July 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: core: make comments fit the code
by Wolfram Sang 19 Jul '11

19 Jul '11

In one comment, cpu_dai was mentioned although codec_dai was used in the code. Also, fix the name for the card dai list which has no seperation into card_dai and codec_dai. Signed-off-by: Wolfram Sang <w.sang(a)pengutronix.de> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d75043e..ba95444 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1631,7 +1631,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) } } cpu_dai->probed = 1; - /* mark cpu_dai as probed and add to card cpu_dai list */ + /* mark cpu_dai as probed and add to card dai list */ list_add(&cpu_dai->card_list, &card->dai_dev_list); } @@ -1672,7 +1672,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) } } - /* mark cpu_dai as probed and add to card cpu_dai list */ + /* mark codec_dai as probed and add to card dai list */ codec_dai->probed = 1; list_add(&codec_dai->card_list, &card->dai_dev_list); } -- 1.7.5.4

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[alsa-devel] [PATCH v5 4/4] ASoC: SAMSUNG: Add I2S0 internal dma driver
by Sangbeom Kim 19 Jul '11

19 Jul '11

I2S in Exynos4 and S5PC110(S5PV210) has a internal dma. It can be used low power audio mode and 2nd channel transfer. This patch can support idma. Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- V2: change idma_new by new pcm_new api add handling different idma address as jassi comments, Modify the idma driver V3: merge idma_reg_int and idma_addr_init V4: move idma setting in i2s.c to idma_hw_params in idma.c move idma_reg_addr_init calling modify idma address handling error modify module information delete useless dev_dbg V5: delete redundant error print and check routine modify irq handler as jassi comment fix the indent error sound/soc/samsung/Makefile | 2 + sound/soc/samsung/idma.c | 453 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/samsung/idma.h | 26 +++ 3 files changed, 481 insertions(+), 0 deletions(-) create mode 100644 sound/soc/samsung/idma.c create mode 100644 sound/soc/samsung/idma.h diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 00ba557..41becdd 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -1,5 +1,6 @@ # S3c24XX Platform Support snd-soc-s3c24xx-objs := dma.o +snd-soc-idma-objs := idma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-ac97-objs := ac97.o @@ -16,6 +17,7 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o obj-$(CONFIG_SND_SAMSUNG_SPDIF) += snd-soc-samsung-spdif.o obj-$(CONFIG_SND_SAMSUNG_PCM) += snd-soc-pcm.o obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-i2s.o +obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-idma.o # S3C24XX Machine Support snd-soc-jive-wm8750-objs := jive_wm8750.o diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c new file mode 100644 index 0000000..ebde074 --- /dev/null +++ b/sound/soc/samsung/idma.c @@ -0,0 +1,453 @@ +/* + * sound/soc/samsung/idma.c + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd. + * http://www.samsung.com + * + * I2S0's Internal DMA driver + * + * 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/interrupt.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "i2s.h" +#include "idma.h" +#include "dma.h" +#include "i2s-regs.h" + +#define ST_RUNNING (1<<0) +#define ST_OPENED (1<<1) + +static const struct snd_pcm_hardware idma_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_U24_LE | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = MAX_IDMA_BUFFER, + .period_bytes_min = 128, + .period_bytes_max = MAX_IDMA_PERIOD, + .periods_min = 1, + .periods_max = 2, +}; + +struct idma_ctrl { + spinlock_t lock; + int state; + dma_addr_t start; + dma_addr_t pos; + dma_addr_t end; + dma_addr_t period; + dma_addr_t periodsz; + void *token; + void (*cb)(void *dt, int bytes_xfer); +}; + +static struct idma_info { + spinlock_t lock; + void __iomem *regs; + dma_addr_t lp_tx_addr; +} idma; + +static void idma_getpos(dma_addr_t *src) +{ + *src = idma.lp_tx_addr + + (readl(idma.regs + I2STRNCNT) & 0xffffff) * 4; +} + +static int idma_enqueue(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = substream->runtime->private_data; + u32 val; + + spin_lock(&prtd->lock); + prtd->token = (void *) substream; + spin_unlock(&prtd->lock); + + /* Internal DMA Level0 Interrupt Address */ + val = idma.lp_tx_addr + prtd->periodsz; + writel(val, idma.regs + I2SLVL0ADDR); + + /* Start address0 of I2S internal DMA operation. */ + val = idma.lp_tx_addr; + writel(val, idma.regs + I2SSTR0); + + /* + * Transfer block size for I2S internal DMA. + * Should decide transfer size before start dma operation + */ + val = readl(idma.regs + I2SSIZE); + val &= ~(I2SSIZE_TRNMSK << I2SSIZE_SHIFT); + val |= (((runtime->dma_bytes >> 2) & + I2SSIZE_TRNMSK) << I2SSIZE_SHIFT); + writel(val, idma.regs + I2SSIZE); + + val = readl(idma.regs + I2SAHB); + val |= AHB_INTENLVL0; + writel(val, idma.regs + I2SAHB); + + return 0; +} + +static void idma_setcallbk(struct snd_pcm_substream *substream, + void (*cb)(void *, int)) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + + spin_lock(&prtd->lock); + prtd->cb = cb; + spin_unlock(&prtd->lock); +} + +static void idma_control(int op) +{ + u32 val = readl(idma.regs + I2SAHB); + + spin_lock(&idma.lock); + + switch (op) { + case LPAM_DMA_START: + val |= (AHB_INTENLVL0 | AHB_DMAEN); + break; + case LPAM_DMA_STOP: + val &= ~(AHB_INTENLVL0 | AHB_DMAEN); + break; + default: + spin_unlock(&idma.lock); + return; + } + + writel(val, idma.regs + I2SAHB); + spin_unlock(&idma.lock); +} + +static void idma_done(void *id, int bytes_xfer) +{ + struct snd_pcm_substream *substream = id; + struct idma_ctrl *prtd = substream->runtime->private_data; + + if (prtd && (prtd->state & ST_RUNNING)) + snd_pcm_period_elapsed(substream); +} + +static int idma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = substream->runtime->private_data; + u32 mod = readl(idma.regs + I2SMOD); + u32 ahb = readl(idma.regs + I2SAHB); + + ahb |= (AHB_DMARLD | AHB_INTMASK); + mod |= MOD_TXS_IDMA; + writel(ahb, idma.regs + I2SAHB); + writel(mod, idma.regs + I2SMOD); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->start = prtd->pos = runtime->dma_addr; + prtd->period = params_periods(params); + prtd->periodsz = params_period_bytes(params); + prtd->end = runtime->dma_addr + runtime->dma_bytes; + + idma_setcallbk(substream, idma_done); + + return 0; +} + +static int idma_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + + return 0; +} + +static int idma_prepare(struct snd_pcm_substream *substream) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + + prtd->pos = prtd->start; + + /* flush the DMA channel */ + idma_control(LPAM_DMA_STOP); + idma_enqueue(substream); + + return 0; +} + +static int idma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + int ret = 0; + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + prtd->state |= ST_RUNNING; + idma_control(LPAM_DMA_START); + break; + + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + prtd->state &= ~ST_RUNNING; + idma_control(LPAM_DMA_STOP); + break; + + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static snd_pcm_uframes_t + idma_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = runtime->private_data; + dma_addr_t src; + unsigned long res; + + spin_lock(&prtd->lock); + + idma_getpos(&src); + res = src - prtd->start; + + spin_unlock(&prtd->lock); + + return bytes_to_frames(substream->runtime, res); +} + +static int idma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long size, offset; + int ret; + + /* From snd_pcm_lib_mmap_iomem */ + vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + vma->vm_flags |= VM_IO; + size = vma->vm_end - vma->vm_start; + offset = vma->vm_pgoff << PAGE_SHIFT; + ret = io_remap_pfn_range(vma, vma->vm_start, + (runtime->dma_addr + offset) >> PAGE_SHIFT, + size, vma->vm_page_prot); + + return ret; +} + +static irqreturn_t iis_irq(int irqno, void *dev_id) +{ + struct idma_ctrl *prtd = (struct idma_ctrl *)dev_id; + u32 iiscon, iisahb, val, addr; + + iisahb = readl(idma.regs + I2SAHB); + iiscon = readl(idma.regs + I2SCON); + + val = (iisahb & AHB_LVL0INT) ? AHB_CLRLVL0INT : 0; + + if (val) { + iisahb |= val; + writel(iisahb, idma.regs + I2SAHB); + + addr = readl(idma.regs + I2SLVL0ADDR) - idma.lp_tx_addr; + addr += prtd->periodsz; + addr %= (prtd->end - prtd->start); + addr += idma.lp_tx_addr; + + writel(addr, idma.regs + I2SLVL0ADDR); + + if (prtd->cb) + prtd->cb(prtd->token, prtd->period); + } + + return IRQ_HANDLED; +} + +static int idma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd; + int ret; + + snd_soc_set_runtime_hwparams(substream, &idma_hardware); + + prtd = kzalloc(sizeof(struct idma_ctrl), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + ret = request_irq(IRQ_I2S0, iis_irq, 0, "i2s", prtd); + if (ret < 0) { + pr_err("fail to claim i2s irq , ret = %d\n", ret); + kfree(prtd); + return ret; + } + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + + return 0; +} + +static int idma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = runtime->private_data; + + free_irq(IRQ_I2S0, prtd); + + if (!prtd) + pr_err("idma_close called with prtd == NULL\n"); + + kfree(prtd); + + return 0; +} + +static struct snd_pcm_ops idma_ops = { + .open = idma_open, + .close = idma_close, + .ioctl = snd_pcm_lib_ioctl, + .trigger = idma_trigger, + .pointer = idma_pointer, + .mmap = idma_mmap, + .hw_params = idma_hw_params, + .hw_free = idma_hw_free, + .prepare = idma_prepare, +}; + +static void idma_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + + substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + if (!substream) + return; + + buf = &substream->dma_buffer; + if (!buf->area) + return; + + iounmap(buf->area); + + buf->area = NULL; + buf->addr = 0; +} + +static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + /* Assign PCM buffer pointers */ + buf->dev.type = SNDRV_DMA_TYPE_CONTINUOUS; + buf->addr = idma.lp_tx_addr; + buf->bytes = idma_hardware.buffer_bytes_max; + buf->area = (unsigned char *)ioremap(buf->addr, buf->bytes); + + return 0; +} + +static u64 idma_mask = DMA_BIT_MASK(32); + +static int idma_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &idma_mask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (dai->driver->playback.channels_min) + ret = preallocate_idma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + + return ret; +} + +void idma_reg_addr_init(void *regs, dma_addr_t addr) +{ + spin_lock_init(&idma.lock); + idma.regs = regs; + idma.lp_tx_addr = addr; +} + +struct snd_soc_platform_driver asoc_idma_platform = { + .ops = &idma_ops, + .pcm_new = idma_new, + .pcm_free = idma_free, +}; + +static int __devinit asoc_idma_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &asoc_idma_platform); +} + +static int __devexit asoc_idma_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver asoc_idma_driver = { + .driver = { + .name = "samsung-idma", + .owner = THIS_MODULE, + }, + + .probe = asoc_idma_platform_probe, + .remove = __devexit_p(asoc_idma_platform_remove), +}; + +static int __init asoc_idma_init(void) +{ + return platform_driver_register(&asoc_idma_driver); +} +module_init(asoc_idma_init); + +static void __exit asoc_idma_exit(void) +{ + platform_driver_unregister(&asoc_idma_driver); +} +module_exit(asoc_idma_exit); + +MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar(a)gmail.com>"); +MODULE_DESCRIPTION("Samsung ASoC IDMA Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/idma.h b/sound/soc/samsung/idma.h new file mode 100644 index 0000000..4827321 --- /dev/null +++ b/sound/soc/samsung/idma.h @@ -0,0 +1,26 @@ +/* + * sound/soc/samsung/idma.h + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd + * http://www.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 __SND_SOC_SAMSUNG_IDMA_H_ +#define __SND_SOC_SAMSUNG_IDMA_H_ + +extern void idma_reg_addr_init(void *regs, dma_addr_t addr); + +/* dma_state */ +#define LPAM_DMA_STOP 0 +#define LPAM_DMA_START 1 + +#define MAX_IDMA_PERIOD (128 * 1024) +#define MAX_IDMA_BUFFER (160 * 1024) + +#endif /* __SND_SOC_SAMSUNG_IDMA_H_ */ -- 1.7.1

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Re: [alsa-devel] HP-path handling on VIA codecs
by alex dot baldacchino dot alsasub at gmail dot com 18 Jul '11

18 Jul '11

2011/7/8 Takashi Iwai <tiwai(a)suse.de>: > Hi, > > while testing with the gathered alsa-info.sh outputs, now I'm > considering how to improve the handling of the headphone (and speaker) > output paths. > > On VIA codecs, it looks like that the loopback mixer always contains > the input from the front DAC, and the front DAC is the only DAC that > can be mixed with loopback. (It might be not always, but on many > codecs, at least). > > I mean, some codecs of other vendors have a connection like: > Input-pin(s) -> Mix1 -> Mix2 -> Output1 > DAC1 -> Mix2 -> Output1 > Input-pin(s) -> Mix3 -> Mix4 -> Output2 > DAC2 -> Mix4 -> Output2 > .... > For such devices, you can use DAC as the volume control of each output > channel (HP, line-out, speaker, etc). So I started implementing in > that way for VIA, too. It's found as two parsed output paths, > spec->hp_out, and spec->hp_dep_out, > > But later I found that VIA has only > Input-pin(s) -> Mix1 -> Output1 > DAC1 -> Mix1 -> Output1 > or > DAC1 -> Output1 > > Input-pin(s) -> Mix1 -> Output1 > DAC1 -> Mix1 -> Output2 > or > DAC2 -> Output2 > ... > > Suppose the output2 is HP, it means that you can't have an individual > volume control via DAC2 if you want to mix the loopback. > Now we face a dilemma: usually an individual volume control is really > nice. But, if we take the individual control, it looses the loopback, > as these are mutual-exclusive for HP (or speaker) outputs. > > Thus we need to decide a policy: either > > A. HP has no volume unless independent mode > > B. Don't mix loopback to HP or others (but front-out) > > C. Allow to turn on/off aa-loopback mode; when loopback is enabled, > switch the path to hp_dep without volume control (only for front). > When loopback is disabled, switch to the direct DAC path. > > A is the old driver behavior, I guess. B is somewhat similar to now. > C would be smart, but need more implementation and add yet another > mixer element. > > What do you think? > > > thanks, > > Takashi > Personally, if I'm not misunderstanding it, I'd vote for C. I had tried something similar for old implementation, and can tell it worked and was easy enough to implement, AFAICT (I can resend the code if found of any interest, since it seems to me my first mail about it went lost, at least that's not in this mailing list archives, though that's based on the old code, and could be useful only to exchange ideas). It was quite simple: I used a few new fields in struct via_spec and two input mux controls, one for hp and one for front (to have a fine grained control and disable/enable aa path independently for those lines - and test if it could work this way), called "Direct Output" and "HP Direct Output" (I chose names thinking about the sound coming straight from analog inputs and going to analog outputs through the AA mixer, but could be, for instance, "({Front, HP}) Output Mode" (or "Output Source") with items labeled "Mixed with input" (or "Stereo mixer") and "Playback only", or the like). Corresponding controls callbacks worked on the mixer connected to each output jack and getting input from a dac (through an audio selector) and from the aa mixer, choosing the right mixer basing on the control they were invoked for, and the corresponding values in via_spec new fields: .get() returned the (negated) value of the mute bit of aa mixer connection in the chosen mixer connection list, and .put() muted/un-muted such a connection if in a different state (a better behavior would be to alternatively mute/unmute the connection to the dac and the connection to the loopback mixer, taking care of hp state (working on aa mixer connection only for hp mixer while in independent mode), and, of course, there could be a single switch/mux control to enable/disable the aa path globally). My reason to test such a solution was some noise I constantly got (and still get with newer code, but for Independent HP in latest version) from my headset microphone to each such line out (front and hp), but I can think, for instance, of someone wishing to listen to something while recording something else from line-in and not aiming to mix both sounds in output. But I guess some problems could arise in actual implementation, unless I'm overvaluing/misunderstanding them. First of all, now there's a pair of PCM Playback controls (switch and volume) attached to aa mixer and I'm not sure if muting connections to it could affect those controls (specially the volume). In such a case, they could be moved to the front dac, but this problem, if existing, could be re-raised for hp path in independent mode (in which case front dac and attached controls would be out of the game). An alternative could be leaving those controls where they're now, but reimplementing callbacks so that they could use a different nid (and work on more than one nid at a time). This way: - the .put callback for turning aa-loopback on/off for a certain nid (or all affected nids at once, if only one such control existed) would be responsible: 1. to record somewhere (accessible by via_spec) the nid(s) to work on (quite easy with separate controls attached to each affected mixer: the nid to gather would be just the actual one being unmuted in the two-entries connection list of passed-in kcontrol nid or the non loopback connection for hp mixer in independent mode), 2. to set the right volume (not just un-muting) for the non-loopback path(s) when switching to it (basing on pcm volume setting) and muting/unmuting the connection(s) to aa mixer as needed; - the .put callback for adjusting pcm volume setting would work as now for aa mixer nid (both to let it work as expected when used as source for output and to gather correct values from it, but also for correct handling of capture from stereo mixer), while performing same operations for each alternative nid, gathered from a proper list and being non-null and different from aa mixer nid; - the .get callback for "PCM Playback Volume" could be left untouched (to always gather infos from the 'base' nid the control is attacthed to). Similar considerations could/should be done for the pcm switch, that (now, in my case) has the effect to mute/unmute front output (only). However, I might be missing the point since I'm not sure how those controls are expected to work properly. Previously, with old code, I had no PCM Playback Switch and a "standalone" PCM Playback Volume, detached from any nid, and such seemed to work for all outputs, as far as I remember (I should re-install old implementation to make a test); alsa-info.sh reported following output for alsactl: control.31 { iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } Now I have both controls attached to 0x21, they seem to have effect on front path only (throughout aa mixer - previously its connection to front dac, for my codec, was just un-muted with an initialization verb) and alsa-info.sh reports: control.9 { iface MIXER name 'PCM Playback Volume' value.0 22 value.1 22 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -150 dbvalue.1 -150 } } control.10 { iface MIXER name 'PCM Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } Similar concerns could be raised for Master Playback controls, given some codecs (but not mine - vt2020) holds them via their aa mixer nid. Another, possibly minor, problem, as pointed out in Lydia's answer, could be the somewhat inconsistence introduced between newer codecs supporting solution C and older ones unable to mute their aa mixer without loosing their front dac (because their aa mixer is just between their front dac and their rear green jack, but also between front dac and hp jack when in redirected mode). Such could be simply disregarded, taken as a consequence of a different set of supported features; therefore, it should be possible to choose automatically whether to create the proper control(s) for turning on/off aa-loopback or not while parsing output paths, looking at the connection lists of nids in a certain path, starting from an output jack and moving towards a dac, and looking if an element (a mixer or the jack nid itself) is found with at least two connections, one of which should be the aa mixer nid and the other one either the front dac nid, or an audio selector connected to it. For hp path the non-loopback connection shouldn't be hp dac directly (not passing through a selector), or the implementation would conflict with redirected output if such a dac were shared, which is the case for several codecs, specially older ones (newer ones seem to use an audio selector, instead, for front and hp), though I'm not sure if that's the same for all such codecs - hp_indep_shared could tell that, so that a few older codecs could support solution C for hp only, but in such a case solution C would behave mostly as B in current implementation, but for redirected mode, given front stream would be 'cloned' for hp dac if not shared, so hp could get loopback mixed in or not at whim, even if such would be inconsistent with front being always mixed with loopback). Regards, Alex

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Re: [alsa-devel] model needed for Clevo P150HM (Realtek ALC892)
by Takashi Iwai 18 Jul '11

18 Jul '11

At Thu, 30 Jun 2011 08:30:10 +0200, Julian Sikorski wrote: > > W dniu 30.06.2011 07:26, Takashi Iwai pisze: > > At Wed, 29 Jun 2011 21:29:38 +0200, > > Julian Sikorski wrote: > >> > >> I also tried merging sound-2.6/master with Linus's tree, but this also > >> did not help. > > > > It's really strange. Are you sure that this is the kernel you > > modified properly? Before further debugging, check two things: > > > > - Run "git log sound/pci/hda/patch_realtek.c" and the first few > > entries, whether they come from sound git tree. For example, > > check the commit below is contained: > > > > commit 7ec9c6ccc6007b14a916021d4ba7ffbcc7822ae3 > > Author: Takashi Iwai <tiwai(a)suse.de> > > Date: Mon Jun 27 15:53:38 2011 +0200 > > > > ALSA: hda - Fix volume-init for ALC259 with invalid widget caps > > commit 0b0c8490670a77b937f502491ae9d5236aa4054a > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Tue Jun 28 12:45:47 2011 +0200 > > ALSA: hda - Create snd_hda_get_conn_index() helper function > > Create snd_hda_get_conn_index() helper function for obtaining the > connection index of the widget. Replaced the similar codes used in > several codec-drivers with this common helper. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > commit 4f574b7b1a1cc8aac617e938459e8f03a641e678 > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Mon Jun 27 16:17:07 2011 +0200 > > ALSA: hda - More volume-init fixes for ALC267 codec > > More similar fixes like previous commits: handle the exceptional case > like ALC267 where no volume amp is found in ADC widget but in the > capsrc widget instead. > > Also minor checks for avoiding possible erros: no connection-select > when the pin has a single selection, and add beep verbs only when the > 0x1d is used for beep. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > To be even more sure, I re-synced your branch again and built a new > kernel. I am attaching patch_realtek.c file being built. Fresh > alsa-info.sh is here: > http://www.alsa-project.org/db/?f=5051b2cb788b1db4273e06e98a78783dd0ec679f OK, then it's correct. > > > > - Confirm the running kernel is what you really built and installed. > > You can modify the suffix via $CONFIG_LOCALVERSION in .config file. > > Name it uniquely, rebuild, re-install and reboot. The new suffix > > should appear in the output of "uname -r". > > belegdol4 is the suffix I added: > $ uname -r > 3.0-0.rc5.git0.1.belegdol4.fc15.x86_64 > > > - Once when these are correct, check /proc/asound/card0/codec#0: > > the widget NID 0x02 should contain "Speaker Playback Volume" instead > > of "Headphone Playback Volume" with the latest driver. > > Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out > Control: name="Headphone Playback Volume", index=0, device=0 > > Perhaps I need to specify a correct model to enable this code? No, you shouldn't pass any model option. Could you apply the patch below to add some debug prints, and give the relevant kernel messages you get? Just grep lines with 'XXX'. Takashi --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7b96dce..3134555 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -18770,8 +18770,10 @@ static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin) static hda_nid_t get_dac_if_single(struct hda_codec *codec, hda_nid_t pin) { hda_nid_t sel = alc_go_down_to_selector(codec, pin); - if (snd_hda_get_conn_list(codec, sel, NULL) == 1) + if (snd_hda_get_conn_list(codec, sel, NULL) == 1) { + printk(KERN_DEBUG "XXX pin %x coupled with a single DAC\n", pin); return alc_auto_look_for_dac(codec, pin); + } return 0; } @@ -18783,6 +18785,7 @@ static int alc662_auto_fill_dac_nids(struct hda_codec *codec) bool redone; int i; + printk(KERN_DEBUG "XXX alc662_auto_fill_dac_nids called\n"); again: spec->multiout.num_dacs = 0; spec->multiout.hp_nid = 0; @@ -18813,6 +18816,7 @@ static int alc662_auto_fill_dac_nids(struct hda_codec *codec) * checking the hard-wired DACs */ redone = true; + printk(KERN_DEBUG "XXX redoing...\n"); goto again; } } @@ -18826,6 +18830,17 @@ static int alc662_auto_fill_dac_nids(struct hda_codec *codec) sizeof(hda_nid_t) * (cfg->line_outs - i - 1)); } + printk(KERN_DEBUG "XXX filled dacs: %x/%x/%x/%x/%x, hp %x, ext %x/%x/%x\n", + spec->private_dac_nids[0], + spec->private_dac_nids[1], + spec->private_dac_nids[2], + spec->private_dac_nids[3], + spec->private_dac_nids[4], + spec->multiout.hp_nid, + spec->multiout.extra_out_nid[0], + spec->multiout.extra_out_nid[1], + spec->multiout.extra_out_nid[2]); + return 0; } @@ -18877,6 +18892,7 @@ static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec, mix = alc_auto_dac_to_mix(codec, pin, nid); if (!mix) continue; + printk(KERN_DEBUG "XXX create mixer ch %d pin %x dac %x mix %x\n", i, pin, nid, mix); name = alc_get_line_out_pfx(spec, i, true, &index); if (!name) { /* Center/LFE */ @@ -19026,6 +19042,7 @@ static int alc_auto_fill_multi_ios(struct hda_codec *codec, dac = alc_auto_look_for_dac(codec, nid); if (!dac) continue; + printk(KERN_DEBUG "XXX added multi-io pin %x, dac %x\n", nid, dac); spec->multi_io[num_pins].pin = nid; spec->multi_io[num_pins].dac = dac; num_pins++; @@ -19128,6 +19145,7 @@ static int alc_auto_add_multi_channel_mode(struct hda_codec *codec, if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT && cfg->hp_outs == 1) { /* use HP as primary out */ + printk(KERN_DEBUG "XXX swap primary out to HP\n"); cfg->speaker_outs = cfg->line_outs; memcpy(cfg->speaker_pins, cfg->line_out_pins, sizeof(cfg->speaker_pins));

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[alsa-devel] [PATCH] ASoC: Mark cache as dirty when suspending
by Mark Brown 18 Jul '11

18 Jul '11

Since quite a few drivers are not managing to flag the cache as needing to be resynced after suspend and it's a reasonable thing to do flag the cache as needing sync automatically when suspending. The expectation is that systems will mainly only keep the CODEC powered when doing audio through the CODEC so we won't actually suspend the device anyway; drivers which want to can override this behaviour when they resume. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e44267f..93109a4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -577,6 +577,7 @@ int snd_soc_suspend(struct device *dev) case SND_SOC_BIAS_OFF: codec->driver->suspend(codec, PMSG_SUSPEND); codec->suspended = 1; + codec->cache_sync = 1; break; default: dev_dbg(codec->dev, "CODEC is on over suspend\n"); -- 1.7.5.4

2 1

[alsa-devel] [RESEND][PATCH v5 4/4] ASoC: SAMSUNG: Add I2S0 internal dma driver
by Sangbeom Kim 18 Jul '11

18 Jul '11

I2S in Exynos4 and S5PC110(S5PV210) has a internal dma. It can be used low power audio mode and 2nd channel transfer. This patch can support idma. Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- V2: change idma_new by new pcm_new api add handling different idma address as jassi comments, Modify the idma driver V3: merge idma_reg_int and idma_addr_init V4: move idma setting in i2s.c to idma_hw_params in idma.c move idma_reg_addr_init calling modify idma address handling error modify module information delete useless dev_dbg V5: delete redundant error print and check routine modify irq handler as jassi comment fix the indent error modify period_bytes_max value to set half size of max buffer sound/soc/samsung/Makefile | 2 + sound/soc/samsung/idma.c | 453 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/samsung/idma.h | 26 +++ 3 files changed, 481 insertions(+), 0 deletions(-) create mode 100644 sound/soc/samsung/idma.c create mode 100644 sound/soc/samsung/idma.h diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 00ba557..41becdd 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -1,5 +1,6 @@ # S3c24XX Platform Support snd-soc-s3c24xx-objs := dma.o +snd-soc-idma-objs := idma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-ac97-objs := ac97.o @@ -16,6 +17,7 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o obj-$(CONFIG_SND_SAMSUNG_SPDIF) += snd-soc-samsung-spdif.o obj-$(CONFIG_SND_SAMSUNG_PCM) += snd-soc-pcm.o obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-i2s.o +obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-idma.o # S3C24XX Machine Support snd-soc-jive-wm8750-objs := jive_wm8750.o diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c new file mode 100644 index 0000000..ebde074 --- /dev/null +++ b/sound/soc/samsung/idma.c @@ -0,0 +1,453 @@ +/* + * sound/soc/samsung/idma.c + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd. + * http://www.samsung.com + * + * I2S0's Internal DMA driver + * + * 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/interrupt.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "i2s.h" +#include "idma.h" +#include "dma.h" +#include "i2s-regs.h" + +#define ST_RUNNING (1<<0) +#define ST_OPENED (1<<1) + +static const struct snd_pcm_hardware idma_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_U24_LE | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = MAX_IDMA_BUFFER, + .period_bytes_min = 128, + .period_bytes_max = MAX_IDMA_BUFFER/2, + .periods_min = 1, + .periods_max = 2, +}; + +struct idma_ctrl { + spinlock_t lock; + int state; + dma_addr_t start; + dma_addr_t pos; + dma_addr_t end; + dma_addr_t period; + dma_addr_t periodsz; + void *token; + void (*cb)(void *dt, int bytes_xfer); +}; + +static struct idma_info { + spinlock_t lock; + void __iomem *regs; + dma_addr_t lp_tx_addr; +} idma; + +static void idma_getpos(dma_addr_t *src) +{ + *src = idma.lp_tx_addr + + (readl(idma.regs + I2STRNCNT) & 0xffffff) * 4; +} + +static int idma_enqueue(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = substream->runtime->private_data; + u32 val; + + spin_lock(&prtd->lock); + prtd->token = (void *) substream; + spin_unlock(&prtd->lock); + + /* Internal DMA Level0 Interrupt Address */ + val = idma.lp_tx_addr + prtd->periodsz; + writel(val, idma.regs + I2SLVL0ADDR); + + /* Start address0 of I2S internal DMA operation. */ + val = idma.lp_tx_addr; + writel(val, idma.regs + I2SSTR0); + + /* + * Transfer block size for I2S internal DMA. + * Should decide transfer size before start dma operation + */ + val = readl(idma.regs + I2SSIZE); + val &= ~(I2SSIZE_TRNMSK << I2SSIZE_SHIFT); + val |= (((runtime->dma_bytes >> 2) & + I2SSIZE_TRNMSK) << I2SSIZE_SHIFT); + writel(val, idma.regs + I2SSIZE); + + val = readl(idma.regs + I2SAHB); + val |= AHB_INTENLVL0; + writel(val, idma.regs + I2SAHB); + + return 0; +} + +static void idma_setcallbk(struct snd_pcm_substream *substream, + void (*cb)(void *, int)) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + + spin_lock(&prtd->lock); + prtd->cb = cb; + spin_unlock(&prtd->lock); +} + +static void idma_control(int op) +{ + u32 val = readl(idma.regs + I2SAHB); + + spin_lock(&idma.lock); + + switch (op) { + case LPAM_DMA_START: + val |= (AHB_INTENLVL0 | AHB_DMAEN); + break; + case LPAM_DMA_STOP: + val &= ~(AHB_INTENLVL0 | AHB_DMAEN); + break; + default: + spin_unlock(&idma.lock); + return; + } + + writel(val, idma.regs + I2SAHB); + spin_unlock(&idma.lock); +} + +static void idma_done(void *id, int bytes_xfer) +{ + struct snd_pcm_substream *substream = id; + struct idma_ctrl *prtd = substream->runtime->private_data; + + if (prtd && (prtd->state & ST_RUNNING)) + snd_pcm_period_elapsed(substream); +} + +static int idma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = substream->runtime->private_data; + u32 mod = readl(idma.regs + I2SMOD); + u32 ahb = readl(idma.regs + I2SAHB); + + ahb |= (AHB_DMARLD | AHB_INTMASK); + mod |= MOD_TXS_IDMA; + writel(ahb, idma.regs + I2SAHB); + writel(mod, idma.regs + I2SMOD); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->start = prtd->pos = runtime->dma_addr; + prtd->period = params_periods(params); + prtd->periodsz = params_period_bytes(params); + prtd->end = runtime->dma_addr + runtime->dma_bytes; + + idma_setcallbk(substream, idma_done); + + return 0; +} + +static int idma_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + + return 0; +} + +static int idma_prepare(struct snd_pcm_substream *substream) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + + prtd->pos = prtd->start; + + /* flush the DMA channel */ + idma_control(LPAM_DMA_STOP); + idma_enqueue(substream); + + return 0; +} + +static int idma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + int ret = 0; + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + prtd->state |= ST_RUNNING; + idma_control(LPAM_DMA_START); + break; + + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + prtd->state &= ~ST_RUNNING; + idma_control(LPAM_DMA_STOP); + break; + + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static snd_pcm_uframes_t + idma_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = runtime->private_data; + dma_addr_t src; + unsigned long res; + + spin_lock(&prtd->lock); + + idma_getpos(&src); + res = src - prtd->start; + + spin_unlock(&prtd->lock); + + return bytes_to_frames(substream->runtime, res); +} + +static int idma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long size, offset; + int ret; + + /* From snd_pcm_lib_mmap_iomem */ + vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + vma->vm_flags |= VM_IO; + size = vma->vm_end - vma->vm_start; + offset = vma->vm_pgoff << PAGE_SHIFT; + ret = io_remap_pfn_range(vma, vma->vm_start, + (runtime->dma_addr + offset) >> PAGE_SHIFT, + size, vma->vm_page_prot); + + return ret; +} + +static irqreturn_t iis_irq(int irqno, void *dev_id) +{ + struct idma_ctrl *prtd = (struct idma_ctrl *)dev_id; + u32 iiscon, iisahb, val, addr; + + iisahb = readl(idma.regs + I2SAHB); + iiscon = readl(idma.regs + I2SCON); + + val = (iisahb & AHB_LVL0INT) ? AHB_CLRLVL0INT : 0; + + if (val) { + iisahb |= val; + writel(iisahb, idma.regs + I2SAHB); + + addr = readl(idma.regs + I2SLVL0ADDR) - idma.lp_tx_addr; + addr += prtd->periodsz; + addr %= (prtd->end - prtd->start); + addr += idma.lp_tx_addr; + + writel(addr, idma.regs + I2SLVL0ADDR); + + if (prtd->cb) + prtd->cb(prtd->token, prtd->period); + } + + return IRQ_HANDLED; +} + +static int idma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd; + int ret; + + snd_soc_set_runtime_hwparams(substream, &idma_hardware); + + prtd = kzalloc(sizeof(struct idma_ctrl), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + ret = request_irq(IRQ_I2S0, iis_irq, 0, "i2s", prtd); + if (ret < 0) { + pr_err("fail to claim i2s irq , ret = %d\n", ret); + kfree(prtd); + return ret; + } + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + + return 0; +} + +static int idma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = runtime->private_data; + + free_irq(IRQ_I2S0, prtd); + + if (!prtd) + pr_err("idma_close called with prtd == NULL\n"); + + kfree(prtd); + + return 0; +} + +static struct snd_pcm_ops idma_ops = { + .open = idma_open, + .close = idma_close, + .ioctl = snd_pcm_lib_ioctl, + .trigger = idma_trigger, + .pointer = idma_pointer, + .mmap = idma_mmap, + .hw_params = idma_hw_params, + .hw_free = idma_hw_free, + .prepare = idma_prepare, +}; + +static void idma_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + + substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + if (!substream) + return; + + buf = &substream->dma_buffer; + if (!buf->area) + return; + + iounmap(buf->area); + + buf->area = NULL; + buf->addr = 0; +} + +static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + /* Assign PCM buffer pointers */ + buf->dev.type = SNDRV_DMA_TYPE_CONTINUOUS; + buf->addr = idma.lp_tx_addr; + buf->bytes = idma_hardware.buffer_bytes_max; + buf->area = (unsigned char *)ioremap(buf->addr, buf->bytes); + + return 0; +} + +static u64 idma_mask = DMA_BIT_MASK(32); + +static int idma_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &idma_mask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (dai->driver->playback.channels_min) + ret = preallocate_idma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + + return ret; +} + +void idma_reg_addr_init(void *regs, dma_addr_t addr) +{ + spin_lock_init(&idma.lock); + idma.regs = regs; + idma.lp_tx_addr = addr; +} + +struct snd_soc_platform_driver asoc_idma_platform = { + .ops = &idma_ops, + .pcm_new = idma_new, + .pcm_free = idma_free, +}; + +static int __devinit asoc_idma_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &asoc_idma_platform); +} + +static int __devexit asoc_idma_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver asoc_idma_driver = { + .driver = { + .name = "samsung-idma", + .owner = THIS_MODULE, + }, + + .probe = asoc_idma_platform_probe, + .remove = __devexit_p(asoc_idma_platform_remove), +}; + +static int __init asoc_idma_init(void) +{ + return platform_driver_register(&asoc_idma_driver); +} +module_init(asoc_idma_init); + +static void __exit asoc_idma_exit(void) +{ + platform_driver_unregister(&asoc_idma_driver); +} +module_exit(asoc_idma_exit); + +MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar(a)gmail.com>"); +MODULE_DESCRIPTION("Samsung ASoC IDMA Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/idma.h b/sound/soc/samsung/idma.h new file mode 100644 index 0000000..4827321 --- /dev/null +++ b/sound/soc/samsung/idma.h @@ -0,0 +1,26 @@ +/* + * sound/soc/samsung/idma.h + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd + * http://www.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 __SND_SOC_SAMSUNG_IDMA_H_ +#define __SND_SOC_SAMSUNG_IDMA_H_ + +extern void idma_reg_addr_init(void *regs, dma_addr_t addr); + +/* dma_state */ +#define LPAM_DMA_STOP 0 +#define LPAM_DMA_START 1 + + +#define MAX_IDMA_BUFFER (128 * 1024) + +#endif /* __SND_SOC_SAMSUNG_IDMA_H_ */ -- 1.7.1

2 3

[alsa-devel] ASoC updates for 3.1
by Mark Brown 18 Jul '11

18 Jul '11

The following changes since commit e320bc42be746b8f006d163b6e340e5fb79ce7b2: Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2011-07-09 11:43:04 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Axel Lin (1): ASoC: wm8900: fix a memory leak if wm8900_set_fll fails Dimitris Papastamos (1): ASoC: WM8983: Initial driver Giridhar Maruthy (1): ASoC: SAMSUNG: 24-bit audio playback on Exynos4210 Jarkko Nikula (2): ASoC: tlv320aic3x: Don't sync first two registers from register cache ASoC: tlv320aic3x: Do soft reset to codec when going to bias off state Johannes Stezenbach (2): ASoC: STA32x: Add mixer controls for biquad coefficients ASoC: STA32x: Preserve reserved register bits Kuninori Morimoto (3): ASoC: sh: fsi-ak4642: fixup snd_soc_card name ASoC: sh: fsi-da7210: fixup snd_soc_card name ASoC: sh: fsi-hdmi: fixup snd_soc_card name Mark Brown (19): Merge branch 'for-3.0' into for-3.1 Merge branch 'for-3.0' into for-3.1 Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-3.1 ASoC: Conditionalize the enable of WM8994 ADC TDM mode ASoC: Use late enable handling for direct voice, speaker and headphone ASoC: Implement DC servo completion IRQ handling for wm_hubs devices ASoC: Hook up DC servo completion IRQ for WM8994 and WM8958 ASoC: Use WM8994 FLL lock interrupt ASoC: Don't warn on low WM8994/58 AIFnCLKs ASoC: Log WM8994 FIFO errors from the interrupt ASoC: Fix shift in WM8958 accessory detection default implementation Merge branch 'for-3.0' into for-3.1 ASoC: Handle spurious wm_hubs DC servo done interrupts ASoC: Handle failed WM8994 FLL lock waits ASoC: Report an error for unknown adav80x formats ASoC: Reduce power consumption for idle DAIs in WM8994 ASoC: Don't use -1 to boostrap subseq so it can be used by drivers ASoC: Correct WM8994 MICBIAS supply widget hookup Merge branch 'for-3.0' into for-3.1 Michael Williamson (1): audio: tlv320aic26: fix PLL register configuration Takashi Iwai (4): Merge branch 'for-3.0' of git://git.kernel.org/.../lrg/asoc-2.6 into fix/asoc Merge branch 'for-3.0' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc ASoC: Don't set invalid name string to snd_card->driver field Merge branch 'for-3.0' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/adav80x.c | 4 +- sound/soc/codecs/sta32x.c | 140 +++++ sound/soc/codecs/tlv320aic26.c | 14 +- sound/soc/codecs/tlv320aic3x.c | 9 +- sound/soc/codecs/wm8900.c | 1 + sound/soc/codecs/wm8983.c | 1203 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8983.h | 1029 +++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 156 +++++- sound/soc/codecs/wm8994.h | 3 + sound/soc/codecs/wm_hubs.c | 33 +- sound/soc/codecs/wm_hubs.h | 8 + sound/soc/samsung/smdk_wm8994.c | 5 +- sound/soc/sh/fsi-ak4642.c | 16 +- sound/soc/sh/fsi-da7210.c | 2 +- sound/soc/sh/fsi-hdmi.c | 4 +- sound/soc/soc-core.c | 5 +- sound/soc/soc-dapm.c | 2 +- 20 files changed, 2593 insertions(+), 49 deletions(-) create mode 100644 sound/soc/codecs/wm8983.c create mode 100644 sound/soc/codecs/wm8983.h

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[alsa-devel] ASoC updates for 3.0
by Mark Brown 18 Jul '11

18 Jul '11

The following changes since commit 18361bbe3180eca62796188d62aefac1519f4c83: Merge branch 'for-3.0' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc (2011-07-09 09:44:09 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.0 Kuninori Morimoto (3): ASoC: sh: fsi-ak4642: fixup snd_soc_card name ASoC: sh: fsi-da7210: fixup snd_soc_card name ASoC: sh: fsi-hdmi: fixup snd_soc_card name Mark Brown (2): ASoC: Fix shift in WM8958 accessory detection default implementation ASoC: Correct WM8994 MICBIAS supply widget hookup sound/soc/codecs/wm8994.c | 9 +++++---- sound/soc/sh/fsi-ak4642.c | 16 ++++++++-------- sound/soc/sh/fsi-da7210.c | 2 +- sound/soc/sh/fsi-hdmi.c | 4 ++-- 4 files changed, 16 insertions(+), 15 deletions(-)

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[alsa-devel] [PATCH] ASoC: Report an error for unknown adav80x formats
by Mark Brown 16 Jul '11

16 Jul '11

Not only fixes error handling but also some uninitialized variable warnings. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/adav80x.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index e30fba6..300c04b 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -456,7 +456,7 @@ static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, val = ADAV80X_CAPTURE_WORD_LEN24; break; default: - break; + return -EINVAL; } snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], @@ -488,7 +488,7 @@ static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec, val = ADAV80X_PLAYBACK_MODE_RIGHT_J_24; break; default: - break; + return -EINVAL; } snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1], -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Don't use -1 to boostrap subseq so it can be used by drivers
by Mark Brown 16 Jul '11

16 Jul '11

Makes life a little easier if you want to add subsequences to an existing driver as you can use -1 to put things at the start of sequences. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 54fa2e5..fbfcda0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -989,7 +989,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, INIT_LIST_HEAD(&pending); cur_sort = -1; - cur_subseq = -1; + cur_subseq = INT_MIN; cur_reg = SND_SOC_NOPM; cur_dapm = NULL; } -- 1.7.5.4

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