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author | Linus Torvalds <torvalds@linux-foundation.org> | 2009-06-12 11:16:27 -0700 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2009-06-12 11:16:27 -0700 |
commit | e349792a385ed47390d156155b1a1e19af1bf163 (patch) | |
tree | f9dc4e3e42769950230eaa58ecdd056eb27b23e8 /sound/soc/fsl | |
parent | 6d21491838a2a9f22843c7530b118596ee9f4d77 (diff) | |
parent | e3f86d3d3ce350144562d9bd035dc8a274fce58e (diff) | |
download | kernel_samsung_smdk4412-e349792a385ed47390d156155b1a1e19af1bf163.zip kernel_samsung_smdk4412-e349792a385ed47390d156155b1a1e19af1bf163.tar.gz kernel_samsung_smdk4412-e349792a385ed47390d156155b1a1e19af1bf163.tar.bz2 |
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (290 commits)
ALSA: pcm - Update document about xrun_debug proc file
ALSA: lx6464es - support standard alsa module parameters
ALSA: snd_usb_caiaq: set mixername
ALSA: hda - add quirk for STAC92xx (SigmaTel STAC9205)
ALSA: use card device as parent for jack input-devices
ALSA: sound/ps3: Correct existing and add missing annotations
ALSA: sound/ps3: Restructure driver source
ALSA: sound/ps3: Fix checkpatch issues
ASoC: Fix lm4857 control
ALSA: ctxfi - Clear PCM resources at hw_params and hw_free
ALSA: ctxfi - Check the presence of SRC instance in PCM pointer callbacks
ALSA: ctxfi - Add missing start check in atc_pcm_playback_start()
ALSA: ctxfi - Add use_system_timer module option
ALSA: usb - Add boot quirk for C-Media 6206 USB Audio
ALSA: ctxfi - Fix wrong model id for UAA
ALSA: ctxfi - Clean up probe routines
ALSA: hda - Fix the previous tagra-8ch patch
ALSA: hda - Add 7.1 support for MSI GX620
ALSA: pcm - A helper function to compose PCM stream name for debug prints
ALSA: emu10k1 - Fix minimum periods for efx playback
...
Diffstat (limited to 'sound/soc/fsl')
-rw-r--r-- | sound/soc/fsl/Kconfig | 32 | ||||
-rw-r--r-- | sound/soc/fsl/Makefile | 7 | ||||
-rw-r--r-- | sound/soc/fsl/efika-audio-fabric.c | 90 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_ssi.c | 11 | ||||
-rw-r--r-- | sound/soc/fsl/mpc5200_dma.c | 564 | ||||
-rw-r--r-- | sound/soc/fsl/mpc5200_dma.h | 80 | ||||
-rw-r--r-- | sound/soc/fsl/mpc5200_psc_ac97.c | 329 | ||||
-rw-r--r-- | sound/soc/fsl/mpc5200_psc_ac97.h | 15 | ||||
-rw-r--r-- | sound/soc/fsl/mpc5200_psc_i2s.c | 754 | ||||
-rw-r--r-- | sound/soc/fsl/mpc5200_psc_i2s.h | 12 | ||||
-rw-r--r-- | sound/soc/fsl/pcm030-audio-fabric.c | 90 |
11 files changed, 1277 insertions, 707 deletions
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 9fc9082..5dbebf8 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,5 +1,8 @@ config SND_SOC_OF_SIMPLE tristate + +config SND_MPC52xx_DMA + tristate # ASoC platform support for the Freescale MPC8610 SOC. This compiles drivers # for the SSI and the Elo DMA controller. You will still need to select @@ -22,7 +25,34 @@ config SND_SOC_MPC8610_HPCD config SND_SOC_MPC5200_I2S tristate "Freescale MPC5200 PSC in I2S mode driver" depends on PPC_MPC52xx && PPC_BESTCOMM - select SND_SOC_OF_SIMPLE + select SND_MPC52xx_DMA select PPC_BESTCOMM_GEN_BD help Say Y here to support the MPC5200 PSCs in I2S mode. + +config SND_SOC_MPC5200_AC97 + tristate "Freescale MPC5200 PSC in AC97 mode driver" + depends on PPC_MPC52xx && PPC_BESTCOMM + select AC97_BUS + select SND_MPC52xx_DMA + select PPC_BESTCOMM_GEN_BD + help + Say Y here to support the MPC5200 PSCs in AC97 mode. + +config SND_MPC52xx_SOC_PCM030 + tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712" + depends on PPC_MPC5200_SIMPLE && BROKEN + select SND_SOC_MPC5200_AC97 + select SND_SOC_WM9712 + help + Say Y if you want to add support for sound on the Phytec pcm030 + baseboard. + +config SND_MPC52xx_SOC_EFIKA + tristate "SoC AC97 Audio support for bbplan Efika and STAC9766" + depends on PPC_EFIKA && BROKEN + select SND_SOC_MPC5200_AC97 + select SND_SOC_STAC9766 + help + Say Y if you want to add support for sound on the Efika. + diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index f85134c..a83a739 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -10,5 +10,12 @@ snd-soc-fsl-ssi-objs := fsl_ssi.o snd-soc-fsl-dma-objs := fsl_dma.o obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o +# MPC5200 Platform Support +obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o +obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o + +# MPC5200 Machine Support +obj-$(CONFIG_SND_MPC52xx_SOC_PCM030) += pcm030-audio-fabric.o +obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c new file mode 100644 index 0000000..85b0e75 --- /dev/null +++ b/sound/soc/fsl/efika-audio-fabric.c @@ -0,0 +1,90 @@ +/* + * Efika driver for the PSC of the Freescale MPC52xx + * configured as AC97 interface + * + * Copyright 2008 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl@gmail.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/of_device.h> +#include <linux/of_platform.h> +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/soc-of-simple.h> + +#include "mpc5200_dma.h" +#include "mpc5200_psc_ac97.h" +#include "../codecs/stac9766.h" + +static struct snd_soc_device device; +static struct snd_soc_card card; + +static struct snd_soc_dai_link efika_fabric_dai[] = { +{ + .name = "AC97", + .stream_name = "AC97 Analog", + .codec_dai = &stac9766_dai[STAC9766_DAI_AC97_ANALOG], + .cpu_dai = &psc_ac97_dai[MPC5200_AC97_NORMAL], +}, +{ + .name = "AC97", + .stream_name = "AC97 IEC958", + .codec_dai = &stac9766_dai[STAC9766_DAI_AC97_DIGITAL], + .cpu_dai = &psc_ac97_dai[MPC5200_AC97_SPDIF], +}, +}; + +static __init int efika_fabric_init(void) +{ + struct platform_device *pdev; + int rc; + + if (!machine_is_compatible("bplan,efika")) + return -ENODEV; + + card.platform = &mpc5200_audio_dma_platform; + card.name = "Efika"; + card.dai_link = efika_fabric_dai; + card.num_links = ARRAY_SIZE(efika_fabric_dai); + + device.card = &card; + device.codec_dev = &soc_codec_dev_stac9766; + + pdev = platform_device_alloc("soc-audio", 1); + if (!pdev) { + pr_err("efika_fabric_init: platform_device_alloc() failed\n"); + return -ENODEV; + } + + platform_set_drvdata(pdev, &device); + device.dev = &pdev->dev; + + rc = platform_device_add(pdev); + if (rc) { + pr_err("efika_fabric_init: platform_device_add() failed\n"); + return -ENODEV; + } + return 0; +} + +module_init(efika_fabric_init); + + +MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>"); +MODULE_DESCRIPTION(DRV_NAME ": mpc5200 Efika fabric driver"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 3711d84..93f0f38 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -375,18 +375,14 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_pcm_runtime *first_runtime = ssi_private->first_stream->runtime; - if (!first_runtime->rate || !first_runtime->sample_bits) { + if (!first_runtime->sample_bits) { dev_err(substream->pcm->card->dev, - "set sample rate and size in %s stream first\n", + "set sample size in %s stream first\n", substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? "capture" : "playback"); return -EAGAIN; } - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - first_runtime->rate, first_runtime->rate); - /* If we're in synchronous mode, then we need to constrain * the sample size as well. We don't support independent sample * rates in asynchronous mode. @@ -674,7 +670,7 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) ssi_private->dev = ssi_info->dev; ssi_private->asynchronous = ssi_info->asynchronous; - ssi_private->dev->driver_data = fsl_ssi_dai; + dev_set_drvdata(ssi_private->dev, fsl_ssi_dai); /* Initialize the the device_attribute structure */ dev_attr->attr.name = "ssi-stats"; @@ -693,6 +689,7 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) fsl_ssi_dai->name = ssi_private->name; fsl_ssi_dai->id = ssi_info->id; fsl_ssi_dai->dev = ssi_info->dev; + fsl_ssi_dai->symmetric_rates = 1; ret = snd_soc_register_dai(fsl_ssi_dai); if (ret != 0) { diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c new file mode 100644 index 0000000..efec33a --- /dev/null +++ b/sound/soc/fsl/mpc5200_dma.c @@ -0,0 +1,564 @@ +/* + * Freescale MPC5200 PSC DMA + * ALSA SoC Platform driver + * + * Copyright (C) 2008 Secret Lab Technologies Ltd. + * Copyright (C) 2009 Jon Smirl, Digispeaker + */ + +#include <linux/module.h> +#include <linux/of_device.h> + +#include <sound/soc.h> + +#include <sysdev/bestcomm/bestcomm.h> +#include <sysdev/bestcomm/gen_bd.h> +#include <asm/mpc52xx_psc.h> + +#include "mpc5200_dma.h" + +/* + * Interrupt handlers + */ +static irqreturn_t psc_dma_status_irq(int irq, void *_psc_dma) +{ + struct psc_dma *psc_dma = _psc_dma; + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + u16 isr; + + isr = in_be16(®s->mpc52xx_psc_isr); + + /* Playback underrun error */ + if (psc_dma->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP)) + psc_dma->stats.underrun_count++; + + /* Capture overrun error */ + if (psc_dma->capture.active && (isr & MPC52xx_PSC_IMR_ORERR)) + psc_dma->stats.overrun_count++; + + out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT); + + return IRQ_HANDLED; +} + +/** + * psc_dma_bcom_enqueue_next_buffer - Enqueue another audio buffer + * @s: pointer to stream private data structure + * + * Enqueues another audio period buffer into the bestcomm queue. + * + * Note: The routine must only be called when there is space available in + * the queue. Otherwise the enqueue will fail and the audio ring buffer + * will get out of sync + */ +static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) +{ + struct bcom_bd *bd; + + /* 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; + 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) +{ + 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); + } +} + +/* Bestcomm DMA irq handler */ +static irqreturn_t psc_dma_bcom_irq_tx(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; + } + 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; + + psc_dma_bcom_enqueue_next_buffer(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 int psc_dma_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + return 0; +} + +/** + * psc_dma_trigger: start and stop the DMA transfer. + * + * This function is called by ALSA to start, stop, pause, and resume the DMA + * transfer of data. + */ +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 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: + 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->active = 1; + + /* track appl_ptr so that we have a better chance of detecting + * end of stream and not over running it. + */ + 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) { + 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 { + bcom_gen_bd_tx_reset(s->bcom_task); + psc_dma_bcom_enqueue_tx(s); + } + + bcom_enable(s->bcom_task); + spin_unlock_irqrestore(&psc_dma->lock, flags); + + out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT); + + break; + + case SNDRV_PCM_TRIGGER_STOP: + s->active = 0; + + spin_lock_irqsave(&psc_dma->lock, flags); + bcom_disable(s->bcom_task); + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + bcom_gen_bd_rx_reset(s->bcom_task); + else + bcom_gen_bd_tx_reset(s->bcom_task); + spin_unlock_irqrestore(&psc_dma->lock, flags); + + break; + + default: + dev_dbg(psc_dma->dev, "invalid command\n"); + return -EINVAL; + } + + /* Update interrupt enable settings */ + imr = 0; + if (psc_dma->playback.active) + imr |= MPC52xx_PSC_IMR_TXEMP; + if (psc_dma->capture.active) + imr |= MPC52xx_PSC_IMR_ORERR; + out_be16(®s->isr_imr.imr, psc_dma->imr | imr); + + return 0; +} + + +/* --------------------------------------------------------------------- + * The PSC DMA 'ASoC platform' driver + * + * Can be referenced by an 'ASoC machine' driver + * This driver only deals with the audio bus; it doesn't have any + * interaction with the attached codec + */ + +static const struct snd_pcm_hardware psc_dma_hardware = { + .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_BATCH, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .period_bytes_max = 1024 * 1024, + .period_bytes_min = 32, + .periods_min = 2, + .periods_max = 256, + .buffer_bytes_max = 2 * 1024 * 1024, + .fifo_size = 512, +}; + +static int psc_dma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + 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; + int rc; + + dev_dbg(psc_dma->dev, "psc_dma_open(substream=%p)\n", substream); + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + s = &psc_dma->capture; + else + s = &psc_dma->playback; + + snd_soc_set_runtime_hwparams(substream, &psc_dma_hardware); + + rc = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (rc < 0) { + dev_err(substream->pcm->card->dev, "invalid buffer size\n"); + return rc; + } + + s->stream = substream; + return 0; +} + +static int psc_dma_close(struct snd_pcm_substream *substream) +{ + 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; + + dev_dbg(psc_dma->dev, "psc_dma_close(substream=%p)\n", substream); + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + s = &psc_dma->capture; + else + s = &psc_dma->playback; + + if (!psc_dma->playback.active && + !psc_dma->capture.active) { + + /* Disable all interrupts and reset the PSC */ + out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); + out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */ + } + s->stream = NULL; + return 0; +} + +static snd_pcm_uframes_t +psc_dma_pointer(struct snd_pcm_substream *substream) +{ + 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; + dma_addr_t count; + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + s = &psc_dma->capture; + else + s = &psc_dma->playback; + + count = s->period_current_pt - s->period_start; + + return bytes_to_frames(substream->runtime, count); +} + +static int +psc_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static struct snd_pcm_ops psc_dma_ops = { + .open = psc_dma_open, + .close = psc_dma_close, + .hw_free = psc_dma_hw_free, + .ioctl = snd_pcm_lib_ioctl, + .pointer = psc_dma_pointer, + .trigger = psc_dma_trigger, + .hw_params = psc_dma_hw_params, +}; + +static u64 psc_dma_dmamask = 0xffffffff; +static int psc_dma_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *rtd = pcm->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + size_t size = psc_dma_hardware.buffer_bytes_max; + int rc = 0; + + dev_dbg(rtd->socdev->dev, "psc_dma_new(card=%p, dai=%p, pcm=%p)\n", + card, dai, pcm); + + if (!card->dev->dma_mask) + card->dev->dma_mask = &psc_dma_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (pcm->streams[0].substream) { + rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev, + size, &pcm->streams[0].substream->dma_buffer); + if (rc) + goto playback_alloc_err; + } + + if (pcm->streams[1].substream) { + rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev, + size, &pcm->streams[1].substream->dma_buffer); + if (rc) + goto capture_alloc_err; + } + + if (rtd->socdev->card->codec->ac97) + rtd->socdev->card->codec->ac97->private_data = psc_dma; + + return 0; + + capture_alloc_err: + if (pcm->streams[0].substream) + snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); + + playback_alloc_err: + dev_err(card->dev, "Cannot allocate buffer(s)\n"); + + return -ENOMEM; +} + +static void psc_dma_free(struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *rtd = pcm->private_data; + struct snd_pcm_substream *substream; + int stream; + + dev_dbg(rtd->socdev->dev, "psc_dma_free(pcm=%p)\n", pcm); + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (substream) { + snd_dma_free_pages(&substream->dma_buffer); + substream->dma_buffer.area = NULL; + substream->dma_buffer.addr = 0; + } + } +} + +struct snd_soc_platform mpc5200_audio_dma_platform = { + .name = "mpc5200-psc-audio", + .pcm_ops = &psc_dma_ops, + .pcm_new = &psc_dma_new, + .pcm_free = &psc_dma_free, +}; +EXPORT_SYMBOL_GPL(mpc5200_audio_dma_platform); + +int mpc5200_audio_dma_create(struct of_device *op) +{ + phys_addr_t fifo; + struct psc_dma *psc_dma; + struct resource res; + int size, irq, rc; + const __be32 *prop; + void __iomem *regs; + + /* Fetch the registers and IRQ of the PSC */ + irq = irq_of_parse_and_map(op->node, 0); + if (of_address_to_resource(op->node, 0, &res)) { + dev_err(&op->dev, "Missing reg property\n"); + return -ENODEV; + } + regs = ioremap(res.start, 1 + res.end - res.start); + if (!regs) { + dev_err(&op->dev, "Could not map registers\n"); + return -ENODEV; + } + + /* Allocate and initialize the driver private data */ + psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL); + if (!psc_dma) { + iounmap(regs); + return -ENOMEM; + } + + /* Get the PSC ID */ + prop = of_get_property(op->node, "cell-index", &size); + if (!prop || size < sizeof *prop) + return -ENODEV; + + spin_lock_init(&psc_dma->lock); + psc_dma->id = be32_to_cpu(*prop); + psc_dma->irq = irq; + psc_dma->psc_regs = regs; + psc_dma->fifo_regs = regs + sizeof *psc_dma->psc_regs; + psc_dma->dev = &op->dev; + psc_dma->playback.psc_dma = psc_dma; + psc_dma->capture.psc_dma = psc_dma; + snprintf(psc_dma->name, sizeof psc_dma->name, "PSC%u", psc_dma->id); + + /* Find the address of the fifo data registers and setup the + * DMA tasks */ + fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32); + psc_dma->capture.bcom_task = + bcom_psc_gen_bd_rx_init(psc_dma->id, 10, fifo, 512); + psc_dma->playback.bcom_task = + bcom_psc_gen_bd_tx_init(psc_dma->id, 10, fifo); + if (!psc_dma->capture.bcom_task || + !psc_dma->playback.bcom_task) { + dev_err(&op->dev, "Could not allocate bestcomm tasks\n"); + iounmap(regs); + kfree(psc_dma); + return -ENODEV; + } + + /* Disable all interrupts and reset the PSC */ + out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); + /* reset receiver */ + out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_RX); + /* reset transmitter */ + out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_TX); + /* reset error */ + out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_ERR_STAT); + /* reset mode */ + out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_SEL_MODE_REG_1); + + /* Set up mode register; + * First write: RxRdy (FIFO Alarm) generates rx FIFO irq + * Second write: register Normal mode for non loopback + */ + out_8(&psc_dma->psc_regs->mode, 0); + out_8(&psc_dma->psc_regs->mode, 0); + + /* Set the TX and RX fifo alarm thresholds */ + out_be16(&psc_dma->fifo_regs->rfalarm, 0x100); + out_8(&psc_dma->fifo_regs->rfcntl, 0x4); + out_be16(&psc_dma->fifo_regs->tfalarm, 0x100); + out_8(&psc_dma->fifo_regs->tfcntl, 0x7); + + /* Lookup the IRQ numbers */ + psc_dma->playback.irq = + bcom_get_task_irq(psc_dma->playback.bcom_task); + psc_dma->capture.irq = + bcom_get_task_irq(psc_dma->capture.bcom_task); + + 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, + "psc-dma-capture", &psc_dma->capture); + rc |= request_irq(psc_dma->playback.irq, + &psc_dma_bcom_irq_tx, IRQF_SHARED, + "psc-dma-playback", &psc_dma->playback); + if (rc) { + free_irq(psc_dma->irq, psc_dma); + free_irq(psc_dma->capture.irq, + &psc_dma->capture); + free_irq(psc_dma->playback.irq, + &psc_dma->playback); + return -ENODEV; + } + + /* Save what we've done so it can be found again later */ + dev_set_drvdata(&op->dev, psc_dma); + + /* Tell the ASoC OF helpers about it */ + return snd_soc_register_platform(&mpc5200_audio_dma_platform); +} +EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create); + +int mpc5200_audio_dma_destroy(struct of_device *op) +{ + struct psc_dma *psc_dma = dev_get_drvdata(&op->dev); + + dev_dbg(&op->dev, "mpc5200_audio_dma_destroy()\n"); + + snd_soc_unregister_platform(&mpc5200_audio_dma_platform); + + bcom_gen_bd_rx_release(psc_dma->capture.bcom_task); + bcom_gen_bd_tx_release(psc_dma->playback.bcom_task); + + /* Release irqs */ + free_irq(psc_dma->irq, psc_dma); + free_irq(psc_dma->capture.irq, &psc_dma->capture); + free_irq(psc_dma->playback.irq, &psc_dma->playback); + + iounmap(psc_dma->psc_regs); + kfree(psc_dma); + dev_set_drvdata(&op->dev, NULL); + + return 0; +} +EXPORT_SYMBOL_GPL(mpc5200_audio_dma_destroy); + +MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>"); +MODULE_DESCRIPTION("Freescale MPC5200 PSC in DMA mode ASoC Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h new file mode 100644 index 0000000..2000803 --- /dev/null +++ b/sound/soc/fsl/mpc5200_dma.h @@ -0,0 +1,80 @@ +/* + * Freescale MPC5200 Audio DMA driver + */ + +#ifndef __SOUND_SOC_FSL_MPC5200_DMA_H__ +#define __SOUND_SOC_FSL_MPC5200_DMA_H__ + +#define PSC_STREAM_NAME_LEN 32 + +/** + * psc_ac97_stream - Data specific to a single stream (playback or capture) + * @active: flag indicating if the stream is active + * @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 + */ +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_bytes; + int period_size; +}; + +/** + * psc_dma - Private driver data + * @name: short name for this device ("PSC0", "PSC1", etc) + * @psc_regs: pointer to the PSC's registers + * @fifo_regs: pointer to the PSC's FIFO registers + * @irq: IRQ of this PSC + * @dev: struct device pointer + * @dai: the CPU DAI for this device + * @sicr: Base value used in serial interface control register; mode is ORed + * with this value. + * @playback: Playback stream context data + * @capture: Capture stream context data + */ +struct psc_dma { + char name[32]; + struct mpc52xx_psc __iomem *psc_regs; + struct mpc52xx_psc_fifo __iomem *fifo_regs; + unsigned int irq; + struct device *dev; + spinlock_t lock; + u32 sicr; + uint sysclk; + int imr; + int id; + unsigned int slots; + + /* per-stream data */ + struct psc_dma_stream playback; + struct psc_dma_stream capture; + + /* Statistics */ + struct { + unsigned long overrun_count; + unsigned long underrun_count; + } stats; +}; + +int mpc5200_audio_dma_create(struct of_device *op); +int mpc5200_audio_dma_destroy(struct of_device *op); + +extern struct snd_soc_platform mpc5200_audio_dma_platform; + +#endif /* __SOUND_SOC_FSL_MPC5200_DMA_H__ */ diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c new file mode 100644 index 0000000..794a247 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -0,0 +1,329 @@ +/* + * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip. + * + * Copyright (C) 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl@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 + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/of_device.h> +#include <linux/of_platform.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/time.h> +#include <asm/delay.h> +#include <asm/mpc52xx_psc.h> + +#include "mpc5200_dma.h" +#include "mpc5200_psc_ac97.h" + +#define DRV_NAME "mpc5200-psc-ac97" + +/* ALSA only supports a single AC97 device so static is recommend here */ +static struct psc_dma *psc_dma; + +static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + int status; + unsigned int val; + + /* Wait for command send status zero = ready */ + status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND), 100, 0); + if (status == 0) { + pr_err("timeout on ac97 bus (rdy)\n"); + return -ENODEV; + } + /* Send the read */ + out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); + + /* Wait for the answer */ + status = spin_event_timeout((in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_DATA_VAL), 100, 0); + if (status == 0) { + pr_err("timeout on ac97 read (val) %x\n", + in_be16(&psc_dma->psc_regs->sr_csr.status)); + return -ENODEV; + } + /* Get the data */ + val = in_be32(&psc_dma->psc_regs->ac97_data); + if (((val >> 24) & 0x7f) != reg) { + pr_err("reg echo error on ac97 read\n"); + return -ENODEV; + } + val = (val >> 8) & 0xffff; + + return (unsigned short) val; +} + +static void psc_ac97_write(struct snd_ac97 *ac97, + unsigned short reg, unsigned short val) +{ + int status; + + /* Wait for command status zero = ready */ + status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND), 100, 0); + if (status == 0) { + pr_err("timeout on ac97 bus (write)\n"); + return; + } + /* Write data */ + out_be32(&psc_dma->psc_regs->ac97_cmd, + ((reg & 0x7f) << 24) | (val << 8)); +} + +static void psc_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + out_be32(®s->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR); + udelay(3); + out_be32(®s->sicr, psc_dma->sicr); +} + +static void psc_ac97_cold_reset(struct snd_ac97 *ac97) +{ + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + /* Do a cold reset */ + out_8(®s->op1, MPC52xx_PSC_OP_RES); + udelay(10); + out_8(®s->op0, MPC52xx_PSC_OP_RES); + udelay(50); + psc_ac97_warm_reset(ac97); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = psc_ac97_read, + .write = psc_ac97_write, + .reset = psc_ac97_cold_reset, + .warm_reset = psc_ac97_warm_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct psc_dma *psc_dma = cpu_dai->private_data; + + dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" + " periods=%i buffer_size=%i buffer_bytes=%i channels=%i" + " rate=%i format=%i\n", + __func__, substream, params_period_size(params), + params_period_bytes(params), params_periods(params), + params_buffer_size(params), params_buffer_bytes(params), + 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); + + return 0; +} + +static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct psc_dma *psc_dma = cpu_dai->private_data; + + if (params_channels(params) == 1) + out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000); + else + out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000); + + return 0; +} + +static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + psc_dma->slots &= 0xFFFF0000; + else + psc_dma->slots &= 0x0000FFFF; + + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + break; + } + return 0; +} + +static int psc_ac97_probe(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct psc_dma *psc_dma = cpu_dai->private_data; + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + /* Go */ + out_8(®s->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE); + return 0; +} + +/* --------------------------------------------------------------------- + * ALSA SoC Bindings + * + * - Digital Audio Interface (DAI) template + * - create/destroy dai hooks + */ + +/** + * psc_ac97_dai_template: template CPU Digital Audio Interface + */ +static struct snd_soc_dai_ops psc_ac97_analog_ops = { + .hw_params = psc_ac97_hw_analog_params, + .trigger = psc_ac97_trigger, +}; + +static struct snd_soc_dai_ops psc_ac97_digital_ops = { + .hw_params = psc_ac97_hw_digital_params, +}; + +struct snd_soc_dai psc_ac97_dai[] = { +{ + .name = "AC97", + .ac97_control = 1, + .probe = psc_ac97_probe, + .playback = { + .channels_min = 1, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .ops = &psc_ac97_analog_ops, +}, +{ + .name = "SPDIF", + .ac97_control = 1, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE, + }, + .ops = &psc_ac97_digital_ops, +} }; +EXPORT_SYMBOL_GPL(psc_ac97_dai); + + + +/* --------------------------------------------------------------------- + * OF platform bus binding code: + * - Probe/remove operations + * - OF device match table + */ +static int __devinit psc_ac97_of_probe(struct of_device *op, + const struct of_device_id *match) +{ + int rc, i; + struct snd_ac97 ac97; + struct mpc52xx_psc __iomem *regs; + + rc = mpc5200_audio_dma_create(op); + if (rc != 0) + return rc; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].dev = &op->dev; + + rc = snd_soc_register_dais(psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai)); + if (rc != 0) { + dev_err(&op->dev, "Failed to register DAI\n"); + return rc; + } + + psc_dma = dev_get_drvdata(&op->dev); + regs = psc_dma->psc_regs; + ac97.private_data = psc_dma; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].private_data = psc_dma; + + psc_dma->imr = 0; + out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); + + /* Configure the serial interface mode to AC97 */ + psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97; + out_be32(®s->sicr, psc_dma->sicr); + + /* No slots active */ + out_be32(®s->ac97_slots, 0x00000000); + + return 0; +} + +static int __devexit psc_ac97_of_remove(struct of_device *op) +{ + return mpc5200_audio_dma_destroy(op); +} + +/* Match table for of_platform binding */ +static struct of_device_id psc_ac97_match[] __devinitdata = { + { .compatible = "fsl,mpc5200-psc-ac97", }, + { .compatible = "fsl,mpc5200b-psc-ac97", }, + {} +}; +MODULE_DEVICE_TABLE(of, psc_ac97_match); + +static struct of_platform_driver psc_ac97_driver = { + .match_table = psc_ac97_match, + .probe = psc_ac97_of_probe, + .remove = __devexit_p(psc_ac97_of_remove), + .driver = { + .name = "mpc5200-psc-ac97", + .owner = THIS_MODULE, + }, +}; + +/* --------------------------------------------------------------------- + * Module setup and teardown; simply register the of_platform driver + * for the PSC in AC97 mode. + */ +static int __init psc_ac97_init(void) +{ + return of_register_platform_driver(&psc_ac97_driver); +} +module_init(psc_ac97_init); + +static void __exit psc_ac97_exit(void) +{ + of_unregister_platform_driver(&psc_ac97_driver); +} +module_exit(psc_ac97_exit); + +MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>"); +MODULE_DESCRIPTION("mpc5200 AC97 module"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h new file mode 100644 index 0000000..4bc18c3 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.h @@ -0,0 +1,15 @@ +/* + * Freescale MPC5200 PSC in AC97 mode + * ALSA SoC Digital Audio Interface (DAI) driver + * + */ + +#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ +#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ + +extern struct snd_soc_dai psc_ac97_dai[]; + +#define MPC5200_AC97_NORMAL 0 +#define MPC5200_AC97_SPDIF 1 + +#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */ diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 1111c71..ce8de90 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -3,31 +3,21 @@ * ALSA SoC Digital Audio Interface (DAI) driver * * Copyright (C) 2008 Secret Lab Technologies Ltd. + * Copyright (C) 2009 Jon Smirl, Digispeaker */ -#include <linux/init.h> #include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/device.h> -#include <linux/delay.h> #include <linux/of_device.h> #include <linux/of_platform.h> -#include <linux/dma-mapping.h> -#include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> -#include <sound/initval.h> #include <sound/soc.h> -#include <sound/soc-of-simple.h> -#include <sysdev/bestcomm/bestcomm.h> -#include <sysdev/bestcomm/gen_bd.h> #include <asm/mpc52xx_psc.h> -MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>"); -MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver"); -MODULE_LICENSE("GPL"); +#include "mpc5200_psc_i2s.h" +#include "mpc5200_dma.h" /** * PSC_I2S_RATES: sample rates supported by the I2S @@ -44,191 +34,17 @@ MODULE_LICENSE("GPL"); * PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode */ #define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \ - SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \ - SNDRV_PCM_FMTBIT_S32_BE) - -/** - * psc_i2s_stream - Data specific to a single stream (playback or capture) - * @active: flag indicating if the stream is active - * @psc_i2s: pointer back to parent psc_i2s 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 - */ -struct psc_i2s_stream { - int active; - struct psc_i2s *psc_i2s; - 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_bytes; -}; - -/** - * psc_i2s - Private driver data - * @name: short name for this device ("PSC0", "PSC1", etc) - * @psc_regs: pointer to the PSC's registers - * @fifo_regs: pointer to the PSC's FIFO registers - * @irq: IRQ of this PSC - * @dev: struct device pointer - * @dai: the CPU DAI for this device - * @sicr: Base value used in serial interface control register; mode is ORed - * with this value. - * @playback: Playback stream context data - * @capture: Capture stream context data - */ -struct psc_i2s { - char name[32]; - struct mpc52xx_psc __iomem *psc_regs; - struct mpc52xx_psc_fifo __iomem *fifo_regs; - unsigned int irq; - struct device *dev; - struct snd_soc_dai dai; - spinlock_t lock; - u32 sicr; - - /* per-stream data */ - struct psc_i2s_stream playback; - struct psc_i2s_stream capture; - - /* Statistics */ - struct { - int overrun_count; - int underrun_count; - } stats; -}; - -/* - * Interrupt handlers - */ -static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s) -{ - struct psc_i2s *psc_i2s = _psc_i2s; - struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs; - u16 isr; - - isr = in_be16(®s->mpc52xx_psc_isr); - - /* Playback underrun error */ - if (psc_i2s->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP)) - psc_i2s->stats.underrun_count++; - - /* Capture overrun error */ - if (psc_i2s->capture.active && (isr & MPC52xx_PSC_IMR_ORERR)) - psc_i2s->stats.overrun_count++; - - out_8(®s->command, 4 << 4); /* reset the error status */ - - return IRQ_HANDLED; -} - -/** - * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer - * @s: pointer to stream private data structure - * - * Enqueues another audio period buffer into the bestcomm queue. - * - * Note: The routine must only be called when there is space available in - * the queue. Otherwise the enqueue will fail and the audio ring buffer - * will get out of sync - */ -static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s) -{ - struct bcom_bd *bd; - - /* 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; - 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; -} - -/* Bestcomm DMA irq handler */ -static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream) -{ - struct psc_i2s_stream *s = _psc_i2s_stream; - - /* 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; - psc_i2s_bcom_enqueue_next_buffer(s); - bcom_enable(s->bcom_task); - } - - /* 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; -} - -/** - * psc_i2s_startup: create a new substream - * - * This is the first function called when a stream is opened. - * - * If this is the first stream open, then grab the IRQ and program most of - * the PSC registers. - */ -static int psc_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; - int rc; - - dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream); - - if (!psc_i2s->playback.active && - !psc_i2s->capture.active) { - /* Setup the IRQs */ - rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED, - "psc-i2s-status", psc_i2s); - rc |= request_irq(psc_i2s->capture.irq, - &psc_i2s_bcom_irq, IRQF_SHARED, - "psc-i2s-capture", &psc_i2s->capture); - rc |= request_irq(psc_i2s->playback.irq, - &psc_i2s_bcom_irq, IRQF_SHARED, - "psc-i2s-playback", &psc_i2s->playback); - if (rc) { - free_irq(psc_i2s->irq, psc_i2s); - free_irq(psc_i2s->capture.irq, - &psc_i2s->capture); - free_irq(psc_i2s->playback.irq, - &psc_i2s->playback); - return -ENODEV; - } - } - - return 0; -} + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) static int psc_i2s_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 psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; u32 mode; - dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i" + dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" " periods=%i buffer_size=%i buffer_bytes=%i\n", __func__, substream, params_period_size(params), params_period_bytes(params), params_periods(params), @@ -248,175 +64,15 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream, mode = MPC52xx_PSC_SICR_SIM_CODEC_32; break; default: - dev_dbg(psc_i2s->dev, "invalid format\n"); - return -EINVAL; - } - out_be32(&psc_i2s->psc_regs->sicr, psc_i2s->sicr | mode); - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - return 0; -} - -static int psc_i2s_hw_free(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - snd_pcm_set_runtime_buffer(substream, NULL); - return 0; -} - -/** - * psc_i2s_trigger: start and stop the DMA transfer. - * - * This function is called by ALSA to start, stop, pause, and resume the DMA - * transfer of data. - */ -static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct psc_i2s_stream *s; - struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs; - u16 imr; - u8 psc_cmd; - unsigned long flags; - - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - s = &psc_i2s->capture; - else - s = &psc_i2s->playback; - - dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)" - " stream_id=%i\n", - substream, cmd, substream->pstr->stream); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - 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->active = 1; - - /* First; reset everything */ - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { - out_8(®s->command, MPC52xx_PSC_RST_RX); - out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT); - } else { - out_8(®s->command, MPC52xx_PSC_RST_TX); - out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT); - } - - /* Next, fill up the bestcomm bd queue and enable DMA. - * This will begin filling the PSC's fifo. */ - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - bcom_gen_bd_rx_reset(s->bcom_task); - else - bcom_gen_bd_tx_reset(s->bcom_task); - while (!bcom_queue_full(s->bcom_task)) - psc_i2s_bcom_enqueue_next_buffer(s); - bcom_enable(s->bcom_task); - - /* Due to errata in the i2s mode; need to line up enabling - * the transmitter with a transition on the frame sync - * line */ - - spin_lock_irqsave(&psc_i2s->lock, flags); - /* first make sure it is low */ - while ((in_8(®s->ipcr_acr.ipcr) & 0x80) != 0) - ; - /* then wait for the transition to high */ - while ((in_8(®s->ipcr_acr.ipcr) & 0x80) == 0) - ; - /* Finally, enable the PSC. - * Receiver must always be enabled; even when we only want - * transmit. (see 15.3.2.3 of MPC5200B User's Guide) */ - psc_cmd = MPC52xx_PSC_RX_ENABLE; - if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK) - psc_cmd |= MPC52xx_PSC_TX_ENABLE; - out_8(®s->command, psc_cmd); - spin_unlock_irqrestore(&psc_i2s->lock, flags); - - break; - - case SNDRV_PCM_TRIGGER_STOP: - /* Turn off the PSC */ - s->active = 0; - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { - if (!psc_i2s->playback.active) { - out_8(®s->command, 2 << 4); /* reset rx */ - out_8(®s->command, 3 << 4); /* reset tx */ - out_8(®s->command, 4 << 4); /* reset err */ - } - } else { - out_8(®s->command, 3 << 4); /* reset tx */ - out_8(®s->command, 4 << 4); /* reset err */ - if (!psc_i2s->capture.active) - out_8(®s->command, 2 << 4); /* reset rx */ - } - - bcom_disable(s->bcom_task); - while (!bcom_queue_empty(s->bcom_task)) - bcom_retrieve_buffer(s->bcom_task, NULL, NULL); - - break; - - default: - dev_dbg(psc_i2s->dev, "invalid command\n"); + dev_dbg(psc_dma->dev, "invalid format\n"); return -EINVAL; } - - /* Update interrupt enable settings */ - imr = 0; - if (psc_i2s->playback.active) - imr |= MPC52xx_PSC_IMR_TXEMP; - if (psc_i2s->capture.active) - imr |= MPC52xx_PSC_IMR_ORERR; - out_be16(®s->isr_imr.imr, imr); + out_be32(&psc_dma->psc_regs->sicr, psc_dma->sicr | mode); return 0; } /** - * psc_i2s_shutdown: shutdown the data transfer on a stream - * - * Shutdown the PSC if there are no other substreams open. - */ -static void psc_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; - - dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream); - - /* - * If this is the last active substream, disable the PSC and release - * the IRQ. - */ - if (!psc_i2s->playback.active && - !psc_i2s->capture.active) { - - /* Disable all interrupts and reset the PSC */ - out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0); - out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset tx */ - out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset rx */ - out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */ - out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */ - - /* Release irqs */ - free_irq(psc_i2s->irq, psc_i2s); - free_irq(psc_i2s->capture.irq, &psc_i2s->capture); - free_irq(psc_i2s->playback.irq, &psc_i2s->playback); - } -} - -/** * psc_i2s_set_sysclk: set the clock frequency and direction * * This function is called by the machine driver to tell us what the clock @@ -433,8 +89,8 @@ static void psc_i2s_shutdown(struct snd_pcm_substream *substream, static int psc_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { - struct psc_i2s *psc_i2s = cpu_dai->private_data; - dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n", + struct psc_dma *psc_dma = cpu_dai->private_data; + dev_dbg(psc_dma->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n", cpu_dai, dir); return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL; } @@ -452,8 +108,8 @@ static int psc_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, */ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) { - struct psc_i2s *psc_i2s = cpu_dai->private_data; - dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n", + struct psc_dma *psc_dma = cpu_dai->private_data; + dev_dbg(psc_dma->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n", cpu_dai, format); return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL; } @@ -469,16 +125,13 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) * psc_i2s_dai_template: template CPU Digital Audio Interface */ static struct snd_soc_dai_ops psc_i2s_dai_ops = { - .startup = psc_i2s_startup, .hw_params = psc_i2s_hw_params, - .hw_free = psc_i2s_hw_free, - .shutdown = psc_i2s_shutdown, - .trigger = psc_i2s_trigger, .set_sysclk = psc_i2s_set_sysclk, .set_fmt = psc_i2s_set_fmt, }; -static struct snd_soc_dai psc_i2s_dai_template = { +struct snd_soc_dai psc_i2s_dai[] = {{ + .name = "I2S", .playback = { .channels_min = 2, .channels_max = 2, @@ -492,223 +145,8 @@ static struct snd_soc_dai psc_i2s_dai_template = { .formats = PSC_I2S_FORMATS, }, .ops = &psc_i2s_dai_ops, -}; - -/* --------------------------------------------------------------------- - * The PSC I2S 'ASoC platform' driver - * - * Can be referenced by an 'ASoC machine' driver - * This driver only deals with the audio bus; it doesn't have any - * interaction with the attached codec - */ - -static const struct snd_pcm_hardware psc_i2s_pcm_hardware = { - .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_BATCH, - .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | - SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .period_bytes_max = 1024 * 1024, - .period_bytes_min = 32, - .periods_min = 2, - .periods_max = 256, - .buffer_bytes_max = 2 * 1024 * 1024, - .fifo_size = 0, -}; - -static int psc_i2s_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; - struct psc_i2s_stream *s; - - dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream); - - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - s = &psc_i2s->capture; - else - s = &psc_i2s->playback; - - snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware); - - s->stream = substream; - return 0; -} - -static int psc_i2s_pcm_close(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; - struct psc_i2s_stream *s; - - dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream); - - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - s = &psc_i2s->capture; - else - s = &psc_i2s->playback; - - s->stream = NULL; - return 0; -} - -static snd_pcm_uframes_t -psc_i2s_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; - struct psc_i2s_stream *s; - dma_addr_t count; - - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - s = &psc_i2s->capture; - else - s = &psc_i2s->playback; - - count = s->period_current_pt - s->period_start; - - return bytes_to_frames(substream->runtime, count); -} - -static struct snd_pcm_ops psc_i2s_pcm_ops = { - .open = psc_i2s_pcm_open, - .close = psc_i2s_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .pointer = psc_i2s_pcm_pointer, -}; - -static u64 psc_i2s_pcm_dmamask = 0xffffffff; -static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) -{ - struct snd_soc_pcm_runtime *rtd = pcm->private_data; - size_t size = psc_i2s_pcm_hardware.buffer_bytes_max; - int rc = 0; - - dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n", - card, dai, pcm); - - if (!card->dev->dma_mask) - card->dev->dma_mask = &psc_i2s_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (pcm->streams[0].substream) { - rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size, - &pcm->streams[0].substream->dma_buffer); - if (rc) - goto playback_alloc_err; - } - - if (pcm->streams[1].substream) { - rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size, - &pcm->streams[1].substream->dma_buffer); - if (rc) - goto capture_alloc_err; - } - - return 0; - - capture_alloc_err: - if (pcm->streams[0].substream) - snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); - playback_alloc_err: - dev_err(card->dev, "Cannot allocate buffer(s)\n"); - return -ENOMEM; -} - -static void psc_i2s_pcm_free(struct snd_pcm *pcm) -{ - struct snd_soc_pcm_runtime *rtd = pcm->private_data; - struct snd_pcm_substream *substream; - int stream; - - dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm); - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (substream) { - snd_dma_free_pages(&substream->dma_buffer); - substream->dma_buffer.area = NULL; - substream->dma_buffer.addr = 0; - } - } -} - -struct snd_soc_platform psc_i2s_pcm_soc_platform = { - .name = "mpc5200-psc-audio", - .pcm_ops = &psc_i2s_pcm_ops, - .pcm_new = &psc_i2s_pcm_new, - .pcm_free = &psc_i2s_pcm_free, -}; - -/* --------------------------------------------------------------------- - * Sysfs attributes for debugging - */ - -static ssize_t psc_i2s_status_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - struct psc_i2s *psc_i2s = dev_get_drvdata(dev); - - return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x " - "tfnum=%i tfstat=0x%.4x\n", - in_be16(&psc_i2s->psc_regs->sr_csr.status), - in_be32(&psc_i2s->psc_regs->sicr), - in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff, - in_be16(&psc_i2s->fifo_regs->rfstat), - in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff, - in_be16(&psc_i2s->fifo_regs->tfstat)); -} - -static int *psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s, const char *name) -{ - if (strcmp(name, "playback_underrun") == 0) - return &psc_i2s->stats.underrun_count; - if (strcmp(name, "capture_overrun") == 0) - return &psc_i2s->stats.overrun_count; - - return NULL; -} - -static ssize_t psc_i2s_stat_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - struct psc_i2s *psc_i2s = dev_get_drvdata(dev); - int *attrib; - - attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name); - if (!attrib) - return 0; - - return sprintf(buf, "%i\n", *attrib); -} - -static ssize_t psc_i2s_stat_store(struct device *dev, - struct device_attribute *attr, - const char *buf, - size_t count) -{ - struct psc_i2s *psc_i2s = dev_get_drvdata(dev); - int *attrib; - - attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name); - if (!attrib) - return 0; - - *attrib = simple_strtoul(buf, NULL, 0); - return count; -} - -static DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL); -static DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show, - psc_i2s_stat_store); -static DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, - psc_i2s_stat_store); +} }; +EXPORT_SYMBOL_GPL(psc_i2s_dai); /* --------------------------------------------------------------------- * OF platform bus binding code: @@ -718,150 +156,65 @@ static DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, static int __devinit psc_i2s_of_probe(struct of_device *op, const struct of_device_id *match) { - phys_addr_t fifo; - struct psc_i2s *psc_i2s; - struct resource res; - int size, psc_id, irq, rc; - const __be32 *prop; - void __iomem *regs; - - dev_dbg(&op->dev, "probing psc i2s device\n"); - - /* Get the PSC ID */ - prop = of_get_property(op->node, "cell-index", &size); - if (!prop || size < sizeof *prop) - return -ENODEV; - psc_id = be32_to_cpu(*prop); - - /* Fetch the registers and IRQ of the PSC */ - irq = irq_of_parse_and_map(op->node, 0); - if (of_address_to_resource(op->node, 0, &res)) { - dev_err(&op->dev, "Missing reg property\n"); - return -ENODEV; - } - regs = ioremap(res.start, 1 + res.end - res.start); - if (!regs) { - dev_err(&op->dev, "Could not map registers\n"); - return -ENODEV; - } + int rc; + struct psc_dma *psc_dma; + struct mpc52xx_psc __iomem *regs; - /* Allocate and initialize the driver private data */ - psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL); - if (!psc_i2s) { - iounmap(regs); - return -ENOMEM; - } - spin_lock_init(&psc_i2s->lock); - psc_i2s->irq = irq; - psc_i2s->psc_regs = regs; - psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs; - psc_i2s->dev = &op->dev; - psc_i2s->playback.psc_i2s = psc_i2s; - psc_i2s->capture.psc_i2s = psc_i2s; - snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1); - - /* Fill out the CPU DAI structure */ - memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai); - psc_i2s->dai.private_data = psc_i2s; - psc_i2s->dai.name = psc_i2s->name; - psc_i2s->dai.id = psc_id; - - /* Find the address of the fifo data registers and setup the - * DMA tasks */ - fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32); - psc_i2s->capture.bcom_task = - bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512); - psc_i2s->playback.bcom_task = - bcom_psc_gen_bd_tx_init(psc_id, 10, fifo); - if (!psc_i2s->capture.bcom_task || - !psc_i2s->playback.bcom_task) { - dev_err(&op->dev, "Could not allocate bestcomm tasks\n"); - iounmap(regs); - kfree(psc_i2s); - return -ENODEV; + rc = mpc5200_audio_dma_create(op); + if (rc != 0) + return rc; + + rc = snd_soc_register_dais(psc_i2s_dai, ARRAY_SIZE(psc_i2s_dai)); + if (rc != 0) { + pr_err("Failed to register DAI\n"); + return 0; } - /* Disable all interrupts and reset the PSC */ - out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0); - out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset transmitter */ - out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset receiver */ - out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */ - out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */ + psc_dma = dev_get_drvdata(&op->dev); + regs = psc_dma->psc_regs; /* Configure the serial interface mode; defaulting to CODEC8 mode */ - psc_i2s->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S | + psc_dma->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S | MPC52xx_PSC_SICR_CLKPOL; - if (of_get_property(op->node, "fsl,cellslave", NULL)) - psc_i2s->sicr |= MPC52xx_PSC_SICR_CELLSLAVE | - MPC52xx_PSC_SICR_GENCLK; - out_be32(&psc_i2s->psc_regs->sicr, - psc_i2s->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8); + out_be32(&psc_dma->psc_regs->sicr, + psc_dma->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8); /* Check for the codec handle. If it is not present then we * are done */ if (!of_get_property(op->node, "codec-handle", NULL)) return 0; - /* Set up mode register; - * First write: RxRdy (FIFO Alarm) generates rx FIFO irq - * Second write: register Normal mode for non loopback - */ - out_8(&psc_i2s->psc_regs->mode, 0); - out_8(&psc_i2s->psc_regs->mode, 0); - - /* Set the TX and RX fifo alarm thresholds */ - out_be16(&psc_i2s->fifo_regs->rfalarm, 0x100); - out_8(&psc_i2s->fifo_regs->rfcntl, 0x4); - out_be16(&psc_i2s->fifo_regs->tfalarm, 0x100); - out_8(&psc_i2s->fifo_regs->tfcntl, 0x7); - - /* Lookup the IRQ numbers */ - psc_i2s->playback.irq = - bcom_get_task_irq(psc_i2s->playback.bcom_task); - psc_i2s->capture.irq = - bcom_get_task_irq(psc_i2s->capture.bcom_task); - - /* Save what we've done so it can be found again later */ - dev_set_drvdata(&op->dev, psc_i2s); - - /* Register the SYSFS files */ - rc = device_create_file(psc_i2s->dev, &dev_attr_status); - rc |= device_create_file(psc_i2s->dev, &dev_attr_capture_overrun); - rc |= device_create_file(psc_i2s->dev, &dev_attr_playback_underrun); - if (rc) - dev_info(psc_i2s->dev, "error creating sysfs files\n"); - - snd_soc_register_platform(&psc_i2s_pcm_soc_platform); - - /* Tell the ASoC OF helpers about it */ - of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node, - &psc_i2s->dai); + /* Due to errata in the dma mode; need to line up enabling + * the transmitter with a transition on the frame sync + * line */ + + /* first make sure it is low */ + while ((in_8(®s->ipcr_acr.ipcr) & 0x80) != 0) + ; + /* then wait for the transition to high */ + while ((in_8(®s->ipcr_acr.ipcr) & 0x80) == 0) + ; + /* Finally, enable the PSC. + * Receiver must always be enabled; even when we only want + * transmit. (see 15.3.2.3 of MPC5200B User's Guide) */ + + /* Go */ + out_8(&psc_dma->psc_regs->command, + MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE); return 0; + } static int __devexit psc_i2s_of_remove(struct of_device *op) { - struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev); - - dev_dbg(&op->dev, "psc_i2s_remove()\n"); - - snd_soc_unregister_platform(&psc_i2s_pcm_soc_platform); - - bcom_gen_bd_rx_release(psc_i2s->capture.bcom_task); - bcom_gen_bd_tx_release(psc_i2s->playback.bcom_task); - - iounmap(psc_i2s->psc_regs); - iounmap(psc_i2s->fifo_regs); - kfree(psc_i2s); - dev_set_drvdata(&op->dev, NULL); - - return 0; + return mpc5200_audio_dma_destroy(op); } /* Match table for of_platform binding */ static struct of_device_id psc_i2s_match[] __devinitdata = { { .compatible = "fsl,mpc5200-psc-i2s", }, + { .compatible = "fsl,mpc5200b-psc-i2s", }, {} }; MODULE_DEVICE_TABLE(of, psc_i2s_match); @@ -892,4 +245,7 @@ static void __exit psc_i2s_exit(void) } module_exit(psc_i2s_exit); +MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>"); +MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/mpc5200_psc_i2s.h b/sound/soc/fsl/mpc5200_psc_i2s.h new file mode 100644 index 0000000..ce55e07 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_i2s.h @@ -0,0 +1,12 @@ +/* + * Freescale MPC5200 PSC in I2S mode + * ALSA SoC Digital Audio Interface (DAI) driver + * + */ + +#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ +#define __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ + +extern struct snd_soc_dai psc_i2s_dai[]; + +#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ */ diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c new file mode 100644 index 0000000..8766f7a --- /dev/null +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -0,0 +1,90 @@ +/* + * Phytec pcm030 driver for the PSC of the Freescale MPC52xx + * configured as AC97 interface + * + * Copyright 2008 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl@gmail.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/of_device.h> +#include <linux/of_platform.h> +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/soc-of-simple.h> + +#include "mpc5200_dma.h" +#include "mpc5200_psc_ac97.h" +#include "../codecs/wm9712.h" + +static struct snd_soc_device device; +static struct snd_soc_card card; + +static struct snd_soc_dai_link pcm030_fabric_dai[] = { +{ + .name = "AC97", + .stream_name = "AC97 Analog", + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], + .cpu_dai = &psc_ac97_dai[MPC5200_AC97_NORMAL], +}, +{ + .name = "AC97", + .stream_name = "AC97 IEC958", + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], + .cpu_dai = &psc_ac97_dai[MPC5200_AC97_SPDIF], +}, +}; + +static __init int pcm030_fabric_init(void) +{ + struct platform_device *pdev; + int rc; + + if (!machine_is_compatible("phytec,pcm030")) + return -ENODEV; + + card.platform = &mpc5200_audio_dma_platform; + card.name = "pcm030"; + card.dai_link = pcm030_fabric_dai; + card.num_links = ARRAY_SIZE(pcm030_fabric_dai); + + device.card = &card; + device.codec_dev = &soc_codec_dev_wm9712; + + pdev = platform_device_alloc("soc-audio", 1); + if (!pdev) { + pr_err("pcm030_fabric_init: platform_device_alloc() failed\n"); + return -ENODEV; + } + + platform_set_drvdata(pdev, &device); + device.dev = &pdev->dev; + + rc = platform_device_add(pdev); + if (rc) { + pr_err("pcm030_fabric_init: platform_device_add() failed\n"); + return -ENODEV; + } + return 0; +} + +module_init(pcm030_fabric_init); + + +MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>"); +MODULE_DESCRIPTION(DRV_NAME ": mpc5200 pcm030 fabric driver"); +MODULE_LICENSE("GPL"); + |