->dev.type = SNDRV_DMA_TYPE_DEV; buf->dev.dev = pcm->card->dev; @@ -347,6 +786,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) int stream; for (stream = 0; stream < 2; stream++) { + struct snd_dma_buffer *iram_dma; substream = pcm->streams[stream].substream; if (!substream) continue; @@ -358,6 +798,11 @@ static void davinci_pcm_free(struct snd_pcm *pcm) dma_free_writecombine(pcm->card->dev, buf->bytes, buf->area, buf->addr); buf->area = NULL; + iram_dma = (struct snd_dma_buffer *)buf->private_data; + if (iram_dma) { + sram_free(iram_dma->area, iram_dma->bytes); + kfree(iram_dma); + } } } @@ -375,14 +820,16 @@ static int davinci_pcm_new(struct snd_card *card, if (dai->playback.channels_min) { ret = davinci_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); + SNDRV_PCM_STREAM_PLAYBACK, + pcm_hardware_playback.buffer_bytes_max); if (ret) return ret; } if (dai->capture.channels_min) { ret = davinci_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); + SNDRV_PCM_STREAM_CAPTURE, + pcm_hardware_capture.buffer_bytes_max); if (ret) return ret; } diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 8746606..0764944 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -20,9 +20,11 @@ struct davinci_pcm_dma_params { int channel; /* sync dma channel ID */ unsigned short acnt; dma_addr_t dma_addr; /* device physical address for DMA */ + unsigned sram_size; enum dma_event_q eventq_no; /* event queue number */ unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; + unsigned int fifo_level; }; diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 6096d22..30ed568 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -58,47 +58,15 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) /* Prepare and enqueue the next buffer descriptor */ bd = bcom_prepare_next_buffer(s->bcom_task); bd->status = s->period_bytes; - bd->data[0] = s->period_next_pt; + bd->data[0] = s->runtime->dma_addr + (s->period_next * s->period_bytes); bcom_submit_next_buffer(s->bcom_task, NULL); /* Update for next period */ - s->period_next_pt += s->period_bytes; - if (s->period_next_pt >= s->period_end) - s->period_next_pt = s->period_start; -} - -static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s) -{ - if (s->appl_ptr > s->runtime->control->appl_ptr) { - /* - * In this case s->runtime->control->appl_ptr has wrapped around. - * Play the data to the end of the boundary, then wrap our own - * appl_ptr back around. - */ - while (s->appl_ptr < s->runtime->boundary) { - if (bcom_queue_full(s->bcom_task)) - return; - - s->appl_ptr += s->period_size; - - psc_dma_bcom_enqueue_next_buffer(s); - } - s->appl_ptr -= s->runtime->boundary; - } - - while (s->appl_ptr < s->runtime->control->appl_ptr) { - - if (bcom_queue_full(s->bcom_task)) - return; - - s->appl_ptr += s->period_size; - - psc_dma_bcom_enqueue_next_buffer(s); - } + s->period_next = (s->period_next + 1) % s->runtime->periods; } /* Bestcomm DMA irq handler */ -static irqreturn_t psc_dma_bcom_irq_tx(int irq, void *_psc_dma_stream) +static irqreturn_t psc_dma_bcom_irq(int irq, void *_psc_dma_stream) { struct psc_dma_stream *s = _psc_dma_stream; @@ -108,34 +76,8 @@ static irqreturn_t psc_dma_bcom_irq_tx(int irq, void *_psc_dma_stream) while (bcom_buffer_done(s->bcom_task)) { bcom_retrieve_buffer(s->bcom_task, NULL, NULL); - s->period_current_pt += s->period_bytes; - if (s->period_current_pt >= s->period_end) - s->period_current_pt = s->period_start; - } - psc_dma_bcom_enqueue_tx(s); - spin_unlock(&s->psc_dma->lock); - - /* If the stream is active, then also inform the PCM middle layer - * of the period finished event. */ - if (s->active) - snd_pcm_period_elapsed(s->stream); - - return IRQ_HANDLED; -} - -static irqreturn_t psc_dma_bcom_irq_rx(int irq, void *_psc_dma_stream) -{ - struct psc_dma_stream *s = _psc_dma_stream; - - spin_lock(&s->psc_dma->lock); - /* For each finished period, dequeue the completed period buffer - * and enqueue a new one in it's place. */ - while (bcom_buffer_done(s->bcom_task)) { - bcom_retrieve_buffer(s->bcom_task, NULL, NULL); - - s->period_current_pt += s->period_bytes; - if (s->period_current_pt >= s->period_end) - s->period_current_pt = s->period_start; + s->period_current = (s->period_current+1) % s->runtime->periods; + s->period_count++; psc_dma_bcom_enqueue_next_buffer(s); } @@ -166,54 +108,38 @@ static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - struct psc_dma_stream *s; + struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma); struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; u16 imr; unsigned long flags; int i; - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - s = &psc_dma->capture; - else - s = &psc_dma->playback; - - dev_dbg(psc_dma->dev, "psc_dma_trigger(substream=%p, cmd=%i)" - " stream_id=%i\n", - substream, cmd, substream->pstr->stream); - switch (cmd) { case SNDRV_PCM_TRIGGER_START: + dev_dbg(psc_dma->dev, "START: stream=%i fbits=%u ps=%u #p=%u\n", + substream->pstr->stream, runtime->frame_bits, + (int)runtime->period_size, runtime->periods); s->period_bytes = frames_to_bytes(runtime, runtime->period_size); - s->period_start = virt_to_phys(runtime->dma_area); - s->period_end = s->period_start + - (s->period_bytes * runtime->periods); - s->period_next_pt = s->period_start; - s->period_current_pt = s->period_start; - s->period_size = runtime->period_size; + s->period_next = 0; + s->period_current = 0; s->active = 1; - - /* track appl_ptr so that we have a better chance of detecting - * end of stream and not over running it. - */ + s->period_count = 0; s->runtime = runtime; - s->appl_ptr = s->runtime->control->appl_ptr - - (runtime->period_size * runtime->periods); /* Fill up the bestcomm bd queue and enable DMA. * This will begin filling the PSC's fifo. */ spin_lock_irqsave(&psc_dma->lock, flags); - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) bcom_gen_bd_rx_reset(s->bcom_task); - for (i = 0; i < runtime->periods; i++) - if (!bcom_queue_full(s->bcom_task)) - psc_dma_bcom_enqueue_next_buffer(s); - } else { + else bcom_gen_bd_tx_reset(s->bcom_task); - psc_dma_bcom_enqueue_tx(s); - } + + for (i = 0; i < runtime->periods; i++) + if (!bcom_queue_full(s->bcom_task)) + psc_dma_bcom_enqueue_next_buffer(s); bcom_enable(s->bcom_task); spin_unlock_irqrestore(&psc_dma->lock, flags); @@ -223,6 +149,8 @@ static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_STOP: + dev_dbg(psc_dma->dev, "STOP: stream=%i periods_count=%i\n", + substream->pstr->stream, s->period_count); s->active = 0; spin_lock_irqsave(&psc_dma->lock, flags); @@ -236,7 +164,8 @@ static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd) break; default: - dev_dbg(psc_dma->dev, "invalid command\n"); + dev_dbg(psc_dma->dev, "unhandled trigger: stream=%i cmd=%i\n", + substream->pstr->stream, cmd); return -EINVAL; } @@ -343,7 +272,7 @@ psc_dma_pointer(struct snd_pcm_substream *substream) else s = &psc_dma->playback; - count = s->period_current_pt - s->period_start; + count = s->period_current * s->period_bytes; return bytes_to_frames(substream->runtime, count); } @@ -532,11 +461,9 @@ int mpc5200_audio_dma_create(struct of_device *op) rc = request_irq(psc_dma->irq, &psc_dma_status_irq, IRQF_SHARED, "psc-dma-status", psc_dma); - rc |= request_irq(psc_dma->capture.irq, - &psc_dma_bcom_irq_rx, IRQF_SHARED, + rc |= request_irq(psc_dma->capture.irq, &psc_dma_bcom_irq, IRQF_SHARED, "psc-dma-capture", &psc_dma->capture); - rc |= request_irq(psc_dma->playback.irq, - &psc_dma_bcom_irq_tx, IRQF_SHARED, + rc |= request_irq(psc_dma->playback.irq, &psc_dma_bcom_irq, IRQF_SHARED, "psc-dma-playback", &psc_dma->playback); if (rc) { ret = -ENODEV; diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index 8d396bb..22208b3 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -13,26 +13,25 @@ * @psc_dma: pointer back to parent psc_dma data structure * @bcom_task: bestcomm task structure * @irq: irq number for bestcomm task - * @period_start: physical address of start of DMA region * @period_end: physical address of end of DMA region * @period_next_pt: physical address of next DMA buffer to enqueue * @period_bytes: size of DMA period in bytes + * @ac97_slot_bits: Enable bits for turning on the correct AC97 slot */ struct psc_dma_stream { struct snd_pcm_runtime *runtime; - snd_pcm_uframes_t appl_ptr; - int active; struct psc_dma *psc_dma; struct bcom_task *bcom_task; int irq; struct snd_pcm_substream *stream; - dma_addr_t period_start; - dma_addr_t period_end; - dma_addr_t period_next_pt; - dma_addr_t period_current_pt; + int period_next; + int period_current; int period_bytes; - int period_size; + int period_count; + + /* AC97 state */ + u32 ac97_slot_bits; }; /** @@ -73,6 +72,15 @@ struct psc_dma { } stats; }; +/* Utility for retrieving psc_dma_stream structure from a substream */ +inline struct psc_dma_stream * +to_psc_dma_stream(struct snd_pcm_substream *substream, struct psc_dma *psc_dma) +{ + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + return &psc_dma->capture; + return &psc_dma->playback; +} + int mpc5200_audio_dma_create(struct of_device *op); int mpc5200_audio_dma_destroy(struct of_device *op); diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index c4ae3e0..3dbc7f7 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -130,6 +130,7 @@ static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct psc_dma *psc_dma = cpu_dai->private_data; + struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma); dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" " periods=%i buffer_size=%i buffer_bytes=%i channels=%i" @@ -140,20 +141,10 @@ static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, params_channels(params), params_rate(params), params_format(params)); - - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { - if (params_channels(params) == 1) - psc_dma->slots |= 0x00000100; - else - psc_dma->slots |= 0x00000300; - } else { - if (params_channels(params) == 1) - psc_dma->slots |= 0x01000000; - else - psc_dma->slots |= 0x03000000; - } - out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); - + /* Determine the set of enable bits to turn on */ + s->ac97_slot_bits = (params_channels(params) == 1) ? 0x100 : 0x300; + if (substream->pstr->stream != SNDRV_PCM_STREAM_CAPTURE) + s->ac97_slot_bits <<= 16; return 0; } @@ -163,6 +154,8 @@ static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream, { struct psc_dma *psc_dma = cpu_dai->private_data; + dev_dbg(psc_dma->dev, "%s(substream=%p)\n", __func__, substream); + if (params_channels(params) == 1) out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000); else @@ -176,14 +169,24 @@ static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma); switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + dev_dbg(psc_dma->dev, "AC97 START: stream=%i\n", + substream->pstr->stream); + + /* Set the slot enable bits */ + psc_dma->slots |= s->ac97_slot_bits; + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + break; + case SNDRV_PCM_TRIGGER_STOP: - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - psc_dma->slots &= 0xFFFF0000; - else - psc_dma->slots &= 0x0000FFFF; + dev_dbg(psc_dma->dev, "AC97 STOP: stream=%i\n", + substream->pstr->stream); + /* Clear the slot enable bits */ + psc_dma->slots &= ~(s->ac97_slot_bits); out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); break; } diff --git a/sound/soc/imx/mx1_mx2-pcm.c b/sound/soc/imx/mx1_mx2-pcm.c index b838665..bffffcd5 100644 --- a/sound/soc/imx/mx1_mx2-pcm.c +++ b/sound/soc/imx/mx1_mx2-pcm.c @@ -322,12 +322,12 @@ static int mx1_mx2_pcm_open(struct snd_pcm_substream *substream) pr_debug("%s: Requesting dma channel (%s)\n", __func__, prtd->dma_params->name); - prtd->dma_ch = imx_dma_request_by_prio(prtd->dma_params->name, - DMA_PRIO_HIGH); - if (prtd->dma_ch < 0) { + ret = imx_dma_request_by_prio(prtd->dma_params->name, DMA_PRIO_HIGH); + if (ret < 0) { printk(KERN_ERR "Error %d requesting dma channel\n", ret); return ret; } + prtd->dma_ch = ret; imx_dma_config_burstlen(prtd->dma_ch, prtd->dma_params->watermark_level); diff --git a/sound/soc/imx/mx27vis_wm8974.c b/sound/soc/imx/mx27vis_wm8974.c index e4dcb53..0267d2d 100644 --- a/sound/soc/imx/mx27vis_wm8974.c +++ b/sound/soc/imx/mx27vis_wm8974.c @@ -157,7 +157,7 @@ static int mx27vis_hifi_hw_params(struct snd_pcm_substream *substream, /* codec PLL input is 25 MHz */ - ret = codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, + ret = codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, IGNORED_ARG, 25000000, pll_out); if (ret < 0) { printk(KERN_ERR "Error when setting PLL input\n"); diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 2dee983..61952aa 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -21,7 +21,18 @@ config SND_OMAP_SOC_AMS_DELTA select SND_OMAP_SOC_MCBSP select SND_SOC_CX20442 help - Say Y if you want to add support for SoC audio on Amstrad Delta. + Say Y if you want to add support for SoC audio device connected to + a handset and a speakerphone found on Amstrad E3 (Delta) videophone. + + Note that in order to get those devices fully supported, you have to + build the kernel with standard serial port driver included and + configured for at least 4 ports. Then, from userspace, you must load + a line discipline #19 on the modem (ttyS3) serial line. The simplest + way to achieve this is to install util-linux-ng and use the included + ldattach utility. This can be started automatically from udev, + a simple rule like this one should do the trick (it does for me): + ACTION=="add", KERNEL=="controlC0", \ + RUN+="/usr/sbin/ldattach 19 /dev/ttyS3" config SND_OMAP_SOC_OSK5912 tristate "SoC Audio support for omap osk5912" @@ -32,12 +43,13 @@ config SND_OMAP_SOC_OSK5912 Say Y if you want to add support for SoC audio on osk5912. config SND_OMAP_SOC_OVERO - tristate "SoC Audio support for Gumstix Overo" - depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OVERO + tristate "SoC Audio support for Gumstix Overo and CompuLab CM-T35" + depends on TWL4030_CORE && SND_OMAP_SOC && (MACH_OVERO || MACH_CM_T35) select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 help - Say Y if you want to add support for SoC audio on the Gumstix Overo. + Say Y if you want to add support for SoC audio on the + Gumstix Overo or CompuLab CM-T35 config SND_OMAP_SOC_OMAP2EVM tristate "SoC Audio support for OMAP2EVM board" @@ -55,6 +67,15 @@ config SND_OMAP_SOC_OMAP3EVM help Say Y if you want to add support for SoC audio on the omap3evm board. +config SND_OMAP_SOC_AM3517EVM + tristate "SoC Audio support for OMAP3517 / AM3517 EVM" + depends on SND_OMAP_SOC && MACH_OMAP3517EVM && I2C + select SND_OMAP_SOC_MCBSP + select SND_SOC_TLV320AIC23 + help + Say Y if you want to add support for SoC audio on the OMAP3517 / AM3517 + EVM. + config SND_OMAP_SOC_SDP3430 tristate "SoC Audio support for Texas Instruments SDP3430" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP @@ -88,3 +109,10 @@ config SND_OMAP_SOC_ZOOM2 help Say Y if you want to add support for Soc audio on Zoom2 board. +config SND_OMAP_SOC_IGEP0020 + tristate "SoC Audio support for IGEP v2" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_IGEP0020 + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for Soc audio on IGEP v2 board. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 02d6947..3db8a6c 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -12,18 +12,22 @@ snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o snd-soc-omap3evm-objs := omap3evm.o +snd-soc-am3517evm-objs := am3517evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o snd-soc-zoom2-objs := zoom2.o +snd-soc-igep0020-objs := igep0020.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o -obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o -obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o +obj-$(CONFIG_SND_OMAP_SOC_IGEP0020) += snd-soc-igep0020.o diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c new file mode 100644 index 0000000..135901b --- /dev/null +++ b/sound/soc/omap/am3517evm.c @@ -0,0 +1,202 @@ +/* + * am3517evm.c -- ALSA SoC support for OMAP3517 / AM3517 EVM + * + * Author: Anuj Aggarwal <anuj.aggarwal@ti.com> + * + * Based on sound/soc/omap/beagle.c by Steve Sakoman + * + * Copyright (C) 2009 Texas Instruments Incorporated + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind, + * whether express or implied; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <plat/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" + +#include "../codecs/tlv320aic23.h" + +#define CODEC_CLOCK 12000000 + +static int am3517evm_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; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + CODEC_CLOCK, SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_CLKR_SRC_CLKX, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_CLKR_SRC_CLKX\n"); + return ret; + } + + snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_FSR_SRC_FSX\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops am3517evm_ops = { + .hw_params = am3517evm_hw_params, +}; + +/* am3517evm machine dapm widgets */ +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_HP("Line Out", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_MIC("Mic In", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Line Out connected to LLOUT, RLOUT */ + {"Line Out", NULL, "LOUT"}, + {"Line Out", NULL, "ROUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, + + {"MICIN", NULL, "Mic In"}, +}; + +static int am3517evm_aic23_init(struct snd_soc_codec *codec) +{ + /* Add am3517-evm specific widgets */ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* Set up davinci-evm specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* always connected */ + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Mic In"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link am3517evm_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &tlv320aic23_dai, + .init = am3517evm_aic23_init, + .ops = &am3517evm_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_am3517evm = { + .name = "am3517evm", + .platform = &omap_soc_platform, + .dai_link = &am3517evm_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device am3517evm_snd_devdata = { + .card = &snd_soc_am3517evm, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static struct platform_device *am3517evm_snd_device; + +static int __init am3517evm_soc_init(void) +{ + int ret; + + if (!machine_is_omap3517evm()) { + pr_err("Not OMAP3517 / AM3517 EVM!\n"); + return -ENODEV; + } + pr_info("OMAP3517 / AM3517 EVM SoC init\n"); + + am3517evm_snd_device = platform_device_alloc("soc-audio", -1); + if (!am3517evm_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(am3517evm_snd_device, &am3517evm_snd_devdata); + am3517evm_snd_devdata.dev = &am3517evm_snd_device->dev; + *(unsigned int *)am3517evm_dai.cpu_dai->private_data = 0; /* McBSP1 */ + + ret = platform_device_add(am3517evm_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(am3517evm_snd_device); + + return ret; +} + +static void __exit am3517evm_soc_exit(void) +{ + platform_device_unregister(am3517evm_snd_device); +} + +module_init(am3517evm_soc_init); +module_exit(am3517evm_soc_exit); + +MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3517 / AM3517 EVM"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 5a5166a..b0f618e 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -31,8 +31,8 @@ #include <asm/mach-types.h> -#include <mach/board-ams-delta.h> -#include <mach/mcbsp.h> +#include <plat/board-ams-delta.h> +#include <plat/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" @@ -40,7 +40,7 @@ /* Board specific DAPM widgets */ - const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = { +static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = { /* Handset */ SND_SOC_DAPM_MIC("Mouthpiece", NULL), SND_SOC_DAPM_HP("Earpiece", NULL), @@ -81,7 +81,7 @@ static const char *ams_delta_audio_mode[] = (1 << AMS_DELTA_SPEAKER)) #define AMS_DELTA_SPEAKERPHONE (AMS_DELTA_HANDSFREE | (1 << AMS_DELTA_AGC)) -unsigned short ams_delta_audio_mode_pins[] = { +static const unsigned short ams_delta_audio_mode_pins[] = { AMS_DELTA_MIXED, AMS_DELTA_HANDSET, AMS_DELTA_HANDSFREE, diff --git a/sound/soc/omap/igep0020.c b/sound/soc/omap/igep0020.c new file mode 100644 index 0000000..3583c42 --- /dev/null +++ b/sound/soc/omap/igep0020.c @@ -0,0 +1,148 @@ +/* + * igep0020.c -- SoC audio for IGEP v2 + * + * Based on sound/soc/omap/overo.c by Steve Sakoman + * + * 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 <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <plat/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int igep2_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; + + /* 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) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + 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) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops igep2_ops = { + .hw_params = igep2_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link igep2_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], + .ops = &igep2_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_card_igep2 = { + .name = "igep2", + .platform = &omap_soc_platform, + .dai_link = &igep2_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device igep2_snd_devdata = { + .card = &snd_soc_card_igep2, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *igep2_snd_device; + +static int __init igep2_soc_init(void) +{ + int ret; + + if (!machine_is_igep0020()) { + pr_debug("Not IGEP v2!\n"); + return -ENODEV; + } + printk(KERN_INFO "IGEP v2 SoC init\n"); + + igep2_snd_device = platform_device_alloc("soc-audio", -1); + if (!igep2_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(igep2_snd_device, &igep2_snd_devdata); + igep2_snd_devdata.dev = &igep2_snd_device->dev; + *(unsigned int *)igep2_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(igep2_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(igep2_snd_device); + + return ret; +} +module_init(igep2_soc_init); + +static void __exit igep2_soc_exit(void) +{ + platform_device_unregister(igep2_snd_device); +} +module_exit(igep2_soc_exit); + +MODULE_AUTHOR("Enric Balletbo i Serra <eballetbo@iseebcn.com>"); +MODULE_DESCRIPTION("ALSA SoC IGEP v2"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 0a50593..08e09d7 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -32,7 +32,7 @@ #include <asm/mach-types.h> #include <mach/hardware.h> #include <linux/gpio.h> -#include <mach/mcbsp.h> +#include <plat/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 3341f49..6bbbd2a 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -31,9 +31,9 @@ #include <sound/initval.h> #include <sound/soc.h> -#include <mach/control.h> -#include <mach/dma.h> -#include <mach/mcbsp.h> +#include <plat/control.h> +#include <plat/dma.h> +#include <plat/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" @@ -49,6 +49,8 @@ struct omap_mcbsp_data { */ int active; int configured; + unsigned int in_freq; + int clk_div; }; #define to_mcbsp(priv) container_of((priv), struct omap_mcbsp_data, bus_id) @@ -257,7 +259,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT; unsigned long port; - unsigned int format; + unsigned int format, div, framesize, master; if (cpu_class_is_omap1()) { dma = omap1_dma_reqs[bus_id][substream->stream]; @@ -294,28 +296,19 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK; wpf = channels = params_channels(params); - switch (channels) { - case 2: - if (format == SND_SOC_DAIFMT_I2S) { - /* Use dual-phase frames */ - regs->rcr2 |= RPHASE; - regs->xcr2 |= XPHASE; - /* Set 1 word per (McBSP) frame for phase1 and phase2 */ - wpf--; - regs->rcr2 |= RFRLEN2(wpf - 1); - regs->xcr2 |= XFRLEN2(wpf - 1); - } - case 1: - case 4: - /* Set word per (McBSP) frame for phase1 */ - regs->rcr1 |= RFRLEN1(wpf - 1); - regs->xcr1 |= XFRLEN1(wpf - 1); - break; - default: - /* Unsupported number of channels */ - return -EINVAL; + if (channels == 2 && format == SND_SOC_DAIFMT_I2S) { + /* Use dual-phase frames */ + regs->rcr2 |= RPHASE; + regs->xcr2 |= XPHASE; + /* Set 1 word per (McBSP) frame for phase1 and phase2 */ + wpf--; + regs->rcr2 |= RFRLEN2(wpf - 1); + regs->xcr2 |= XFRLEN2(wpf - 1); } + regs->rcr1 |= RFRLEN1(wpf - 1); + regs->xcr1 |= XFRLEN1(wpf - 1); + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: /* Set word lengths */ @@ -330,15 +323,30 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + /* In McBSP master modes, FRAME (i.e. sample rate) is generated + * by _counting_ BCLKs. Calculate frame size in BCLKs */ + master = mcbsp_data->fmt & SND_SOC_DAIFMT_MASTER_MASK; + if (master == SND_SOC_DAIFMT_CBS_CFS) { + div = mcbsp_data->clk_div ? mcbsp_data->clk_div : 1; + framesize = (mcbsp_data->in_freq / div) / params_rate(params); + + if (framesize < wlen * channels) { + printk(KERN_ERR "%s: not enough bandwidth for desired rate and " + "channels\n", __func__); + return -EINVAL; + } + } else + framesize = wlen * channels; + /* Set FS period and length in terms of bit clock periods */ switch (format) { case SND_SOC_DAIFMT_I2S: - regs->srgr2 |= FPER(wlen * channels - 1); - regs->srgr1 |= FWID(wlen - 1); + regs->srgr2 |= FPER(framesize - 1); + regs->srgr1 |= FWID((framesize >> 1) - 1); break; case SND_SOC_DAIFMT_DSP_A: case SND_SOC_DAIFMT_DSP_B: - regs->srgr2 |= FPER(wlen * channels - 1); + regs->srgr2 |= FPER(framesize - 1); regs->srgr1 |= FWID(0); break; } @@ -454,6 +462,7 @@ static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, if (div_id != OMAP_MCBSP_CLKGDV) return -ENODEV; + mcbsp_data->clk_div = div; regs->srgr1 |= CLKGDV(div - 1); return 0; @@ -554,6 +563,8 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int err = 0; + mcbsp_data->in_freq = freq; + switch (clk_id) { case OMAP_MCBSP_SYSCLK_CLK: regs->srgr2 |= CLKSM; @@ -598,13 +609,13 @@ static struct snd_soc_dai_ops omap_mcbsp_dai_ops = { .id = (link_id), \ .playback = { \ .channels_min = 1, \ - .channels_max = 4, \ + .channels_max = 16, \ .rates = OMAP_MCBSP_RATES, \ .formats = SNDRV_PCM_FMTBIT_S16_LE, \ }, \ .capture = { \ .channels_min = 1, \ - .channels_max = 4, \ + .channels_max = 16, \ .rates = OMAP_MCBSP_RATES, \ .formats = SNDRV_PCM_FMTBIT_S16_LE, \ }, \ diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 5735945..9db2770 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -28,7 +28,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> -#include <mach/dma.h> +#include <plat/dma.h> #include "omap-pcm.h" static const struct snd_pcm_hardware omap_pcm_hardware = { @@ -195,8 +195,12 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) else omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ); - omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); - omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); + if (!(cpu_class_is_omap1())) { + omap_set_dma_src_burst_mode(prtd->dma_ch, + OMAP_DMA_DATA_BURST_16); + omap_set_dma_dest_burst_mode(prtd->dma_ch, + OMAP_DMA_DATA_BURST_16); + } return 0; } diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c index 027e1a4..c7adea3 100644 --- a/sound/soc/omap/omap2evm.c +++ b/sound/soc/omap/omap2evm.c @@ -31,7 +31,7 @@ #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> -#include <mach/mcbsp.h> +#include <plat/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index b0cff9f..d88ad5c 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -29,7 +29,7 @@ #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> -#include <mach/mcbsp.h> +#include <plat/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 9114c26..dfcb344 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -27,7 +27,7 @@ #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> -#include <mach/mcbsp.h> +#include <plat/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" @@ -93,10 +93,17 @@ static struct snd_soc_card snd_soc_omap3evm = { .num_links = 1, }; +/* twl4030 setup */ +static struct twl4030_setup_data twl4030_setup = { + .ramp_delay_value = 4, + .sysclk = 26000, +}; + /* Audio subsystem */ static struct snd_soc_device omap3evm_snd_devdata = { .card = &snd_soc_omap3evm, .codec_dev = &soc_codec_dev_twl4030, + .codec_data = &twl4030_setup, }; static struct platform_device *omap3evm_snd_device; @@ -144,4 +151,4 @@ module_exit(omap3evm_soc_exit); MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>"); MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM"); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index ad219aa..71b2c16 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -40,9 +40,12 @@ #define PREFIX "ASoC omap3pandora: " -static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai, - struct snd_soc_dai *cpu_dai, unsigned int fmt) +static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, unsigned int fmt) { + 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; /* Set codec DAI configuration */ @@ -68,8 +71,9 @@ static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai, } /* Set McBSP clock to external */ - ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0, - SND_SOC_CLOCK_IN); + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, + 256 * params_rate(params), + SND_SOC_CLOCK_IN); if (ret < 0) { pr_err(PREFIX "can't set cpu system clock\n"); return ret; @@ -87,11 +91,7 @@ static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai, static int omap3pandora_out_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; - - return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + return omap3pandora_cmn_hw_params(substream, params, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -100,11 +100,7 @@ static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream, static int omap3pandora_in_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; - - return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + return omap3pandora_cmn_hw_params(substream, params, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -134,7 +130,7 @@ static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, * |P| <--- TWL4030 <--------- Line In and MICs */ static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = { - SND_SOC_DAPM_DAC("PCM DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("PCM DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM, 0, 0, NULL, 0, omap3pandora_hp_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -181,6 +177,7 @@ static int omap3pandora_out_init(struct snd_soc_codec *codec) snd_soc_dapm_nc_pin(codec, "CARKITR"); snd_soc_dapm_nc_pin(codec, "HFL"); snd_soc_dapm_nc_pin(codec, "HFR"); + snd_soc_dapm_nc_pin(codec, "VIBRA"); ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets, ARRAY_SIZE(omap3pandora_out_dapm_widgets)); diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index a4e149b7..498ca2e 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -31,7 +31,7 @@ #include <asm/mach-types.h> #include <mach/hardware.h> #include <linux/gpio.h> -#include <mach/mcbsp.h> +#include <plat/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index ec4f8fd..c25f527 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -29,7 +29,7 @@ #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> -#include <mach/mcbsp.h> +#include <plat/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" @@ -107,8 +107,8 @@ static int __init overo_soc_init(void) { int ret; - if (!machine_is_overo()) { - pr_debug("Not Overo!\n"); + if (!(machine_is_overo() || machine_is_cm_t35())) { + pr_debug("Incomatible machine!\n"); return -ENODEV; } printk(KERN_INFO "overo SoC init\n"); diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 4a3f62d..c071f96 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -34,7 +34,7 @@ #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> -#include <mach/mcbsp.h> +#include <plat/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index f90b45f..f90a2ac 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -29,7 +29,7 @@ #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> -#include <mach/mcbsp.h> +#include <plat/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index dcb3181b..376e14a 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -90,7 +90,8 @@ config SND_PXA2XX_SOC_E800 config SND_PXA2XX_SOC_EM_X270 tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300" - depends on SND_PXA2XX_SOC && MACH_EM_X270 + depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \ + MACH_CM_X300) select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -117,6 +118,15 @@ config SND_SOC_ZYLONITE Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. +config SND_SOC_RAUMFELD + tristate "SoC Audio support Raumfeld audio adapter" + depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) + select SND_PXA_SOC_SSP + select SND_SOC_CS4270 + select SND_SOC_AK4104 + help + Say Y if you want to add support for SoC audio on Raumfeld devices + config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" depends on SND_PXA2XX_SOC && MACH_MAGICIAN diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 6e096b4..f3e08fd 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -23,6 +23,7 @@ snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o snd-soc-imote2-objs := imote2.o +snd-soc-raumfeld-objs := raumfeld.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -37,3 +38,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o +obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 9f7c61e..4c8d99a 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -213,7 +213,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, return ret; /* set SSP audio pll clock */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, acps); + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps); if (ret < 0) return ret; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index d11a6d7..3bd7712 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -305,8 +305,8 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, /* * Configure the PLL frequency pxa27x and (afaik - pxa320 only) */ -static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; @@ -760,13 +760,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -780,13 +780,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -801,13 +801,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -822,13 +822,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c new file mode 100644 index 0000000..acfce1c --- /dev/null +++ b/sound/soc/pxa/raumfeld.c @@ -0,0 +1,335 @@ +/* + * raumfeld_audio.c -- SoC audio for Raumfeld audio devices + * + * Copyright (c) 2009 Daniel Mack <daniel@caiaq.de> + * + * based on code from: + * + * Wolfson Microelectronics PLC. + * Openedhand Ltd. + * Liam Girdwood <lrg@slimlogic.co.uk> + * Richard Purdie <richard@openedhand.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "../codecs/cs4270.h" +#include "../codecs/ak4104.h" +#include "pxa2xx-pcm.h" +#include "pxa-ssp.h" + +#define GPIO_SPDIF_RESET (38) +#define GPIO_MCLK_RESET (111) +#define GPIO_CODEC_RESET (120) + +static struct i2c_client *max9486_client; +static struct i2c_board_info max9486_hwmon_info = { + I2C_BOARD_INFO("max9485", 0x63), +}; + +#define MAX9485_MCLK_FREQ_112896 0x22 +#define MAX9485_MCLK_FREQ_122880 0x23 + +static void set_max9485_clk(char clk) +{ + i2c_master_send(max9486_client, &clk, 1); +} + +static void raumfeld_enable_audio(bool en) +{ + if (en) { + gpio_set_value(GPIO_MCLK_RESET, 1); + + /* wait some time to let the clocks become stable */ + msleep(100); + + gpio_set_value(GPIO_SPDIF_RESET, 1); + gpio_set_value(GPIO_CODEC_RESET, 1); + } else { + gpio_set_value(GPIO_MCLK_RESET, 0); + gpio_set_value(GPIO_SPDIF_RESET, 0); + gpio_set_value(GPIO_CODEC_RESET, 0); + } +} + +/* CS4270 */ +static int raumfeld_cs4270_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; + + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + + return snd_soc_dai_set_sysclk(codec_dai, 0, 11289600, 0); +} + +static int raumfeld_cs4270_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; + unsigned int fmt, clk = 0; + int ret = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; + } + + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_cs4270_ops = { + .startup = raumfeld_cs4270_startup, + .hw_params = raumfeld_cs4270_hw_params, +}; + +static int raumfeld_line_suspend(struct platform_device *pdev, pm_message_t state) +{ + raumfeld_enable_audio(false); + return 0; +} + +static int raumfeld_line_resume(struct platform_device *pdev) +{ + raumfeld_enable_audio(true); + return 0; +} + +static struct snd_soc_dai_link raumfeld_line_dai = { + .name = "CS4270", + .stream_name = "CS4270", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .codec_dai = &cs4270_dai, + .ops = &raumfeld_cs4270_ops, +}; + +static struct snd_soc_card snd_soc_line_raumfeld = { + .name = "Raumfeld analog", + .platform = &pxa2xx_soc_platform, + .dai_link = &raumfeld_line_dai, + .suspend_post = raumfeld_line_suspend, + .resume_pre = raumfeld_line_resume, + .num_links = 1, +}; + + +/* AK4104 */ + +static int raumfeld_ak4104_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 fmt, ret = 0, clk = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; + } + + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_ak4104_ops = { + .hw_params = raumfeld_ak4104_hw_params, +}; + +static struct snd_soc_dai_link raumfeld_spdif_dai = { + .name = "ak4104", + .stream_name = "Playback", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP2], + .codec_dai = &ak4104_dai, + .ops = &raumfeld_ak4104_ops, +}; + +static struct snd_soc_card snd_soc_spdif_raumfeld = { + .name = "Raumfeld S/PDIF", + .platform = &pxa2xx_soc_platform, + .dai_link = &raumfeld_spdif_dai, + .num_links = 1 +}; + +/* raumfeld_audio audio subsystem */ +static struct snd_soc_device raumfeld_line_devdata = { + .card = &snd_soc_line_raumfeld, + .codec_dev = &soc_codec_device_cs4270, +}; + +static struct snd_soc_device raumfeld_spdif_devdata = { + .card = &snd_soc_spdif_raumfeld, + .codec_dev = &soc_codec_device_ak4104, +}; + +static struct platform_device *raumfeld_audio_line_device; +static struct platform_device *raumfeld_audio_spdif_device; + +static int __init raumfeld_audio_init(void) +{ + int ret; + + if (!machine_is_raumfeld_speaker() && + !machine_is_raumfeld_connector()) + return 0; + + max9486_client = i2c_new_device(i2c_get_adapter(0), + &max9486_hwmon_info); + + if (!max9486_client) + return -ENOMEM; + + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + + /* LINE */ + raumfeld_audio_line_device = platform_device_alloc("soc-audio", 0); + if (!raumfeld_audio_line_device) + return -ENOMEM; + + platform_set_drvdata(raumfeld_audio_line_device, + &raumfeld_line_devdata); + raumfeld_line_devdata.dev = &raumfeld_audio_line_device->dev; + ret = platform_device_add(raumfeld_audio_line_device); + if (ret) + platform_device_put(raumfeld_audio_line_device); + + /* no S/PDIF on Speakers */ + if (machine_is_raumfeld_speaker()) + return ret; + + /* S/PDIF */ + raumfeld_audio_spdif_device = platform_device_alloc("soc-audio", 1); + if (!raumfeld_audio_spdif_device) { + platform_device_put(raumfeld_audio_line_device); + return -ENOMEM; + } + + platform_set_drvdata(raumfeld_audio_spdif_device, + &raumfeld_spdif_devdata); + raumfeld_spdif_devdata.dev = &raumfeld_audio_spdif_device->dev; + ret = platform_device_add(raumfeld_audio_spdif_device); + if (ret) { + platform_device_put(raumfeld_audio_line_device); + platform_device_put(raumfeld_audio_spdif_device); + } + + raumfeld_enable_audio(true); + + return ret; +} + +static void __exit raumfeld_audio_exit(void) +{ + raumfeld_enable_audio(false); + + platform_device_unregister(raumfeld_audio_line_device); + + if (machine_is_raumfeld_connector()) + platform_device_unregister(raumfeld_audio_spdif_device); + + i2c_unregister_device(max9486_client); + + gpio_free(GPIO_MCLK_RESET); + gpio_free(GPIO_CODEC_RESET); + gpio_free(GPIO_SPDIF_RESET); +} + +module_init(raumfeld_audio_init); +module_exit(raumfeld_audio_exit); + +/* Module information */ +MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>"); +MODULE_DESCRIPTION("Raumfeld audio SoC"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 9a386b4..dd678ae 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -74,7 +74,8 @@ static const struct snd_soc_dapm_route audio_map[] = { static int zylonite_wm9713_init(struct snd_soc_codec *codec) { if (clk_pout) - snd_soc_dai_set_pll(&codec->dai[0], 0, clk_get_rate(pout), 0); + snd_soc_dai_set_pll(&codec->dai[0], 0, 0, + clk_get_rate(pout), 0); snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets, ARRAY_SIZE(zylonite_dapm_widgets)); @@ -128,7 +129,7 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, pll_out); + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out); if (ret < 0) return ret; diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 923428f..b489f1a 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -24,6 +24,9 @@ config SND_S3C64XX_SOC_I2S select SND_S3C_I2SV2_SOC select S3C64XX_DMA +config SND_S3C_SOC_PCM + tristate + config SND_S3C2443_SOC_AC97 tristate select S3C2410_DMA @@ -56,6 +59,15 @@ config SND_S3C24XX_SOC_JIVE_WM8750 help Sat Y if you want to add support for SoC audio on the Jive. +config SND_S3C64XX_SOC_WM8580 + tristate "SoC I2S Audio support for WM8580 on SMDK64XX" + depends on SND_S3C24XX_SOC && (MACH_SMDK6400 || MACH_SMDK6410) + depends on BROKEN + select SND_SOC_WM8580 + select SND_S3C64XX_SOC_I2S + help + Sat Y if you want to add support for SoC audio on the SMDK64XX. + config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_S3C24XX_SOC && MACH_SMDK2443 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 99f5a7d..b744657 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -1,10 +1,11 @@ # S3c24XX Platform Support -snd-soc-s3c24xx-objs := s3c24xx-pcm.o +snd-soc-s3c24xx-objs := s3c-dma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o +snd-soc-s3c-pcm-objs := s3c-pcm.o obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o @@ -12,6 +13,7 @@ obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o +obj-$(CONFIG_SND_S3C_SOC_PCM) += snd-soc-s3c-pcm.o # S3C24XX Machine Support snd-soc-jive-wm8750-objs := jive_wm8750.o @@ -23,6 +25,7 @@ snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o +snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -33,4 +36,5 @@ obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o +obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c index 93e6c87..59dc2c6 100644 --- a/sound/soc/s3c24xx/jive_wm8750.c +++ b/sound/soc/s3c24xx/jive_wm8750.c @@ -25,7 +25,7 @@ #include <asm/mach-types.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c2412-i2s.h" #include "../codecs/wm8750.h" diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c index 12c7148..d00d359 100644 --- a/sound/soc/s3c24xx/ln2440sbc_alc650.c +++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c @@ -24,7 +24,7 @@ #include <sound/soc-dapm.h> #include "../codecs/ac97.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-ac97.h" static struct snd_soc_card ln2440sbc; diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c index 0c52e36..dea83d3 100644 --- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c @@ -32,7 +32,7 @@ #include <asm/io.h> #include <mach/gta02.h> #include "../codecs/wm8753.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" static struct snd_soc_card neo1973_gta02; @@ -119,7 +119,7 @@ static int neo1973_gta02_hifi_hw_params(struct snd_pcm_substream *substream, return ret; /* codec PLL input is PCLK/4 */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, iis_clkrate / 4, pll_out); if (ret < 0) return ret; @@ -133,7 +133,7 @@ static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); } /* @@ -183,7 +183,7 @@ static int neo1973_gta02_voice_hw_params( return ret; /* configue and enable PLL for 12.288MHz output */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, iis_clkrate / 4, 12288000); if (ret < 0) return ret; @@ -197,7 +197,7 @@ static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); } static struct snd_soc_ops neo1973_gta02_voice_ops = { diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 906709e..0cb4f86 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -29,7 +29,6 @@ #include <mach/regs-clock.h> #include <mach/regs-gpio.h> #include <mach/hardware.h> -#include <plat/audio.h> #include <linux/io.h> #include <mach/spi-gpio.h> @@ -37,7 +36,7 @@ #include "../codecs/wm8753.h" #include "lm4857.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" /* define the scenarios */ @@ -137,7 +136,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, return ret; /* codec PLL input is PCLK/4 */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, iis_clkrate / 4, pll_out); if (ret < 0) return ret; @@ -153,7 +152,7 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); } /* @@ -203,7 +202,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, return ret; /* configue and enable PLL for 12.288MHz output */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, iis_clkrate / 4, 12288000); if (ret < 0) return ret; @@ -219,7 +218,7 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); } static struct snd_soc_ops neo1973_voice_ops = { diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c-dma.c index 5cbbdc8..7725e26 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c-dma.c @@ -1,5 +1,5 @@ /* - * s3c24xx-pcm.c -- ALSA Soc Audio Layer + * s3c-dma.c -- ALSA Soc Audio Layer * * (c) 2006 Wolfson Microelectronics PLC. * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com @@ -29,11 +29,10 @@ #include <asm/dma.h> #include <mach/hardware.h> #include <mach/dma.h> -#include <plat/audio.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" -static const struct snd_pcm_hardware s3c24xx_pcm_hardware = { +static const struct snd_pcm_hardware s3c_dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | @@ -63,23 +62,32 @@ struct s3c24xx_runtime_data { dma_addr_t dma_start; dma_addr_t dma_pos; dma_addr_t dma_end; - struct s3c24xx_pcm_dma_params *params; + struct s3c_dma_params *params; }; -/* s3c24xx_pcm_enqueue +/* s3c_dma_enqueue * * place a dma buffer onto the queue for the dma system * to handle. */ -static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream) +static void s3c_dma_enqueue(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; dma_addr_t pos = prtd->dma_pos; + unsigned int limit; int ret; pr_debug("Entered %s\n", __func__); - while (prtd->dma_loaded < prtd->dma_limit) { + if (s3c_dma_has_circular()) + limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; + else + limit = prtd->dma_limit; + + pr_debug("%s: loaded %d, limit %d\n", + __func__, prtd->dma_loaded, limit); + + while (prtd->dma_loaded < limit) { unsigned long len = prtd->dma_period; pr_debug("dma_loaded: %d\n", prtd->dma_loaded); @@ -123,21 +131,21 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, snd_pcm_period_elapsed(substream); spin_lock(&prtd->lock); - if (prtd->state & ST_RUNNING) { + if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) { prtd->dma_loaded--; - s3c24xx_pcm_enqueue(substream); + s3c_dma_enqueue(substream); } spin_unlock(&prtd->lock); } -static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, +static int s3c_dma_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c24xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data; + struct s3c_dma_params *dma = rtd->dai->cpu_dai->dma_data; unsigned long totbytes = params_buffer_bytes(params); int ret = 0; @@ -164,6 +172,11 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, printk(KERN_ERR "failed to get dma channel\n"); return ret; } + + /* use the circular buffering if we have it available. */ + if (s3c_dma_has_circular()) + s3c2410_dma_setflags(prtd->params->channel, + S3C2410_DMAF_CIRCULAR); } s3c2410_dma_set_buffdone_fn(prtd->params->channel, @@ -185,7 +198,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream) +static int s3c_dma_hw_free(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; @@ -202,7 +215,7 @@ static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) +static int s3c_dma_prepare(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; int ret = 0; @@ -235,12 +248,12 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) prtd->dma_pos = prtd->dma_start; /* enqueue dma buffers */ - s3c24xx_pcm_enqueue(substream); + s3c_dma_enqueue(substream); return ret; } -static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int s3c_dma_trigger(struct snd_pcm_substream *substream, int cmd) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; int ret = 0; @@ -275,7 +288,7 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } static snd_pcm_uframes_t -s3c24xx_pcm_pointer(struct snd_pcm_substream *substream) +s3c_dma_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; @@ -310,7 +323,7 @@ s3c24xx_pcm_pointer(struct snd_pcm_substream *substream) return bytes_to_frames(substream->runtime, res); } -static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) +static int s3c_dma_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd; @@ -318,7 +331,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware); + snd_soc_set_runtime_hwparams(substream, &s3c_dma_hardware); prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); if (prtd == NULL) @@ -330,7 +343,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) return 0; } -static int s3c24xx_pcm_close(struct snd_pcm_substream *substream) +static int s3c_dma_close(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; @@ -338,14 +351,14 @@ static int s3c24xx_pcm_close(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); if (!prtd) - pr_debug("s3c24xx_pcm_close called with prtd == NULL\n"); + pr_debug("s3c_dma_close called with prtd == NULL\n"); kfree(prtd); return 0; } -static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream, +static int s3c_dma_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -358,23 +371,23 @@ static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream, runtime->dma_bytes); } -static struct snd_pcm_ops s3c24xx_pcm_ops = { - .open = s3c24xx_pcm_open, - .close = s3c24xx_pcm_close, +static struct snd_pcm_ops s3c_dma_ops = { + .open = s3c_dma_open, + .close = s3c_dma_close, .ioctl = snd_pcm_lib_ioctl, - .hw_params = s3c24xx_pcm_hw_params, - .hw_free = s3c24xx_pcm_hw_free, - .prepare = s3c24xx_pcm_prepare, - .trigger = s3c24xx_pcm_trigger, - .pointer = s3c24xx_pcm_pointer, - .mmap = s3c24xx_pcm_mmap, + .hw_params = s3c_dma_hw_params, + .hw_free = s3c_dma_hw_free, + .prepare = s3c_dma_prepare, + .trigger = s3c_dma_trigger, + .pointer = s3c_dma_pointer, + .mmap = s3c_dma_mmap, }; -static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +static int s3c_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { struct snd_pcm_substream *substream = pcm->streams[stream].substream; struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = s3c24xx_pcm_hardware.buffer_bytes_max; + size_t size = s3c_dma_hardware.buffer_bytes_max; pr_debug("Entered %s\n", __func__); @@ -389,7 +402,7 @@ static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) return 0; } -static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) +static void s3c_dma_free_dma_buffers(struct snd_pcm *pcm) { struct snd_pcm_substream *substream; struct snd_dma_buffer *buf; @@ -412,9 +425,9 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) } } -static u64 s3c24xx_pcm_dmamask = DMA_BIT_MASK(32); +static u64 s3c_dma_mask = DMA_BIT_MASK(32); -static int s3c24xx_pcm_new(struct snd_card *card, +static int s3c_dma_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -422,19 +435,19 @@ static int s3c24xx_pcm_new(struct snd_card *card, pr_debug("Entered %s\n", __func__); if (!card->dev->dma_mask) - card->dev->dma_mask = &s3c24xx_pcm_dmamask; + card->dev->dma_mask = &s3c_dma_mask; if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = 0xffffffff; if (dai->playback.channels_min) { - ret = s3c24xx_pcm_preallocate_dma_buffer(pcm, + ret = s3c_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto out; } if (dai->capture.channels_min) { - ret = s3c24xx_pcm_preallocate_dma_buffer(pcm, + ret = s3c_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) goto out; @@ -445,9 +458,9 @@ static int s3c24xx_pcm_new(struct snd_card *card, struct snd_soc_platform s3c24xx_soc_platform = { .name = "s3c24xx-audio", - .pcm_ops = &s3c24xx_pcm_ops, - .pcm_new = s3c24xx_pcm_new, - .pcm_free = s3c24xx_pcm_free_dma_buffers, + .pcm_ops = &s3c_dma_ops, + .pcm_new = s3c_dma_new, + .pcm_free = s3c_dma_free_dma_buffers, }; EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); @@ -464,5 +477,5 @@ static void __exit s3c24xx_soc_platform_exit(void) module_exit(s3c24xx_soc_platform_exit); MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>"); -MODULE_DESCRIPTION("Samsung S3C24XX PCM DMA module"); +MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.h b/sound/soc/s3c24xx/s3c-dma.h index 0088c79..69bb6bf 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.h +++ b/sound/soc/s3c24xx/s3c-dma.h @@ -1,5 +1,5 @@ /* - * s3c24xx-pcm.h -- + * s3c-dma.h -- * * 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 @@ -9,13 +9,13 @@ * ALSA PCM interface for the Samsung S3C24xx CPU */ -#ifndef _S3C24XX_PCM_H -#define _S3C24XX_PCM_H +#ifndef _S3C_AUDIO_H +#define _S3C_AUDIO_H #define ST_RUNNING (1<<0) #define ST_OPENED (1<<1) -struct s3c24xx_pcm_dma_params { +struct s3c_dma_params { struct s3c2410_dma_client *client; /* stream identifier */ int channel; /* Channel ID */ dma_addr_t dma_addr; diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 9bc4aa3..e994d83 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -32,11 +32,10 @@ #include <plat/regs-s3c2412-iis.h> -#include <plat/audio.h> #include <mach/dma.h> #include "s3c-i2s-v2.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #undef S3C_IIS_V2_SUPPORTED @@ -312,12 +311,15 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_RIGHT_J: + iismod |= S3C2412_IISMOD_LR_RLOW; iismod |= S3C2412_IISMOD_SDF_MSB; break; case SND_SOC_DAIFMT_LEFT_J: + iismod |= S3C2412_IISMOD_LR_RLOW; iismod |= S3C2412_IISMOD_SDF_LSB; break; case SND_SOC_DAIFMT_I2S: + iismod &= ~S3C2412_IISMOD_LR_RLOW; iismod |= S3C2412_IISMOD_SDF_IIS; break; default: @@ -392,7 +394,7 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; int ret = 0; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; pr_debug("Entered %s\n", __func__); @@ -467,6 +469,31 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, switch (div_id) { case S3C_I2SV2_DIV_BCLK: + if (div > 3) { + /* convert value to bit field */ + + switch (div) { + case 16: + div = S3C2412_IISMOD_BCLK_16FS; + break; + + case 32: + div = S3C2412_IISMOD_BCLK_32FS; + break; + + case 24: + div = S3C2412_IISMOD_BCLK_24FS; + break; + + case 48: + div = S3C2412_IISMOD_BCLK_48FS; + break; + + default: + return -EINVAL; + } + } + reg = readl(i2s->regs + S3C2412_IISMOD); reg &= ~S3C2412_IISMOD_BCLK_MASK; writel(reg | div, i2s->regs + S3C2412_IISMOD); @@ -626,7 +653,7 @@ int s3c_i2sv2_probe(struct platform_device *pdev, } i2s->iis_pclk = clk_get(dev, "iis"); - if (i2s->iis_pclk == NULL) { + if (IS_ERR(i2s->iis_pclk)) { dev_err(dev, "failed to get iis_clock\n"); iounmap(i2s->regs); return -ENOENT; diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h index f66854a..ecf8eaa 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.h +++ b/sound/soc/s3c24xx/s3c-i2s-v2.h @@ -49,8 +49,8 @@ struct s3c_i2sv2_info { unsigned char master; - struct s3c24xx_pcm_dma_params *dma_playback; - struct s3c24xx_pcm_dma_params *dma_capture; + struct s3c_dma_params *dma_playback; + struct s3c_dma_params *dma_capture; u32 suspend_iismod; u32 suspend_iiscon; diff --git a/sound/soc/s3c24xx/s3c-pcm.c b/sound/soc/s3c24xx/s3c-pcm.c new file mode 100644 index 0000000..9e61a7c --- /dev/null +++ b/sound/soc/s3c24xx/s3c-pcm.c @@ -0,0 +1,552 @@ +/* sound/soc/s3c24xx/s3c-pcm.c + * + * ALSA SoC Audio Layer - S3C PCM-Controller driver + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar@samsung.com> + * based upon I2S drivers by Ben Dooks. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/kernel.h> +#include <linux/gpio.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <plat/audio.h> +#include <plat/dma.h> + +#include "s3c-dma.h" +#include "s3c-pcm.h" + +static struct s3c2410_dma_client s3c_pcm_dma_client_out = { + .name = "PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c_pcm_dma_client_in = { + .name = "PCM Stereo in" +}; + +static struct s3c_dma_params s3c_pcm_stereo_out[] = { + [0] = { + .client = &s3c_pcm_dma_client_out, + .dma_size = 4, + }, + [1] = { + .client = &s3c_pcm_dma_client_out, + .dma_size = 4, + }, +}; + +static struct s3c_dma_params s3c_pcm_stereo_in[] = { + [0] = { + .client = &s3c_pcm_dma_client_in, + .dma_size = 4, + }, + [1] = { + .client = &s3c_pcm_dma_client_in, + .dma_size = 4, + }, +}; + +static struct s3c_pcm_info s3c_pcm[2]; + +static inline struct s3c_pcm_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static void s3c_pcm_snd_txctrl(struct s3c_pcm_info *pcm, int on) +{ + void __iomem *regs = pcm->regs; + u32 ctl, clkctl; + + clkctl = readl(regs + S3C_PCM_CLKCTL); + ctl = readl(regs + S3C_PCM_CTL); + ctl &= ~(S3C_PCM_CTL_TXDIPSTICK_MASK + << S3C_PCM_CTL_TXDIPSTICK_SHIFT); + + if (on) { + ctl |= S3C_PCM_CTL_TXDMA_EN; + ctl |= S3C_PCM_CTL_TXFIFO_EN; + ctl |= S3C_PCM_CTL_ENABLE; + ctl |= (0x20<<S3C_PCM_CTL_TXDIPSTICK_SHIFT); + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } else { + ctl &= ~S3C_PCM_CTL_TXDMA_EN; + ctl &= ~S3C_PCM_CTL_TXFIFO_EN; + + if (!(ctl & S3C_PCM_CTL_RXFIFO_EN)) { + ctl &= ~S3C_PCM_CTL_ENABLE; + if (!pcm->idleclk) + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } + } + + writel(clkctl, regs + S3C_PCM_CLKCTL); + writel(ctl, regs + S3C_PCM_CTL); +} + +static void s3c_pcm_snd_rxctrl(struct s3c_pcm_info *pcm, int on) +{ + void __iomem *regs = pcm->regs; + u32 ctl, clkctl; + + ctl = readl(regs + S3C_PCM_CTL); + clkctl = readl(regs + S3C_PCM_CLKCTL); + + if (on) { + ctl |= S3C_PCM_CTL_RXDMA_EN; + ctl |= S3C_PCM_CTL_RXFIFO_EN; + ctl |= S3C_PCM_CTL_ENABLE; + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } else { + ctl &= ~S3C_PCM_CTL_RXDMA_EN; + ctl &= ~S3C_PCM_CTL_RXFIFO_EN; + + if (!(ctl & S3C_PCM_CTL_TXFIFO_EN)) { + ctl &= ~S3C_PCM_CTL_ENABLE; + if (!pcm->idleclk) + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } + } + + writel(clkctl, regs + S3C_PCM_CLKCTL); + writel(ctl, regs + S3C_PCM_CTL); +} + +static int s3c_pcm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_pcm_info *pcm = to_info(rtd->dai->cpu_dai); + unsigned long flags; + + dev_dbg(pcm->dev, "Entered %s\n", __func__); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&pcm->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + s3c_pcm_snd_rxctrl(pcm, 1); + else + s3c_pcm_snd_txctrl(pcm, 1); + + spin_unlock_irqrestore(&pcm->lock, flags); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&pcm->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + s3c_pcm_snd_rxctrl(pcm, 0); + else + s3c_pcm_snd_txctrl(pcm, 0); + + spin_unlock_irqrestore(&pcm->lock, flags); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *socdai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_link *dai = rtd->dai; + struct s3c_pcm_info *pcm = to_info(dai->cpu_dai); + void __iomem *regs = pcm->regs; + struct clk *clk; + int sclk_div, sync_div; + unsigned long flags; + u32 clkctl; + + dev_dbg(pcm->dev, "Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dai->cpu_dai->dma_data = pcm->dma_playback; + else + dai->cpu_dai->dma_data = pcm->dma_capture; + + /* Strictly check for sample size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + default: + return -EINVAL; + } + + spin_lock_irqsave(&pcm->lock, flags); + + /* Get hold of the PCMSOURCE_CLK */ + clkctl = readl(regs + S3C_PCM_CLKCTL); + if (clkctl & S3C_PCM_CLKCTL_SERCLKSEL_PCLK) + clk = pcm->pclk; + else + clk = pcm->cclk; + + /* Set the SCLK divider */ + sclk_div = clk_get_rate(clk) / pcm->sclk_per_fs / + params_rate(params) / 2 - 1; + + clkctl &= ~(S3C_PCM_CLKCTL_SCLKDIV_MASK + << S3C_PCM_CLKCTL_SCLKDIV_SHIFT); + clkctl |= ((sclk_div & S3C_PCM_CLKCTL_SCLKDIV_MASK) + << S3C_PCM_CLKCTL_SCLKDIV_SHIFT); + + /* Set the SYNC divider */ + sync_div = pcm->sclk_per_fs - 1; + + clkctl &= ~(S3C_PCM_CLKCTL_SYNCDIV_MASK + << S3C_PCM_CLKCTL_SYNCDIV_SHIFT); + clkctl |= ((sync_div & S3C_PCM_CLKCTL_SYNCDIV_MASK) + << S3C_PCM_CLKCTL_SYNCDIV_SHIFT); + + writel(clkctl, regs + S3C_PCM_CLKCTL); + + spin_unlock_irqrestore(&pcm->lock, flags); + + dev_dbg(pcm->dev, "PCMSOURCE_CLK-%lu SCLK=%ufs \ + SCLK_DIV=%d SYNC_DIV=%d\n", + clk_get_rate(clk), pcm->sclk_per_fs, + sclk_div, sync_div); + + return 0; +} + +static int s3c_pcm_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct s3c_pcm_info *pcm = to_info(cpu_dai); + void __iomem *regs = pcm->regs; + unsigned long flags; + int ret = 0; + u32 ctl; + + dev_dbg(pcm->dev, "Entered %s\n", __func__); + + spin_lock_irqsave(&pcm->lock, flags); + + ctl = readl(regs + S3C_PCM_CTL); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Nothing to do, NB_NF by default */ + break; + default: + dev_err(pcm->dev, "Unsupported clock inversion!\n"); + ret = -EINVAL; + goto exit; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* Nothing to do, Master by default */ + break; + default: + dev_err(pcm->dev, "Unsupported master/slave format!\n"); + ret = -EINVAL; + goto exit; + } + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + pcm->idleclk = 1; + break; + case SND_SOC_DAIFMT_GATED: + pcm->idleclk = 0; + break; + default: + dev_err(pcm->dev, "Invalid Clock gating request!\n"); + ret = -EINVAL; + goto exit; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + ctl |= S3C_PCM_CTL_TXMSB_AFTER_FSYNC; + ctl |= S3C_PCM_CTL_RXMSB_AFTER_FSYNC; + break; + case SND_SOC_DAIFMT_DSP_B: + ctl &= ~S3C_PCM_CTL_TXMSB_AFTER_FSYNC; + ctl &= ~S3C_PCM_CTL_RXMSB_AFTER_FSYNC; + break; + default: + dev_err(pcm->dev, "Unsupported data format!\n"); + ret = -EINVAL; + goto exit; + } + + writel(ctl, regs + S3C_PCM_CTL); + +exit: + spin_unlock_irqrestore(&pcm->lock, flags); + + return ret; +} + +static int s3c_pcm_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct s3c_pcm_info *pcm = to_info(cpu_dai); + + switch (div_id) { + case S3C_PCM_SCLK_PER_FS: + pcm->sclk_per_fs = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int s3c_pcm_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct s3c_pcm_info *pcm = to_info(cpu_dai); + void __iomem *regs = pcm->regs; + u32 clkctl = readl(regs + S3C_PCM_CLKCTL); + + switch (clk_id) { + case S3C_PCM_CLKSRC_PCLK: + clkctl |= S3C_PCM_CLKCTL_SERCLKSEL_PCLK; + break; + + case S3C_PCM_CLKSRC_MUX: + clkctl &= ~S3C_PCM_CLKCTL_SERCLKSEL_PCLK; + + if (clk_get_rate(pcm->cclk) != freq) + clk_set_rate(pcm->cclk, freq); + + break; + + default: + return -EINVAL; + } + + writel(clkctl, regs + S3C_PCM_CLKCTL); + + return 0; +} + +static struct snd_soc_dai_ops s3c_pcm_dai_ops = { + .set_sysclk = s3c_pcm_set_sysclk, + .set_clkdiv = s3c_pcm_set_clkdiv, + .trigger = s3c_pcm_trigger, + .hw_params = s3c_pcm_hw_params, + .set_fmt = s3c_pcm_set_fmt, +}; + +#define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 + +#define S3C_PCM_DECLARE(n) \ +{ \ + .name = "samsung-pcm", \ + .id = (n), \ + .symmetric_rates = 1, \ + .ops = &s3c_pcm_dai_ops, \ + .playback = { \ + .channels_min = 2, \ + .channels_max = 2, \ + .rates = S3C_PCM_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ + .capture = { \ + .channels_min = 2, \ + .channels_max = 2, \ + .rates = S3C_PCM_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ +} + +struct snd_soc_dai s3c_pcm_dai[] = { + S3C_PCM_DECLARE(0), + S3C_PCM_DECLARE(1), +}; +EXPORT_SYMBOL_GPL(s3c_pcm_dai); + +static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) +{ + struct s3c_pcm_info *pcm; + struct snd_soc_dai *dai; + struct resource *mem_res, *dmatx_res, *dmarx_res; + struct s3c_audio_pdata *pcm_pdata; + int ret; + + /* Check for valid device index */ + if ((pdev->id < 0) || pdev->id >= ARRAY_SIZE(s3c_pcm)) { + dev_err(&pdev->dev, "id %d out of range\n", pdev->id); + return -EINVAL; + } + + pcm_pdata = pdev->dev.platform_data; + + /* Check for availability of necessary resource */ + dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmatx_res) { + dev_err(&pdev->dev, "Unable to get PCM-TX dma resource\n"); + return -ENXIO; + } + + dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmarx_res) { + dev_err(&pdev->dev, "Unable to get PCM-RX dma resource\n"); + return -ENXIO; + } + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem_res) { + dev_err(&pdev->dev, "Unable to get register resource\n"); + return -ENXIO; + } + + if (pcm_pdata && pcm_pdata->cfg_gpio && pcm_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + return -EINVAL; + } + + pcm = &s3c_pcm[pdev->id]; + pcm->dev = &pdev->dev; + + spin_lock_init(&pcm->lock); + + dai = &s3c_pcm_dai[pdev->id]; + dai->dev = &pdev->dev; + + /* Default is 128fs */ + pcm->sclk_per_fs = 128; + + pcm->cclk = clk_get(&pdev->dev, "audio-bus"); + if (IS_ERR(pcm->cclk)) { + dev_err(&pdev->dev, "failed to get audio-bus\n"); + ret = PTR_ERR(pcm->cclk); + goto err1; + } + clk_enable(pcm->cclk); + + /* record our pcm structure for later use in the callbacks */ + dai->private_data = pcm; + + if (!request_mem_region(mem_res->start, + resource_size(mem_res), "samsung-pcm")) { + dev_err(&pdev->dev, "Unable to request register region\n"); + ret = -EBUSY; + goto err2; + } + + pcm->regs = ioremap(mem_res->start, 0x100); + if (pcm->regs == NULL) { + dev_err(&pdev->dev, "cannot ioremap registers\n"); + ret = -ENXIO; + goto err3; + } + + pcm->pclk = clk_get(&pdev->dev, "pcm"); + if (IS_ERR(pcm->pclk)) { + dev_err(&pdev->dev, "failed to get pcm_clock\n"); + ret = -ENOENT; + goto err4; + } + clk_enable(pcm->pclk); + + ret = snd_soc_register_dai(dai); + if (ret != 0) { + dev_err(&pdev->dev, "failed to get pcm_clock\n"); + goto err5; + } + + s3c_pcm_stereo_in[pdev->id].dma_addr = mem_res->start + + S3C_PCM_RXFIFO; + s3c_pcm_stereo_out[pdev->id].dma_addr = mem_res->start + + S3C_PCM_TXFIFO; + + s3c_pcm_stereo_in[pdev->id].channel = dmarx_res->start; + s3c_pcm_stereo_out[pdev->id].channel = dmatx_res->start; + + pcm->dma_capture = &s3c_pcm_stereo_in[pdev->id]; + pcm->dma_playback = &s3c_pcm_stereo_out[pdev->id]; + + return 0; + +err5: + clk_disable(pcm->pclk); + clk_put(pcm->pclk); +err4: + iounmap(pcm->regs); +err3: + release_mem_region(mem_res->start, resource_size(mem_res)); +err2: + clk_disable(pcm->cclk); + clk_put(pcm->cclk); +err1: + return ret; +} + +static __devexit int s3c_pcm_dev_remove(struct platform_device *pdev) +{ + struct s3c_pcm_info *pcm = &s3c_pcm[pdev->id]; + struct resource *mem_res; + + iounmap(pcm->regs); + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(mem_res->start, resource_size(mem_res)); + + clk_disable(pcm->cclk); + clk_disable(pcm->pclk); + clk_put(pcm->pclk); + clk_put(pcm->cclk); + + return 0; +} + +static struct platform_driver s3c_pcm_driver = { + .probe = s3c_pcm_dev_probe, + .remove = s3c_pcm_dev_remove, + .driver = { + .name = "samsung-pcm", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c_pcm_init(void) +{ + return platform_driver_register(&s3c_pcm_driver); +} +module_init(s3c_pcm_init); + +static void __exit s3c_pcm_exit(void) +{ + platform_driver_unregister(&s3c_pcm_driver); +} +module_exit(s3c_pcm_exit); + +/* Module information */ +MODULE_AUTHOR("Jaswinder Singh, <jassi.brar@samsung.com>"); +MODULE_DESCRIPTION("S3C PCM Controller Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-pcm.h b/sound/soc/s3c24xx/s3c-pcm.h new file mode 100644 index 0000000..69ff997 --- /dev/null +++ b/sound/soc/s3c24xx/s3c-pcm.h @@ -0,0 +1,123 @@ +/* sound/soc/s3c24xx/s3c-pcm.h + * + * 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 __S3C_PCM_H +#define __S3C_PCM_H __FILE__ + +/*Register Offsets */ +#define S3C_PCM_CTL (0x00) +#define S3C_PCM_CLKCTL (0x04) +#define S3C_PCM_TXFIFO (0x08) +#define S3C_PCM_RXFIFO (0x0C) +#define S3C_PCM_IRQCTL (0x10) +#define S3C_PCM_IRQSTAT (0x14) +#define S3C_PCM_FIFOSTAT (0x18) +#define S3C_PCM_CLRINT (0x20) + +/* PCM_CTL Bit-Fields */ +#define S3C_PCM_CTL_TXDIPSTICK_MASK (0x3f) +#define S3C_PCM_CTL_TXDIPSTICK_SHIFT (13) +#define S3C_PCM_CTL_RXDIPSTICK_MSK (0x3f<<7) +#define S3C_PCM_CTL_TXDMA_EN (0x1<<6) +#define S3C_PCM_CTL_RXDMA_EN (0x1<<5) +#define S3C_PCM_CTL_TXMSB_AFTER_FSYNC (0x1<<4) +#define S3C_PCM_CTL_RXMSB_AFTER_FSYNC (0x1<<3) +#define S3C_PCM_CTL_TXFIFO_EN (0x1<<2) +#define S3C_PCM_CTL_RXFIFO_EN (0x1<<1) +#define S3C_PCM_CTL_ENABLE (0x1<<0) + +/* PCM_CLKCTL Bit-Fields */ +#define S3C_PCM_CLKCTL_SERCLK_EN (0x1<<19) +#define S3C_PCM_CLKCTL_SERCLKSEL_PCLK (0x1<<18) +#define S3C_PCM_CLKCTL_SCLKDIV_MASK (0x1ff) +#define S3C_PCM_CLKCTL_SYNCDIV_MASK (0x1ff) +#define S3C_PCM_CLKCTL_SCLKDIV_SHIFT (9) +#define S3C_PCM_CLKCTL_SYNCDIV_SHIFT (0) + +/* PCM_TXFIFO Bit-Fields */ +#define S3C_PCM_TXFIFO_DVALID (0x1<<16) +#define S3C_PCM_TXFIFO_DATA_MSK (0xffff<<0) + +/* PCM_RXFIFO Bit-Fields */ +#define S3C_PCM_RXFIFO_DVALID (0x1<<16) +#define S3C_PCM_RXFIFO_DATA_MSK (0xffff<<0) + +/* PCM_IRQCTL Bit-Fields */ +#define S3C_PCM_IRQCTL_IRQEN (0x1<<14) +#define S3C_PCM_IRQCTL_WRDEN (0x1<<12) +#define S3C_PCM_IRQCTL_TXEMPTYEN (0x1<<11) +#define S3C_PCM_IRQCTL_TXALMSTEMPTYEN (0x1<<10) +#define S3C_PCM_IRQCTL_TXFULLEN (0x1<<9) +#define S3C_PCM_IRQCTL_TXALMSTFULLEN (0x1<<8) +#define S3C_PCM_IRQCTL_TXSTARVEN (0x1<<7) +#define S3C_PCM_IRQCTL_TXERROVRFLEN (0x1<<6) +#define S3C_PCM_IRQCTL_RXEMPTEN (0x1<<5) +#define S3C_PCM_IRQCTL_RXALMSTEMPTEN (0x1<<4) +#define S3C_PCM_IRQCTL_RXFULLEN (0x1<<3) +#define S3C_PCM_IRQCTL_RXALMSTFULLEN (0x1<<2) +#define S3C_PCM_IRQCTL_RXSTARVEN (0x1<<1) +#define S3C_PCM_IRQCTL_RXERROVRFLEN (0x1<<0) + +/* PCM_IRQSTAT Bit-Fields */ +#define S3C_PCM_IRQSTAT_IRQPND (0x1<<13) +#define S3C_PCM_IRQSTAT_WRD_XFER (0x1<<12) +#define S3C_PCM_IRQSTAT_TXEMPTY (0x1<<11) +#define S3C_PCM_IRQSTAT_TXALMSTEMPTY (0x1<<10) +#define S3C_PCM_IRQSTAT_TXFULL (0x1<<9) +#define S3C_PCM_IRQSTAT_TXALMSTFULL (0x1<<8) +#define S3C_PCM_IRQSTAT_TXSTARV (0x1<<7) +#define S3C_PCM_IRQSTAT_TXERROVRFL (0x1<<6) +#define S3C_PCM_IRQSTAT_RXEMPT (0x1<<5) +#define S3C_PCM_IRQSTAT_RXALMSTEMPT (0x1<<4) +#define S3C_PCM_IRQSTAT_RXFULL (0x1<<3) +#define S3C_PCM_IRQSTAT_RXALMSTFULL (0x1<<2) +#define S3C_PCM_IRQSTAT_RXSTARV (0x1<<1) +#define S3C_PCM_IRQSTAT_RXERROVRFL (0x1<<0) + +/* PCM_FIFOSTAT Bit-Fields */ +#define S3C_PCM_FIFOSTAT_TXCNT_MSK (0x3f<<14) +#define S3C_PCM_FIFOSTAT_TXFIFOEMPTY (0x1<<13) +#define S3C_PCM_FIFOSTAT_TXFIFOALMSTEMPTY (0x1<<12) +#define S3C_PCM_FIFOSTAT_TXFIFOFULL (0x1<<11) +#define S3C_PCM_FIFOSTAT_TXFIFOALMSTFULL (0x1<<10) +#define S3C_PCM_FIFOSTAT_RXCNT_MSK (0x3f<<4) +#define S3C_PCM_FIFOSTAT_RXFIFOEMPTY (0x1<<3) +#define S3C_PCM_FIFOSTAT_RXFIFOALMSTEMPTY (0x1<<2) +#define S3C_PCM_FIFOSTAT_RXFIFOFULL (0x1<<1) +#define S3C_PCM_FIFOSTAT_RXFIFOALMSTFULL (0x1<<0) + +#define S3C_PCM_CLKSRC_PCLK 0 +#define S3C_PCM_CLKSRC_MUX 1 + +#define S3C_PCM_SCLK_PER_FS 0 + +/** + * struct s3c_pcm_info - S3C PCM Controller information + * @dev: The parent device passed to use from the probe. + * @regs: The pointer to the device register block. + * @dma_playback: DMA information for playback channel. + * @dma_capture: DMA information for capture channel. + */ +struct s3c_pcm_info { + spinlock_t lock; + struct device *dev; + void __iomem *regs; + + unsigned int sclk_per_fs; + + /* Whether to keep PCMSCLK enabled even when idle(no active xfer) */ + unsigned int idleclk; + + struct clk *pclk; + struct clk *cclk; + + struct s3c_dma_params *dma_playback; + struct s3c_dma_params *dma_capture; +}; + +#endif /* __S3C_PCM_H */ diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index a587ec4..359e593 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -34,11 +34,10 @@ #include <plat/regs-s3c2412-iis.h> -#include <plat/audio.h> #include <mach/regs-gpio.h> #include <mach/dma.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c2412-i2s.h" #define S3C2412_I2S_DEBUG 0 @@ -51,14 +50,14 @@ static struct s3c2410_dma_client s3c2412_dma_client_in = { .name = "I2S PCM Stereo in" }; -static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_out = { +static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = { .client = &s3c2412_dma_client_out, .channel = DMACH_I2S_OUT, .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, .dma_size = 4, }; -static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_in = { +static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = { .client = &s3c2412_dma_client_in, .channel = DMACH_I2S_IN, .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index fc1beb0..0191e3a 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -32,11 +32,10 @@ #include <plat/regs-ac97.h> #include <mach/regs-gpio.h> #include <mach/regs-clock.h> -#include <plat/audio.h> #include <asm/dma.h> #include <mach/dma.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-ac97.h" struct s3c24xx_ac97_info { @@ -189,21 +188,21 @@ static struct s3c2410_dma_client s3c2443_dma_client_micin = { .name = "AC97 Mic Mono in" }; -static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_out = { +static struct s3c_dma_params s3c2443_ac97_pcm_stereo_out = { .client = &s3c2443_dma_client_out, .channel = DMACH_PCM_OUT, .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA, .dma_size = 4, }; -static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_in = { +static struct s3c_dma_params s3c2443_ac97_pcm_stereo_in = { .client = &s3c2443_dma_client_in, .channel = DMACH_PCM_IN, .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA, .dma_size = 4, }; -static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = { +static struct s3c_dma_params s3c2443_ac97_mic_mono_in = { .client = &s3c2443_dma_client_micin, .channel = DMACH_MIC_IN, .dma_addr = S3C2440_PA_AC97 + S3C_AC97_MIC_DATA, @@ -291,7 +290,7 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd, { u32 ac_glbctrl; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); @@ -340,7 +339,7 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, { u32 ac_glbctrl; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 40e2c47..0bc5950 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -32,13 +32,13 @@ #include <mach/hardware.h> #include <mach/regs-gpio.h> #include <mach/regs-clock.h> -#include <plat/audio.h> + #include <asm/dma.h> #include <mach/dma.h> #include <plat/regs-iis.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" static struct s3c2410_dma_client s3c24xx_dma_client_out = { @@ -49,14 +49,14 @@ static struct s3c2410_dma_client s3c24xx_dma_client_in = { .name = "I2S PCM Stereo in" }; -static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_out = { +static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = { .client = &s3c24xx_dma_client_out, .channel = DMACH_I2S_OUT, .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, .dma_size = 2, }; -static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_in = { +static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = { .client = &s3c24xx_dma_client_in, .channel = DMACH_I2S_IN, .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, @@ -258,12 +258,12 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: iismod &= ~S3C2410_IISMOD_16BIT; - ((struct s3c24xx_pcm_dma_params *) + ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->dma_size = 1; break; case SNDRV_PCM_FORMAT_S16_LE: iismod |= S3C2410_IISMOD_16BIT; - ((struct s3c24xx_pcm_dma_params *) + ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->dma_size = 2; break; default: @@ -280,7 +280,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; pr_debug("Entered %s\n", __func__); diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c index 1966e0d..d441c3b 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec.c @@ -21,7 +21,7 @@ #include <plat/audio-simtec.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" #include "s3c24xx_simtec.h" @@ -270,7 +270,7 @@ static int attach_gpio_amp(struct device *dev, gpio_direction_output(pd->amp_gain[1], 0); } - /* note, curently we assume GPA0 isn't valid amp */ + /* note, currently we assume GPA0 isn't valid amp */ if (pdata->amp_gpio > 0) { ret = gpio_request(pd->amp_gpio, "gpio-amp"); if (ret) { diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c index 8346bd9..bdf8951 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c @@ -18,7 +18,7 @@ #include <plat/audio-simtec.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" #include "s3c24xx_simtec.h" diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c index 25797e0..185c0ac 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c @@ -18,7 +18,7 @@ #include <plat/audio-simtec.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" #include "s3c24xx_simtec.h" diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c index c215d32..052d596 100644 --- a/sound/soc/s3c24xx/s3c24xx_uda134x.c +++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c @@ -24,7 +24,7 @@ #include <plat/regs-iis.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" #include "../codecs/uda134x.h" diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 3c06c40..cc7edb5 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -31,12 +31,11 @@ #include <plat/gpio-bank-d.h> #include <plat/gpio-bank-e.h> #include <plat/gpio-cfg.h> -#include <plat/audio.h> #include <mach/map.h> #include <mach/dma.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c64xx-i2s.h" static struct s3c2410_dma_client s3c64xx_dma_client_out = { @@ -47,7 +46,7 @@ static struct s3c2410_dma_client s3c64xx_dma_client_in = { .name = "I2S PCM Stereo in" }; -static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { [0] = { .channel = DMACH_I2S0_OUT, .client = &s3c64xx_dma_client_out, @@ -62,7 +61,7 @@ static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { }, }; -static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_in[2] = { +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_in[2] = { [0] = { .channel = DMACH_I2S0_IN, .client = &s3c64xx_dma_client_in, @@ -99,6 +98,19 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, iismod |= S3C64XX_IISMOD_IMS_SYSMUX; break; + case S3C64XX_CLKSRC_CDCLK: + switch (dir) { + case SND_SOC_CLOCK_IN: + iismod |= S3C64XX_IISMOD_CDCLKCON; + break; + case SND_SOC_CLOCK_OUT: + iismod &= ~S3C64XX_IISMOD_CDCLKCON; + break; + default: + return -EINVAL; + } + break; + default: return -EINVAL; } @@ -111,8 +123,12 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai) { struct s3c_i2sv2_info *i2s = to_info(dai); + u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - return i2s->iis_cclk; + if (iismod & S3C64XX_IISMOD_IMS_SYSMUX) + return i2s->iis_cclk; + else + return i2s->iis_pclk; } EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clock); @@ -220,6 +236,8 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) goto err; } + clk_enable(i2s->iis_cclk); + ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); if (ret) goto err_clk; diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h index 02148ce..abe7253 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.h +++ b/sound/soc/s3c24xx/s3c64xx-i2s.h @@ -25,6 +25,7 @@ struct clk; #define S3C64XX_CLKSRC_PCLK (0) #define S3C64XX_CLKSRC_MUX (1) +#define S3C64XX_CLKSRC_CDCLK (2) extern struct snd_soc_dai s3c64xx_i2s_dai[]; diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c index a2a4f53..12b783b1 100644 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -20,7 +20,7 @@ #include <sound/soc-dapm.h> #include "../codecs/ac97.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-ac97.h" static struct snd_soc_card smdk2443; diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c new file mode 100644 index 0000000..efe4901 --- /dev/null +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -0,0 +1,268 @@ +/* + * smdk64xx_wm8580.c + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar@samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/platform_device.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm8580.h" +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +#define S3C64XX_I2S_V4 2 + +/* SMDK64XX has a 12MHZ crystal attached to WM8580 */ +#define SMDK64XX_WM8580_FREQ 12000000 + +static int smdk64xx_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; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int pll_out; + int bfs, rfs, ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + bfs = 16; + break; + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + bfs = 32; + break; + default: + return -EINVAL; + } + + /* The Fvco for WM8580 PLLs must fall within [90,100]MHz. + * This criterion can't be met if we request PLL output + * as {8000x256, 64000x256, 11025x256}Hz. + * As a wayout, we rather change rfs to a minimum value that + * results in (params_rate(params) * rfs), and itself, acceptable + * to both - the CODEC and the CPU. + */ + switch (params_rate(params)) { + case 16000: + case 22050: + case 32000: + case 44100: + case 48000: + case 88200: + case 96000: + rfs = 256; + break; + case 64000: + rfs = 384; + break; + case 8000: + case 11025: + rfs = 512; + break; + default: + return -EINVAL; + } + pll_out = params_rate(params) * rfs; + + /* Set the 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 the AP 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; + + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_CDCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* We use PCLK for basic ops in SoC-Slave mode */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set WM8580 to drive MCLK from its PLLA */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, + WM8580_CLKSRC_PLLA); + if (ret < 0) + return ret; + + /* Explicitly set WM8580-DAC to source from MCLK */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL, + WM8580_CLKSRC_MCLK); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, + SMDK64XX_WM8580_FREQ, pll_out); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs); + if (ret < 0) + return ret; + + return 0; +} + +/* + * SMDK64XX WM8580 DAI operations. + */ +static struct snd_soc_ops smdk64xx_ops = { + .hw_params = smdk64xx_hw_params, +}; + +/* SMDK64xx Playback widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = { + SND_SOC_DAPM_HP("Front-L/R", NULL), + SND_SOC_DAPM_HP("Center/Sub", NULL), + SND_SOC_DAPM_HP("Rear-L/R", NULL), +}; + +/* SMDK64xx Capture widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = { + SND_SOC_DAPM_MIC("MicIn", NULL), + SND_SOC_DAPM_LINE("LineIn", NULL), +}; + +/* SMDK-PAIFTX connections */ +static const struct snd_soc_dapm_route audio_map_tx[] = { + /* MicIn feeds AINL */ + {"AINL", NULL, "MicIn"}, + + /* LineIn feeds AINL/R */ + {"AINL", NULL, "LineIn"}, + {"AINR", NULL, "LineIn"}, +}; + +/* SMDK-PAIFRX connections */ +static const struct snd_soc_dapm_route audio_map_rx[] = { + /* Front Left/Right are fed VOUT1L/R */ + {"Front-L/R", NULL, "VOUT1L"}, + {"Front-L/R", NULL, "VOUT1R"}, + + /* Center/Sub are fed VOUT2L/R */ + {"Center/Sub", NULL, "VOUT2L"}, + {"Center/Sub", NULL, "VOUT2R"}, + + /* Rear Left/Right are fed VOUT3L/R */ + {"Rear-L/R", NULL, "VOUT3L"}, + {"Rear-L/R", NULL, "VOUT3R"}, +}; + +static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Capture widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt, + ARRAY_SIZE(wm8580_dapm_widgets_cpt)); + + /* Set up PAIFTX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx)); + + /* Enabling the microphone requires the fitting of a 0R + * resistor to connect the line from the microphone jack. + */ + snd_soc_dapm_disable_pin(codec, "MicIn"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Playback widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk, + ARRAY_SIZE(wm8580_dapm_widgets_pbk)); + + /* Set up PAIFRX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx)); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link smdk64xx_dai[] = { +{ /* Primary Playback i/f */ + .name = "WM8580 PAIF RX", + .stream_name = "Playback", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], + .init = smdk64xx_wm8580_init_paifrx, + .ops = &smdk64xx_ops, +}, +{ /* Primary Capture i/f */ + .name = "WM8580 PAIF TX", + .stream_name = "Capture", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], + .init = smdk64xx_wm8580_init_paiftx, + .ops = &smdk64xx_ops, +}, +}; + +static struct snd_soc_card smdk64xx = { + .name = "smdk64xx", + .platform = &s3c24xx_soc_platform, + .dai_link = smdk64xx_dai, + .num_links = ARRAY_SIZE(smdk64xx_dai), +}; + +static struct snd_soc_device smdk64xx_snd_devdata = { + .card = &smdk64xx, + .codec_dev = &soc_codec_dev_wm8580, +}; + +static struct platform_device *smdk64xx_snd_device; + +static int __init smdk64xx_audio_init(void) +{ + int ret; + + smdk64xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!smdk64xx_snd_device) + return -ENOMEM; + + platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata); + smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev; + ret = platform_device_add(smdk64xx_snd_device); + + if (ret) + platform_device_put(smdk64xx_snd_device); + + return ret; +} +module_init(smdk64xx_audio_init); + +MODULE_AUTHOR("Jaswinder Singh, jassi.brar@samsung.com"); +MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 83b8028..1d61109 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -196,7 +196,7 @@ static int s6000_pcm_start(struct snd_pcm_substream *substream) 0 /* destination skip after chunk (impossible) */, 4 /* 16 byte burst size */, -1 /* don't conserve bandwidth */, - 0 /* low watermark irq descriptor theshold */, + 0 /* low watermark irq descriptor threshold */, 0 /* disable hardware timestamps */, 1 /* enable channel */); @@ -423,7 +423,7 @@ static void s6000_pcm_free(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -static u64 s6000_pcm_dmamask = DMA_32BIT_MASK; +static u64 s6000_pcm_dmamask = DMA_BIT_MASK(32); static int s6000_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) @@ -435,7 +435,7 @@ static int s6000_pcm_new(struct snd_card *card, if (!card->dev->dma_mask) card->dev->dma_mask = &s6000_pcm_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_32BIT_MASK; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (params->dma_in) { s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_in), diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 9154b43..9e69765 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -23,7 +23,6 @@ config SND_SOC_SH4_SSI config SND_SOC_SH4_FSI tristate "SH4 FSI support" depends on CPU_SUBTYPE_SH7724 - select SH_DMA help This option enables FSI sound support diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 4412324..9c49c11 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -17,7 +17,7 @@ #include <linux/platform_device.h> #include <linux/delay.h> #include <linux/list.h> -#include <linux/clk.h> +#include <linux/pm_runtime.h> #include <linux/io.h> #include <sound/core.h> #include <sound/pcm.h> @@ -26,8 +26,6 @@ #include <sound/pcm_params.h> #include <sound/sh_fsi.h> #include <asm/atomic.h> -#include <asm/dma.h> -#include <asm/dma-sh.h> #define DO_FMT 0x0000 #define DOFF_CTL 0x0004 @@ -97,7 +95,6 @@ struct fsi_priv { int fifo_max; int chan; - int dma_chan; int byte_offset; int period_len; @@ -108,7 +105,6 @@ struct fsi_priv { struct fsi_master { void __iomem *base; int irq; - struct clk *clk; struct fsi_priv fsia; struct fsi_priv fsib; struct sh_fsi_platform_info *info; @@ -308,62 +304,6 @@ static int fsi_get_fifo_residue(struct fsi_priv *fsi, int is_play) return residue; } -static int fsi_get_residue(struct fsi_priv *fsi, int is_play) -{ - int residue; - int width; - struct snd_pcm_runtime *runtime; - - runtime = fsi->substream->runtime; - - /* get 1 channel data width */ - width = frames_to_bytes(runtime, 1) / fsi->chan; - - if (2 == width) - residue = fsi_get_fifo_residue(fsi, is_play); - else - residue = get_dma_residue(fsi->dma_chan); - - return residue; -} - -/************************************************************************ - - - basic dma function - - -************************************************************************/ -#define PORTA_DMA 0 -#define PORTB_DMA 1 - -static int fsi_get_dma_chan(void) -{ - if (0 != request_dma(PORTA_DMA, "fsia")) - return -EIO; - - if (0 != request_dma(PORTB_DMA, "fsib")) { - free_dma(PORTA_DMA); - return -EIO; - } - - master->fsia.dma_chan = PORTA_DMA; - master->fsib.dma_chan = PORTB_DMA; - - return 0; -} - -static void fsi_free_dma_chan(void) -{ - dma_wait_for_completion(PORTA_DMA); - dma_wait_for_completion(PORTB_DMA); - free_dma(PORTA_DMA); - free_dma(PORTB_DMA); - - master->fsia.dma_chan = -1; - master->fsib.dma_chan = -1; -} - /************************************************************************ @@ -435,44 +375,6 @@ static void fsi_soft_all_reset(void) mdelay(10); } -static void fsi_16data_push(struct fsi_priv *fsi, - struct snd_pcm_runtime *runtime, - int send) -{ - u16 *dma_start; - u32 snd; - int i; - - /* get dma start position for FSI */ - dma_start = (u16 *)runtime->dma_area; - dma_start += fsi->byte_offset / 2; - - /* - * soft dma - * FSI can not use DMA when 16bpp - */ - for (i = 0; i < send; i++) { - snd = (u32)dma_start[i]; - fsi_reg_write(fsi, DODT, snd << 8); - } -} - -static void fsi_32data_push(struct fsi_priv *fsi, - struct snd_pcm_runtime *runtime, - int send) -{ - u32 *dma_start; - - /* get dma start position for FSI */ - dma_start = (u32 *)runtime->dma_area; - dma_start += fsi->byte_offset / 4; - - dma_wait_for_completion(fsi->dma_chan); - dma_configure_channel(fsi->dma_chan, (SM_INC|0x400|TS_32|TM_BUR)); - dma_write(fsi->dma_chan, (u32)dma_start, - (u32)(fsi->base + DODT), send * 4); -} - /* playback interrupt */ static int fsi_data_push(struct fsi_priv *fsi) { @@ -481,6 +383,8 @@ static int fsi_data_push(struct fsi_priv *fsi) int send; int fifo_free; int width; + u8 *start; + int i; if (!fsi || !fsi->substream || @@ -515,12 +419,22 @@ static int fsi_data_push(struct fsi_priv *fsi) if (fifo_free < send) send = fifo_free; - if (2 == width) - fsi_16data_push(fsi, runtime, send); - else if (4 == width) - fsi_32data_push(fsi, runtime, send); - else + start = runtime->dma_area; + start += fsi->byte_offset; + + switch (width) { + case 2: + for (i = 0; i < send; i++) + fsi_reg_write(fsi, DODT, + ((u32)*((u16 *)start + i) << 8)); + break; + case 4: + for (i = 0; i < send; i++) + fsi_reg_write(fsi, DODT, *((u32 *)start + i)); + break; + default: return -EINVAL; + } fsi->byte_offset += send * width; @@ -532,6 +446,75 @@ static int fsi_data_push(struct fsi_priv *fsi) return 0; } +static int fsi_data_pop(struct fsi_priv *fsi) +{ + struct snd_pcm_runtime *runtime; + struct snd_pcm_substream *substream = NULL; + int free; + int fifo_fill; + int width; + u8 *start; + int i; + + if (!fsi || + !fsi->substream || + !fsi->substream->runtime) + return -EINVAL; + + runtime = fsi->substream->runtime; + + /* FSI FIFO has limit. + * So, this driver can not send periods data at a time + */ + if (fsi->byte_offset >= + fsi->period_len * (fsi->periods + 1)) { + + substream = fsi->substream; + fsi->periods = (fsi->periods + 1) % runtime->periods; + + if (0 == fsi->periods) + fsi->byte_offset = 0; + } + + /* get 1 channel data width */ + width = frames_to_bytes(runtime, 1) / fsi->chan; + + /* get free space for alsa */ + free = (fsi->buffer_len - fsi->byte_offset) / width; + + /* get recv size */ + fifo_fill = fsi_get_fifo_residue(fsi, 0); + + if (free < fifo_fill) + fifo_fill = free; + + start = runtime->dma_area; + start += fsi->byte_offset; + + switch (width) { + case 2: + for (i = 0; i < fifo_fill; i++) + *((u16 *)start + i) = + (u16)(fsi_reg_read(fsi, DIDT) >> 8); + break; + case 4: + for (i = 0; i < fifo_fill; i++) + *((u32 *)start + i) = fsi_reg_read(fsi, DIDT); + break; + default: + return -EINVAL; + } + + fsi->byte_offset += fifo_fill * width; + + fsi_irq_enable(fsi, 0); + + if (substream) + snd_pcm_period_elapsed(substream); + + return 0; +} + static irqreturn_t fsi_interrupt(int irq, void *data) { u32 status = fsi_master_read(SOFT_RST) & ~0x00000010; @@ -545,6 +528,10 @@ static irqreturn_t fsi_interrupt(int irq, void *data) fsi_data_push(&master->fsia); if (int_st & INT_B_OUT) fsi_data_push(&master->fsib); + if (int_st & INT_A_IN) + fsi_data_pop(&master->fsia); + if (int_st & INT_B_IN) + fsi_data_pop(&master->fsib); fsi_master_write(INT_ST, 0x0000000); @@ -571,7 +558,7 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, int is_master; int ret = 0; - clk_enable(master->clk); + pm_runtime_get_sync(dai->dev); /* CKG1 */ data = is_play ? (1 << 0) : (1 << 4); @@ -664,8 +651,6 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, } fsi_reg_write(fsi, reg, data); - dev_dbg(dai->dev, "use %s format (%d channel) use %d DMAC\n", - msg, fsi->chan, fsi->dma_chan); /* * clear clk reset if master mode @@ -688,7 +673,7 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream, fsi_irq_disable(fsi, is_play); fsi_clk_ctrl(fsi, 0); - clk_disable(master->clk); + pm_runtime_put_sync(dai->dev); } static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -699,16 +684,12 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int ret = 0; - /* capture not supported */ - if (!is_play) - return -ENODEV; - switch (cmd) { case SNDRV_PCM_TRIGGER_START: fsi_stream_push(fsi, substream, frames_to_bytes(runtime, runtime->buffer_size), frames_to_bytes(runtime, runtime->period_size)); - ret = fsi_data_push(fsi); + ret = is_play ? fsi_data_push(fsi) : fsi_data_pop(fsi); break; case SNDRV_PCM_TRIGGER_STOP: fsi_irq_disable(fsi, is_play); @@ -780,10 +761,9 @@ static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsi_priv *fsi = fsi_get(substream); - int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; long location; - location = (fsi->byte_offset - 1) - fsi_get_residue(fsi, is_play); + location = (fsi->byte_offset - 1); if (location < 0) location = 0; @@ -845,7 +825,12 @@ struct snd_soc_dai fsi_soc_dai[] = { .channels_min = 1, .channels_max = 8, }, - /* capture not supported */ + .capture = { + .rates = FSI_RATES, + .formats = FSI_FMTS, + .channels_min = 1, + .channels_max = 8, + }, .ops = &fsi_dai_ops, }, { @@ -857,7 +842,12 @@ struct snd_soc_dai fsi_soc_dai[] = { .channels_min = 1, .channels_max = 8, }, - /* capture not supported */ + .capture = { + .rates = FSI_RATES, + .formats = FSI_FMTS, + .channels_min = 1, + .channels_max = 8, + }, .ops = &fsi_dai_ops, }, }; @@ -881,7 +871,6 @@ EXPORT_SYMBOL_GPL(fsi_soc_platform); static int fsi_probe(struct platform_device *pdev) { struct resource *res; - char clk_name[8]; unsigned int irq; int ret; @@ -912,23 +901,8 @@ static int fsi_probe(struct platform_device *pdev) master->fsia.base = master->base; master->fsib.base = master->base + 0x40; - master->fsia.dma_chan = -1; - master->fsib.dma_chan = -1; - - ret = fsi_get_dma_chan(); - if (ret < 0) { - dev_err(&pdev->dev, "cannot get dma api\n"); - goto exit_iounmap; - } - - /* FSI is based on SPU mstp */ - snprintf(clk_name, sizeof(clk_name), "spu%d", pdev->id); - master->clk = clk_get(NULL, clk_name); - if (IS_ERR(master->clk)) { - dev_err(&pdev->dev, "cannot get %s mstp\n", clk_name); - ret = -EIO; - goto exit_free_dma; - } + pm_runtime_enable(&pdev->dev); + pm_runtime_resume(&pdev->dev); fsi_soc_dai[0].dev = &pdev->dev; fsi_soc_dai[1].dev = &pdev->dev; @@ -938,7 +912,7 @@ static int fsi_probe(struct platform_device *pdev) ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, "fsi", master); if (ret) { dev_err(&pdev->dev, "irq request err\n"); - goto exit_free_dma; + goto exit_iounmap; } ret = snd_soc_register_platform(&fsi_soc_platform); @@ -951,10 +925,9 @@ static int fsi_probe(struct platform_device *pdev) exit_free_irq: free_irq(irq, master); -exit_free_dma: - fsi_free_dma_chan(); exit_iounmap: iounmap(master->base); + pm_runtime_disable(&pdev->dev); exit_kfree: kfree(master); master = NULL; @@ -967,9 +940,7 @@ static int fsi_remove(struct platform_device *pdev) snd_soc_unregister_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); snd_soc_unregister_platform(&fsi_soc_platform); - clk_put(master->clk); - - fsi_free_dma_chan(); + pm_runtime_disable(&pdev->dev); free_irq(master->irq, master); @@ -979,9 +950,27 @@ static int fsi_remove(struct platform_device *pdev) return 0; } +static int fsi_runtime_nop(struct device *dev) +{ + /* Runtime PM callback shared between ->runtime_suspend() + * and ->runtime_resume(). Simply returns success. + * + * This driver re-initializes all registers after + * pm_runtime_get_sync() anyway so there is no need + * to save and restore registers here. + */ + return 0; +} + +static struct dev_pm_ops fsi_pm_ops = { + .runtime_suspend = fsi_runtime_nop, + .runtime_resume = fsi_runtime_nop, +}; + static struct platform_driver fsi_driver = { .driver = { .name = "sh_fsi", + .pm = &fsi_pm_ops, }, .probe = fsi_probe, .remove = fsi_remove, diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index c8ceddc..d2505e8 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -77,6 +77,35 @@ static int snd_soc_7_9_spi_write(void *control_data, const char *data, #define snd_soc_7_9_spi_write NULL #endif +static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 *cache = codec->reg_cache; + u8 data[2]; + + BUG_ON(codec->volatile_register); + + data[0] = reg & 0xff; + data[1] = value & 0xff; + + if (reg < codec->reg_cache_size) + cache[reg] = value; + + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + if (reg >= codec->reg_cache_size) + return -1; + return cache[reg]; +} + static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -150,9 +179,20 @@ static struct { unsigned int (*read)(struct snd_soc_codec *, unsigned int); unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); } io_types[] = { - { 7, 9, snd_soc_7_9_write, snd_soc_7_9_spi_write, snd_soc_7_9_read }, - { 8, 16, snd_soc_8_16_write, NULL, snd_soc_8_16_read, - snd_soc_8_16_read_i2c }, + { + .addr_bits = 7, .data_bits = 9, + .write = snd_soc_7_9_write, .read = snd_soc_7_9_read, + .spi_write = snd_soc_7_9_spi_write + }, + { + .addr_bits = 8, .data_bits = 8, + .write = snd_soc_8_8_write, .read = snd_soc_8_8_read, + }, + { + .addr_bits = 8, .data_bits = 16, + .write = snd_soc_8_16_write, .read = snd_soc_8_16_read, + .i2c_read = snd_soc_8_16_read_i2c, + }, }; /** diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7ff04ad..ef8f282 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -37,7 +37,6 @@ #include <sound/initval.h> static DEFINE_MUTEX(pcm_mutex); -static DEFINE_MUTEX(io_mutex); static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); #ifdef CONFIG_DEBUG_FS @@ -81,6 +80,173 @@ static int run_delayed_work(struct delayed_work *dwork) return ret; } +/* codec register dump */ +static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) +{ + int i, step = 1, count = 0; + + if (!codec->reg_cache_size) + return 0; + + if (codec->reg_cache_step) + step = codec->reg_cache_step; + + count += sprintf(buf, "%s registers\n", codec->name); + for (i = 0; i < codec->reg_cache_size; i += step) { + if (codec->readable_register && !codec->readable_register(i)) + continue; + + count += sprintf(buf + count, "%2x: ", i); + if (count >= PAGE_SIZE - 1) + break; + + if (codec->display_register) + count += codec->display_register(codec, buf + count, + PAGE_SIZE - count, i); + else + count += snprintf(buf + count, PAGE_SIZE - count, + "%4x", codec->read(codec, i)); + + if (count >= PAGE_SIZE - 1) + break; + + count += snprintf(buf + count, PAGE_SIZE - count, "\n"); + if (count >= PAGE_SIZE - 1) + break; + } + + /* Truncate count; min() would cause a warning */ + if (count >= PAGE_SIZE) + count = PAGE_SIZE - 1; + + return count; +} +static ssize_t codec_reg_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_soc_device *devdata = dev_get_drvdata(dev); + return soc_codec_reg_show(devdata->card->codec, buf); +} + +static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); + +#ifdef CONFIG_DEBUG_FS +static int codec_reg_open_file(struct inode *inode, struct file *file) +{ + file->private_data = inode->i_private; + return 0; +} + +static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + ssize_t ret; + struct snd_soc_codec *codec = file->private_data; + char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + ret = soc_codec_reg_show(codec, buf); + if (ret >= 0) + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); + kfree(buf); + return ret; +} + +static ssize_t codec_reg_write_file(struct file *file, + const char __user *user_buf, size_t count, loff_t *ppos) +{ + char buf[32]; + int buf_size; + char *start = buf; + unsigned long reg, value; + int step = 1; + struct snd_soc_codec *codec = file->private_data; + + buf_size = min(count, (sizeof(buf)-1)); + if (copy_from_user(buf, user_buf, buf_size)) + return -EFAULT; + buf[buf_size] = 0; + + if (codec->reg_cache_step) + step = codec->reg_cache_step; + + while (*start == ' ') + start++; + reg = simple_strtoul(start, &start, 16); + if ((reg >= codec->reg_cache_size) || (reg % step)) + return -EINVAL; + while (*start == ' ') + start++; + if (strict_strtoul(start, 16, &value)) + return -EINVAL; + codec->write(codec, reg, value); + return buf_size; +} + +static const struct file_operations codec_reg_fops = { + .open = codec_reg_open_file, + .read = codec_reg_read_file, + .write = codec_reg_write_file, +}; + +static void soc_init_codec_debugfs(struct snd_soc_codec *codec) +{ + char codec_root[128]; + + if (codec->dev) + snprintf(codec_root, sizeof(codec_root), + "%s.%s", codec->name, dev_name(codec->dev)); + else + snprintf(codec_root, sizeof(codec_root), + "%s", codec->name); + + codec->debugfs_codec_root = debugfs_create_dir(codec_root, + debugfs_root); + if (!codec->debugfs_codec_root) { + printk(KERN_WARNING + "ASoC: Failed to create codec debugfs directory\n"); + return; + } + + codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, + codec->debugfs_codec_root, + codec, &codec_reg_fops); + if (!codec->debugfs_reg) + printk(KERN_WARNING + "ASoC: Failed to create codec register debugfs file\n"); + + codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744, + codec->debugfs_codec_root, + &codec->pop_time); + if (!codec->debugfs_pop_time) + printk(KERN_WARNING + "Failed to create pop time debugfs file\n"); + + codec->debugfs_dapm = debugfs_create_dir("dapm", + codec->debugfs_codec_root); + if (!codec->debugfs_dapm) + printk(KERN_WARNING + "Failed to create DAPM debugfs directory\n"); + + snd_soc_dapm_debugfs_init(codec); +} + +static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) +{ + debugfs_remove_recursive(codec->debugfs_codec_root); +} + +#else + +static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec) +{ +} + +static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) +{ +} +#endif + #ifdef CONFIG_SND_SOC_AC97_BUS /* unregister ac97 codec */ static int soc_ac97_dev_unregister(struct snd_soc_codec *codec) @@ -790,56 +956,21 @@ static int soc_resume(struct device *dev) return 0; } - -/** - * snd_soc_suspend_device: Notify core of device suspend - * - * @dev: Device being suspended. - * - * In order to ensure that the entire audio subsystem is suspended in a - * coordinated fashion ASoC devices should suspend themselves when - * called by ASoC. When the standard kernel suspend process asks the - * device to suspend it should call this function to initiate a suspend - * of the entire ASoC card. - * - * \note Currently this function is stubbed out. - */ -int snd_soc_suspend_device(struct device *dev) -{ - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_suspend_device); - -/** - * snd_soc_resume_device: Notify core of device resume - * - * @dev: Device being resumed. - * - * In order to ensure that the entire audio subsystem is resumed in a - * coordinated fashion ASoC devices should resume themselves when called - * by ASoC. When the standard kernel resume process asks the device - * to resume it should call this function. Once all the components of - * the card have notified that they are ready to be resumed the card - * will be resumed. - * - * \note Currently this function is stubbed out. - */ -int snd_soc_resume_device(struct device *dev) -{ - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_resume_device); #else #define soc_suspend NULL #define soc_resume NULL #endif +static struct snd_soc_dai_ops null_dai_ops = { +}; + static void snd_soc_instantiate_card(struct snd_soc_card *card) { struct platform_device *pdev = container_of(card->dev, struct platform_device, dev); struct snd_soc_codec_device *codec_dev = card->socdev->codec_dev; + struct snd_soc_codec *codec; struct snd_soc_platform *platform; struct snd_soc_dai *dai; int i, found, ret, ac97; @@ -877,6 +1008,11 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) ac97 = 1; } + for (i = 0; i < card->num_links; i++) { + if (!card->dai_link[i].codec_dai->ops) + card->dai_link[i].codec_dai->ops = &null_dai_ops; + } + /* If we have AC97 in the system then don't wait for the * codec. This will need revisiting if we have to handle * systems with mixed AC97 and non-AC97 parts. Only check for @@ -923,6 +1059,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) if (ret < 0) goto cpu_dai_err; } + codec = card->codec; if (platform->probe) { ret = platform->probe(pdev); @@ -937,10 +1074,69 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) INIT_WORK(&card->deferred_resume_work, soc_resume_deferred); #endif + for (i = 0; i < card->num_links; i++) { + if (card->dai_link[i].init) { + ret = card->dai_link[i].init(codec); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to init %s\n", + card->dai_link[i].stream_name); + continue; + } + } + if (card->dai_link[i].codec_dai->ac97_control) + ac97 = 1; + } + + snprintf(codec->card->shortname, sizeof(codec->card->shortname), + "%s", card->name); + snprintf(codec->card->longname, sizeof(codec->card->longname), + "%s (%s)", card->name, codec->name); + + /* Make sure all DAPM widgets are instantiated */ + snd_soc_dapm_new_widgets(codec); + + ret = snd_card_register(codec->card); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to register soundcard for %s\n", + codec->name); + goto card_err; + } + + mutex_lock(&codec->mutex); +#ifdef CONFIG_SND_SOC_AC97_BUS + /* Only instantiate AC97 if not already done by the adaptor + * for the generic AC97 subsystem. + */ + if (ac97 && strcmp(codec->name, "AC97") != 0) { + ret = soc_ac97_dev_register(codec); + if (ret < 0) { + printk(KERN_ERR "asoc: AC97 device register failed\n"); + snd_card_free(codec->card); + mutex_unlock(&codec->mutex); + goto card_err; + } + } +#endif + + ret = snd_soc_dapm_sys_add(card->socdev->dev); + if (ret < 0) + printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n"); + + ret = device_create_file(card->socdev->dev, &dev_attr_codec_reg); + if (ret < 0) + printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); + + soc_init_codec_debugfs(codec); + mutex_unlock(&codec->mutex); + card->instantiated = 1; return; +card_err: + if (platform->remove) + platform->remove(pdev); + platform_err: if (codec_dev->remove) codec_dev->remove(pdev); @@ -1143,157 +1339,6 @@ int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg) } EXPORT_SYMBOL_GPL(snd_soc_codec_volatile_register); -/* codec register dump */ -static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) -{ - int i, step = 1, count = 0; - - if (!codec->reg_cache_size) - return 0; - - if (codec->reg_cache_step) - step = codec->reg_cache_step; - - count += sprintf(buf, "%s registers\n", codec->name); - for (i = 0; i < codec->reg_cache_size; i += step) { - if (codec->readable_register && !codec->readable_register(i)) - continue; - - count += sprintf(buf + count, "%2x: ", i); - if (count >= PAGE_SIZE - 1) - break; - - if (codec->display_register) - count += codec->display_register(codec, buf + count, - PAGE_SIZE - count, i); - else - count += snprintf(buf + count, PAGE_SIZE - count, - "%4x", codec->read(codec, i)); - - if (count >= PAGE_SIZE - 1) - break; - - count += snprintf(buf + count, PAGE_SIZE - count, "\n"); - if (count >= PAGE_SIZE - 1) - break; - } - - /* Truncate count; min() would cause a warning */ - if (count >= PAGE_SIZE) - count = PAGE_SIZE - 1; - - return count; -} -static ssize_t codec_reg_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - struct snd_soc_device *devdata = dev_get_drvdata(dev); - return soc_codec_reg_show(devdata->card->codec, buf); -} - -static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); - -#ifdef CONFIG_DEBUG_FS -static int codec_reg_open_file(struct inode *inode, struct file *file) -{ - file->private_data = inode->i_private; - return 0; -} - -static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, - size_t count, loff_t *ppos) -{ - ssize_t ret; - struct snd_soc_codec *codec = file->private_data; - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); - if (!buf) - return -ENOMEM; - ret = soc_codec_reg_show(codec, buf); - if (ret >= 0) - ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); - kfree(buf); - return ret; -} - -static ssize_t codec_reg_write_file(struct file *file, - const char __user *user_buf, size_t count, loff_t *ppos) -{ - char buf[32]; - int buf_size; - char *start = buf; - unsigned long reg, value; - int step = 1; - struct snd_soc_codec *codec = file->private_data; - - buf_size = min(count, (sizeof(buf)-1)); - if (copy_from_user(buf, user_buf, buf_size)) - return -EFAULT; - buf[buf_size] = 0; - - if (codec->reg_cache_step) - step = codec->reg_cache_step; - - while (*start == ' ') - start++; - reg = simple_strtoul(start, &start, 16); - if ((reg >= codec->reg_cache_size) || (reg % step)) - return -EINVAL; - while (*start == ' ') - start++; - if (strict_strtoul(start, 16, &value)) - return -EINVAL; - codec->write(codec, reg, value); - return buf_size; -} - -static const struct file_operations codec_reg_fops = { - .open = codec_reg_open_file, - .read = codec_reg_read_file, - .write = codec_reg_write_file, -}; - -static void soc_init_codec_debugfs(struct snd_soc_codec *codec) -{ - codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, - debugfs_root, codec, - &codec_reg_fops); - if (!codec->debugfs_reg) - printk(KERN_WARNING - "ASoC: Failed to create codec register debugfs file\n"); - - codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744, - debugfs_root, - &codec->pop_time); - if (!codec->debugfs_pop_time) - printk(KERN_WARNING - "Failed to create pop time debugfs file\n"); - - codec->debugfs_dapm = debugfs_create_dir("dapm", debugfs_root); - if (!codec->debugfs_dapm) - printk(KERN_WARNING - "Failed to create DAPM debugfs directory\n"); - - snd_soc_dapm_debugfs_init(codec); -} - -static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) -{ - debugfs_remove_recursive(codec->debugfs_dapm); - debugfs_remove(codec->debugfs_pop_time); - debugfs_remove(codec->debugfs_reg); -} - -#else - -static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec) -{ -} - -static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) -{ -} -#endif - /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec @@ -1361,19 +1406,41 @@ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, int change; unsigned int old, new; - mutex_lock(&io_mutex); old = snd_soc_read(codec, reg); new = (old & ~mask) | value; change = old != new; if (change) snd_soc_write(codec, reg, new); - mutex_unlock(&io_mutex); return change; } EXPORT_SYMBOL_GPL(snd_soc_update_bits); /** + * snd_soc_update_bits_locked - update codec register bits + * @codec: audio codec + * @reg: codec register + * @mask: register mask + * @value: new value + * + * Writes new register value, and takes the codec mutex. + * + * Returns 1 for change else 0. + */ +static int snd_soc_update_bits_locked(struct snd_soc_codec *codec, + unsigned short reg, unsigned int mask, + unsigned int value) +{ + int change; + + mutex_lock(&codec->mutex); + change = snd_soc_update_bits(codec, reg, mask, value); + mutex_unlock(&codec->mutex); + + return change; +} + +/** * snd_soc_test_bits - test register for change * @codec: audio codec * @reg: codec register @@ -1391,11 +1458,9 @@ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, int change; unsigned int old, new; - mutex_lock(&io_mutex); old = snd_soc_read(codec, reg); new = (old & ~mask) | value; change = old != new; - mutex_unlock(&io_mutex); return change; } @@ -1442,89 +1507,16 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) mutex_unlock(&codec->mutex); return ret; } - } - - mutex_unlock(&codec->mutex); - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_new_pcms); - -/** - * snd_soc_init_card - register sound card - * @socdev: the SoC audio device - * - * Register a SoC sound card. Also registers an AC97 device if the - * codec is AC97 for ad hoc devices. - * - * Returns 0 for success, else error. - */ -int snd_soc_init_card(struct snd_soc_device *socdev) -{ - struct snd_soc_card *card = socdev->card; - struct snd_soc_codec *codec = card->codec; - int ret = 0, i, ac97 = 0, err = 0; - - for (i = 0; i < card->num_links; i++) { - if (card->dai_link[i].init) { - err = card->dai_link[i].init(codec); - if (err < 0) { - printk(KERN_ERR "asoc: failed to init %s\n", - card->dai_link[i].stream_name); - continue; - } - } if (card->dai_link[i].codec_dai->ac97_control) { - ac97 = 1; snd_ac97_dev_add_pdata(codec->ac97, card->dai_link[i].cpu_dai->ac97_pdata); } } - snprintf(codec->card->shortname, sizeof(codec->card->shortname), - "%s", card->name); - snprintf(codec->card->longname, sizeof(codec->card->longname), - "%s (%s)", card->name, codec->name); - /* Make sure all DAPM widgets are instantiated */ - snd_soc_dapm_new_widgets(codec); - - ret = snd_card_register(codec->card); - if (ret < 0) { - printk(KERN_ERR "asoc: failed to register soundcard for %s\n", - codec->name); - goto out; - } - - mutex_lock(&codec->mutex); -#ifdef CONFIG_SND_SOC_AC97_BUS - /* Only instantiate AC97 if not already done by the adaptor - * for the generic AC97 subsystem. - */ - if (ac97 && strcmp(codec->name, "AC97") != 0) { - ret = soc_ac97_dev_register(codec); - if (ret < 0) { - printk(KERN_ERR "asoc: AC97 device register failed\n"); - snd_card_free(codec->card); - mutex_unlock(&codec->mutex); - goto out; - } - } -#endif - - err = snd_soc_dapm_sys_add(socdev->dev); - if (err < 0) - printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n"); - - err = device_create_file(socdev->dev, &dev_attr_codec_reg); - if (err < 0) - printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); - - soc_init_codec_debugfs(codec); mutex_unlock(&codec->mutex); - -out: return ret; } -EXPORT_SYMBOL_GPL(snd_soc_init_card); +EXPORT_SYMBOL_GPL(snd_soc_new_pcms); /** * snd_soc_free_pcms - free sound card and pcms @@ -1726,7 +1718,7 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, mask |= (bitmask - 1) << e->shift_r; } - return snd_soc_update_bits(codec, e->reg, mask, val); + return snd_soc_update_bits_locked(codec, e->reg, mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); @@ -1800,7 +1792,7 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, mask |= e->mask << e->shift_r; } - return snd_soc_update_bits(codec, e->reg, mask, val); + return snd_soc_update_bits_locked(codec, e->reg, mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_value_enum_double); @@ -1961,7 +1953,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, val_mask |= mask << rshift; val |= val2 << rshift; } - return snd_soc_update_bits(codec, reg, val_mask, val); + return snd_soc_update_bits_locked(codec, reg, val_mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_volsw); @@ -2067,11 +2059,11 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, val = val << shift; val2 = val2 << shift; - err = snd_soc_update_bits(codec, reg, val_mask, val); + err = snd_soc_update_bits_locked(codec, reg, val_mask, val); if (err < 0) return err; - err = snd_soc_update_bits(codec, reg2, val_mask, val2); + err = snd_soc_update_bits_locked(codec, reg2, val_mask, val2); return err; } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r); @@ -2150,7 +2142,7 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, val = (ucontrol->value.integer.value[0]+min) & 0xff; val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; - return snd_soc_update_bits(codec, reg, 0xffff, val); + return snd_soc_update_bits_locked(codec, reg, 0xffff, val); } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); @@ -2197,16 +2189,18 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); * snd_soc_dai_set_pll - configure DAI PLL. * @dai: DAI * @pll_id: DAI specific PLL ID + * @source: DAI specific source for the PLL * @freq_in: PLL input clock frequency in Hz * @freq_out: requested PLL output clock frequency in Hz * * Configures and enables PLL to generate output clock based on input clock. */ -int snd_soc_dai_set_pll(struct snd_soc_dai *dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) { if (dai->ops && dai->ops->set_pll) - return dai->ops->set_pll(dai, pll_id, freq_in, freq_out); + return dai->ops->set_pll(dai, pll_id, source, + freq_in, freq_out); else return -EINVAL; } @@ -2251,6 +2245,30 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); /** + * snd_soc_dai_set_channel_map - configure DAI audio channel map + * @dai: DAI + * @tx_num: how many TX channels + * @tx_slot: pointer to an array which imply the TX slot number channel + * 0~num-1 uses + * @rx_num: how many RX channels + * @rx_slot: pointer to an array which imply the RX slot number channel + * 0~num-1 uses + * + * configure the relationship between channel number and TDM slot number. + */ +int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + if (dai->ops && dai->ops->set_channel_map) + return dai->ops->set_channel_map(dai, tx_num, tx_slot, + rx_num, rx_slot); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map); + +/** * snd_soc_dai_set_tristate - configure DAI system or master clock. * @dai: DAI * @tristate: tristate enable @@ -2329,9 +2347,6 @@ static int snd_soc_unregister_card(struct snd_soc_card *card) return 0; } -static struct snd_soc_dai_ops null_dai_ops = { -}; - /** * snd_soc_register_dai - Register a DAI with the ASoC core * diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8de6f9d..0d294ef 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -719,6 +719,10 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) /* Check if one of our outputs is connected */ list_for_each_entry(path, &w->sinks, list_source) { + if (path->connected && + !path->connected(path->source, path->sink)) + continue; + if (path->sink && path->sink->power_check && path->sink->power_check(path->sink)) { power = 1; @@ -973,9 +977,19 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) if (!w->power_check) continue; - power = w->power_check(w); - if (power) - sys_power = 1; + /* If we're suspending then pull down all the + * power. */ + switch (event) { + case SND_SOC_DAPM_STREAM_SUSPEND: + power = 0; + break; + + default: + power = w->power_check(w); + if (power) + sys_power = 1; + break; + } if (w->power == power) continue; @@ -999,8 +1013,12 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) case SND_SOC_DAPM_STREAM_RESUME: sys_power = 1; break; + case SND_SOC_DAPM_STREAM_SUSPEND: + sys_power = 0; + break; case SND_SOC_DAPM_STREAM_NOP: sys_power = codec->bias_level != SND_SOC_BIAS_STANDBY; + break; default: break; } @@ -1138,6 +1156,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, w->active ? "active" : "inactive"); list_for_each_entry(p, &w->sources, list_sink) { + if (p->connected && !p->connected(w, p->sink)) + continue; + if (p->connect) ret += snprintf(buf + ret, PAGE_SIZE - ret, " in %s %s\n", @@ -1145,6 +1166,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, p->source->name); } list_for_each_entry(p, &w->sinks, list_source) { + if (p->connected && !p->connected(w, p->sink)) + continue; + if (p->connect) ret += snprintf(buf + ret, PAGE_SIZE - ret, " out %s %s\n", @@ -1192,8 +1216,8 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec) /* test and update the power status of a mux widget */ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kcontrol, int mask, - int mux, int val, struct soc_enum *e) + struct snd_kcontrol *kcontrol, int change, + int mux, struct soc_enum *e) { struct snd_soc_dapm_path *path; int found = 0; @@ -1202,7 +1226,7 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, widget->id != snd_soc_dapm_value_mux) return -ENODEV; - if (!snd_soc_test_bits(widget->codec, e->reg, mask, val)) + if (!change) return 0; /* find dapm widget path assoc with kcontrol */ @@ -1387,10 +1411,13 @@ int snd_soc_dapm_sync(struct snd_soc_codec *codec) EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, - const char *sink, const char *control, const char *source) + const struct snd_soc_dapm_route *route) { struct snd_soc_dapm_path *path; struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w; + const char *sink = route->sink; + const char *control = route->control; + const char *source = route->source; int ret = 0; /* find src and dest widgets */ @@ -1414,6 +1441,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, path->source = wsource; path->sink = wsink; + path->connected = route->connected; INIT_LIST_HEAD(&path->list); INIT_LIST_HEAD(&path->list_source); INIT_LIST_HEAD(&path->list_sink); @@ -1514,8 +1542,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, int i, ret; for (i = 0; i < num; i++) { - ret = snd_soc_dapm_add_route(codec, route->sink, - route->control, route->source); + ret = snd_soc_dapm_add_route(codec, route); if (ret < 0) { printk(KERN_ERR "Failed to add route %s->%s\n", route->source, @@ -1752,7 +1779,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, mux; + unsigned int val, mux, change; unsigned int mask, bitmask; int ret = 0; @@ -1772,20 +1799,21 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mutex_lock(&widget->codec->mutex); widget->value = val; - dapm_mux_update_power(widget, kcontrol, mask, mux, val, e); - if (widget->event) { - if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { - ret = widget->event(widget, - kcontrol, SND_SOC_DAPM_PRE_REG); - if (ret < 0) - goto out; - } - ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); - if (widget->event_flags & SND_SOC_DAPM_POST_REG) - ret = widget->event(widget, - kcontrol, SND_SOC_DAPM_POST_REG); - } else - ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); + change = snd_soc_test_bits(widget->codec, e->reg, mask, val); + dapm_mux_update_power(widget, kcontrol, change, mux, e); + + if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { + ret = widget->event(widget, + kcontrol, SND_SOC_DAPM_PRE_REG); + if (ret < 0) + goto out; + } + + ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); + + if (widget->event_flags & SND_SOC_DAPM_POST_REG) + ret = widget->event(widget, + kcontrol, SND_SOC_DAPM_POST_REG); out: mutex_unlock(&widget->codec->mutex); @@ -1794,6 +1822,54 @@ out: EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); /** + * snd_soc_dapm_get_enum_virt - Get virtual DAPM mux + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Returns 0 for success. + */ +int snd_soc_dapm_get_enum_virt(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = widget->value; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_virt); + +/** + * snd_soc_dapm_put_enum_virt - Set virtual DAPM mux + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Returns 0 for success. + */ +int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = + (struct soc_enum *)kcontrol->private_value; + int change; + int ret = 0; + + if (ucontrol->value.enumerated.item[0] >= e->max) + return -EINVAL; + + mutex_lock(&widget->codec->mutex); + + change = widget->value != ucontrol->value.enumerated.item[0]; + widget->value = ucontrol->value.enumerated.item[0]; + dapm_mux_update_power(widget, kcontrol, change, widget->value, e); + + mutex_unlock(&widget->codec->mutex); + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt); + +/** * snd_soc_dapm_get_value_enum_double - dapm semi enumerated double mixer get * callback * @kcontrol: mixer control @@ -1851,7 +1927,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, mux; + unsigned int val, mux, change; unsigned int mask; int ret = 0; @@ -1869,20 +1945,21 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, mutex_lock(&widget->codec->mutex); widget->value = val; - dapm_mux_update_power(widget, kcontrol, mask, mux, val, e); - if (widget->event) { - if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { - ret = widget->event(widget, - kcontrol, SND_SOC_DAPM_PRE_REG); - if (ret < 0) - goto out; - } - ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); - if (widget->event_flags & SND_SOC_DAPM_POST_REG) - ret = widget->event(widget, - kcontrol, SND_SOC_DAPM_POST_REG); - } else - ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); + change = snd_soc_test_bits(widget->codec, e->reg, mask, val); + dapm_mux_update_power(widget, kcontrol, change, mux, e); + + if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { + ret = widget->event(widget, + kcontrol, SND_SOC_DAPM_PRE_REG); + if (ret < 0) + goto out; + } + + ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); + + if (widget->event_flags & SND_SOC_DAPM_POST_REG) + ret = widget->event(widget, + kcontrol, SND_SOC_DAPM_POST_REG); out: mutex_unlock(&widget->codec->mutex); @@ -2072,9 +2149,9 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, } } } - mutex_unlock(&codec->mutex); dapm_power_widgets(codec, event); + mutex_unlock(&codec->mutex); dump_dapm(codec, __func__); return 0; } diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 1d455ab7..3c07a94 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -58,7 +58,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_new); */ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) { - struct snd_soc_codec *codec = jack->card->codec; + struct snd_soc_codec *codec; struct snd_soc_jack_pin *pin; int enable; int oldstatus; @@ -67,6 +67,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) WARN_ON_ONCE(!jack); return; } + codec = jack->card->codec; mutex_lock(&codec->mutex); @@ -162,6 +163,9 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) else report = 0; + if (gpio->jack_status_check) + report = gpio->jack_status_check(); + snd_soc_jack_report(jack, report, gpio->report); } diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c new file mode 100644 index 0000000..1d07b93 --- /dev/null +++ b/sound/soc/soc-utils.c @@ -0,0 +1,74 @@ +/* + * soc-util.c -- ALSA SoC Audio Layer utility functions + * + * Copyright 2009 Wolfson Microelectronics PLC. + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * Liam Girdwood <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 + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots) +{ + return sample_size * channels * tdm_slots; +} +EXPORT_SYMBOL_GPL(snd_soc_calc_frame_size); + +int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params) +{ + int sample_size; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + sample_size = 16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + case SNDRV_PCM_FORMAT_S20_3BE: + sample_size = 20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_BE: + sample_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + case SNDRV_PCM_FORMAT_S32_BE: + sample_size = 32; + break; + default: + return -ENOTSUPP; + } + + return snd_soc_calc_frame_size(sample_size, params_channels(params), + 1); +} +EXPORT_SYMBOL_GPL(snd_soc_params_to_frame_size); + +int snd_soc_calc_bclk(int fs, int sample_size, int channels, int tdm_slots) +{ + return fs * snd_soc_calc_frame_size(sample_size, channels, tdm_slots); +} +EXPORT_SYMBOL_GPL(snd_soc_calc_bclk); + +int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params) +{ + int ret; + + ret = snd_soc_params_to_frame_size(params); + + if (ret > 0) + return ret * params_rate(params); + else + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk); diff --git a/sound/sound_core.c b/sound/sound_core.c index 49c9981..dbca7c9 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -353,7 +353,7 @@ static struct sound_unit *chains[SOUND_STEP]; * @dev: device pointer * * Allocate a special sound device by minor number from the sound - * subsystem. The allocated number is returned on succes. On failure + * subsystem. The allocated number is returned on success. On failure * a negative error code is returned. */ diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 63c8f45..67c9123 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -374,7 +374,7 @@ sf_zone_new(struct snd_sf_list *sflist, struct snd_soundfont *sf) /* - * increment sample couter + * increment sample counter */ static void set_sample_counter(struct snd_sf_list *sflist, struct snd_soundfont *sf, diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 8db0374..b074a59 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2893,7 +2893,9 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && altsd->bInterfaceSubClass == USB_SUBCLASS_MIDI_STREAMING) { - if (snd_usb_create_midi_interface(chip, iface, NULL) < 0) { + int err = snd_usbmidi_create(chip->card, iface, + &chip->midi_list, NULL); + if (err < 0) { snd_printk(KERN_ERR "%d:%u:%d: cannot create sequencer device\n", dev->devnum, ctrlif, j); continue; } @@ -3038,12 +3040,11 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = &uaxx_ep }; - if (chip->usb_id == USB_ID(0x0582, 0x002b)) - return snd_usb_create_midi_interface(chip, iface, - &ua700_quirk); - else - return snd_usb_create_midi_interface(chip, iface, - &uaxx_quirk); + const struct snd_usb_audio_quirk *quirk = + chip->usb_id == USB_ID(0x0582, 0x002b) + ? &ua700_quirk : &uaxx_quirk; + return snd_usbmidi_create(chip->card, iface, + &chip->midi_list, quirk); } if (altsd->bNumEndpoints != 1) @@ -3370,6 +3371,13 @@ static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip, return 0; /* keep this altsetting */ } +static int create_any_midi_quirk(struct snd_usb_audio *chip, + struct usb_interface *intf, + const struct snd_usb_audio_quirk *quirk) +{ + return snd_usbmidi_create(chip->card, intf, &chip->midi_list, quirk); +} + /* * audio-interface quirks * @@ -3387,14 +3395,14 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, static const quirk_func_t quirk_funcs[] = { [QUIRK_IGNORE_INTERFACE] = ignore_interface_quirk, [QUIRK_COMPOSITE] = create_composite_quirk, - [QUIRK_MIDI_STANDARD_INTERFACE] = snd_usb_create_midi_interface, - [QUIRK_MIDI_FIXED_ENDPOINT] = snd_usb_create_midi_interface, - [QUIRK_MIDI_YAMAHA] = snd_usb_create_midi_interface, - [QUIRK_MIDI_MIDIMAN] = snd_usb_create_midi_interface, - [QUIRK_MIDI_NOVATION] = snd_usb_create_midi_interface, - [QUIRK_MIDI_FASTLANE] = snd_usb_create_midi_interface, - [QUIRK_MIDI_EMAGIC] = snd_usb_create_midi_interface, - [QUIRK_MIDI_CME] = snd_usb_create_midi_interface, + [QUIRK_MIDI_STANDARD_INTERFACE] = create_any_midi_quirk, + [QUIRK_MIDI_FIXED_ENDPOINT] = create_any_midi_quirk, + [QUIRK_MIDI_YAMAHA] = create_any_midi_quirk, + [QUIRK_MIDI_MIDIMAN] = create_any_midi_quirk, + [QUIRK_MIDI_NOVATION] = create_any_midi_quirk, + [QUIRK_MIDI_FASTLANE] = create_any_midi_quirk, + [QUIRK_MIDI_EMAGIC] = create_any_midi_quirk, + [QUIRK_MIDI_CME] = create_any_midi_quirk, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 8e7f789..40ba811 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -132,7 +132,6 @@ struct snd_usb_audio { int pcm_devs; struct list_head midi_list; /* list of midi interfaces */ - int next_midi_device; struct list_head mixer_list; /* list of mixer interfaces */ }; @@ -210,7 +209,7 @@ struct snd_usb_midi_endpoint_info { /* */ -#define combine_word(s) ((*s) | ((unsigned int)(s)[1] << 8)) +#define combine_word(s) ((*(s)) | ((unsigned int)(s)[1] << 8)) #define combine_triple(s) (combine_word(s) | ((unsigned int)(s)[2] << 16)) #define combine_quad(s) (combine_triple(s) | ((unsigned int)(s)[3] << 24)) @@ -227,8 +226,10 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, int ignore_error); void snd_usb_mixer_disconnect(struct list_head *p); -int snd_usb_create_midi_interface(struct snd_usb_audio *chip, struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk); +int snd_usbmidi_create(struct snd_card *card, + struct usb_interface *iface, + struct list_head *midi_list, + const struct snd_usb_audio_quirk *quirk); void snd_usbmidi_input_stop(struct list_head* p); void snd_usbmidi_input_start(struct list_head* p); void snd_usbmidi_disconnect(struct list_head *p); diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 0eff19c..6e89b83 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1,7 +1,7 @@ /* * usbmidi.c - ALSA USB MIDI driver * - * Copyright (c) 2002-2007 Clemens Ladisch + * Copyright (c) 2002-2009 Clemens Ladisch * All rights reserved. * * Based on the OSS usb-midi driver by NAGANO Daisuke, @@ -47,6 +47,7 @@ #include <linux/usb.h> #include <linux/wait.h> #include <sound/core.h> +#include <sound/control.h> #include <sound/rawmidi.h> #include <sound/asequencer.h> #include "usbaudio.h" @@ -101,7 +102,8 @@ struct usb_protocol_ops { }; struct snd_usb_midi { - struct snd_usb_audio *chip; + struct usb_device *dev; + struct snd_card *card; struct usb_interface *iface; const struct snd_usb_audio_quirk *quirk; struct snd_rawmidi *rmidi; @@ -109,13 +111,19 @@ struct snd_usb_midi { struct list_head list; struct timer_list error_timer; spinlock_t disc_lock; + struct mutex mutex; + u32 usb_id; + int next_midi_device; struct snd_usb_midi_endpoint { struct snd_usb_midi_out_endpoint *out; struct snd_usb_midi_in_endpoint *in; } endpoints[MIDI_MAX_ENDPOINTS]; unsigned long input_triggered; + unsigned int opened; unsigned char disconnected; + + struct snd_kcontrol *roland_load_ctl; }; struct snd_usb_midi_out_endpoint { @@ -255,7 +263,7 @@ static void snd_usbmidi_in_urb_complete(struct urb* urb) } } - urb->dev = ep->umidi->chip->dev; + urb->dev = ep->umidi->dev; snd_usbmidi_submit_urb(urb, GFP_ATOMIC); } @@ -296,7 +304,7 @@ static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint* ep) unsigned long flags; spin_lock_irqsave(&ep->buffer_lock, flags); - if (ep->umidi->chip->shutdown) { + if (ep->umidi->disconnected) { spin_unlock_irqrestore(&ep->buffer_lock, flags); return; } @@ -312,7 +320,7 @@ static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint* ep) dump_urb("sending", urb->transfer_buffer, urb->transfer_buffer_length); - urb->dev = ep->umidi->chip->dev; + urb->dev = ep->umidi->dev; if (snd_usbmidi_submit_urb(urb, GFP_ATOMIC) < 0) break; ep->active_urbs |= 1 << urb_index; @@ -349,7 +357,7 @@ static void snd_usbmidi_error_timer(unsigned long data) if (in && in->error_resubmit) { in->error_resubmit = 0; for (j = 0; j < INPUT_URBS; ++j) { - in->urbs[j]->dev = umidi->chip->dev; + in->urbs[j]->dev = umidi->dev; snd_usbmidi_submit_urb(in->urbs[j], GFP_ATOMIC); } } @@ -369,7 +377,7 @@ static int send_bulk_static_data(struct snd_usb_midi_out_endpoint* ep, return -ENOMEM; dump_urb("sending", buf, len); if (ep->urbs[0].urb) - err = usb_bulk_msg(ep->umidi->chip->dev, ep->urbs[0].urb->pipe, + err = usb_bulk_msg(ep->umidi->dev, ep->urbs[0].urb->pipe, buf, len, NULL, 250); kfree(buf); return err; @@ -724,8 +732,7 @@ static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep, if (!ep->ports[0].active) return; - count = snd_usb_get_speed(ep->umidi->chip->dev) == USB_SPEED_HIGH - ? 1 : 2; + count = snd_usb_get_speed(ep->umidi->dev) == USB_SPEED_HIGH ? 1 : 2; count = snd_rawmidi_transmit(ep->ports[0].substream, urb->transfer_buffer, count); @@ -879,6 +886,50 @@ static struct usb_protocol_ops snd_usbmidi_emagic_ops = { }; +static void update_roland_altsetting(struct snd_usb_midi* umidi) +{ + struct usb_interface *intf; + struct usb_host_interface *hostif; + struct usb_interface_descriptor *intfd; + int is_light_load; + + intf = umidi->iface; + is_light_load = intf->cur_altsetting != intf->altsetting; + if (umidi->roland_load_ctl->private_value == is_light_load) + return; + hostif = &intf->altsetting[umidi->roland_load_ctl->private_value]; + intfd = get_iface_desc(hostif); + snd_usbmidi_input_stop(&umidi->list); + usb_set_interface(umidi->dev, intfd->bInterfaceNumber, + intfd->bAlternateSetting); + snd_usbmidi_input_start(&umidi->list); +} + +static void substream_open(struct snd_rawmidi_substream *substream, int open) +{ + struct snd_usb_midi* umidi = substream->rmidi->private_data; + struct snd_kcontrol *ctl; + + mutex_lock(&umidi->mutex); + if (open) { + if (umidi->opened++ == 0 && umidi->roland_load_ctl) { + ctl = umidi->roland_load_ctl; + ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(umidi->card, + SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); + update_roland_altsetting(umidi); + } + } else { + if (--umidi->opened == 0 && umidi->roland_load_ctl) { + ctl = umidi->roland_load_ctl; + ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(umidi->card, + SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); + } + } + mutex_unlock(&umidi->mutex); +} + static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) { struct snd_usb_midi* umidi = substream->rmidi->private_data; @@ -898,11 +949,13 @@ static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) } substream->runtime->private_data = port; port->state = STATE_UNKNOWN; + substream_open(substream, 1); return 0; } static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream) { + substream_open(substream, 0); return 0; } @@ -912,7 +965,7 @@ static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream, port->active = up; if (up) { - if (port->ep->umidi->chip->shutdown) { + if (port->ep->umidi->disconnected) { /* gobble up remaining bytes to prevent wait in * snd_rawmidi_drain_output */ while (!snd_rawmidi_transmit_empty(substream)) @@ -954,11 +1007,13 @@ static void snd_usbmidi_output_drain(struct snd_rawmidi_substream *substream) static int snd_usbmidi_input_open(struct snd_rawmidi_substream *substream) { + substream_open(substream, 1); return 0; } static int snd_usbmidi_input_close(struct snd_rawmidi_substream *substream) { + substream_open(substream, 0); return 0; } @@ -988,7 +1043,7 @@ static struct snd_rawmidi_ops snd_usbmidi_input_ops = { static void free_urb_and_buffer(struct snd_usb_midi *umidi, struct urb *urb, unsigned int buffer_length) { - usb_buffer_free(umidi->chip->dev, buffer_length, + usb_buffer_free(umidi->dev, buffer_length, urb->transfer_buffer, urb->transfer_dma); usb_free_urb(urb); } @@ -1035,24 +1090,24 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi, } } if (ep_info->in_interval) - pipe = usb_rcvintpipe(umidi->chip->dev, ep_info->in_ep); + pipe = usb_rcvintpipe(umidi->dev, ep_info->in_ep); else - pipe = usb_rcvbulkpipe(umidi->chip->dev, ep_info->in_ep); - length = usb_maxpacket(umidi->chip->dev, pipe, 0); + pipe = usb_rcvbulkpipe(umidi->dev, ep_info->in_ep); + length = usb_maxpacket(umidi->dev, pipe, 0); for (i = 0; i < INPUT_URBS; ++i) { - buffer = usb_buffer_alloc(umidi->chip->dev, length, GFP_KERNEL, + buffer = usb_buffer_alloc(umidi->dev, length, GFP_KERNEL, &ep->urbs[i]->transfer_dma); if (!buffer) { snd_usbmidi_in_endpoint_delete(ep); return -ENOMEM; } if (ep_info->in_interval) - usb_fill_int_urb(ep->urbs[i], umidi->chip->dev, + usb_fill_int_urb(ep->urbs[i], umidi->dev, pipe, buffer, length, snd_usbmidi_in_urb_complete, ep, ep_info->in_interval); else - usb_fill_bulk_urb(ep->urbs[i], umidi->chip->dev, + usb_fill_bulk_urb(ep->urbs[i], umidi->dev, pipe, buffer, length, snd_usbmidi_in_urb_complete, ep); ep->urbs[i]->transfer_flags = URB_NO_TRANSFER_DMA_MAP; @@ -1062,15 +1117,6 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi, return 0; } -static unsigned int snd_usbmidi_count_bits(unsigned int x) -{ - unsigned int bits; - - for (bits = 0; x; ++bits) - x &= x - 1; - return bits; -} - /* * Frees an output endpoint. * May be called when ep hasn't been initialized completely. @@ -1113,15 +1159,15 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, ep->urbs[i].ep = ep; } if (ep_info->out_interval) - pipe = usb_sndintpipe(umidi->chip->dev, ep_info->out_ep); + pipe = usb_sndintpipe(umidi->dev, ep_info->out_ep); else - pipe = usb_sndbulkpipe(umidi->chip->dev, ep_info->out_ep); - if (umidi->chip->usb_id == USB_ID(0x0a92, 0x1020)) /* ESI M4U */ + pipe = usb_sndbulkpipe(umidi->dev, ep_info->out_ep); + if (umidi->usb_id == USB_ID(0x0a92, 0x1020)) /* ESI M4U */ ep->max_transfer = 4; else - ep->max_transfer = usb_maxpacket(umidi->chip->dev, pipe, 1); + ep->max_transfer = usb_maxpacket(umidi->dev, pipe, 1); for (i = 0; i < OUTPUT_URBS; ++i) { - buffer = usb_buffer_alloc(umidi->chip->dev, + buffer = usb_buffer_alloc(umidi->dev, ep->max_transfer, GFP_KERNEL, &ep->urbs[i].urb->transfer_dma); if (!buffer) { @@ -1129,12 +1175,12 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, return -ENOMEM; } if (ep_info->out_interval) - usb_fill_int_urb(ep->urbs[i].urb, umidi->chip->dev, + usb_fill_int_urb(ep->urbs[i].urb, umidi->dev, pipe, buffer, ep->max_transfer, snd_usbmidi_out_urb_complete, &ep->urbs[i], ep_info->out_interval); else - usb_fill_bulk_urb(ep->urbs[i].urb, umidi->chip->dev, + usb_fill_bulk_urb(ep->urbs[i].urb, umidi->dev, pipe, buffer, ep->max_transfer, snd_usbmidi_out_urb_complete, &ep->urbs[i]); @@ -1172,6 +1218,7 @@ static void snd_usbmidi_free(struct snd_usb_midi* umidi) if (ep->in) snd_usbmidi_in_endpoint_delete(ep->in); } + mutex_destroy(&umidi->mutex); kfree(umidi); } @@ -1367,7 +1414,7 @@ static struct port_info *find_port_info(struct snd_usb_midi* umidi, int number) int i; for (i = 0; i < ARRAY_SIZE(snd_usbmidi_port_info); ++i) { - if (snd_usbmidi_port_info[i].id == umidi->chip->usb_id && + if (snd_usbmidi_port_info[i].id == umidi->usb_id && snd_usbmidi_port_info[i].port == number) return &snd_usbmidi_port_info[i]; } @@ -1405,7 +1452,7 @@ static void snd_usbmidi_init_substream(struct snd_usb_midi* umidi, port_info = find_port_info(umidi, number); name_format = port_info ? port_info->name : "%s MIDI %d"; snprintf(substream->name, sizeof(substream->name), - name_format, umidi->chip->card->shortname, number + 1); + name_format, umidi->card->shortname, number + 1); *rsubstream = substream; } @@ -1503,7 +1550,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, endpoints[epidx].out_ep = usb_endpoint_num(ep); if (usb_endpoint_xfer_int(ep)) endpoints[epidx].out_interval = ep->bInterval; - else if (snd_usb_get_speed(umidi->chip->dev) == USB_SPEED_LOW) + else if (snd_usb_get_speed(umidi->dev) == USB_SPEED_LOW) /* * Low speed bulk transfers don't exist, so * force interrupt transfers for devices like @@ -1523,7 +1570,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, endpoints[epidx].in_ep = usb_endpoint_num(ep); if (usb_endpoint_xfer_int(ep)) endpoints[epidx].in_interval = ep->bInterval; - else if (snd_usb_get_speed(umidi->chip->dev) == USB_SPEED_LOW) + else if (snd_usb_get_speed(umidi->dev) == USB_SPEED_LOW) endpoints[epidx].in_interval = 1; endpoints[epidx].in_cables = (1 << ms_ep->bNumEmbMIDIJack) - 1; snd_printdd(KERN_INFO "EP %02X: %d jack(s)\n", @@ -1533,6 +1580,52 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, return 0; } +static int roland_load_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *info) +{ + static const char *const names[] = { "High Load", "Light Load" }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 2; + if (info->value.enumerated.item > 1) + info->value.enumerated.item = 1; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int roland_load_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *value) +{ + value->value.enumerated.item[0] = kcontrol->private_value; + return 0; +} + +static int roland_load_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *value) +{ + struct snd_usb_midi* umidi = kcontrol->private_data; + int changed; + + if (value->value.enumerated.item[0] > 1) + return -EINVAL; + mutex_lock(&umidi->mutex); + changed = value->value.enumerated.item[0] != kcontrol->private_value; + if (changed) + kcontrol->private_value = value->value.enumerated.item[0]; + mutex_unlock(&umidi->mutex); + return changed; +} + +static struct snd_kcontrol_new roland_load_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "MIDI Input Mode", + .info = roland_load_info, + .get = roland_load_get, + .put = roland_load_put, + .private_value = 1, +}; + /* * On Roland devices, use the second alternate setting to be able to use * the interrupt input endpoint. @@ -1556,8 +1649,12 @@ static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi* umidi) snd_printdd(KERN_INFO "switching to altsetting %d with int ep\n", intfd->bAlternateSetting); - usb_set_interface(umidi->chip->dev, intfd->bInterfaceNumber, + usb_set_interface(umidi->dev, intfd->bInterfaceNumber, intfd->bAlternateSetting); + + umidi->roland_load_ctl = snd_ctl_new1(&roland_load_ctl, umidi); + if (snd_ctl_add(umidi->card, umidi->roland_load_ctl) < 0) + umidi->roland_load_ctl = NULL; } /* @@ -1573,7 +1670,7 @@ static int snd_usbmidi_detect_endpoints(struct snd_usb_midi* umidi, struct usb_endpoint_descriptor* epd; int i, out_eps = 0, in_eps = 0; - if (USB_ID_VENDOR(umidi->chip->usb_id) == 0x0582) + if (USB_ID_VENDOR(umidi->usb_id) == 0x0582) snd_usbmidi_switch_roland_altsetting(umidi); if (endpoint[0].out_ep || endpoint[0].in_ep) @@ -1760,12 +1857,12 @@ static int snd_usbmidi_create_rawmidi(struct snd_usb_midi* umidi, struct snd_rawmidi *rmidi; int err; - err = snd_rawmidi_new(umidi->chip->card, "USB MIDI", - umidi->chip->next_midi_device++, + err = snd_rawmidi_new(umidi->card, "USB MIDI", + umidi->next_midi_device++, out_ports, in_ports, &rmidi); if (err < 0) return err; - strcpy(rmidi->name, umidi->chip->card->shortname); + strcpy(rmidi->name, umidi->card->shortname); rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT | SNDRV_RAWMIDI_INFO_INPUT | SNDRV_RAWMIDI_INFO_DUPLEX; @@ -1804,7 +1901,7 @@ static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint* ep) return; for (i = 0; i < INPUT_URBS; ++i) { struct urb* urb = ep->urbs[i]; - urb->dev = ep->umidi->chip->dev; + urb->dev = ep->umidi->dev; snd_usbmidi_submit_urb(urb, GFP_KERNEL); } } @@ -1825,9 +1922,10 @@ void snd_usbmidi_input_start(struct list_head* p) /* * Creates and registers everything needed for a MIDI streaming interface. */ -int snd_usb_create_midi_interface(struct snd_usb_audio* chip, - struct usb_interface* iface, - const struct snd_usb_audio_quirk* quirk) +int snd_usbmidi_create(struct snd_card *card, + struct usb_interface* iface, + struct list_head *midi_list, + const struct snd_usb_audio_quirk* quirk) { struct snd_usb_midi* umidi; struct snd_usb_midi_endpoint_info endpoints[MIDI_MAX_ENDPOINTS]; @@ -1837,12 +1935,16 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, umidi = kzalloc(sizeof(*umidi), GFP_KERNEL); if (!umidi) return -ENOMEM; - umidi->chip = chip; + umidi->dev = interface_to_usbdev(iface); + umidi->card = card; umidi->iface = iface; umidi->quirk = quirk; umidi->usb_protocol_ops = &snd_usbmidi_standard_ops; init_timer(&umidi->error_timer); spin_lock_init(&umidi->disc_lock); + mutex_init(&umidi->mutex); + umidi->usb_id = USB_ID(le16_to_cpu(umidi->dev->descriptor.idVendor), + le16_to_cpu(umidi->dev->descriptor.idProduct)); umidi->error_timer.function = snd_usbmidi_error_timer; umidi->error_timer.data = (unsigned long)umidi; @@ -1851,7 +1953,7 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, switch (quirk ? quirk->type : QUIRK_MIDI_STANDARD_INTERFACE) { case QUIRK_MIDI_STANDARD_INTERFACE: err = snd_usbmidi_get_ms_info(umidi, endpoints); - if (chip->usb_id == USB_ID(0x0763, 0x0150)) /* M-Audio Uno */ + if (umidi->usb_id == USB_ID(0x0763, 0x0150)) /* M-Audio Uno */ umidi->usb_protocol_ops = &snd_usbmidi_maudio_broken_running_status_ops; break; @@ -1887,7 +1989,7 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, * interface 0, so we have to make sure that the USB core looks * again at interface 0 by calling usb_set_interface() on it. */ - usb_set_interface(umidi->chip->dev, 0, 0); + usb_set_interface(umidi->dev, 0, 0); err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; case QUIRK_MIDI_EMAGIC: @@ -1914,8 +2016,8 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, out_ports = 0; in_ports = 0; for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { - out_ports += snd_usbmidi_count_bits(endpoints[i].out_cables); - in_ports += snd_usbmidi_count_bits(endpoints[i].in_cables); + out_ports += hweight16(endpoints[i].out_cables); + in_ports += hweight16(endpoints[i].in_cables); } err = snd_usbmidi_create_rawmidi(umidi, out_ports, in_ports); if (err < 0) { @@ -1933,14 +2035,14 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, return err; } - list_add(&umidi->list, &umidi->chip->midi_list); + list_add_tail(&umidi->list, midi_list); for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) snd_usbmidi_input_start_ep(umidi->endpoints[i].in); return 0; } -EXPORT_SYMBOL(snd_usb_create_midi_interface); +EXPORT_SYMBOL(snd_usbmidi_create); EXPORT_SYMBOL(snd_usbmidi_input_stop); EXPORT_SYMBOL(snd_usbmidi_input_start); EXPORT_SYMBOL(snd_usbmidi_disconnect); diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 9efcfd0..c998220 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -1071,6 +1071,15 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unsig channels = (ftr[0] - 7) / csize - 1; master_bits = snd_usb_combine_bytes(ftr + 6, csize); + /* master configuration quirks */ + switch (state->chip->usb_id) { + case USB_ID(0x08bb, 0x2702): + snd_printk(KERN_INFO + "usbmixer: master volume quirk for PCM2702 chip\n"); + /* disable non-functional volume control */ + master_bits &= ~(1 << (USB_FEATURE_VOLUME - 1)); + break; + } if (channels > 0) first_ch_bits = snd_usb_combine_bytes(ftr + 6 + csize, csize); else diff --git a/sound/usb/usbmixer_maps.c b/sound/usb/usbmixer_maps.c index 3e5d66c..77c3588 100644 --- a/sound/usb/usbmixer_maps.c +++ b/sound/usb/usbmixer_maps.c @@ -277,6 +277,22 @@ static struct usbmix_name_map scratch_live_map[] = { { 0 } /* terminator */ }; +/* "Gamesurround Muse Pocket LT" looks same like "Sound Blaster MP3+" + * most importand difference is SU[8], it should be set to "Capture Source" + * to make alsamixer and PA working properly. + * FIXME: or mp3plus_map should use "Capture Source" too, + * so this maps can be merget + */ +static struct usbmix_name_map hercules_usb51_map[] = { + { 8, "Capture Source" }, /* SU, default "PCM Capture Source" */ + { 9, "Master Playback" }, /* FU, default "Speaker Playback" */ + { 10, "Mic Boost", 7 }, /* FU, default "Auto Gain Input" */ + { 11, "Line Capture" }, /* FU, default "PCM Capture" */ + { 13, "Mic Bypass Playback" }, /* FU, default "Mic Playback" */ + { 14, "Line Bypass Playback" }, /* FU, default "Line Playback" */ + { 0 } /* terminator */ +}; + /* * Control map entries */ @@ -316,6 +332,13 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .ignore_ctl_error = 1, }, { + /* Hercules Gamesurround Muse Pocket LT + * (USB 5.1 Channel Audio Adapter) + */ + .id = USB_ID(0x06f8, 0xc000), + .map = hercules_usb51_map, + }, + { .id = USB_ID(0x08bb, 0x2702), .map = linex_map, .ignore_ctl_error = 1, diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index f6f201e..a892bda 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1563,6 +1563,29 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* has ID 0x00ea when not in Advanced Driver mode */ + USB_DEVICE_VENDOR_SPEC(0x0582, 0x00e9), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Roland", */ + /* .product_name = "UA-1G", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, /* Guillemot devices */ { diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 00cd54c..91bb296 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -59,11 +59,11 @@ static int us122l_create_usbmidi(struct snd_card *card) .type = QUIRK_MIDI_US122L, .data = &quirk_data }; - struct usb_device *dev = US122L(card)->chip.dev; + struct usb_device *dev = US122L(card)->dev; struct usb_interface *iface = usb_ifnum_to_if(dev, 1); - return snd_usb_create_midi_interface(&US122L(card)->chip, - iface, &quirk); + return snd_usbmidi_create(card, iface, + &US122L(card)->midi_list, &quirk); } static int us144_create_usbmidi(struct snd_card *card) @@ -81,11 +81,11 @@ static int us144_create_usbmidi(struct snd_card *card) .type = QUIRK_MIDI_US122L, .data = &quirk_data }; - struct usb_device *dev = US122L(card)->chip.dev; + struct usb_device *dev = US122L(card)->dev; struct usb_interface *iface = usb_ifnum_to_if(dev, 0); - return snd_usb_create_midi_interface(&US122L(card)->chip, - iface, &quirk); + return snd_usbmidi_create(card, iface, + &US122L(card)->midi_list, &quirk); } /* @@ -194,11 +194,12 @@ static int usb_stream_hwdep_open(struct snd_hwdep *hw, struct file *file) if (!us122l->first) us122l->first = file; - if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) { - iface = usb_ifnum_to_if(us122l->chip.dev, 0); + if (us122l->dev->descriptor.idProduct == USB_ID_US144 || + us122l->dev->descriptor.idProduct == USB_ID_US144MKII) { + iface = usb_ifnum_to_if(us122l->dev, 0); usb_autopm_get_interface(iface); } - iface = usb_ifnum_to_if(us122l->chip.dev, 1); + iface = usb_ifnum_to_if(us122l->dev, 1); usb_autopm_get_interface(iface); return 0; } @@ -209,11 +210,12 @@ static int usb_stream_hwdep_release(struct snd_hwdep *hw, struct file *file) struct usb_interface *iface; snd_printdd(KERN_DEBUG "%p %p\n", hw, file); - if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) { - iface = usb_ifnum_to_if(us122l->chip.dev, 0); + if (us122l->dev->descriptor.idProduct == USB_ID_US144 || + us122l->dev->descriptor.idProduct == USB_ID_US144MKII) { + iface = usb_ifnum_to_if(us122l->dev, 0); usb_autopm_put_interface(iface); } - iface = usb_ifnum_to_if(us122l->chip.dev, 1); + iface = usb_ifnum_to_if(us122l->dev, 1); usb_autopm_put_interface(iface); if (us122l->first == file) us122l->first = NULL; @@ -297,7 +299,7 @@ static unsigned int usb_stream_hwdep_poll(struct snd_hwdep *hw, static void us122l_stop(struct us122l *us122l) { struct list_head *p; - list_for_each(p, &us122l->chip.midi_list) + list_for_each(p, &us122l->midi_list) snd_usbmidi_input_stop(p); usb_stream_stop(&us122l->sk); @@ -330,7 +332,7 @@ static bool us122l_start(struct us122l *us122l, unsigned use_packsize = 0; bool success = false; - if (us122l->chip.dev->speed == USB_SPEED_HIGH) { + if (us122l->dev->speed == USB_SPEED_HIGH) { /* The us-122l's descriptor defaults to iso max_packsize 78, which isn't needed for samplerates <= 48000. Lets save some memory: @@ -347,11 +349,11 @@ static bool us122l_start(struct us122l *us122l, break; } } - if (!usb_stream_new(&us122l->sk, us122l->chip.dev, 1, 2, + if (!usb_stream_new(&us122l->sk, us122l->dev, 1, 2, rate, use_packsize, period_frames, 6)) goto out; - err = us122l_set_sample_rate(us122l->chip.dev, rate); + err = us122l_set_sample_rate(us122l->dev, rate); if (err < 0) { us122l_stop(us122l); snd_printk(KERN_ERR "us122l_set_sample_rate error \n"); @@ -363,7 +365,7 @@ static bool us122l_start(struct us122l *us122l, snd_printk(KERN_ERR "us122l_start error %i \n", err); goto out; } - list_for_each(p, &us122l->chip.midi_list) + list_for_each(p, &us122l->midi_list) snd_usbmidi_input_start(p); success = true; out: @@ -390,7 +392,7 @@ static int usb_stream_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, err = -ENXIO; goto free; } - high_speed = us122l->chip.dev->speed == USB_SPEED_HIGH; + high_speed = us122l->dev->speed == USB_SPEED_HIGH; if ((cfg->sample_rate != 44100 && cfg->sample_rate != 48000 && (!high_speed || (cfg->sample_rate != 88200 && cfg->sample_rate != 96000))) || @@ -450,7 +452,7 @@ static int usb_stream_hwdep_new(struct snd_card *card) { int err; struct snd_hwdep *hw; - struct usb_device *dev = US122L(card)->chip.dev; + struct usb_device *dev = US122L(card)->dev; err = snd_hwdep_new(card, SND_USB_STREAM_ID, 0, &hw); if (err < 0) @@ -476,26 +478,28 @@ static bool us122l_create_card(struct snd_card *card) int err; struct us122l *us122l = US122L(card); - if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) { - err = usb_set_interface(us122l->chip.dev, 0, 1); + if (us122l->dev->descriptor.idProduct == USB_ID_US144 || + us122l->dev->descriptor.idProduct == USB_ID_US144MKII) { + err = usb_set_interface(us122l->dev, 0, 1); if (err) { snd_printk(KERN_ERR "usb_set_interface error \n"); return false; } } - err = usb_set_interface(us122l->chip.dev, 1, 1); + err = usb_set_interface(us122l->dev, 1, 1); if (err) { snd_printk(KERN_ERR "usb_set_interface error \n"); return false; } - pt_info_set(us122l->chip.dev, 0x11); - pt_info_set(us122l->chip.dev, 0x10); + pt_info_set(us122l->dev, 0x11); + pt_info_set(us122l->dev, 0x10); if (!us122l_start(us122l, 44100, 256)) return false; - if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) + if (us122l->dev->descriptor.idProduct == USB_ID_US144 || + us122l->dev->descriptor.idProduct == USB_ID_US144MKII) err = us144_create_usbmidi(card); else err = us122l_create_usbmidi(card); @@ -508,7 +512,7 @@ static bool us122l_create_card(struct snd_card *card) if (err < 0) { /* release the midi resources */ struct list_head *p; - list_for_each(p, &us122l->chip.midi_list) + list_for_each(p, &us122l->midi_list) snd_usbmidi_disconnect(p); us122l_stop(us122l); @@ -520,7 +524,7 @@ static bool us122l_create_card(struct snd_card *card) static void snd_us122l_free(struct snd_card *card) { struct us122l *us122l = US122L(card); - int index = us122l->chip.index; + int index = us122l->card_index; if (index >= 0 && index < SNDRV_CARDS) snd_us122l_card_used[index] = 0; } @@ -540,13 +544,12 @@ static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp) sizeof(struct us122l), &card); if (err < 0) return err; - snd_us122l_card_used[US122L(card)->chip.index = dev] = 1; + snd_us122l_card_used[US122L(card)->card_index = dev] = 1; card->private_free = snd_us122l_free; - US122L(card)->chip.dev = device; - US122L(card)->chip.card = card; + US122L(card)->dev = device; mutex_init(&US122L(card)->mutex); init_waitqueue_head(&US122L(card)->sk.sleep); - INIT_LIST_HEAD(&US122L(card)->chip.midi_list); + INIT_LIST_HEAD(&US122L(card)->midi_list); strcpy(card->driver, "USB "NAME_ALLCAPS""); sprintf(card->shortname, "TASCAM "NAME_ALLCAPS""); sprintf(card->longname, "%s (%x:%x if %d at %03d/%03d)", @@ -554,8 +557,8 @@ static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp) le16_to_cpu(device->descriptor.idVendor), le16_to_cpu(device->descriptor.idProduct), 0, - US122L(card)->chip.dev->bus->busnum, - US122L(card)->chip.dev->devnum + US122L(card)->dev->bus->busnum, + US122L(card)->dev->devnum ); *cardp = card; return 0; @@ -598,7 +601,8 @@ static int snd_us122l_probe(struct usb_interface *intf, struct snd_card *card; int err; - if (device->descriptor.idProduct == USB_ID_US144 + if ((device->descriptor.idProduct == USB_ID_US144 || + device->descriptor.idProduct == USB_ID_US144MKII) && device->speed == USB_SPEED_HIGH) { snd_printk(KERN_ERR "disable ehci-hcd to run US-144 \n"); return -ENODEV; @@ -635,16 +639,15 @@ static void snd_us122l_disconnect(struct usb_interface *intf) mutex_lock(&us122l->mutex); us122l_stop(us122l); mutex_unlock(&us122l->mutex); - us122l->chip.shutdown = 1; /* release the midi resources */ - list_for_each(p, &us122l->chip.midi_list) { + list_for_each(p, &us122l->midi_list) { snd_usbmidi_disconnect(p); } - usb_put_intf(usb_ifnum_to_if(us122l->chip.dev, 0)); - usb_put_intf(usb_ifnum_to_if(us122l->chip.dev, 1)); - usb_put_dev(us122l->chip.dev); + usb_put_intf(usb_ifnum_to_if(us122l->dev, 0)); + usb_put_intf(usb_ifnum_to_if(us122l->dev, 1)); + usb_put_dev(us122l->dev); while (atomic_read(&us122l->mmap_count)) msleep(500); @@ -667,7 +670,7 @@ static int snd_us122l_suspend(struct usb_interface *intf, pm_message_t message) if (!us122l) return 0; - list_for_each(p, &us122l->chip.midi_list) + list_for_each(p, &us122l->midi_list) snd_usbmidi_input_stop(p); mutex_lock(&us122l->mutex); @@ -694,23 +697,24 @@ static int snd_us122l_resume(struct usb_interface *intf) mutex_lock(&us122l->mutex); /* needed, doesn't restart without: */ - if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) { - err = usb_set_interface(us122l->chip.dev, 0, 1); + if (us122l->dev->descriptor.idProduct == USB_ID_US144 || + us122l->dev->descriptor.idProduct == USB_ID_US144MKII) { + err = usb_set_interface(us122l->dev, 0, 1); if (err) { snd_printk(KERN_ERR "usb_set_interface error \n"); goto unlock; } } - err = usb_set_interface(us122l->chip.dev, 1, 1); + err = usb_set_interface(us122l->dev, 1, 1); if (err) { snd_printk(KERN_ERR "usb_set_interface error \n"); goto unlock; } - pt_info_set(us122l->chip.dev, 0x11); - pt_info_set(us122l->chip.dev, 0x10); + pt_info_set(us122l->dev, 0x11); + pt_info_set(us122l->dev, 0x10); - err = us122l_set_sample_rate(us122l->chip.dev, + err = us122l_set_sample_rate(us122l->dev, us122l->sk.s->cfg.sample_rate); if (err < 0) { snd_printk(KERN_ERR "us122l_set_sample_rate error \n"); @@ -720,7 +724,7 @@ static int snd_us122l_resume(struct usb_interface *intf) if (err) goto unlock; - list_for_each(p, &us122l->chip.midi_list) + list_for_each(p, &us122l->midi_list) snd_usbmidi_input_start(p); unlock: mutex_unlock(&us122l->mutex); @@ -739,6 +743,16 @@ static struct usb_device_id snd_us122l_usb_id_table[] = { .idVendor = 0x0644, .idProduct = USB_ID_US144 }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x0644, + .idProduct = USB_ID_US122MKII + }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x0644, + .idProduct = USB_ID_US144MKII + }, { /* terminator */ } }; diff --git a/sound/usb/usx2y/us122l.h b/sound/usb/usx2y/us122l.h index 3d10c4b..f263b3f 100644 --- a/sound/usb/usx2y/us122l.h +++ b/sound/usb/usx2y/us122l.h @@ -3,7 +3,8 @@ struct us122l { - struct snd_usb_audio chip; + struct usb_device *dev; + int card_index; int stride; struct usb_stream_kernel sk; @@ -12,6 +13,7 @@ struct us122l { unsigned second_periods_polled; struct file *master; struct file *slave; + struct list_head midi_list; atomic_t mmap_count; }; @@ -23,5 +25,7 @@ struct us122l { #define USB_ID_US122L 0x800E #define USB_ID_US144 0x800F +#define USB_ID_US122MKII 0x8021 +#define USB_ID_US144MKII 0x8020 #endif diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index 52e04b2..1879b72 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -114,7 +114,7 @@ static int snd_usX2Y_hwdep_dsp_status(struct snd_hwdep *hw, struct usX2Ydev *us428 = hw->private_data; int id = -1; - switch (le16_to_cpu(us428->chip.dev->descriptor.idProduct)) { + switch (le16_to_cpu(us428->dev->descriptor.idProduct)) { case USB_ID_US122: id = USX2Y_TYPE_122; break; @@ -164,14 +164,14 @@ static int usX2Y_create_usbmidi(struct snd_card *card) .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = &quirk_data_2 }; - struct usb_device *dev = usX2Y(card)->chip.dev; + struct usb_device *dev = usX2Y(card)->dev; struct usb_interface *iface = usb_ifnum_to_if(dev, 0); struct snd_usb_audio_quirk *quirk = le16_to_cpu(dev->descriptor.idProduct) == USB_ID_US428 ? &quirk_2 : &quirk_1; snd_printdd("usX2Y_create_usbmidi \n"); - return snd_usb_create_midi_interface(&usX2Y(card)->chip, iface, quirk); + return snd_usbmidi_create(card, iface, &usX2Y(card)->midi_list, quirk); } static int usX2Y_create_alsa_devices(struct snd_card *card) @@ -202,7 +202,7 @@ static int snd_usX2Y_hwdep_dsp_load(struct snd_hwdep *hw, snd_printdd( "dsp_load %s\n", dsp->name); if (access_ok(VERIFY_READ, dsp->image, dsp->length)) { - struct usb_device* dev = priv->chip.dev; + struct usb_device* dev = priv->dev; char *buf; buf = memdup_user(dsp->image, dsp->length); diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index cb4bb83..c42350e 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -239,8 +239,8 @@ static void i_usX2Y_In04Int(struct urb *urb) for (j = 0; j < URBS_AsyncSeq && !err; ++j) if (0 == usX2Y->AS04.urb[j]->status) { struct us428_p4out *p4out = us428ctls->p4out + send; // FIXME if more than 1 p4out is new, 1 gets lost. - usb_fill_bulk_urb(usX2Y->AS04.urb[j], usX2Y->chip.dev, - usb_sndbulkpipe(usX2Y->chip.dev, 0x04), &p4out->val.vol, + usb_fill_bulk_urb(usX2Y->AS04.urb[j], usX2Y->dev, + usb_sndbulkpipe(usX2Y->dev, 0x04), &p4out->val.vol, p4out->type == eLT_Light ? sizeof(struct us428_lights) : 5, i_usX2Y_Out04Int, usX2Y); err = usb_submit_urb(usX2Y->AS04.urb[j], GFP_ATOMIC); @@ -253,7 +253,7 @@ static void i_usX2Y_In04Int(struct urb *urb) if (err) snd_printk(KERN_ERR "In04Int() usb_submit_urb err=%i\n", err); - urb->dev = usX2Y->chip.dev; + urb->dev = usX2Y->dev; usb_submit_urb(urb, GFP_ATOMIC); } @@ -273,8 +273,8 @@ int usX2Y_AsyncSeq04_init(struct usX2Ydev *usX2Y) err = -ENOMEM; break; } - usb_fill_bulk_urb( usX2Y->AS04.urb[i], usX2Y->chip.dev, - usb_sndbulkpipe(usX2Y->chip.dev, 0x04), + usb_fill_bulk_urb( usX2Y->AS04.urb[i], usX2Y->dev, + usb_sndbulkpipe(usX2Y->dev, 0x04), usX2Y->AS04.buffer + URB_DataLen_AsyncSeq*i, 0, i_usX2Y_Out04Int, usX2Y ); @@ -293,7 +293,7 @@ int usX2Y_In04_init(struct usX2Ydev *usX2Y) } init_waitqueue_head(&usX2Y->In04WaitQueue); - usb_fill_int_urb(usX2Y->In04urb, usX2Y->chip.dev, usb_rcvintpipe(usX2Y->chip.dev, 0x4), + usb_fill_int_urb(usX2Y->In04urb, usX2Y->dev, usb_rcvintpipe(usX2Y->dev, 0x4), usX2Y->In04Buf, 21, i_usX2Y_In04Int, usX2Y, 10); @@ -348,13 +348,12 @@ static int usX2Y_create_card(struct usb_device *device, struct snd_card **cardp) sizeof(struct usX2Ydev), &card); if (err < 0) return err; - snd_usX2Y_card_used[usX2Y(card)->chip.index = dev] = 1; + snd_usX2Y_card_used[usX2Y(card)->card_index = dev] = 1; card->private_free = snd_usX2Y_card_private_free; - usX2Y(card)->chip.dev = device; - usX2Y(card)->chip.card = card; + usX2Y(card)->dev = device; init_waitqueue_head(&usX2Y(card)->prepare_wait_queue); mutex_init(&usX2Y(card)->prepare_mutex); - INIT_LIST_HEAD(&usX2Y(card)->chip.midi_list); + INIT_LIST_HEAD(&usX2Y(card)->midi_list); strcpy(card->driver, "USB "NAME_ALLCAPS""); sprintf(card->shortname, "TASCAM "NAME_ALLCAPS""); sprintf(card->longname, "%s (%x:%x if %d at %03d/%03d)", @@ -362,7 +361,7 @@ static int usX2Y_create_card(struct usb_device *device, struct snd_card **cardp) le16_to_cpu(device->descriptor.idVendor), le16_to_cpu(device->descriptor.idProduct), 0,//us428(card)->usbmidi.ifnum, - usX2Y(card)->chip.dev->bus->busnum, usX2Y(card)->chip.dev->devnum + usX2Y(card)->dev->bus->busnum, usX2Y(card)->dev->devnum ); *cardp = card; return 0; @@ -432,8 +431,8 @@ static void snd_usX2Y_card_private_free(struct snd_card *card) usb_free_urb(usX2Y(card)->In04urb); if (usX2Y(card)->us428ctls_sharedmem) snd_free_pages(usX2Y(card)->us428ctls_sharedmem, sizeof(*usX2Y(card)->us428ctls_sharedmem)); - if (usX2Y(card)->chip.index >= 0 && usX2Y(card)->chip.index < SNDRV_CARDS) - snd_usX2Y_card_used[usX2Y(card)->chip.index] = 0; + if (usX2Y(card)->card_index >= 0 && usX2Y(card)->card_index < SNDRV_CARDS) + snd_usX2Y_card_used[usX2Y(card)->card_index] = 0; } /* @@ -445,13 +444,12 @@ static void usX2Y_usb_disconnect(struct usb_device *device, void* ptr) struct snd_card *card = ptr; struct usX2Ydev *usX2Y = usX2Y(card); struct list_head *p; - usX2Y->chip.shutdown = 1; usX2Y->chip_status = USX2Y_STAT_CHIP_HUP; usX2Y_unlinkSeq(&usX2Y->AS04); usb_kill_urb(usX2Y->In04urb); snd_card_disconnect(card); /* release the midi resources */ - list_for_each(p, &usX2Y->chip.midi_list) { + list_for_each(p, &usX2Y->midi_list) { snd_usbmidi_disconnect(p); } if (usX2Y->us428ctls_sharedmem) diff --git a/sound/usb/usx2y/usbusx2y.h b/sound/usb/usx2y/usbusx2y.h index 456b5fd..1d174ce 100644 --- a/sound/usb/usx2y/usbusx2y.h +++ b/sound/usb/usx2y/usbusx2y.h @@ -22,7 +22,8 @@ struct snd_usX2Y_urbSeq { #include "usx2yhwdeppcm.h" struct usX2Ydev { - struct snd_usb_audio chip; + struct usb_device *dev; + int card_index; int stride; struct urb *In04urb; void *In04Buf; @@ -42,6 +43,9 @@ struct usX2Ydev { struct snd_usX2Y_substream *subs[4]; struct snd_usX2Y_substream * volatile prepare_subs; wait_queue_head_t prepare_wait_queue; + struct list_head midi_list; + struct list_head pcm_list; + int pcm_devs; }; diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 9efd27f..74a67a8 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -199,7 +199,7 @@ static int usX2Y_urb_submit(struct snd_usX2Y_substream *subs, struct urb *urb, i return -ENODEV; urb->start_frame = (frame + NRURBS * nr_of_packs()); // let hcd do rollover sanity checks urb->hcpriv = NULL; - urb->dev = subs->usX2Y->chip.dev; /* we need to set this at each time */ + urb->dev = subs->usX2Y->dev; /* we need to set this at each time */ if ((err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { snd_printk(KERN_ERR "usb_submit_urb() returned %i\n", err); return err; @@ -300,7 +300,7 @@ static void usX2Y_error_sequence(struct usX2Ydev *usX2Y, "Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" "Most propably some urb of usb-frame %i is still missing.\n" "Cause could be too long delays in usb-hcd interrupt handling.\n", - usb_get_current_frame_number(usX2Y->chip.dev), + usb_get_current_frame_number(usX2Y->dev), subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", usX2Y->wait_iso_frame, urb->start_frame, usX2Y->wait_iso_frame); usX2Y_clients_stop(usX2Y); @@ -313,7 +313,7 @@ static void i_usX2Y_urb_complete(struct urb *urb) if (unlikely(atomic_read(&subs->state) < state_PREPARED)) { snd_printdd("hcd_frame=%i ep=%i%s status=%i start_frame=%i\n", - usb_get_current_frame_number(usX2Y->chip.dev), + usb_get_current_frame_number(usX2Y->dev), subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", urb->status, urb->start_frame); return; @@ -424,7 +424,7 @@ static int usX2Y_urbs_allocate(struct snd_usX2Y_substream *subs) int i; unsigned int pipe; int is_playback = subs == subs->usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; - struct usb_device *dev = subs->usX2Y->chip.dev; + struct usb_device *dev = subs->usX2Y->dev; pipe = is_playback ? usb_sndisocpipe(dev, subs->endpoint) : usb_rcvisocpipe(dev, subs->endpoint); @@ -500,7 +500,7 @@ static int usX2Y_urbs_start(struct snd_usX2Y_substream *subs) unsigned long pack; if (0 == i) atomic_set(&subs->state, state_STARTING3); - urb->dev = usX2Y->chip.dev; + urb->dev = usX2Y->dev; urb->transfer_flags = URB_ISO_ASAP; for (pack = 0; pack < nr_of_packs(); pack++) { urb->iso_frame_desc[pack].offset = subs->maxpacksize * pack; @@ -692,7 +692,7 @@ static int usX2Y_rate_set(struct usX2Ydev *usX2Y, int rate) } ((char*)(usbdata + i))[0] = ra[i].c1; ((char*)(usbdata + i))[1] = ra[i].c2; - usb_fill_bulk_urb(us->urb[i], usX2Y->chip.dev, usb_sndbulkpipe(usX2Y->chip.dev, 4), + usb_fill_bulk_urb(us->urb[i], usX2Y->dev, usb_sndbulkpipe(usX2Y->dev, 4), usbdata + i, 2, i_usX2Y_04Int, usX2Y); #ifdef OLD_USB us->urb[i]->transfer_flags = USB_QUEUE_BULK; @@ -740,17 +740,17 @@ static int usX2Y_format_set(struct usX2Ydev *usX2Y, snd_pcm_format_t format) alternate = 1; usX2Y->stride = 4; } - list_for_each(p, &usX2Y->chip.midi_list) { + list_for_each(p, &usX2Y->midi_list) { snd_usbmidi_input_stop(p); } usb_kill_urb(usX2Y->In04urb); - if ((err = usb_set_interface(usX2Y->chip.dev, 0, alternate))) { + if ((err = usb_set_interface(usX2Y->dev, 0, alternate))) { snd_printk(KERN_ERR "usb_set_interface error \n"); return err; } - usX2Y->In04urb->dev = usX2Y->chip.dev; + usX2Y->In04urb->dev = usX2Y->dev; err = usb_submit_urb(usX2Y->In04urb, GFP_KERNEL); - list_for_each(p, &usX2Y->chip.midi_list) { + list_for_each(p, &usX2Y->midi_list) { snd_usbmidi_input_start(p); } usX2Y->format = format; @@ -955,7 +955,7 @@ static int usX2Y_audio_stream_new(struct snd_card *card, int playback_endpoint, struct snd_pcm *pcm; int err, i; struct snd_usX2Y_substream **usX2Y_substream = - usX2Y(card)->subs + 2 * usX2Y(card)->chip.pcm_devs; + usX2Y(card)->subs + 2 * usX2Y(card)->pcm_devs; for (i = playback_endpoint ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; i <= SNDRV_PCM_STREAM_CAPTURE; ++i) { @@ -971,7 +971,7 @@ static int usX2Y_audio_stream_new(struct snd_card *card, int playback_endpoint, usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]->endpoint = playback_endpoint; usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE]->endpoint = capture_endpoint; - err = snd_pcm_new(card, NAME_ALLCAPS" Audio", usX2Y(card)->chip.pcm_devs, + err = snd_pcm_new(card, NAME_ALLCAPS" Audio", usX2Y(card)->pcm_devs, playback_endpoint ? 1 : 0, 1, &pcm); if (err < 0) { @@ -987,7 +987,7 @@ static int usX2Y_audio_stream_new(struct snd_card *card, int playback_endpoint, pcm->private_free = snd_usX2Y_pcm_private_free; pcm->info_flags = 0; - sprintf(pcm->name, NAME_ALLCAPS" Audio #%d", usX2Y(card)->chip.pcm_devs); + sprintf(pcm->name, NAME_ALLCAPS" Audio #%d", usX2Y(card)->pcm_devs); if ((playback_endpoint && 0 > (err = snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream, @@ -1001,7 +1001,7 @@ static int usX2Y_audio_stream_new(struct snd_card *card, int playback_endpoint, snd_usX2Y_pcm_private_free(pcm); return err; } - usX2Y(card)->chip.pcm_devs++; + usX2Y(card)->pcm_devs++; return 0; } @@ -1013,14 +1013,14 @@ int usX2Y_audio_create(struct snd_card *card) { int err = 0; - INIT_LIST_HEAD(&usX2Y(card)->chip.pcm_list); + INIT_LIST_HEAD(&usX2Y(card)->pcm_list); if (0 > (err = usX2Y_audio_stream_new(card, 0xA, 0x8))) return err; - if (le16_to_cpu(usX2Y(card)->chip.dev->descriptor.idProduct) == USB_ID_US428) + if (le16_to_cpu(usX2Y(card)->dev->descriptor.idProduct) == USB_ID_US428) if (0 > (err = usX2Y_audio_stream_new(card, 0, 0xA))) return err; - if (le16_to_cpu(usX2Y(card)->chip.dev->descriptor.idProduct) != USB_ID_US122) + if (le16_to_cpu(usX2Y(card)->dev->descriptor.idProduct) != USB_ID_US122) err = usX2Y_rate_set(usX2Y(card), 44100); // Lets us428 recognize output-volume settings, disturbs us122. return err; } diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 4b2304c..9ed6c39 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -234,7 +234,7 @@ static void i_usX2Y_usbpcm_urb_complete(struct urb *urb) if (unlikely(atomic_read(&subs->state) < state_PREPARED)) { snd_printdd("hcd_frame=%i ep=%i%s status=%i start_frame=%i\n", - usb_get_current_frame_number(usX2Y->chip.dev), + usb_get_current_frame_number(usX2Y->dev), subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", urb->status, urb->start_frame); return; @@ -318,7 +318,7 @@ static int usX2Y_usbpcm_urbs_allocate(struct snd_usX2Y_substream *subs) int i; unsigned int pipe; int is_playback = subs == subs->usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; - struct usb_device *dev = subs->usX2Y->chip.dev; + struct usb_device *dev = subs->usX2Y->dev; pipe = is_playback ? usb_sndisocpipe(dev, subs->endpoint) : usb_rcvisocpipe(dev, subs->endpoint); @@ -441,7 +441,7 @@ static int usX2Y_usbpcm_urbs_start(struct snd_usX2Y_substream *subs) unsigned long pack; if (0 == u) atomic_set(&subs->state, state_STARTING3); - urb->dev = usX2Y->chip.dev; + urb->dev = usX2Y->dev; urb->transfer_flags = URB_ISO_ASAP; for (pack = 0; pack < nr_of_packs(); pack++) { urb->iso_frame_desc[pack].offset = subs->maxpacksize * (pack + u * nr_of_packs()); @@ -741,7 +741,7 @@ int usX2Y_hwdep_pcm_new(struct snd_card *card) int err; struct snd_hwdep *hw; struct snd_pcm *pcm; - struct usb_device *dev = usX2Y(card)->chip.dev; + struct usb_device *dev = usX2Y(card)->dev; if (1 != nr_of_packs()) return 0; |