Basic split of mpc5200 DMA code out from i2s into a standalone file.

Signed-off-by: Jon Smirl jonsmirl@gmail.com --- sound/soc/fsl/Kconfig | 4 sound/soc/fsl/Makefile | 2 sound/soc/fsl/mpc5200_dma.c | 457 +++++++++++++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_dma.h | 81 +++++++ sound/soc/fsl/mpc5200_psc_i2s.c | 484 --------------------------------------- 5 files changed, 546 insertions(+), 482 deletions(-) create mode 100644 sound/soc/fsl/mpc5200_dma.c create mode 100644 sound/soc/fsl/mpc5200_dma.h

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 9fc9082..dc79bdf 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 @@ -23,6 +26,7 @@ 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. diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index f85134c..7731ef2 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -10,5 +10,7 @@ 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

diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c new file mode 100644 index 0000000..cccaff4 --- /dev/null +++ b/sound/soc/fsl/mpc5200_dma.c @@ -0,0 +1,457 @@ +/* + * Freescale MPC5200 PSC DMA + * ALSA SoC Platform driver + * + * Copyright (C) 2008 Secret Lab Technologies Ltd. + */ + +#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> + +#include "mpc5200_dma.h" + +MODULE_AUTHOR("Grant Likely grant.likely@secretlab.ca"); +MODULE_DESCRIPTION("Freescale MPC5200 PSC in DMA mode ASoC Driver"); +MODULE_LICENSE("GPL"); + +/* + * 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(&regs->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(&regs->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. + */ +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; +} + +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. + */ +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(&regs->command, MPC52xx_PSC_RST_RX); + out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT); + } else { + out_8(&regs->command, MPC52xx_PSC_RST_TX); + out_8(&regs->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(&regs->ipcr_acr.ipcr) & 0x80) != 0) + ; + /* then wait for the transition to high */ + while ((in_8(&regs->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(&regs->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(&regs->command, 2 << 4); /* reset rx */ + out_8(&regs->command, 3 << 4); /* reset tx */ + out_8(&regs->command, 4 << 4); /* reset err */ + } + } else { + out_8(&regs->command, 3 << 4); /* reset tx */ + out_8(&regs->command, 4 << 4); /* reset err */ + if (!psc_i2s->capture.active) + out_8(&regs->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"); + 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(&regs->isr_imr.imr, imr); + + return 0; +} + +/** + * psc_i2s_shutdown: shutdown the data transfer on a stream + * + * Shutdown the PSC if there are no other substreams open. + */ +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); + } +} + +/* --------------------------------------------------------------------- + * 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_i2s_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER, + .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, +}; + diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h new file mode 100644 index 0000000..9a19e8a --- /dev/null +++ b/sound/soc/fsl/mpc5200_dma.h @@ -0,0 +1,81 @@ +/* + * Freescale MPC5200 Audio DMA driver + */ + +#ifndef __SOUND_SOC_FSL_MPC5200_DMA_H__ +#define __SOUND_SOC_FSL_MPC5200_DMA_H__ + +/** + * 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; +}; + + +int psc_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); + +int psc_i2s_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); + +void psc_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); + +int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai); + +extern struct snd_soc_platform psc_i2s_pcm_soc_platform; + +#endif /* __SOUND_SOC_FSL_MPC5200_DMA_H__ */ diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 3aa729d..8974b53 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -25,6 +25,8 @@ #include <sysdev/bestcomm/gen_bd.h> #include <asm/mpc52xx_psc.h>

+#include "mpc5200_dma.h" + MODULE_AUTHOR("Grant Likely grant.likely@secretlab.ca"); MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver"); MODULE_LICENSE("GPL"); @@ -47,179 +49,6 @@ MODULE_LICENSE("GPL"); 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(&regs->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(&regs->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; -} - static int psc_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -258,164 +87,6 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream, 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(&regs->command, MPC52xx_PSC_RST_RX); - out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT); - } else { - out_8(&regs->command, MPC52xx_PSC_RST_TX); - out_8(&regs->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(&regs->ipcr_acr.ipcr) & 0x80) != 0) - ; - /* then wait for the transition to high */ - while ((in_8(&regs->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(&regs->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(&regs->command, 2 << 4); /* reset rx */ - out_8(&regs->command, 3 << 4); /* reset tx */ - out_8(&regs->command, 4 << 4); /* reset err */ - } - } else { - out_8(&regs->command, 3 << 4); /* reset tx */ - out_8(&regs->command, 4 << 4); /* reset err */ - if (!psc_i2s->capture.active) - out_8(&regs->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"); - 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(&regs->isr_imr.imr, imr); - - 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 * @@ -495,157 +166,6 @@ static struct snd_soc_dai psc_i2s_dai_template = { };

/* --------------------------------------------------------------------- - * 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, - .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 */

Add a few more mpc5200 PSC defines. More bit fields defines for mpc5200 PSC registers

Signed-off-by: Jon Smirl jonsmirl@gmail.com --- arch/powerpc/include/asm/mpc52xx_psc.h | 10 ++++++++++ 1 files changed, 10 insertions(+), 0 deletions(-)

diff --git a/arch/powerpc/include/asm/mpc52xx_psc.h b/arch/powerpc/include/asm/mpc52xx_psc.h index a218da6..c323f4f 100644 --- a/arch/powerpc/include/asm/mpc52xx_psc.h +++ b/arch/powerpc/include/asm/mpc52xx_psc.h @@ -28,6 +28,10 @@ #define MPC52xx_PSC_MAXNUM 6

/* Programmable Serial Controller (PSC) status register bits */ +#define MPC52xx_PSC_SR_UNEX_RX 0x0001 +#define MPC52xx_PSC_SR_DATA_VAL 0x0002 +#define MPC52xx_PSC_SR_DATA_OVR 0x0004 +#define MPC52xx_PSC_SR_CMDSEND 0x0008 #define MPC52xx_PSC_SR_CDE 0x0080 #define MPC52xx_PSC_SR_RXRDY 0x0100 #define MPC52xx_PSC_SR_RXFULL 0x0200 @@ -61,6 +65,12 @@ #define MPC52xx_PSC_RXTX_FIFO_EMPTY 0x0001

/* PSC interrupt status/mask bits */ +#define MPC52xx_PSC_IMR_UNEX_RX_SLOT 0x0001 +#define MPC52xx_PSC_IMR_DATA_VALID 0x0002 +#define MPC52xx_PSC_IMR_DATA_OVR 0x0004 +#define MPC52xx_PSC_IMR_CMD_SEND 0x0008 +#define MPC52xx_PSC_IMR_ERROR 0x0040 +#define MPC52xx_PSC_IMR_DEOF 0x0080 #define MPC52xx_PSC_IMR_TXRDY 0x0100 #define MPC52xx_PSC_IMR_RXRDY 0x0200 #define MPC52xx_PSC_IMR_DB 0x0400

Allow a custom ASOC fabric driver with soc-of-simple. Register multiple DAIs. If no custom fabric driver, supply a default one so that ASOC binding will complete.

Signed-off-by: Jon Smirl jonsmirl@gmail.com --- include/sound/soc-of-simple.h | 16 +++- sound/last.c | 6 + sound/soc/codecs/tlv320aic26.c | 2 sound/soc/fsl/Kconfig | 2 sound/soc/fsl/mpc5200_psc_i2s.c | 4 - sound/soc/fsl/soc-of-simple.c | 160 ++++++++++++++++++++++++++++++++------- 6 files changed, 154 insertions(+), 36 deletions(-)

diff --git a/include/sound/soc-of-simple.h b/include/sound/soc-of-simple.h index a064e19..97d04af 100644 --- a/include/sound/soc-of-simple.h +++ b/include/sound/soc-of-simple.h @@ -12,13 +12,21 @@ #include <linux/of.h> #include <sound/soc.h>

+#define SOC_OF_SIMPLE_MAX_DAI 2 + int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev, void *codec_data, struct snd_soc_dai *dai, - struct device_node *node); + int count, struct device_node *node); + +int of_snd_soc_register_cpu_dai(struct device_node *node, + struct snd_soc_dai *cpu_dai, int count); + +int of_snd_soc_register_platform(struct snd_soc_platform *platform); + +int of_snd_soc_register_fabric(const char *name, struct snd_soc_ops *ops, + int (*init)(struct snd_soc_codec *codec));

-int of_snd_soc_register_platform(struct snd_soc_platform *platform, - struct device_node *node, - struct snd_soc_dai *cpu_dai); +int of_snd_soc_register_default_fabric(void);

#endif

diff --git a/sound/last.c b/sound/last.c index bdd0857..b6efece 100644 --- a/sound/last.c +++ b/sound/last.c @@ -22,11 +22,15 @@ #define SNDRV_MAIN_OBJECT_FILE #include <linux/init.h> #include <sound/core.h> +#include <sound/soc-of-simple.h>

static int __init alsa_sound_last_init(void) { int idx, ok = 0; - + +#if defined(CONFIG_SND_SOC_OF_SIMPLE) + of_snd_soc_register_default_fabric(); +#endif printk(KERN_INFO "ALSA device list:\n"); for (idx = 0; idx < SNDRV_CARDS; idx++) if (snd_cards[idx] != NULL) { diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 3387d9e..95a7ac9 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -487,7 +487,7 @@ static int aic26_spi_probe(struct spi_device *spi)

#if defined(CONFIG_SND_SOC_OF_SIMPLE) /* Tell the of_soc helper about this codec */ - of_snd_soc_register_codec(&aic26_soc_codec_dev, aic26, &aic26_dai, + of_snd_soc_register_codec(&aic26_soc_codec_dev, aic26, &aic26_dai, 1, spi->dev.archdata.of_node); #endif

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index dc79bdf..8060b95 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,5 +1,5 @@ config SND_SOC_OF_SIMPLE - tristate + bool config SND_MPC52xx_DMA tristate diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 12a7917..5dbf1bc 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -353,8 +353,8 @@ static int __devinit psc_i2s_of_probe(struct of_device *op, snd_soc_register_platform(&psc_dma_pcm_soc_platform);

/* Tell the ASoC OF helpers about it */ - of_snd_soc_register_platform(&psc_dma_pcm_soc_platform, op->node, - &psc_dma->dai); + of_snd_soc_register_platform(&mpc5200_audio_dma_platform); + of_snd_soc_register_cpu_dai(op->node, &psc_dma->dai, 1);

return 0; } diff --git a/sound/soc/fsl/soc-of-simple.c b/sound/soc/fsl/soc-of-simple.c index 8bc5cd9..bf307aa 100644 --- a/sound/soc/fsl/soc-of-simple.c +++ b/sound/soc/fsl/soc-of-simple.c @@ -32,18 +32,23 @@ struct of_snd_soc_device { struct list_head list; struct snd_soc_device device; struct snd_soc_card card; - struct snd_soc_dai_link dai_link; + struct snd_soc_dai_link dai_link[SOC_OF_SIMPLE_MAX_DAI]; struct platform_device *pdev; - struct device_node *platform_node; + struct device_node *cpu_dai_node; struct device_node *codec_node; };

-static struct snd_soc_ops of_snd_soc_ops = { -}; +/* template values */ +struct snd_soc_platform *template_platform; +const char *template_name = NULL; +struct snd_soc_ops *template_ops; +int (*template_init)(struct snd_soc_codec *codec);

static struct of_snd_soc_device * of_snd_soc_get_device(struct device_node *codec_node) { + int i; + struct of_snd_soc_device *of_soc;

list_for_each_entry(of_soc, &of_snd_soc_device_list, list) { @@ -58,10 +63,16 @@ of_snd_soc_get_device(struct device_node *codec_node) /* Initialize the structure and add it to the global list */ of_soc->codec_node = codec_node; of_soc->id = of_snd_soc_next_index++; - of_soc->card.dai_link = &of_soc->dai_link; - of_soc->card.num_links = 1; + of_soc->card.dai_link = of_soc->dai_link; of_soc->device.card = &of_soc->card; - of_soc->dai_link.ops = &of_snd_soc_ops; + of_soc->card.num_links = SOC_OF_SIMPLE_MAX_DAI; + for (i = 0; i < SOC_OF_SIMPLE_MAX_DAI; i++) { + of_soc->dai_link[i].ops = template_ops; + of_soc->dai_link[i].init = template_init; + } + of_soc->card.name = (char *)template_name; + of_soc->card.platform = template_platform; + list_add(&of_soc->list, &of_snd_soc_device_list);

return of_soc; @@ -74,10 +85,11 @@ static void of_snd_soc_register_device(struct of_snd_soc_device *of_soc)

/* Only register the device if both the codec and platform have * been registered */ - if ((!of_soc->device.codec_data) || (!of_soc->platform_node)) + if ((!of_soc->device.codec_data) || (!of_soc->cpu_dai_node) || + !of_soc->card.platform || !of_soc->card.name) return;

- pr_info("platform<-->codec match achieved; registering machine\n"); + pr_info("platform<-->codec match achieved; registering fabric\n");

pdev = platform_device_alloc("soc-audio", of_soc->id); if (!pdev) { @@ -100,10 +112,10 @@ static void of_snd_soc_register_device(struct of_snd_soc_device *of_soc)

int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev, void *codec_data, struct snd_soc_dai *dai, - struct device_node *node) + int count, struct device_node *node) { struct of_snd_soc_device *of_soc; - int rc = 0; + int i, rc = 0;

pr_info("registering ASoC codec driver: %s\n", node->full_name);

@@ -117,10 +129,11 @@ int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev, /* Store the codec data */ of_soc->device.codec_data = codec_data; of_soc->device.codec_dev = codec_dev; - of_soc->dai_link.name = (char *)node->name; - of_soc->dai_link.stream_name = (char *)node->name; - of_soc->dai_link.codec_dai = dai; - + of_soc->card.num_links = min(count, of_soc->card.num_links); + for (i = 0; i < of_soc->card.num_links; i++) { + of_soc->dai_link[i].name = dai[i].name; + of_soc->dai_link[i].codec_dai = &dai[i]; + } /* Now try to register the SoC device */ of_snd_soc_register_device(of_soc);

@@ -130,21 +143,51 @@ int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev, } EXPORT_SYMBOL_GPL(of_snd_soc_register_codec);

-int of_snd_soc_register_platform(struct snd_soc_platform *platform, - struct device_node *node, - struct snd_soc_dai *cpu_dai) +int of_snd_soc_register_cpu_dai(struct device_node *node, + struct snd_soc_dai *cpu_dai, int count) { struct of_snd_soc_device *of_soc; struct device_node *codec_node; const phandle *handle; - int len, rc = 0; + int i, len, rc = 0;

- pr_info("registering ASoC platform driver: %s\n", node->full_name); + pr_info("registering ASoC CPU DAI driver: %s\n", node->full_name);

handle = of_get_property(node, "codec-handle", &len); - if (!handle || len < sizeof(handle)) - return -ENODEV; - codec_node = of_find_node_by_phandle(*handle); + if (handle && len >= sizeof(handle)) + codec_node = of_find_node_by_phandle(*handle); + else { + /* Check for the codec child nodes */ + for_each_child_of_node(node, codec_node) { + struct platform_device *pdev; + struct dev_archdata dev_ad = {}; + char name[MODULE_NAME_LEN]; + const u32 *addr; + int len; + + if (of_modalias_node(codec_node, name, sizeof(name)) < 0) + continue; + + addr = of_get_property(codec_node, "reg", &len); + if (!addr || len < sizeof(int) || *addr > (1 << 10) - 1) { + pr_err("invalid codec reg in device tree\n"); + continue; + } + request_module("%s", name); + + pdev = platform_device_alloc(name, 0); + + dev_archdata_set_node(&dev_ad, codec_node); + pdev->dev.archdata = dev_ad; + + rc = platform_device_add(pdev); + if (rc) { + platform_device_put(pdev); + return rc; + } + break; + } + } if (!codec_node) return -ENODEV; pr_info("looking for codec: %s\n", codec_node->full_name); @@ -156,10 +199,12 @@ int of_snd_soc_register_platform(struct snd_soc_platform *platform, goto out; }

- of_soc->platform_node = node; - of_soc->dai_link.cpu_dai = cpu_dai; - of_soc->card.platform = platform; - of_soc->card.name = of_soc->dai_link.cpu_dai->name; + of_soc->cpu_dai_node = node; + of_soc->card.num_links = min(count, of_soc->card.num_links); + for (i = 0; i < of_soc->card.num_links; i++) { + of_soc->dai_link[i].stream_name = cpu_dai[i].name; + of_soc->dai_link[i].cpu_dai = &cpu_dai[i]; + }

/* Now try to register the SoC device */ of_snd_soc_register_device(of_soc); @@ -168,4 +213,65 @@ int of_snd_soc_register_platform(struct snd_soc_platform *platform, mutex_unlock(&of_snd_soc_mutex); return rc; } +EXPORT_SYMBOL_GPL(of_snd_soc_register_cpu_dai); + +int of_snd_soc_register_platform(struct snd_soc_platform *platform) +{ + struct of_snd_soc_device *of_soc; + int rc = 0; + + pr_info("registering ASoC platform driver: %s\n", platform->name); + template_platform = platform; + + mutex_lock(&of_snd_soc_mutex); + list_for_each_entry(of_soc, &of_snd_soc_device_list, list) { + of_soc->card.platform = platform; + of_snd_soc_register_device(of_soc); + } + mutex_unlock(&of_snd_soc_mutex); + return rc; +} EXPORT_SYMBOL_GPL(of_snd_soc_register_platform); + +int of_snd_soc_register_fabric(const char *name, struct snd_soc_ops *ops, + int (*init)(struct snd_soc_codec *codec)) +{ + int i; + struct of_snd_soc_device *of_soc; + + pr_info("registering ASoC fabric driver: %s\n", name); + template_name = name; + template_ops = ops; + template_init = init; + + mutex_lock(&of_snd_soc_mutex); + list_for_each_entry(of_soc, &of_snd_soc_device_list, list) { + for (i = 0; i < SOC_OF_SIMPLE_MAX_DAI; i++) { + of_soc->dai_link[i].ops = ops; + of_soc->dai_link[i].init = init; + } + of_soc->card.name = (char *)name; + of_snd_soc_register_device(of_soc); + } + mutex_unlock(&of_snd_soc_mutex); + return 0; +} +EXPORT_SYMBOL_GPL(of_snd_soc_register_fabric); + +/* If no board specific fabric driver has been registered, register a default one */ +int of_snd_soc_register_default_fabric(void) +{ + struct device_node *root; + const char *model = ""; + + if (template_name != NULL) + return 0; + + root = of_find_node_by_path("/"); + if (root) + model = of_get_property(root, "model", NULL); + + return of_snd_soc_register_fabric(model, NULL, NULL); +} +EXPORT_SYMBOL_GPL(of_snd_soc_register_default_fabric); +

Add SNDRV_PCM_FMTBIT_S32_BE as a valid AC97 format

Signed-off-by: Jon Smirl jonsmirl@gmail.com --- include/sound/soc-dai.h | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-)

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 496dc30..352d7ee 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -79,7 +79,8 @@ struct snd_pcm_substream; #define SND_SOC_CLOCK_OUT 1

#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S32_LE) + SNDRV_PCM_FMTBIT_S32_LE |\ + SNDRV_PCM_FMTBIT_S32_BE)

struct snd_soc_dai_ops; struct snd_soc_dai;

Have the WM9712 driver self register itself. This is needed to support device tree driver binding via the soc_of_simple code.

Signed-off-by: Jon Smirl jonsmirl@gmail.com --- sound/soc/codecs/wm9712.c | 34 ++++++++++++++++++++++++++++++++++ 1 files changed, 34 insertions(+), 0 deletions(-)

diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 550c903..cb0e679 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -20,6 +20,7 @@ #include <sound/initval.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <sound/soc-of-simple.h> #include "wm9712.h"

#define WM9712_VERSION "0.4" @@ -740,6 +741,39 @@ struct snd_soc_codec_device soc_codec_dev_wm9712 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm9712);

+static int __init wm9712_probe(struct platform_device *pdev) +{ + snd_soc_register_dais(wm9712_dai, ARRAY_SIZE(wm9712_dai)); +#if defined(CONFIG_SND_SOC_OF_SIMPLE) + /* Tell the of_soc helper about this codec */ + of_snd_soc_register_codec(&soc_codec_dev_wm9712, pdev->dev.archdata.of_node, + wm9712_dai, ARRAY_SIZE(wm9712_dai), + pdev->dev.archdata.of_node); +#endif + return 0; +} + +static struct platform_driver wm9712_driver = +{ + .probe = wm9712_probe, + .driver = { + .name = "wm9712", + }, +}; + +static __init int wm9712_driver_init(void) +{ + return platform_driver_register(&wm9712_driver); +} + +static __exit void wm9712_driver_exit(void) +{ +} + +module_init(wm9712_driver_init); +module_exit(wm9712_driver_exit); + + MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver"); MODULE_AUTHOR("Liam Girdwood"); MODULE_LICENSE("GPL");

Rewrite the mpc5200 audio DMA code to support both I2S and AC97. Make it more robust.

Signed-off-by: Jon Smirl jonsmirl@gmail.com --- sound/soc/fsl/mpc5200_dma.c | 503 +++++++++++++++++++++++++++++---------- sound/soc/fsl/mpc5200_dma.h | 32 +- sound/soc/fsl/mpc5200_psc_i2s.c | 198 +-------------- 3 files changed, 396 insertions(+), 337 deletions(-)

diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 6850392..5ac2693 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -23,6 +23,8 @@

#include <sysdev/bestcomm/bestcomm.h> #include <sysdev/bestcomm/gen_bd.h> +#include <asm/time.h> +#include <asm/mpc52xx.h> #include <asm/mpc52xx_psc.h>

#include "mpc5200_dma.h" @@ -50,7 +52,7 @@ static irqreturn_t psc_dma_status_irq(int irq, void *_psc_dma) if (psc_dma->capture.active && (isr & MPC52xx_PSC_IMR_ORERR)) psc_dma->stats.overrun_count++;

- out_8(&regs->command, 4 << 4); /* reset the error status */ + out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);

return IRQ_HANDLED; } @@ -81,8 +83,21 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) 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(int irq, void *_psc_dma_stream) +static irqreturn_t psc_dma_bcom_irq_tx(int irq, void *_psc_dma_stream) { struct psc_dma_stream *s = _psc_dma_stream;

@@ -90,12 +105,12 @@ static irqreturn_t psc_dma_bcom_irq(int irq, void *_psc_dma_stream) * 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); - bcom_enable(s->bcom_task); } + psc_dma_bcom_enqueue_tx(s);

/* If the stream is active, then also inform the PCM middle layer * of the period finished event. */ @@ -105,49 +120,31 @@ static irqreturn_t psc_dma_bcom_irq(int irq, void *_psc_dma_stream) return IRQ_HANDLED; }

-/** - * psc_dma_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. - */ -int psc_dma_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static irqreturn_t psc_dma_bcom_irq_rx(int irq, void *_psc_dma_stream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; - int rc; + struct psc_dma_stream *s = _psc_dma_stream;

- dev_dbg(psc_dma->dev, "psc_dma_startup(substream=%p)\n", substream); + /* 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);

- if (!psc_dma->playback.active && - !psc_dma->capture.active) { - /* Setup the IRQs */ - 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, IRQF_SHARED, - "psc-dma-capture", &psc_dma->capture); - rc |= request_irq(psc_dma->playback.irq, - &psc_dma_bcom_irq, 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; - } + 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); }

- return 0; + /* 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; }

-int psc_dma_hw_free(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int psc_dma_hw_free(struct snd_pcm_substream *substream) { snd_pcm_set_runtime_buffer(substream, NULL); return 0; @@ -159,8 +156,7 @@ int psc_dma_hw_free(struct snd_pcm_substream *substream, * This function is called by ALSA to start, stop, pause, and resume the DMA * transfer of data. */ -int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) +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; @@ -170,6 +166,7 @@ int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd, u16 imr; u8 psc_cmd; unsigned long flags; + int i;

if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) s = &psc_dma->capture; @@ -189,25 +186,24 @@ int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd, (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;

- /* First; reset everything */ - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { - out_8(&regs->command, MPC52xx_PSC_RST_RX); - out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT); - } else { - out_8(&regs->command, MPC52xx_PSC_RST_TX); - out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT); - } + s->runtime = runtime; + s->appl_ptr = s->runtime->control->appl_ptr - (runtime->period_size * runtime->periods);

- /* Next, fill up the bestcomm bd queue and enable DMA. + /* 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) + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { bcom_gen_bd_rx_reset(s->bcom_task); - else + 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); - while (!bcom_queue_full(s->bcom_task)) - psc_dma_bcom_enqueue_next_buffer(s); + psc_dma_bcom_enqueue_tx(s); + } + bcom_enable(s->bcom_task);

/* Due to errata in the dma mode; need to line up enabling @@ -227,30 +223,22 @@ int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd, psc_cmd = MPC52xx_PSC_RX_ENABLE; if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK) psc_cmd |= MPC52xx_PSC_TX_ENABLE; - out_8(&regs->command, psc_cmd); + //out_8(&regs->command, psc_cmd); + + out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT); + spin_unlock_irqrestore(&psc_dma->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_dma->playback.active) { - out_8(&regs->command, 2 << 4); /* reset rx */ - out_8(&regs->command, 3 << 4); /* reset tx */ - out_8(&regs->command, 4 << 4); /* reset err */ - } - } else { - out_8(&regs->command, 3 << 4); /* reset tx */ - out_8(&regs->command, 4 << 4); /* reset err */ - if (!psc_dma->capture.active) - out_8(&regs->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); + 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);

break;

@@ -265,44 +253,11 @@ int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd, imr |= MPC52xx_PSC_IMR_TXEMP; if (psc_dma->capture.active) imr |= MPC52xx_PSC_IMR_ORERR; - out_be16(&regs->isr_imr.imr, imr); + out_be16(&regs->isr_imr.imr, psc_dma->imr | imr);

return 0; }

-/** - * psc_dma_shutdown: shutdown the data transfer on a stream - * - * Shutdown the PSC if there are no other substreams open. - */ -void psc_dma_shutdown(struct snd_pcm_substream *substream, - 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; - - dev_dbg(psc_dma->dev, "psc_dma_shutdown(substream=%p)\n", substream); - - /* - * If this is the last active substream, disable the PSC and release - * the IRQ. - */ - 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, 0); - out_8(&psc_dma->psc_regs->command, 3 << 4); /* reset tx */ - out_8(&psc_dma->psc_regs->command, 2 << 4); /* reset rx */ - out_8(&psc_dma->psc_regs->command, 1 << 4); /* reset mode */ - out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */ - - /* 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); - } -}

/* --------------------------------------------------------------------- * The PSC DMA 'ASoC platform' driver @@ -312,67 +267,93 @@ void psc_dma_shutdown(struct snd_pcm_substream *substream, * interaction with the attached codec */

-static const struct snd_pcm_hardware psc_dma_pcm_hardware = { +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, + .formats = SNDRV_PCM_FMTBIT_S32_BE, .rate_min = 8000, .rate_max = 48000, - .channels_min = 2, + .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 = 0, + .fifo_size = 512, };

-static int psc_dma_pcm_open(struct snd_pcm_substream *substream) +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_pcm_open(substream=%p)\n", substream); + 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_pcm_hardware); + 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; + } + + if (!psc_dma->playback.active && + !psc_dma->capture.active) { + /* Setup the IRQs */ + }

s->stream = substream; return 0; }

-static int psc_dma_pcm_close(struct snd_pcm_substream *substream) +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_pcm_close(substream=%p)\n", substream); + 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, 3 << 4); /* reset tx */ + //out_8(&psc_dma->psc_regs->command, 2 << 4); /* reset rx */ + //out_8(&psc_dma->psc_regs->command, 1 << 4); /* reset mode */ + //out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */ + + } + s->stream = NULL; return 0; }

static snd_pcm_uframes_t -psc_dma_pcm_pointer(struct snd_pcm_substream *substream) +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; + snd_pcm_uframes_t frames;

if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) s = &psc_dma->capture; @@ -381,46 +362,63 @@ psc_dma_pcm_pointer(struct snd_pcm_substream *substream)

count = s->period_current_pt - s->period_start;

- return bytes_to_frames(substream->runtime, count); + frames = bytes_to_frames(substream->runtime, count); + return frames; +} + +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_pcm_ops = { - .open = psc_dma_pcm_open, - .close = psc_dma_pcm_close, +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_pcm_pointer, + .pointer = psc_dma_pointer, + .trigger = psc_dma_trigger, + .hw_params = psc_dma_hw_params, };

-static u64 psc_dma_pcm_dmamask = 0xffffffff; -static int psc_dma_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, +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; - size_t size = psc_dma_pcm_hardware.buffer_bytes_max; + 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_pcm_new(card=%p, dai=%p, pcm=%p)\n", + 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_pcm_dmamask; + 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->dev, size, + 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->dev, size, + 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: @@ -431,13 +429,13 @@ static int psc_dma_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, return -ENOMEM; }

-static void psc_dma_pcm_free(struct snd_pcm *pcm) +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_pcm_free(pcm=%p)\n", pcm); + dev_dbg(rtd->socdev->dev, "psc_dma_free(pcm=%p)\n", pcm);

for (stream = 0; stream < 2; stream++) { substream = pcm->streams[stream].substream; @@ -449,10 +447,247 @@ static void psc_dma_pcm_free(struct snd_pcm *pcm) } }

-struct snd_soc_platform psc_dma_pcm_soc_platform = { +struct snd_soc_platform mpc5200_audio_dma_platform = { .name = "mpc5200-psc-audio", - .pcm_ops = &psc_dma_pcm_ops, - .pcm_new = &psc_dma_pcm_new, - .pcm_free = &psc_dma_pcm_free, + .pcm_ops = &psc_dma_ops, + .pcm_new = &psc_dma_new, + .pcm_free = &psc_dma_free, }; +EXPORT_SYMBOL_GPL(mpc5200_audio_dma_platform); + +/* --------------------------------------------------------------------- + * Sysfs attributes for debugging + */ + +static ssize_t psc_dma_status_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct psc_dma *psc_dma = 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_dma->psc_regs->sr_csr.status), + in_be32(&psc_dma->psc_regs->sicr), + in_be16(&psc_dma->fifo_regs->rfnum) & 0x1ff, + in_be16(&psc_dma->fifo_regs->rfstat), + in_be16(&psc_dma->fifo_regs->tfnum) & 0x1ff, + in_be16(&psc_dma->fifo_regs->tfstat)); +} + +static int *psc_dma_get_stat_attr(struct psc_dma *psc_dma, const char *name) +{ + if (strcmp(name, "playback_underrun") == 0) + return &psc_dma->stats.underrun_count; + if (strcmp(name, "capture_overrun") == 0) + return &psc_dma->stats.overrun_count; + + return NULL; +} + +static ssize_t psc_dma_stat_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct psc_dma *psc_dma = dev_get_drvdata(dev); + int *attrib; + + attrib = psc_dma_get_stat_attr(psc_dma, attr->attr.name); + if (!attrib) + return 0; + + return sprintf(buf, "%i\n", *attrib); +} + +static ssize_t psc_dma_stat_store(struct device *dev, + struct device_attribute *attr, + const char *buf, + size_t count) +{ + struct psc_dma *psc_dma = dev_get_drvdata(dev); + int *attrib; + + attrib = psc_dma_get_stat_attr(psc_dma, attr->attr.name); + if (!attrib) + return 0; + + *attrib = simple_strtoul(buf, NULL, 0); + return count; +} + +static DEVICE_ATTR(status, 0644, psc_dma_status_show, NULL); +static DEVICE_ATTR(playback_underrun, 0644, psc_dma_stat_show, + psc_dma_stat_store); +static DEVICE_ATTR(capture_overrun, 0644, psc_dma_stat_show, + psc_dma_stat_store); + + +int mpc5200_audio_dma_create(struct of_device *op, struct snd_soc_dai *template, int tsize) +{ + phys_addr_t fifo; + struct psc_dma *psc_dma; + struct resource res; + int i, nDAI, size, psc_id, irq, rc; + const __be32 *prop; + void __iomem *regs; + + /* 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; + } + + /* Allocate and initialize the driver private data */ + psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL); + if (!psc_dma) { + iounmap(regs); + return -ENOMEM; + } + + spin_lock_init(&psc_dma->lock); + 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 AC97", psc_id+1); + + /* Fill out the CPU DAI structure */ + nDAI = min(tsize, SOC_OF_SIMPLE_MAX_DAI); + for (i = 0; i < nDAI; i++) { + memcpy(&psc_dma->dai[i], &template[i], sizeof(struct snd_soc_dai)); + psc_dma->dai[i].private_data = psc_dma; + snprintf(psc_dma->stream_name[i], PSC_STREAM_NAME_LEN, template[i].name, psc_dma->name); + psc_dma->dai[i].name = psc_dma->stream_name[i]; + psc_dma->dai[i].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_dma->capture.bcom_task = + bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512); + psc_dma->playback.bcom_task = + bcom_psc_gen_bd_tx_init(psc_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); + out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_RX); /* reset receiver */ + out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_TX); /* reset transmitter */ + out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_ERR_STAT); /* reset error */ + out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_SEL_MODE_REG_1); /* reset mode */ + + /* 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); + + /* Register the SYSFS files */ + rc = device_create_file(psc_dma->dev, &dev_attr_status); + rc |= device_create_file(psc_dma->dev, &dev_attr_capture_overrun); + rc |= device_create_file(psc_dma->dev, &dev_attr_playback_underrun); + if (rc) + dev_info(psc_dma->dev, "error creating sysfs files\n"); + + rc = snd_soc_register_dais(psc_dma->dai, nDAI); + if (rc != 0) { + pr_err("Failed to register DAI\n"); + return 0; + } + + /* Tell the ASoC OF helpers about it */ + of_snd_soc_register_cpu_dai(op->node, psc_dma->dai, nDAI); + + return 0; +} +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"); + + 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); + iounmap(psc_dma->fifo_regs); + kfree(psc_dma); + dev_set_drvdata(&op->dev, NULL); + + return 0; +} +EXPORT_SYMBOL_GPL(mpc5200_audio_dma_destroy); + +static int __init mpc5200_soc_platform_init(void) +{ + /* Tell the ASoC OF helpers about it */ + of_snd_soc_register_platform(&mpc5200_audio_dma_platform); + return snd_soc_register_platform(&mpc5200_audio_dma_platform); +} +module_init(mpc5200_soc_platform_init); + +static void __exit mpc5200_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&mpc5200_audio_dma_platform); +} +module_exit(mpc5200_soc_platform_exit);

diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index a33232c..d71e503 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -5,8 +5,12 @@ #ifndef __SOUND_SOC_FSL_MPC5200_DMA_H__ #define __SOUND_SOC_FSL_MPC5200_DMA_H__

+#include <sound/soc-of-simple.h> + +#define PSC_STREAM_NAME_LEN 32 + /** - * psc_dma_stream - Data specific to a single stream (playback or capture) + * 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 @@ -17,6 +21,9 @@ * @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; @@ -27,6 +34,7 @@ struct psc_dma_stream { dma_addr_t period_next_pt; dma_addr_t period_current_pt; int period_bytes; + int period_size; };

/** @@ -48,9 +56,13 @@ struct psc_dma { struct mpc52xx_psc_fifo __iomem *fifo_regs; unsigned int irq; struct device *dev; - struct snd_soc_dai dai; + struct snd_soc_dai dai[SOC_OF_SIMPLE_MAX_DAI]; + char stream_name[SOC_OF_SIMPLE_MAX_DAI][PSC_STREAM_NAME_LEN]; spinlock_t lock; u32 sicr; + uint sysclk; + int imr; + unsigned int slots;

/* per-stream data */ struct psc_dma_stream playback; @@ -63,19 +75,9 @@ struct psc_dma { } stats; };

+int mpc5200_audio_dma_create(struct of_device *op, struct snd_soc_dai *template, int tsize); +int mpc5200_audio_dma_destroy(struct of_device *op);

-int psc_dma_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai); - -int psc_dma_hw_free(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai); - -void psc_dma_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai); - -int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai); - -extern struct snd_soc_platform psc_dma_pcm_soc_platform; +extern struct snd_soc_platform mpc5200_audio_dma_platform;

#endif /* __SOUND_SOC_FSL_MPC5200_DMA_H__ */ diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 5dbf1bc..513d68d 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -82,8 +82,6 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream, } out_be32(&psc_dma->psc_regs->sicr, psc_dma->sicr | mode);

- snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - return 0; }

@@ -140,11 +138,7 @@ 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_dma_startup, .hw_params = psc_i2s_hw_params, - .hw_free = psc_dma_hw_free, - .shutdown = psc_dma_shutdown, - .trigger = psc_dma_trigger, .set_sysclk = psc_i2s_set_sysclk, .set_fmt = psc_i2s_set_fmt, }; @@ -166,70 +160,6 @@ static struct snd_soc_dai psc_i2s_dai_template = { };

/* --------------------------------------------------------------------- - * Sysfs attributes for debugging - */ - -static ssize_t psc_i2s_status_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - struct psc_dma *psc_dma = 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_dma->psc_regs->sr_csr.status), - in_be32(&psc_dma->psc_regs->sicr), - in_be16(&psc_dma->fifo_regs->rfnum) & 0x1ff, - in_be16(&psc_dma->fifo_regs->rfstat), - in_be16(&psc_dma->fifo_regs->tfnum) & 0x1ff, - in_be16(&psc_dma->fifo_regs->tfstat)); -} - -static int *psc_i2s_get_stat_attr(struct psc_dma *psc_dma, const char *name) -{ - if (strcmp(name, "playback_underrun") == 0) - return &psc_dma->stats.underrun_count; - if (strcmp(name, "capture_overrun") == 0) - return &psc_dma->stats.overrun_count; - - return NULL; -} - -static ssize_t psc_i2s_stat_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - struct psc_dma *psc_dma = dev_get_drvdata(dev); - int *attrib; - - attrib = psc_i2s_get_stat_attr(psc_dma, 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_dma *psc_dma = dev_get_drvdata(dev); - int *attrib; - - attrib = psc_i2s_get_stat_attr(psc_dma, 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); - -/* --------------------------------------------------------------------- * OF platform bus binding code: * - Probe/remove operations * - OF device match table @@ -237,82 +167,18 @@ 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; + int rc; struct psc_dma *psc_dma; - 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; - }

- /* Allocate and initialize the driver private data */ - psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL); - if (!psc_dma) { - iounmap(regs); - return -ENOMEM; - } - spin_lock_init(&psc_dma->lock); - 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_id+1); - - /* Fill out the CPU DAI structure */ - memcpy(&psc_dma->dai, &psc_i2s_dai_template, sizeof psc_dma->dai); - psc_dma->dai.private_data = psc_dma; - psc_dma->dai.name = psc_dma->name; - psc_dma->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_dma->capture.bcom_task = - bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512); - psc_dma->playback.bcom_task = - bcom_psc_gen_bd_tx_init(psc_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; - } + rc = mpc5200_audio_dma_create(op, &psc_i2s_dai_template, 1); + if (rc != 0) + return rc;

- /* Disable all interrupts and reset the PSC */ - out_be16(&psc_dma->psc_regs->isr_imr.imr, 0); - out_8(&psc_dma->psc_regs->command, 3 << 4); /* reset transmitter */ - out_8(&psc_dma->psc_regs->command, 2 << 4); /* reset receiver */ - out_8(&psc_dma->psc_regs->command, 1 << 4); /* reset mode */ - out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */ + psc_dma = dev_get_drvdata(&op->dev);

/* Configure the serial interface mode; defaulting to CODEC8 mode */ 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_dma->sicr |= MPC52xx_PSC_SICR_CELLSLAVE | - MPC52xx_PSC_SICR_GENCLK; out_be32(&psc_dma->psc_regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8);

@@ -321,61 +187,17 @@ static int __devinit psc_i2s_of_probe(struct of_device *op, 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_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); - - /* Save what we've done so it can be found again later */ - dev_set_drvdata(&op->dev, psc_dma); - - /* Register the SYSFS files */ - rc = device_create_file(psc_dma->dev, &dev_attr_status); - rc |= device_create_file(psc_dma->dev, &dev_attr_capture_overrun); - rc |= device_create_file(psc_dma->dev, &dev_attr_playback_underrun); - if (rc) - dev_info(psc_dma->dev, "error creating sysfs files\n"); - - snd_soc_register_platform(&psc_dma_pcm_soc_platform); - - /* Tell the ASoC OF helpers about it */ - of_snd_soc_register_platform(&mpc5200_audio_dma_platform); - of_snd_soc_register_cpu_dai(op->node, &psc_dma->dai, 1); + /* Go */ + out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_TX_ENABLE); + out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RX_ENABLE);

return 0; + }

static int __devexit psc_i2s_of_remove(struct of_device *op) { - struct psc_dma *psc_dma = dev_get_drvdata(&op->dev); - - dev_dbg(&op->dev, "psc_i2s_remove()\n"); - - snd_soc_unregister_platform(&psc_dma_pcm_soc_platform); - - bcom_gen_bd_rx_release(psc_dma->capture.bcom_task); - bcom_gen_bd_tx_release(psc_dma->playback.bcom_task); - - iounmap(psc_dma->psc_regs); - iounmap(psc_dma->fifo_regs); - kfree(psc_dma); - dev_set_drvdata(&op->dev, NULL); - - return 0; + return mpc5200_audio_dma_destroy(op); }

/* Match table for of_platform binding */

AC97 codec for STAC9766 used on the Efika. Support registration from device trees. Datasheet: http://www.idt.com/products/getDoc.cfm?docID=13134007

Signed-off-by: Jon Smirl jonsmirl@gmail.com --- sound/soc/codecs/Kconfig | 4 sound/soc/codecs/Makefile | 2 sound/soc/codecs/stac9766.c | 491 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/stac9766.h | 21 ++ 4 files changed, 518 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/stac9766.c create mode 100644 sound/soc/codecs/stac9766.h

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1c19ad5..d86b0de 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 select SND_SOC_SSM2602 if I2C + select SND_SOC_STAC9766 select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C @@ -92,6 +93,9 @@ config SND_SOC_PCM3008 config SND_SOC_SSM2602 tristate

+config SND_SOC_STAC9766 + tristate + config SND_SOC_TLV320AIC23 tristate

diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3d31b6b..c6998e4 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -41,6 +42,7 @@ obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c new file mode 100644 index 0000000..9f9dfa8 --- /dev/null +++ b/sound/soc/codecs/stac9766.c @@ -0,0 +1,491 @@ +/* + * stac9766.c -- ALSA SoC STAC9766 codec support + * + * Copyright 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 as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Features:- + * + * o Support for AC97 Codec, S/PDIF + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/soc-of-simple.h> + +#include "stac9766.h" + +#define STAC9766_VERSION "0.10" + +/* + * STAC9766 register cache + */ +static const u16 stac9766_reg[] = { + 0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */ + 0x0000, 0x0000, 0x8008, 0x8008, /* e */ + 0x8808, 0x8808, 0x8808, 0x8808, /* 16 */ + 0x8808, 0x0000, 0x8000, 0x0000, /* 1e */ + 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */ + 0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */ + 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */ + 0x0000, 0x2000, 0x0000, 0x0100, /* 3e */ + 0x0000, 0x0000, 0x0080, 0x0000, /* 46 */ + 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ + 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */ + 0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ + 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */ +}; + +static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; +static const char *stac9766_mono_mux[] = {"Mix", "Mic"}; +static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"}; +static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"}; +static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"}; +static const char *stac9766_record_all_mux[] = {"All analog", "Analog plus DAC"}; +static const char *stac9766_boost1[] = {"0dB", "10dB"}; +static const char *stac9766_boost2[] = {"0dB", "20dB"}; +static const char *stac9766_stereo_mic[] = {"Off", "On"}; + +static const struct soc_enum stac9766_record_enum = + SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux); +static const struct soc_enum stac9766_mono_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux); +static const struct soc_enum stac9766_mic_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux); +static const struct soc_enum stac9766_SPDIF_enum = + SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux); +static const struct soc_enum stac9766_popbypass_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux); +static const struct soc_enum stac9766_record_all_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, stac9766_record_all_mux); +static const struct soc_enum stac9766_boost1_enum = + SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */ +static const struct soc_enum stac9766_boost2_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */ +static const struct soc_enum stac9766_stereo_mic_enum = + SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic); + +static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0); +static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250); +static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0); +static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200); + +static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { + SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1), + SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1), + SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, master_tlv), + SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1), + + SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv), + SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1), + + + SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv), + SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1), + SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1), + SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv), + SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1), + + SOC_ENUM("Mic Boost1", stac9766_boost1_enum), + SOC_ENUM("Mic Boost2", stac9766_boost2_enum), + SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv), + SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), + SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum), + + SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1), + SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1), + SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1), + SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1), + + SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1), + SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0), + SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1), + SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0), + + SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum), + SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum), + SOC_ENUM("Record All Mux", stac9766_record_all_enum), + SOC_ENUM("Record Mux", stac9766_record_enum), + SOC_ENUM("Mono Mux", stac9766_mono_enum), + SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum), +}; + +int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + u16 *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + soc_ac97_ops.write(codec->ac97, reg, val); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return 0; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + soc_ac97_ops.write(codec->ac97, reg, val); + cache[reg / 2] = val; + return 0; +} + +unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) +{ + u16 val = 0, *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return val; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || + reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || + reg == AC97_VENDOR_ID2) { + + val = soc_ac97_ops.read(codec->ac97, reg); + return val; + } + return cache[reg / 2]; +} + +static int ac97_analog_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + + vra |= 0x1; /* enable variable rate audio */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + stac9766_ac97_write(codec, AC97_SPDIF, 0x2002); + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + + vra |= 0x5; /* Enable VRA and SPDIF out */ + printk("var is %x\n", vra); + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + reg = AC97_PCM_FRONT_DAC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned short vra; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra &= !0x04; + //stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + break; + } + return 0; +} + +static int stac9766_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: /* full On */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); + break; + case SND_SOC_BIAS_PREPARE: /* partial On */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); + break; + case SND_SOC_BIAS_STANDBY: /* Off, with power */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); + break; + case SND_SOC_BIAS_OFF: /* Off, without power */ + /* disable everything including AC link */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +static int stac9766_codec_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int stac9766_codec_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + u16 id; + + /* give the codec an AC97 warm reset to start the link */ + codec->ac97->bus->ops->warm_reset(codec->ac97); + id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2); + if (id != 0x4c13) { + printk(KERN_ERR "stac9766 failed to resume"); + return -EIO; + } + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + stac9766_set_bias_level(codec, SND_SOC_BIAS_ON); + + return 0; +} + +static struct snd_soc_dai_ops stac9766_dai_ops_analog = +{ + .prepare = ac97_analog_prepare, +}; + +static struct snd_soc_dai_ops stac9766_dai_ops_digital = +{ + .prepare = ac97_digital_prepare, + .trigger = ac97_digital_trigger, +}; + +struct snd_soc_dai stac9766_dai[] = { +{ + .name = "stac9766 analog", + .id = 0, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, + }, + .capture = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_analog, +}, +{ + .name = "stac9766 IEC958", + .id = 1, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 IEC958", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_digital, +}}; +EXPORT_SYMBOL_GPL(stac9766_dai); + +int stac9766_reset(struct snd_soc_codec *codec, int try_warm) +{ + if (try_warm && soc_ac97_ops.warm_reset) { + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) + return 1; + } + + soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops.warm_reset) + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) + return -EIO; + return 0; +} + +static int stac9766_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); + + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->card->codec == NULL) + return -ENOMEM; + codec = socdev->card->codec; + mutex_init(&codec->mutex); + + codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + codec->reg_cache_size = sizeof(stac9766_reg); + codec->reg_cache_step = 2; + + codec->name = "STAC9766"; + codec->owner = THIS_MODULE; + codec->dai = stac9766_dai; + codec->num_dai = ARRAY_SIZE(stac9766_dai); + codec->write = stac9766_ac97_write; + codec->read = stac9766_ac97_read; + codec->set_bias_level = stac9766_set_bias_level; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) + goto codec_err; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + /* do a cold reset for the controller and then try + * a warm reset followed by an optional cold reset for codec */ + stac9766_reset(codec, 0); + ret = stac9766_reset(codec, 1); + if (ret < 0) { + printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n"); + goto reset_err; + } + + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE( + stac9766_snd_ac97_controls)); + + ret = snd_soc_init_card(socdev); + if (ret < 0) + goto reset_err; + return 0; + +reset_err: + snd_soc_free_pcms(socdev); +pcm_err: + snd_soc_free_ac97_codec(codec); +codec_err: + kfree(codec->private_data); +cache_err: + kfree(socdev->card->codec); + socdev->card->codec = NULL; + return ret; +} + +static int stac9766_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec == NULL) + return 0; + + snd_soc_free_pcms(socdev); + snd_soc_free_ac97_codec(codec); + kfree(codec->reg_cache); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_stac9766 = +{ + .probe = stac9766_codec_probe, + .remove = stac9766_codec_remove, + .suspend = stac9766_codec_suspend, + .resume = stac9766_codec_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766); + +static int __init stac9766_probe(struct platform_device *pdev) +{ + snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +#if defined(CONFIG_SND_SOC_OF_SIMPLE) + /* Tell the of_soc helper about this codec */ + of_snd_soc_register_codec(&soc_codec_dev_stac9766, pdev->dev.archdata.of_node, + stac9766_dai, ARRAY_SIZE(stac9766_dai), + pdev->dev.archdata.of_node); +#endif + return 0; +} + +static struct platform_driver stac9766_driver = +{ + .probe = stac9766_probe, + .driver = { + .name = "stac9766", + }, +}; + +static __init int stac9766_driver_init(void) +{ + return platform_driver_register(&stac9766_driver); +} + +static __exit void stac9766_driver_exit(void) +{ +} + +module_init(stac9766_driver_init); +module_exit(stac9766_driver_exit); + +MODULE_DESCRIPTION("ASoC stac9766 driver"); +MODULE_AUTHOR("Jon Smirl jonsmirl@gmail.com"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h new file mode 100644 index 0000000..65642eb --- /dev/null +++ b/sound/soc/codecs/stac9766.h @@ -0,0 +1,21 @@ +/* + * stac9766.h -- STAC9766 Soc Audio driver + */ + +#ifndef _STAC9766_H +#define _STAC9766_H + +#define AC97_STAC_PAGE0 0x1000 +#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A) +#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E) +#define AC97_STAC_STEREO_MIC 0x78 + +/* STAC9766 DAI ID's */ +#define STAC9766_DAI_AC97_ANALOG 0 +#define STAC9766_DAI_AC97_DIGITAL 1 + +extern struct snd_soc_dai stac9766_dai[]; +extern struct snd_soc_codec_device soc_codec_dev_stac9766; + + +#endif

AC97 driver for mpc5200

Signed-off-by: Jon Smirl jonsmirl@gmail.com --- sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 1 sound/soc/fsl/mpc5200_psc_ac97.c | 379 ++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_psc_ac97.h | 13 + 4 files changed, 404 insertions(+), 0 deletions(-) create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 8060b95..d8f63f5 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -30,3 +30,14 @@ config SND_SOC_MPC5200_I2S 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 SND_SOC_OF_SIMPLE + select SND_MPC52xx_DMA + select PPC_BESTCOMM_GEN_BD + help + Say Y here to support the MPC5200 PSCs in AC97 mode. + + diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 7731ef2..14631a1 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -13,4 +13,5 @@ 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

diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c new file mode 100644 index 0000000..aa37c8d --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -0,0 +1,379 @@ +/* + * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip. + * + * Copyright 2008 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/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/time.h> +#include <asm/mpc52xx.h> +#include <asm/mpc52xx_psc.h> + +#include "mpc5200_dma.h" +#include "mpc5200_psc_ac97.h" + +MODULE_AUTHOR("Jon Smirl jonsmirl@gmail.com"); +MODULE_DESCRIPTION("mpc5200 AC97 module"); +MODULE_LICENSE("GPL"); + +#define DRV_NAME "mpc5200-psc-ac97" + +static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + struct psc_dma *psc_dma = ac97->private_data; + int timeout; + unsigned int val; + + spin_lock(&psc_dma->lock); + + /* Wait for it to be ready */ + timeout = 1000; + while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND) ) + udelay(10); + + if (!timeout) { + pr_err(timeout on ac97 bus (rdy)\n"); + return 0xffff; + } + + /* Do the read */ + out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); + + /* Wait for the answer */ + timeout = 1000; + while ((--timeout) && !(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_DATA_VAL) ) + udelay(10); + + if (!timeout) { + pr_err(timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); + return 0xffff; + } + + /* 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 0xffff; + } + val = (val >> 8) & 0xffff; + + spin_unlock(&psc_dma->lock); + return (unsigned short) val; +} + +static void psc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) +{ + struct psc_dma *psc_dma = ac97->private_data; + int timeout; + + spin_lock(&psc_dma->lock); + + /* Wait for it to be ready */ + timeout = 1000; + while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND) ) + udelay(10); + + if (!timeout) { + pr_err(timeout on ac97 write\n"); + return; + } + + /* Write data */ + out_be32(&psc_dma->psc_regs->ac97_cmd, ((reg & 0x7f) << 24) | (val << 8)); + + spin_unlock(&psc_dma->lock); +} + +static void psc_ac97_cold_reset(struct snd_ac97 *ac97) +{ + struct psc_dma *psc_dma = ac97->private_data; + + /* Do a cold reset */ + out_8(&psc_dma->psc_regs->op1, MPC52xx_PSC_OP_RES); + udelay(10); + out_8(&psc_dma->psc_regs->op0, MPC52xx_PSC_OP_RES); + udelay(50); + + /* PSC recover from cold reset (cfr user manual, not sure if useful) */ + out_be32(&psc_dma->psc_regs->sicr, in_be32(&psc_dma->psc_regs->sicr)); +} + +static void psc_ac97_warm_reset(struct snd_ac97 *ac97) +{ + pr_info("psc_ac97_warm_reset\n"); +} + +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); + +#ifdef CONFIG_PM +static int psc_ac97_suspend(struct snd_soc_dai *dai) +{ + return 0; +} + +static int psc_ac97_resume(struct snd_soc_dai *dai) +{ + return 0; +} + +#else +#define psc_ac97_suspend NULL +#define psc_ac97_resume NULL +#endif + +static int psc_ac97_hw_analog_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_dma *psc_dma = rtd->dai->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; + } + + spin_lock(&psc_dma->lock); + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + spin_unlock(&psc_dma->lock); + + return 0; +} + +static int psc_ac97_hw_digital_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_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + spin_lock(&psc_dma->lock); + if (params_channels(params) == 1) + out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000); + else + out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000); + spin_unlock(&psc_dma->lock); + + 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; + + spin_lock(&psc_dma->lock); + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + spin_unlock(&psc_dma->lock); + break; + } + return 0; +} + +/** + * psc_ac97_set_fmt: set the serial format. + * + * This function is called by the machine driver to tell us what serial + * format to use. + * + * This driver only supports AC97 mode. Return an error if the format is + * not SND_SOC_DAIFMT_AC97. + * + * @format: one of SND_SOC_DAIFMT_xxx + */ +static int psc_ac97_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) +{ + struct psc_dma *psc_dma = cpu_dai->private_data; + dev_dbg(psc_dma->dev, "psc_ac97_set_fmt(cpu_dai=%p, format=%i)\n", + cpu_dai, format); + + return (format == SND_SOC_DAIFMT_AC97) ? 0 : -EINVAL; +} + +/* --------------------------------------------------------------------- + * 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, + .set_fmt = psc_ac97_set_fmt, + .trigger = psc_ac97_trigger, +}; + +static struct snd_soc_dai_ops psc_ac97_digital_ops = { + .hw_params = psc_ac97_hw_digital_params, + .set_fmt = psc_ac97_set_fmt, +}; + +static struct snd_soc_dai psc_ac97_dai_template[] = { +{ + .name = "%s analog", + .suspend = psc_ac97_suspend, + .resume = psc_ac97_resume, + .playback = { + .channels_min = 1, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, + }, + .ops = &psc_ac97_analog_ops, +}, +{ + .name = "%s digital", + .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, +}}; + + +/* --------------------------------------------------------------------- + * 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; + struct psc_dma *psc_dma; + + rc = mpc5200_audio_dma_create(op, psc_ac97_dai_template, ARRAY_SIZE(psc_ac97_dai_template)); + if (rc != 0) + return rc; + + psc_dma = dev_get_drvdata(&op->dev); + + psc_dma->imr = 0; + out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); + + out_8(&psc_dma->psc_regs->op1, MPC52xx_PSC_OP_RES); + udelay(10); + out_8(&psc_dma->psc_regs->op0, MPC52xx_PSC_OP_RES); + udelay(50); + + /* Configure the serial interface mode to AC97 */ + psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97; + out_be32(&psc_dma->psc_regs->sicr, psc_dma->sicr); + + /* No slots active */ + out_be32(&psc_dma->psc_regs->ac97_slots, 0x00000000); + + /* Go */ + out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_TX_ENABLE); + out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RX_ENABLE); + + 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); diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h new file mode 100644 index 0000000..d028c55 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.h @@ -0,0 +1,13 @@ +/* + * 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__ + +#define MPC5200_AC97_ANALOG 0 +#define MPC5200_AC97_DIGITAL 1 + +#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */

Support for AC97 on Phytec pmc030 base board. This uses the generic fabric driver binding.

Signed-off-by: Jon Smirl jonsmirl@gmail.com --- arch/powerpc/boot/dts/pcm030.dts | 8 ++++++++ sound/soc/fsl/Kconfig | 7 +++++++ 2 files changed, 15 insertions(+), 0 deletions(-)

diff --git a/arch/powerpc/boot/dts/pcm030.dts b/arch/powerpc/boot/dts/pcm030.dts index 8958347..001c8ef 100644 --- a/arch/powerpc/boot/dts/pcm030.dts +++ b/arch/powerpc/boot/dts/pcm030.dts @@ -190,6 +190,14 @@ cell-index = <0>; reg = <0x2000 0x100>; interrupts = <2 1 0>; + + #address-cells = <1>; + #size-cells = <0>; + + codec@0 { + compatible = "wlf,wm9712"; + reg = <0x0>; + }; };

/* PSC2 port is used by CAN1/2 */ diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 266c6b7..11735e4 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -48,4 +48,11 @@ config SND_MPC52xx_SOC_EFIKA help Say Y if you want to add support for sound on the Efika.

+config SND_MPC52xx_SOC_PCM030 + tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712" + depends on PPC_MPC5200_SIMPLE + 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.