[alsa-devel] [PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers
From: Grant Likely grant.likely@secretlab.ca
Simple utility layer for creating ASoC machine instances based on data in the OpenFirmware device tree. OF aware platform drivers and codec drivers register themselves with this framework and the framework automatically instantiates a machine driver.
This is most likely temporary glue code to work around limitations in the ASoC v1 framework. I expect ASoC v2 won't need this. ---
sound/soc/Kconfig | 6 ++ sound/soc/Makefile | 1 sound/soc/soc-of.c | 171 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 178 insertions(+), 0 deletions(-)
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 18f28ac..c5736e5 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -23,6 +23,12 @@ config SND_SOC This ASoC audio support can also be built as a module. If so, the module will be called snd-soc-core.
+config SND_SOC_OF + tristate "OF helpers for SoC audio support" + depends on SND_SOC + ---help--- + Add support for OpenFirmware device tree descriptions of sound device + # All the supported Soc's source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 782db21..191c2e5 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -2,3 +2,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o
obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/ +obj-$(CONFIG_SND_SOC_OF) += soc-of.o diff --git a/sound/soc/soc-of.c b/sound/soc/soc-of.c new file mode 100644 index 0000000..9694979 --- /dev/null +++ b/sound/soc/soc-of.c @@ -0,0 +1,171 @@ +/* + * OF helpers for ALSA SoC Layer + * + * Copyright (C) 2008, Secret Lab Technologies Ltd. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/bitops.h> +#include <linux/platform_device.h> +#include <linux/of.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-of.h> +#include <sound/initval.h> + +MODULE_AUTHOR("Grant Likely grant.likely@secretlab.ca"); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("ALSA SoC OpenFirmware bindings"); + +DEFINE_MUTEX(of_snd_soc_mutex); +LIST_HEAD(of_snd_soc_device_list); +static int of_snd_soc_next_index; + +struct of_snd_soc_device { + int id; + struct list_head list; + struct snd_soc_device device; + struct snd_soc_machine machine; + struct snd_soc_dai_link dai_link; + struct platform_device *pdev; + struct device_node *platform_node; + struct device_node *codec_node; +}; + +static struct snd_soc_ops of_snd_soc_ops = { +}; + +static struct of_snd_soc_device * +of_snd_soc_get_device(struct device_node *codec_node) +{ + struct of_snd_soc_device *of_soc; + + list_for_each_entry(of_soc, &of_snd_soc_device_list, list) { + if (of_soc->codec_node == codec_node) + return of_soc; + } + + of_soc = kzalloc(sizeof(struct of_snd_soc_device), GFP_KERNEL); + if (!of_soc) + return NULL; + + /* 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->machine.dai_link = &of_soc->dai_link; + of_soc->machine.num_links = 1; + of_soc->device.machine = &of_soc->machine; + of_soc->dai_link.ops = &of_snd_soc_ops; + list_add(&of_soc->list, &of_snd_soc_device_list); + + return of_soc; +} + +static void of_snd_soc_register_device(struct of_snd_soc_device *of_soc) +{ + struct platform_device *pdev; + int rc; + + /* Only register the device if both the codec and platform have + * been registered */ + if ((!of_soc->device.codec_data) || (!of_soc->platform_node)) + return; + + pr_info("platform<-->codec match achieved; registering machine\n"); + + pdev = platform_device_alloc("soc-audio", of_soc->id); + if (!pdev) { + pr_err("of_soc: platform_device_alloc() failed\n"); + return; + } + + pdev->dev.platform_data = of_soc; + platform_set_drvdata(pdev, &of_soc->device); + of_soc->device.dev = &pdev->dev; + + /* The ASoC device is complete; register it */ + rc = platform_device_add(pdev); + if (rc) { + pr_err("of_soc: platform_device_add() failed\n"); + return; + } + +} + +int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev, + void *codec_data, struct snd_soc_codec_dai *dai, + struct device_node *node) +{ + struct of_snd_soc_device *of_soc; + int rc = 0; + + pr_info("registering ASoC codec driver: %s\n", node->full_name); + + mutex_lock(&of_snd_soc_mutex); + of_soc = of_snd_soc_get_device(node); + if (!of_soc) { + rc = -ENOMEM; + goto out; + } + + /* Store the codec data */ + of_soc->device.codec_data = codec_data; + of_soc->device.codec_dev = codec_dev; + of_soc->dai_link.name = node->name; + of_soc->dai_link.stream_name = node->name; + of_soc->dai_link.codec_dai = dai; + + /* Now try to register the SoC device */ + of_snd_soc_register_device(of_soc); + + out: + mutex_unlock(&of_snd_soc_mutex); + return rc; +} +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_cpu_dai *cpu_dai) +{ + struct of_snd_soc_device *of_soc; + struct device_node *codec_node; + const phandle *handle; + int len, rc = 0; + + pr_info("registering ASoC platform 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 (!codec_node) + return -ENODEV; + pr_info("looking for codec: %s\n", codec_node->full_name); + + mutex_lock(&of_snd_soc_mutex); + of_soc = of_snd_soc_get_device(codec_node); + if (!of_soc) { + rc = -ENOMEM; + goto out; + } + + of_soc->platform_node = node; + of_soc->dai_link.cpu_dai = cpu_dai; + of_soc->device.platform = platform; + of_soc->machine.name = of_soc->dai_link.cpu_dai->name; + + /* Now try to register the SoC device */ + of_snd_soc_register_device(of_soc); + + out: + mutex_unlock(&of_snd_soc_mutex); + return rc; +} +EXPORT_SYMBOL_GPL(of_snd_soc_register_platform);
From: Grant Likely grant.likely@secretlab.ca
This is an I2S bus driver for the MPC5200 PSC device. It is probably will not be merged as-is because it uses v1 of the ASoC API, but I want to get it out there for comments. ---
sound/soc/fsl/Kconfig | 6 sound/soc/fsl/Makefile | 2 sound/soc/fsl/mpc5200_psc_i2s.c | 899 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 907 insertions(+), 0 deletions(-)
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 257101f..5daa8d3 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -17,4 +17,10 @@ config SND_SOC_MPC8610_HPCD help Say Y if you want to enable audio on the Freescale MPC8610 HPCD.
+config SND_SOC_MPC5200_I2S + bool "Freescale MPC5200 PSC in I2S mode driver" + depends on SND_SOC && PPC_MPC52xx + help + Say Y here to support the MPC5200 PSCs in I2S mode. + endmenu diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 62f680a..98729a1 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -4,3 +4,5 @@ obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o # MPC8610 Platform Support obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o
+obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o + diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c new file mode 100644 index 0000000..81d0933 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -0,0 +1,899 @@ +/* + * Freescale MPC5200 PSC in I2S mode + * ALSA SoC Digital Audio Interface (DAI) 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.h> + +#include <sysdev/bestcomm/bestcomm.h> +#include <sysdev/bestcomm/gen_bd.h> +#include <asm/mpc52xx_psc.h> + +MODULE_AUTHOR("Grant Likely grant.likely@secretlab.ca"); +MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver"); +MODULE_LICENSE("GPL"); + +/** + * PSC_I2S_RATES: sample rates supported by the I2S + * + * This driver currently only supports the PSC running in I2S slave mode, + * which means the codec determines the sample rate. Therefore, we tell + * ALSA that we support all rates and let the codec driver decide what rates + * are really supported. + */ +#define PSC_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \ + SNDRV_PCM_RATE_CONTINUOUS) + +/** + * PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode + */ +#define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_S32_BE) + +/** + * psc_i2s_stream - Data specific to a single stream (playback or capture) + * @active: flag indicating if the stream is active + * @psc_i2s: pointer back to parent psc_i2s data structure + * @bcom_task: bestcomm task structure + * @irq: irq number for bestcomm task + * @period_start: physical address of start of DMA region + * @period_end: physical address of end of DMA region + * @period_next_pt: physical address of next DMA buffer to enqueue + * @period_bytes: size of DMA period in bytes + */ +struct psc_i2s_stream { + int active; + struct psc_i2s *psc_i2s; + struct bcom_task *bcom_task; + int irq; + struct snd_pcm_substream *stream; + dma_addr_t period_start; + dma_addr_t period_end; + dma_addr_t period_next_pt; + dma_addr_t period_current_pt; + int period_bytes; +}; + +/** + * psc_i2s - Private driver data + * @name: short name for this device ("PSC0", "PSC1", etc) + * @psc_regs: pointer to the PSC's registers + * @fifo_regs: pointer to the PSC's FIFO registers + * @irq: IRQ of this PSC + * @dev: struct device pointer + * @playback: the number of playback streams opened + * @capture: the number of capture streams opened + * @dai: the CPU DAI for this device + * @playback_stream: Playback stream context data + * @capture_stream: 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_cpu_dai dai; + spinlock_t lock; + + /* per-stream data */ + struct psc_i2s_stream playback_stream; + struct psc_i2s_stream capture_stream; + + /* 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 imr; + u16 isr; + + isr = in_be16(®s->mpc52xx_psc_isr); + imr = in_be16(®s->mpc52xx_psc_imr); + + /* Playback underrun error */ + if (isr & imr & MPC52xx_PSC_IMR_TXEMP) + psc_i2s->stats.underrun_count++; + + /* Capture overrun error */ + if (isr & imr & MPC52xx_PSC_IMR_ORERR) + psc_i2s->stats.overrun_count++; + + out_8(®s->command, 4 << 4); /* reset the error status */ + + return IRQ_HANDLED; +} + +/** + * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer + * @s: pointer to stream private data structure + * + * Enqueues another audio period buffer into the bestcomm queue. + * + * Note: The routine must only be called when there is space available in + * the queue. Otherwise the enqueue will fail and the audio ring buffer + * will get out of sync + */ +static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s) +{ + struct bcom_bd *bd; + + /* Prepare and enqueue the next buffer descriptor */ + bd = bcom_prepare_next_buffer(s->bcom_task); + bd->status = s->period_bytes; + bd->data[0] = s->period_next_pt; + bcom_submit_next_buffer(s->bcom_task, NULL); + + /* Update for next period */ + s->period_next_pt += s->period_bytes; + if (s->period_next_pt >= s->period_end) + s->period_next_pt = s->period_start; +} + +/* Bestcomm DMA irq handler */ +static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream) +{ + struct psc_i2s_stream *s = _psc_i2s_stream; + + //spin_lock(&s->psc_i2s->lock); + + /* For each finished period, dequeue the completed period buffer + * and enqueue a new one in it's place. */ + while (bcom_buffer_done(s->bcom_task)) { + bcom_retrieve_buffer(s->bcom_task, NULL, NULL); + s->period_current_pt += s->period_bytes; + if (s->period_current_pt >= s->period_end) + s->period_current_pt = s->period_start; + psc_i2s_bcom_enqueue_next_buffer(s); + bcom_enable(s->bcom_task); + } + + //spin_unlock(&s->psc_i2s->lock); + + /* If the stream is active, then also inform the PCM middle layer + * of the period finished event. */ + if (s->active) + snd_pcm_period_elapsed(s->stream); + + return IRQ_HANDLED; +} + +/** + * 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) +{ + int playback_irq, capture_irq, rc; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; + struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs; + struct mpc52xx_psc_fifo __iomem *fiforegs = psc_i2s->fifo_regs; + + dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream); + + /* Disable all interrupts and reset the PSC */ + out_be16(®s->mpc52xx_psc_imr, 0); + out_8(®s->command, 3 << 4); /* reset transmitter */ + out_8(®s->command, 2 << 4); /* reset receiver */ + out_8(®s->command, 1 << 4); /* reset mode */ + out_8(®s->command, 4 << 4); /* reset error */ + + /* Default to CODEC8 mode */ + out_be32(®s->sicr, + MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S | + MPC52xx_PSC_SICR_CLKPOL | MPC52xx_PSC_SICR_SIM_CODEC_8); + + /* First write: RxRdy (FIFO Alarm) generates receive FIFO interrupt */ + /* Second write to mode: register Normal mode for non loopback */ + out_8(®s->mode, 0); + out_8(®s->mode, 0); + + /* Set the TX and RX fifo alarm thresholds */ + out_be16(&fiforegs->rfalarm, 0x100); /* set RFALARM level */ + out_8(&fiforegs->rfcntl, 0x4); /* set RFGRAN level (bytes) */ + out_be16(&fiforegs->tfalarm, 0x100); /* set TFALARM level */ + out_8(&fiforegs->tfcntl, 0x7); /* set TFGRAN level (bytes*4) */ + + /* Setup the IRQs */ + playback_irq = bcom_get_task_irq(psc_i2s->playback_stream.bcom_task); + capture_irq = bcom_get_task_irq(psc_i2s->capture_stream.bcom_task); + rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED, + "psc-i2s-status", psc_i2s); + rc |= request_irq(capture_irq, &psc_i2s_bcom_irq, IRQF_SHARED, + "psc-i2s-capture", &psc_i2s->capture_stream); + rc |= request_irq(playback_irq, &psc_i2s_bcom_irq, IRQF_SHARED, + "psc-i2s-playback", &psc_i2s->playback_stream); + if (rc) { + free_irq(psc_i2s->irq, psc_i2s); + free_irq(capture_irq, &psc_i2s->capture_stream); + free_irq(playback_irq, &psc_i2s->playback_stream); + return -ENODEV; + } + + return 0; +} + +static int psc_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; + u32 sicr; + + dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i" + " periods=%i buffer_size=%i buffer_bytes=%i\n", + __FUNCTION__, substream, params_period_size(params), + params_period_bytes(params), params_periods(params), + params_buffer_size(params), params_buffer_bytes(params)); + + sicr = MPC52xx_PSC_SICR_DTS1 | + MPC52xx_PSC_SICR_I2S | MPC52xx_PSC_SICR_CLKPOL; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + sicr |= MPC52xx_PSC_SICR_SIM_CODEC_8; + break; + case SNDRV_PCM_FORMAT_S16_BE: + sicr |= MPC52xx_PSC_SICR_SIM_CODEC_16; + break; + case SNDRV_PCM_FORMAT_S24_BE: + sicr |= MPC52xx_PSC_SICR_SIM_CODEC_24; + break; + case SNDRV_PCM_FORMAT_S32_BE: + sicr |= MPC52xx_PSC_SICR_SIM_CODEC_32; + break; + default: + dev_dbg(psc_i2s->dev, "invalid format\n"); + return -EINVAL; + } + out_be32(&psc_i2s->psc_regs->sicr, sicr); + + //rc = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + //if (rc) { + // dev_err(psc_i2s->dev, "could not allocate dma buffer\n"); + // return rc; + //} + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static int psc_i2s_hw_free(struct snd_pcm_substream *substream) +{ + //return snd_pcm_lib_free_pages(substream); + 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_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; + long flags; + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + s = &psc_i2s->capture_stream; + else + s = &psc_i2s->playback_stream; + + dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)" + " stream_id=%i\n", + substream, cmd, substream->pstr->stream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + s->period_bytes = frames_to_bytes(runtime, + runtime->period_size); + s->period_start = virt_to_phys(runtime->dma_area); + s->period_end = s->period_start + + (s->period_bytes * runtime->periods); + s->period_next_pt = s->period_start; + s->period_current_pt = s->period_start; + s->active = 1; + + /* First; reset everything */ + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + out_8(®s->command, MPC52xx_PSC_RST_RX); + out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT); + } else { + out_8(®s->command, MPC52xx_PSC_RST_TX); + out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT); + } + + /* Next, fill up the bestcomm bd queue and enable DMA. + * This will begin filling the PSC's fifo. */ + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + bcom_gen_bd_rx_reset(s->bcom_task); + else + bcom_gen_bd_tx_reset(s->bcom_task); + while (!bcom_queue_full(s->bcom_task)) + psc_i2s_bcom_enqueue_next_buffer(s); + bcom_enable(s->bcom_task); + + /* Update interrupt enable settings. This must be done + * before the PSC is enabled so that TX underrun events + * are not missed. */ + imr = 0; + if (psc_i2s->playback_stream.active) + imr |= MPC52xx_PSC_IMR_TXEMP; + if (psc_i2s->capture_stream.active) + imr |= MPC52xx_PSC_IMR_ORERR; + out_be16(®s->isr_imr.imr, imr); + + /* Due to errata in the i2s mode; need to line up enabling + * the transmitter with a transition on the frame sync + * line */ + + spin_lock_irqsave(&psc_i2s->lock, flags); + /* first make sure it is low */ + while ((in_8(®s->ipcr_acr.ipcr) & 0x80) != 0); + /* then wait for the transition to high */ + while ((in_8(®s->ipcr_acr.ipcr) & 0x80) == 0); + /* Finally, enable the PSC. + * Receiver must always be enabled; even when we only want + * transmit. (see 15.3.2.3 of MPC5200B User's Guide) */ + psc_cmd = MPC52xx_PSC_RX_ENABLE; + if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK) + psc_cmd |= MPC52xx_PSC_TX_ENABLE; + out_8(®s->command, psc_cmd); + spin_unlock_irqrestore(&psc_i2s->lock, flags); + + break; + + case SNDRV_PCM_TRIGGER_STOP: + /* Turn off the PSC */ + s->active = 0; + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (!psc_i2s->playback_stream.active) { + out_8(®s->command, 2 << 4); /* reset rx */ + out_8(®s->command, 3 << 4); /* reset tx */ + out_8(®s->command, 4 << 4); /* reset err */ + } + } else { + out_8(®s->command, 3 << 4); /* reset tx */ + out_8(®s->command, 4 << 4); /* reset err */ + if (!psc_i2s->capture_stream.active) + out_8(®s->command, 2 << 4); /* reset rx */ + } + + bcom_disable(s->bcom_task); + while (!bcom_queue_empty(s->bcom_task)) + bcom_retrieve_buffer(s->bcom_task, NULL, NULL); + + break; + + default: + dev_dbg(psc_i2s->dev, "invalid command\n"); + return -EINVAL; + } + + /* Update interrupt enable settings */ + imr = 0; + if (psc_i2s->playback_stream.active) imr |= MPC52xx_PSC_IMR_TXEMP; + if (psc_i2s->capture_stream.active) imr |= MPC52xx_PSC_IMR_ORERR; + out_be16(®s->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_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_stream.active && + !psc_i2s->capture_stream.active) { + /* TODO: shut off channels */ + free_irq(psc_i2s->irq, psc_i2s); + free_irq(bcom_get_task_irq(psc_i2s->capture_stream.bcom_task), + &psc_i2s->capture_stream); + free_irq(bcom_get_task_irq(psc_i2s->playback_stream.bcom_task), + &psc_i2s->playback_stream); + } +} + +/** + * psc_i2s_set_sysclk: set the clock frequency and direction + * + * This function is called by the machine driver to tell us what the clock + * frequency and direction are. + * + * Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN), + * and we don't care about the frequency. Return an error if the direction + * is not SND_SOC_CLOCK_IN. + * + * @clk_id: reserved, should be zero + * @freq: the frequency of the given clock ID, currently ignored + * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master) + */ +static int psc_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct psc_i2s *psc_i2s = cpu_dai->private_data; + dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n", + cpu_dai, dir); + return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL; +} + +/** + * psc_i2s_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 I2S mode. Return an error if the format is + * not SND_SOC_DAIFMT_I2S. + * + * @format: one of SND_SOC_DAIFMT_xxx + */ +static int psc_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format) +{ + struct psc_i2s *psc_i2s = cpu_dai->private_data; + dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n", + cpu_dai, format); + return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL; +} + +/* --------------------------------------------------------------------- + * ALSA SoC Bindings + * + * - Digital Audio Interface (DAI) template + * - create/destroy dai hooks + */ + +/** + * psc_i2s_dai_template: template CPU Digital Audio Interface + */ +static struct snd_soc_cpu_dai psc_i2s_dai_template = { + .type = SND_SOC_DAI_I2S, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = PSC_I2S_RATES, + .formats = PSC_I2S_FORMATS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = PSC_I2S_RATES, + .formats = PSC_I2S_FORMATS, + }, + .ops = { + .startup = psc_i2s_startup, + .hw_params = psc_i2s_hw_params, + .hw_free = psc_i2s_hw_free, + .shutdown = psc_i2s_shutdown, + .trigger = psc_i2s_trigger, + }, + .dai_ops = { + .set_sysclk = psc_i2s_set_sysclk, + .set_fmt = psc_i2s_set_fmt, + }, +}; + +/* --------------------------------------------------------------------- + * 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, + .period_bytes_max = 1024 * 1024, + .periods_min = 2, + .periods_max = 256, + .buffer_bytes_max = 2 * 1024 * 1024, + .fifo_size = 0, +}; + +static unsigned int psc_i2s_fixed_rates[] = { + 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, +}; + +static struct snd_pcm_hw_constraint_list psc_i2s_constraints_rates = { + .count = ARRAY_SIZE(psc_i2s_fixed_rates), + .list = psc_i2s_fixed_rates, + .mask = 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; + int rc; + + 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_stream; + else + s = &psc_i2s->playback_stream; + + snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware); + + rc = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (rc < 0) { + dev_err(psc_i2s->dev, "invalid buffer size\n"); + return rc; + } + rc = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &psc_i2s_constraints_rates); + if (rc < 0) { + dev_err(psc_i2s->dev, "invalid rate\n"); + return rc; + } + + 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_stream; + else + s = &psc_i2s->playback_stream; + + 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_stream; + else + s = &psc_i2s->playback_stream; + + /*FIXME: count = s->sdma->bd[s->sdma->outdex].data - s->period_start;*/ + 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_codec_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; + + rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size, + &pcm->streams[0].substream->dma_buffer); + if (rc) { + dev_err(card->dev, "Cannot alloc playback DMA buffer\n"); + return -ENOMEM; + } + + rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size, + &pcm->streams[1].substream->dma_buffer); + if (rc) { + snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); + dev_err(card->dev, "Can't allocate capture DMA buffer\n"); + return -ENOMEM; + } + + return 0; +} + +static void psc_i2s_pcm_free(struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *rtd = pcm->private_data; + struct snd_pcm_substream *substream; + int stream; + + dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm); + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (substream) { + snd_dma_free_pages(&substream->dma_buffer); + substream->dma_buffer.area = NULL; + substream->dma_buffer.addr = 0; + } + } +} + +struct snd_soc_platform psc_i2s_pcm_soc_platform = { + .name = "mpc5200-psc-audio", + .pcm_ops = &psc_i2s_pcm_ops, + .pcm_new = &psc_i2s_pcm_new, + .pcm_free = &psc_i2s_pcm_free, +}; + +/* --------------------------------------------------------------------- + * Sysfs attributes for debugging + */ + +static ssize_t psc_i2s_status_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct psc_i2s *psc_i2s = dev_get_drvdata(dev); + + return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x tfnum=%i tfstat=0x%.4x\n", + in_be16(&psc_i2s->psc_regs->sr_csr.status), + in_be32(&psc_i2s->psc_regs->sicr), + in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff, + in_be16(&psc_i2s->fifo_regs->rfstat), + in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff, + in_be16(&psc_i2s->fifo_regs->tfstat)); +} + +static int * psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s, + const char *name) +{ + if (strcmp(name, "playback_underrun") == 0) + return &psc_i2s->stats.underrun_count; + if (strcmp(name, "capture_overrun") == 0) + return &psc_i2s->stats.overrun_count; + + return NULL; +} + +static ssize_t psc_i2s_stat_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct psc_i2s *psc_i2s = dev_get_drvdata(dev); + int *attrib; + + attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name); + if (!attrib) + return 0; + + return sprintf(buf, "%i\n", *attrib); +} + +static ssize_t psc_i2s_stat_store(struct device *dev, + struct device_attribute *attr, + const char *buf, + size_t count) +{ + struct psc_i2s *psc_i2s = dev_get_drvdata(dev); + int *attrib; + + attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name); + if (!attrib) + return 0; + + *attrib = simple_strtoul(buf, NULL, 0); + return count; +} + +DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL); +DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show,psc_i2s_stat_store); +DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store); + +/* --------------------------------------------------------------------- + * OF platform bus binding code: + * - Probe/remove operations + * - OF device match table + */ +static int __devinit psc_i2s_of_probe(struct of_device *op, + const struct of_device_id *match) +{ + phys_addr_t fifo; + struct psc_i2s *psc_i2s; + struct resource res; + int size, psc_id, irq, rc; + const __be32 *prop; + void __iomem *regs; + + dev_dbg(&op->dev, "probing psc i2s device\n"); + + /* Get the PSC ID */ + prop = of_get_property(op->node, "cell-index", &size); + if (!prop || size < sizeof *prop) + return -ENODEV; + psc_id = be32_to_cpu(*prop); + + /* Fetch the registers and IRQ of the PSC */ + irq = irq_of_parse_and_map(op->node, 0); + if (of_address_to_resource(op->node, 0, &res)) { + dev_err(&op->dev, "Missing reg property\n"); + return -ENODEV; + } + regs = ioremap(res.start, 1 + res.end - res.start); + if (!regs) { + dev_err(&op->dev, "Could not map registers\n"); + return -ENODEV; + } + + /* Allocate and initialize the driver private data */ + psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL); + if (!psc_i2s) { + iounmap(regs); + return -ENOMEM; + } + spin_lock_init(&psc_i2s->lock); + psc_i2s->irq = irq; + psc_i2s->psc_regs = regs; + psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs; + psc_i2s->dev = &op->dev; + psc_i2s->playback_stream.psc_i2s = psc_i2s; + psc_i2s->capture_stream.psc_i2s = psc_i2s; + snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1); + + /* Fill out the CPU DAI structure */ + memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai); + psc_i2s->dai.private_data = psc_i2s; + psc_i2s->dai.name = psc_i2s->name; + psc_i2s->dai.id = psc_id; + + /* Find the address of the fifo data registers and setup the + * DMA tasks */ + fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32); + psc_i2s->capture_stream.bcom_task = + bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512); + psc_i2s->playback_stream.bcom_task = + bcom_psc_gen_bd_tx_init(psc_id, 10, fifo); + if (!psc_i2s->capture_stream.bcom_task || + !psc_i2s->playback_stream.bcom_task) { + dev_err(&op->dev, "Could not allocate bestcomm tasks\n"); + iounmap(regs); + kfree(psc_i2s); + return -ENODEV; + } + + /* Save what we've done so it can be found again later */ + dev_set_drvdata(&op->dev, psc_i2s); + + /* Register the SYSFS files */ + rc = device_create_file(psc_i2s->dev, &dev_attr_status); + rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun); + rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun); + if (rc) + dev_info(psc_i2s->dev, "error creating sysfs files\n"); + + /* Tell the ASoC OF helpers about it */ + of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node, + &psc_i2s->dai); + + return 0; +} + +static int __devexit psc_i2s_of_remove(struct of_device *op) +{ + struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev); + + dev_dbg(&op->dev, "psc_i2s_remove()\n"); + + bcom_gen_bd_rx_release(psc_i2s->capture_stream.bcom_task); + bcom_gen_bd_tx_release(psc_i2s->playback_stream.bcom_task); + + iounmap(psc_i2s->psc_regs); + iounmap(psc_i2s->fifo_regs); + kfree(psc_i2s); + dev_set_drvdata(&op->dev, NULL); + + return 0; +} + +/* Match table for of_platform binding */ +static struct of_device_id psc_i2s_match[] __devinitdata = { + { .compatible = "fsl,mpc5200-psc-i2s", }, + {} +}; +MODULE_DEVICE_TABLE(of, psc_i2s_match); + +static struct of_platform_driver psc_i2s_driver = { + .match_table = psc_i2s_match, + .probe = psc_i2s_of_probe, + .remove = __devexit_p(psc_i2s_of_remove), + .driver = { + .name = "mpc5200-psc-i2s", + .owner = THIS_MODULE, + }, +}; + +/* --------------------------------------------------------------------- + * Module setup and teardown; simply register the of_platform driver + * for the PSC in I2S mode. + */ +static int __init psc_i2s_init(void) +{ + return of_register_platform_driver(&psc_i2s_driver); +} +module_init(psc_i2s_init); + +static void __exit psc_i2s_exit(void) +{ + of_unregister_platform_driver(&psc_i2s_driver); +} +module_exit(psc_i2s_exit); + +
On Tue, 2008-07-01 at 17:53 -0600, Grant Likely wrote:
From: Grant Likely grant.likely@secretlab.ca
This is an I2S bus driver for the MPC5200 PSC device. It is probably will not be merged as-is because it uses v1 of the ASoC API, but I want to get it out there for comments.
Looks good, just minor comments.
sound/soc/fsl/Kconfig | 6 sound/soc/fsl/Makefile | 2 sound/soc/fsl/mpc5200_psc_i2s.c | 899 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 907 insertions(+), 0 deletions(-)
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 257101f..5daa8d3 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -17,4 +17,10 @@ config SND_SOC_MPC8610_HPCD help Say Y if you want to enable audio on the Freescale MPC8610 HPCD.
+config SND_SOC_MPC5200_I2S
- bool "Freescale MPC5200 PSC in I2S mode driver"
Is this built-in only ?
- depends on SND_SOC && PPC_MPC52xx
- help
Say Y here to support the MPC5200 PSCs in I2S mode.endmenu diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 62f680a..98729a1 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -4,3 +4,5 @@ obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o # MPC8610 Platform Support obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o
+obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c new file mode 100644 index 0000000..81d0933 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -0,0 +1,899 @@ +/*
- Freescale MPC5200 PSC in I2S mode
- ALSA SoC Digital Audio Interface (DAI) 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.h>
+#include <sysdev/bestcomm/bestcomm.h> +#include <sysdev/bestcomm/gen_bd.h> +#include <asm/mpc52xx_psc.h>
+MODULE_AUTHOR("Grant Likely grant.likely@secretlab.ca"); +MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver"); +MODULE_LICENSE("GPL");
+/**
- PSC_I2S_RATES: sample rates supported by the I2S
- This driver currently only supports the PSC running in I2S slave mode,
- which means the codec determines the sample rate. Therefore, we tell
- ALSA that we support all rates and let the codec driver decide what rates
- are really supported.
- */
+#define PSC_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \
SNDRV_PCM_RATE_CONTINUOUS)+/**
- PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode
- */
+#define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \SNDRV_PCM_FMTBIT_S32_BE)+/**
- psc_i2s_stream - Data specific to a single stream (playback or capture)
- @active: flag indicating if the stream is active
- @psc_i2s: pointer back to parent psc_i2s data structure
- @bcom_task: bestcomm task structure
- @irq: irq number for bestcomm task
- @period_start: physical address of start of DMA region
- @period_end: physical address of end of DMA region
- @period_next_pt: physical address of next DMA buffer to enqueue
- @period_bytes: size of DMA period in bytes
- */
+struct psc_i2s_stream {
- int active;
- struct psc_i2s *psc_i2s;
- struct bcom_task *bcom_task;
- int irq;
- struct snd_pcm_substream *stream;
- dma_addr_t period_start;
- dma_addr_t period_end;
- dma_addr_t period_next_pt;
- dma_addr_t period_current_pt;
- int period_bytes;
+};
+/**
- psc_i2s - Private driver data
- @name: short name for this device ("PSC0", "PSC1", etc)
- @psc_regs: pointer to the PSC's registers
- @fifo_regs: pointer to the PSC's FIFO registers
- @irq: IRQ of this PSC
- @dev: struct device pointer
- @playback: the number of playback streams opened
- @capture: the number of capture streams opened
- @dai: the CPU DAI for this device
- @playback_stream: Playback stream context data
- @capture_stream: 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_cpu_dai dai;
- spinlock_t lock;
- /* per-stream data */
- struct psc_i2s_stream playback_stream;
- struct psc_i2s_stream capture_stream;
- /* 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 imr;
- u16 isr;
- isr = in_be16(®s->mpc52xx_psc_isr);
- imr = in_be16(®s->mpc52xx_psc_imr);
- /* Playback underrun error */
- if (isr & imr & MPC52xx_PSC_IMR_TXEMP)
psc_i2s->stats.underrun_count++;- /* Capture overrun error */
- if (isr & imr & MPC52xx_PSC_IMR_ORERR)
psc_i2s->stats.overrun_count++;- out_8(®s->command, 4 << 4); /* reset the error status */
- return IRQ_HANDLED;
+}
+/**
- psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer
- @s: pointer to stream private data structure
- Enqueues another audio period buffer into the bestcomm queue.
- Note: The routine must only be called when there is space available in
- the queue. Otherwise the enqueue will fail and the audio ring buffer
- will get out of sync
- */
+static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s) +{
- struct bcom_bd *bd;
- /* Prepare and enqueue the next buffer descriptor */
- bd = bcom_prepare_next_buffer(s->bcom_task);
- bd->status = s->period_bytes;
- bd->data[0] = s->period_next_pt;
- bcom_submit_next_buffer(s->bcom_task, NULL);
- /* Update for next period */
- s->period_next_pt += s->period_bytes;
- if (s->period_next_pt >= s->period_end)
s->period_next_pt = s->period_start;+}
+/* Bestcomm DMA irq handler */ +static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream) +{
- struct psc_i2s_stream *s = _psc_i2s_stream;
- //spin_lock(&s->psc_i2s->lock);
- /* For each finished period, dequeue the completed period buffer
* and enqueue a new one in it's place. */- while (bcom_buffer_done(s->bcom_task)) {
bcom_retrieve_buffer(s->bcom_task, NULL, NULL);s->period_current_pt += s->period_bytes;if (s->period_current_pt >= s->period_end)s->period_current_pt = s->period_start;psc_i2s_bcom_enqueue_next_buffer(s);bcom_enable(s->bcom_task);- }
- //spin_unlock(&s->psc_i2s->lock);
- /* If the stream is active, then also inform the PCM middle layer
* of the period finished event. */- if (s->active)
snd_pcm_period_elapsed(s->stream);- return IRQ_HANDLED;
+}
+/**
- 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) +{
- int playback_irq, capture_irq, rc;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
- struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
- struct mpc52xx_psc_fifo __iomem *fiforegs = psc_i2s->fifo_regs;
- dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream);
- /* Disable all interrupts and reset the PSC */
- out_be16(®s->mpc52xx_psc_imr, 0);
- out_8(®s->command, 3 << 4); /* reset transmitter */
- out_8(®s->command, 2 << 4); /* reset receiver */
- out_8(®s->command, 1 << 4); /* reset mode */
- out_8(®s->command, 4 << 4); /* reset error */
- /* Default to CODEC8 mode */
- out_be32(®s->sicr,
MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |MPC52xx_PSC_SICR_CLKPOL | MPC52xx_PSC_SICR_SIM_CODEC_8);- /* First write: RxRdy (FIFO Alarm) generates receive FIFO interrupt */
- /* Second write to mode: register Normal mode for non loopback */
- out_8(®s->mode, 0);
- out_8(®s->mode, 0);
- /* Set the TX and RX fifo alarm thresholds */
- out_be16(&fiforegs->rfalarm, 0x100); /* set RFALARM level */
- out_8(&fiforegs->rfcntl, 0x4); /* set RFGRAN level (bytes) */
- out_be16(&fiforegs->tfalarm, 0x100); /* set TFALARM level */
- out_8(&fiforegs->tfcntl, 0x7); /* set TFGRAN level (bytes*4) */
- /* Setup the IRQs */
- playback_irq = bcom_get_task_irq(psc_i2s->playback_stream.bcom_task);
- capture_irq = bcom_get_task_irq(psc_i2s->capture_stream.bcom_task);
- rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED,
"psc-i2s-status", psc_i2s);- rc |= request_irq(capture_irq, &psc_i2s_bcom_irq, IRQF_SHARED,
"psc-i2s-capture", &psc_i2s->capture_stream);- rc |= request_irq(playback_irq, &psc_i2s_bcom_irq, IRQF_SHARED,
"psc-i2s-playback", &psc_i2s->playback_stream);- if (rc) {
free_irq(psc_i2s->irq, psc_i2s);free_irq(capture_irq, &psc_i2s->capture_stream);free_irq(playback_irq, &psc_i2s->playback_stream);return -ENODEV;- }
- return 0;
+}
+static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)+{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
- u32 sicr;
- dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
" periods=%i buffer_size=%i buffer_bytes=%i\n",__FUNCTION__, substream, params_period_size(params),params_period_bytes(params), params_periods(params),params_buffer_size(params), params_buffer_bytes(params));- sicr = MPC52xx_PSC_SICR_DTS1 |
MPC52xx_PSC_SICR_I2S | MPC52xx_PSC_SICR_CLKPOL;- switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8:sicr |= MPC52xx_PSC_SICR_SIM_CODEC_8;break;case SNDRV_PCM_FORMAT_S16_BE:sicr |= MPC52xx_PSC_SICR_SIM_CODEC_16;break;case SNDRV_PCM_FORMAT_S24_BE:sicr |= MPC52xx_PSC_SICR_SIM_CODEC_24;break;case SNDRV_PCM_FORMAT_S32_BE:sicr |= MPC52xx_PSC_SICR_SIM_CODEC_32;break;default:dev_dbg(psc_i2s->dev, "invalid format\n");return -EINVAL;- }
case should have the same indent level as switch.
- out_be32(&psc_i2s->psc_regs->sicr, sicr);
- //rc = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
- //if (rc) {
- // dev_err(psc_i2s->dev, "could not allocate dma buffer\n");
- // return rc;
- //}
- snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
- return 0;
+}
+static int psc_i2s_hw_free(struct snd_pcm_substream *substream) +{
- //return snd_pcm_lib_free_pages(substream);
- 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_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;
- long flags;
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
s = &psc_i2s->capture_stream;- else
s = &psc_i2s->playback_stream;- 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:
ditto re case indent.
s->period_bytes = frames_to_bytes(runtime,runtime->period_size);s->period_start = virt_to_phys(runtime->dma_area);s->period_end = s->period_start +(s->period_bytes * runtime->periods);s->period_next_pt = s->period_start;s->period_current_pt = s->period_start;s->active = 1;/* First; reset everything */if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {out_8(®s->command, MPC52xx_PSC_RST_RX);out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT);} else {out_8(®s->command, MPC52xx_PSC_RST_TX);out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT);}/* Next, fill up the bestcomm bd queue and enable DMA.* This will begin filling the PSC's fifo. */if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)bcom_gen_bd_rx_reset(s->bcom_task);elsebcom_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);/* Update interrupt enable settings. This must be done* before the PSC is enabled so that TX underrun events* are not missed. */imr = 0;if (psc_i2s->playback_stream.active)imr |= MPC52xx_PSC_IMR_TXEMP;if (psc_i2s->capture_stream.active)imr |= MPC52xx_PSC_IMR_ORERR;out_be16(®s->isr_imr.imr, imr);/* Due to errata in the i2s mode; need to line up enabling* the transmitter with a transition on the frame sync* line */spin_lock_irqsave(&psc_i2s->lock, flags);/* first make sure it is low */while ((in_8(®s->ipcr_acr.ipcr) & 0x80) != 0);/* then wait for the transition to high */while ((in_8(®s->ipcr_acr.ipcr) & 0x80) == 0);
We should be able to exit both while loops if the conditions are not met within a certain time limit.
/* Finally, enable the PSC.* Receiver must always be enabled; even when we only want* transmit. (see 15.3.2.3 of MPC5200B User's Guide) */psc_cmd = MPC52xx_PSC_RX_ENABLE;if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK)psc_cmd |= MPC52xx_PSC_TX_ENABLE;out_8(®s->command, psc_cmd);spin_unlock_irqrestore(&psc_i2s->lock, flags);break;case SNDRV_PCM_TRIGGER_STOP:/* Turn off the PSC */s->active = 0;if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {if (!psc_i2s->playback_stream.active) {out_8(®s->command, 2 << 4); /* reset rx */out_8(®s->command, 3 << 4); /* reset tx */out_8(®s->command, 4 << 4); /* reset err */}} else {out_8(®s->command, 3 << 4); /* reset tx */out_8(®s->command, 4 << 4); /* reset err */if (!psc_i2s->capture_stream.active)out_8(®s->command, 2 << 4); /* reset rx */}bcom_disable(s->bcom_task);while (!bcom_queue_empty(s->bcom_task))bcom_retrieve_buffer(s->bcom_task, NULL, NULL);break;default:dev_dbg(psc_i2s->dev, "invalid command\n");return -EINVAL;- }
- /* Update interrupt enable settings */
- imr = 0;
- if (psc_i2s->playback_stream.active) imr |= MPC52xx_PSC_IMR_TXEMP;
- if (psc_i2s->capture_stream.active) imr |= MPC52xx_PSC_IMR_ORERR;
- out_be16(®s->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_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_stream.active &&
!psc_i2s->capture_stream.active) {/* TODO: shut off channels */free_irq(psc_i2s->irq, psc_i2s);free_irq(bcom_get_task_irq(psc_i2s->capture_stream.bcom_task),&psc_i2s->capture_stream);free_irq(bcom_get_task_irq(psc_i2s->playback_stream.bcom_task),&psc_i2s->playback_stream);- }
+}
+/**
- psc_i2s_set_sysclk: set the clock frequency and direction
- This function is called by the machine driver to tell us what the clock
- frequency and direction are.
- Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN),
- and we don't care about the frequency. Return an error if the direction
- is not SND_SOC_CLOCK_IN.
- @clk_id: reserved, should be zero
- @freq: the frequency of the given clock ID, currently ignored
- @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master)
- */
+static int psc_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
int clk_id, unsigned int freq, int dir)+{
- struct psc_i2s *psc_i2s = cpu_dai->private_data;
- dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
cpu_dai, dir);- return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;
+}
+/**
- psc_i2s_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 I2S mode. Return an error if the format is
- not SND_SOC_DAIFMT_I2S.
- @format: one of SND_SOC_DAIFMT_xxx
- */
+static int psc_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format) +{
- struct psc_i2s *psc_i2s = cpu_dai->private_data;
- dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
cpu_dai, format);- return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
+}
+/* ---------------------------------------------------------------------
- ALSA SoC Bindings
- Digital Audio Interface (DAI) template
- create/destroy dai hooks
- */
+/**
- psc_i2s_dai_template: template CPU Digital Audio Interface
- */
+static struct snd_soc_cpu_dai psc_i2s_dai_template = {
- .type = SND_SOC_DAI_I2S,
- .playback = {
.channels_min = 2,.channels_max = 2,.rates = PSC_I2S_RATES,.formats = PSC_I2S_FORMATS,- },
- .capture = {
.channels_min = 2,.channels_max = 2,.rates = PSC_I2S_RATES,.formats = PSC_I2S_FORMATS,- },
- .ops = {
.startup = psc_i2s_startup,.hw_params = psc_i2s_hw_params,.hw_free = psc_i2s_hw_free,.shutdown = psc_i2s_shutdown,.trigger = psc_i2s_trigger,- },
- .dai_ops = {
.set_sysclk = psc_i2s_set_sysclk,.set_fmt = psc_i2s_set_fmt,- },
+};
+/* ---------------------------------------------------------------------
- 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,
- .period_bytes_max = 1024 * 1024,
- .periods_min = 2,
- .periods_max = 256,
- .buffer_bytes_max = 2 * 1024 * 1024,
- .fifo_size = 0,
+};
Fwiw, I usually separate out the DMA from I2S so it can be used by AC97, PCM interfaces etc. If your hardware only has I2S the it's not required.
+static unsigned int psc_i2s_fixed_rates[] = {
- 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
+};
+static struct snd_pcm_hw_constraint_list psc_i2s_constraints_rates = {
- .count = ARRAY_SIZE(psc_i2s_fixed_rates),
- .list = psc_i2s_fixed_rates,
- .mask = 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;
- int rc;
- 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_stream;- else
s = &psc_i2s->playback_stream;- snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware);
- rc = snd_pcm_hw_constraint_integer(substream->runtime,
SNDRV_PCM_HW_PARAM_PERIODS);- if (rc < 0) {
dev_err(psc_i2s->dev, "invalid buffer size\n");return rc;- }
- rc = snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,&psc_i2s_constraints_rates);- if (rc < 0) {
dev_err(psc_i2s->dev, "invalid rate\n");return rc;- }
- 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_stream;- else
s = &psc_i2s->playback_stream;- 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_stream;- else
s = &psc_i2s->playback_stream;- /*FIXME: count = s->sdma->bd[s->sdma->outdex].data - s->period_start;*/
- 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_codec_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;- rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
&pcm->streams[0].substream->dma_buffer);- if (rc) {
dev_err(card->dev, "Cannot alloc playback DMA buffer\n");return -ENOMEM;- }
- rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
&pcm->streams[1].substream->dma_buffer);- if (rc) {
snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);dev_err(card->dev, "Can't allocate capture DMA buffer\n");return -ENOMEM;- }
- return 0;
+}
+static void psc_i2s_pcm_free(struct snd_pcm *pcm) +{
- struct snd_soc_pcm_runtime *rtd = pcm->private_data;
- struct snd_pcm_substream *substream;
- int stream;
- dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm);
- for (stream = 0; stream < 2; stream++) {
substream = pcm->streams[stream].substream;if (substream) {snd_dma_free_pages(&substream->dma_buffer);substream->dma_buffer.area = NULL;substream->dma_buffer.addr = 0;}- }
+}
+struct snd_soc_platform psc_i2s_pcm_soc_platform = {
- .name = "mpc5200-psc-audio",
- .pcm_ops = &psc_i2s_pcm_ops,
- .pcm_new = &psc_i2s_pcm_new,
- .pcm_free = &psc_i2s_pcm_free,
+};
+/* ---------------------------------------------------------------------
- Sysfs attributes for debugging
- */
+static ssize_t psc_i2s_status_show(struct device *dev,
struct device_attribute *attr, char *buf)+{
- struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
- return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x tfnum=%i tfstat=0x%.4x\n",
in_be16(&psc_i2s->psc_regs->sr_csr.status),in_be32(&psc_i2s->psc_regs->sicr),in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff,in_be16(&psc_i2s->fifo_regs->rfstat),in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff,in_be16(&psc_i2s->fifo_regs->tfstat));+}
+static int * psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s,
const char *name)+{
- if (strcmp(name, "playback_underrun") == 0)
return &psc_i2s->stats.underrun_count;- if (strcmp(name, "capture_overrun") == 0)
return &psc_i2s->stats.overrun_count;- return NULL;
+}
+static ssize_t psc_i2s_stat_show(struct device *dev,
struct device_attribute *attr, char *buf)+{
- struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
- int *attrib;
- attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
- if (!attrib)
return 0;- return sprintf(buf, "%i\n", *attrib);
+}
+static ssize_t psc_i2s_stat_store(struct device *dev,
struct device_attribute *attr,const char *buf,size_t count)+{
- struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
- int *attrib;
- attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
- if (!attrib)
return 0;- *attrib = simple_strtoul(buf, NULL, 0);
- return count;
+}
+DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL); +DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show,psc_i2s_stat_store); +DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store);
+/* ---------------------------------------------------------------------
- OF platform bus binding code:
- Probe/remove operations
- OF device match table
- */
+static int __devinit psc_i2s_of_probe(struct of_device *op,
const struct of_device_id *match)+{
- phys_addr_t fifo;
- struct psc_i2s *psc_i2s;
- struct resource res;
- int size, psc_id, irq, rc;
- const __be32 *prop;
- void __iomem *regs;
- dev_dbg(&op->dev, "probing psc i2s device\n");
- /* Get the PSC ID */
- prop = of_get_property(op->node, "cell-index", &size);
- if (!prop || size < sizeof *prop)
return -ENODEV;- psc_id = be32_to_cpu(*prop);
- /* Fetch the registers and IRQ of the PSC */
- irq = irq_of_parse_and_map(op->node, 0);
- if (of_address_to_resource(op->node, 0, &res)) {
dev_err(&op->dev, "Missing reg property\n");return -ENODEV;- }
- regs = ioremap(res.start, 1 + res.end - res.start);
- if (!regs) {
dev_err(&op->dev, "Could not map registers\n");return -ENODEV;- }
- /* Allocate and initialize the driver private data */
- psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL);
- if (!psc_i2s) {
iounmap(regs);return -ENOMEM;- }
- spin_lock_init(&psc_i2s->lock);
- psc_i2s->irq = irq;
- psc_i2s->psc_regs = regs;
- psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs;
- psc_i2s->dev = &op->dev;
- psc_i2s->playback_stream.psc_i2s = psc_i2s;
- psc_i2s->capture_stream.psc_i2s = psc_i2s;
- snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1);
- /* Fill out the CPU DAI structure */
- memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai);
- psc_i2s->dai.private_data = psc_i2s;
- psc_i2s->dai.name = psc_i2s->name;
- psc_i2s->dai.id = psc_id;
- /* Find the address of the fifo data registers and setup the
* DMA tasks */- fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
- psc_i2s->capture_stream.bcom_task =
bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512);- psc_i2s->playback_stream.bcom_task =
bcom_psc_gen_bd_tx_init(psc_id, 10, fifo);- if (!psc_i2s->capture_stream.bcom_task ||
!psc_i2s->playback_stream.bcom_task) {dev_err(&op->dev, "Could not allocate bestcomm tasks\n");iounmap(regs);kfree(psc_i2s);return -ENODEV;- }
- /* Save what we've done so it can be found again later */
- dev_set_drvdata(&op->dev, psc_i2s);
- /* Register the SYSFS files */
- rc = device_create_file(psc_i2s->dev, &dev_attr_status);
- rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);
- rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);
- if (rc)
dev_info(psc_i2s->dev, "error creating sysfs files\n");- /* Tell the ASoC OF helpers about it */
- of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
&psc_i2s->dai);- return 0;
+}
+static int __devexit psc_i2s_of_remove(struct of_device *op) +{
- struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev);
- dev_dbg(&op->dev, "psc_i2s_remove()\n");
- bcom_gen_bd_rx_release(psc_i2s->capture_stream.bcom_task);
- bcom_gen_bd_tx_release(psc_i2s->playback_stream.bcom_task);
- iounmap(psc_i2s->psc_regs);
- iounmap(psc_i2s->fifo_regs);
- kfree(psc_i2s);
- dev_set_drvdata(&op->dev, NULL);
- return 0;
+}
+/* Match table for of_platform binding */ +static struct of_device_id psc_i2s_match[] __devinitdata = {
- { .compatible = "fsl,mpc5200-psc-i2s", },
- {}
+}; +MODULE_DEVICE_TABLE(of, psc_i2s_match);
+static struct of_platform_driver psc_i2s_driver = {
- .match_table = psc_i2s_match,
- .probe = psc_i2s_of_probe,
- .remove = __devexit_p(psc_i2s_of_remove),
- .driver = {
.name = "mpc5200-psc-i2s",.owner = THIS_MODULE,- },
+};
+/* ---------------------------------------------------------------------
- Module setup and teardown; simply register the of_platform driver
- for the PSC in I2S mode.
- */
+static int __init psc_i2s_init(void) +{
- return of_register_platform_driver(&psc_i2s_driver);
+} +module_init(psc_i2s_init);
+static void __exit psc_i2s_exit(void) +{
- of_unregister_platform_driver(&psc_i2s_driver);
+} +module_exit(psc_i2s_exit);
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DMA, needs to be split out. Efika is AC97 on the MPC5200 and needs to share DMA code. The new Phytec pcm030 baseboard is AC97 too.
What does the device tree look like?
On 7/1/08, Grant Likely grant.likely@secretlab.ca wrote:
From: Grant Likely grant.likely@secretlab.ca
This is an I2S bus driver for the MPC5200 PSC device. It is probably will not be merged as-is because it uses v1 of the ASoC API, but I want to get it out there for comments.
sound/soc/fsl/Kconfig | 6 sound/soc/fsl/Makefile | 2 sound/soc/fsl/mpc5200_psc_i2s.c | 899 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 907 insertions(+), 0 deletions(-)
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 257101f..5daa8d3 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -17,4 +17,10 @@ config SND_SOC_MPC8610_HPCD help Say Y if you want to enable audio on the Freescale MPC8610 HPCD.
+config SND_SOC_MPC5200_I2S
bool "Freescale MPC5200 PSC in I2S mode driver"depends on SND_SOC && PPC_MPC52xxhelpSay Y here to support the MPC5200 PSCs in I2S mode.endmenu diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 62f680a..98729a1 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -4,3 +4,5 @@ obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o # MPC8610 Platform Support obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o
+obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c new file mode 100644 index 0000000..81d0933 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -0,0 +1,899 @@ +/*
- Freescale MPC5200 PSC in I2S mode
- ALSA SoC Digital Audio Interface (DAI) 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.h>
+#include <sysdev/bestcomm/bestcomm.h> +#include <sysdev/bestcomm/gen_bd.h> +#include <asm/mpc52xx_psc.h>
+MODULE_AUTHOR("Grant Likely grant.likely@secretlab.ca"); +MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver"); +MODULE_LICENSE("GPL");
+/**
- PSC_I2S_RATES: sample rates supported by the I2S
- This driver currently only supports the PSC running in I2S slave mode,
- which means the codec determines the sample rate. Therefore, we tell
- ALSA that we support all rates and let the codec driver decide what rates
- are really supported.
- */
+#define PSC_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \
SNDRV_PCM_RATE_CONTINUOUS)+/**
- PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode
- */
+#define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \SNDRV_PCM_FMTBIT_S32_BE)+/**
- psc_i2s_stream - Data specific to a single stream (playback or capture)
- @active: flag indicating if the stream is active
- @psc_i2s: pointer back to parent psc_i2s data structure
- @bcom_task: bestcomm task structure
- @irq: irq number for bestcomm task
- @period_start: physical address of start of DMA region
- @period_end: physical address of end of DMA region
- @period_next_pt: physical address of next DMA buffer to enqueue
- @period_bytes: size of DMA period in bytes
- */
+struct psc_i2s_stream {
int active;struct psc_i2s *psc_i2s;struct bcom_task *bcom_task;int irq;struct snd_pcm_substream *stream;dma_addr_t period_start;dma_addr_t period_end;dma_addr_t period_next_pt;dma_addr_t period_current_pt;int period_bytes;+};
+/**
- psc_i2s - Private driver data
- @name: short name for this device ("PSC0", "PSC1", etc)
- @psc_regs: pointer to the PSC's registers
- @fifo_regs: pointer to the PSC's FIFO registers
- @irq: IRQ of this PSC
- @dev: struct device pointer
- @playback: the number of playback streams opened
- @capture: the number of capture streams opened
- @dai: the CPU DAI for this device
- @playback_stream: Playback stream context data
- @capture_stream: 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_cpu_dai dai;spinlock_t lock;/* per-stream data */struct psc_i2s_stream playback_stream;struct psc_i2s_stream capture_stream;/* 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 imr;u16 isr;isr = in_be16(®s->mpc52xx_psc_isr);imr = in_be16(®s->mpc52xx_psc_imr);/* Playback underrun error */if (isr & imr & MPC52xx_PSC_IMR_TXEMP)psc_i2s->stats.underrun_count++;/* Capture overrun error */if (isr & imr & MPC52xx_PSC_IMR_ORERR)psc_i2s->stats.overrun_count++;out_8(®s->command, 4 << 4); /* reset the error status */return IRQ_HANDLED;+}
+/**
- psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer
- @s: pointer to stream private data structure
- Enqueues another audio period buffer into the bestcomm queue.
- Note: The routine must only be called when there is space available in
- the queue. Otherwise the enqueue will fail and the audio ring buffer
- will get out of sync
- */
+static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s) +{
struct bcom_bd *bd;/* Prepare and enqueue the next buffer descriptor */bd = bcom_prepare_next_buffer(s->bcom_task);bd->status = s->period_bytes;bd->data[0] = s->period_next_pt;bcom_submit_next_buffer(s->bcom_task, NULL);/* Update for next period */s->period_next_pt += s->period_bytes;if (s->period_next_pt >= s->period_end)s->period_next_pt = s->period_start;+}
+/* Bestcomm DMA irq handler */ +static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream) +{
struct psc_i2s_stream *s = _psc_i2s_stream;//spin_lock(&s->psc_i2s->lock);/* For each finished period, dequeue the completed period buffer* and enqueue a new one in it's place. */while (bcom_buffer_done(s->bcom_task)) {bcom_retrieve_buffer(s->bcom_task, NULL, NULL);s->period_current_pt += s->period_bytes;if (s->period_current_pt >= s->period_end)s->period_current_pt = s->period_start;psc_i2s_bcom_enqueue_next_buffer(s);bcom_enable(s->bcom_task);}//spin_unlock(&s->psc_i2s->lock);/* If the stream is active, then also inform the PCM middle layer* of the period finished event. */if (s->active)snd_pcm_period_elapsed(s->stream);return IRQ_HANDLED;+}
+/**
- 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) +{
int playback_irq, capture_irq, rc;struct snd_soc_pcm_runtime *rtd = substream->private_data;struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;struct mpc52xx_psc_fifo __iomem *fiforegs = psc_i2s->fifo_regs;dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream);/* Disable all interrupts and reset the PSC */out_be16(®s->mpc52xx_psc_imr, 0);out_8(®s->command, 3 << 4); /* reset transmitter */out_8(®s->command, 2 << 4); /* reset receiver */out_8(®s->command, 1 << 4); /* reset mode */out_8(®s->command, 4 << 4); /* reset error *//* Default to CODEC8 mode */out_be32(®s->sicr,MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |MPC52xx_PSC_SICR_CLKPOL | MPC52xx_PSC_SICR_SIM_CODEC_8);/* First write: RxRdy (FIFO Alarm) generates receive FIFO interrupt *//* Second write to mode: register Normal mode for non loopback */out_8(®s->mode, 0);out_8(®s->mode, 0);/* Set the TX and RX fifo alarm thresholds */out_be16(&fiforegs->rfalarm, 0x100); /* set RFALARM level */out_8(&fiforegs->rfcntl, 0x4); /* set RFGRAN level (bytes) */out_be16(&fiforegs->tfalarm, 0x100); /* set TFALARM level */out_8(&fiforegs->tfcntl, 0x7); /* set TFGRAN level (bytes*4) *//* Setup the IRQs */playback_irq = bcom_get_task_irq(psc_i2s->playback_stream.bcom_task);capture_irq = bcom_get_task_irq(psc_i2s->capture_stream.bcom_task);rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED,"psc-i2s-status", psc_i2s);rc |= request_irq(capture_irq, &psc_i2s_bcom_irq, IRQF_SHARED,"psc-i2s-capture", &psc_i2s->capture_stream);rc |= request_irq(playback_irq, &psc_i2s_bcom_irq, IRQF_SHARED,"psc-i2s-playback", &psc_i2s->playback_stream);if (rc) {free_irq(psc_i2s->irq, psc_i2s);free_irq(capture_irq, &psc_i2s->capture_stream);free_irq(playback_irq, &psc_i2s->playback_stream);return -ENODEV;}return 0;+}
+static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)+{
struct snd_soc_pcm_runtime *rtd = substream->private_data;struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;u32 sicr;dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i"" periods=%i buffer_size=%i buffer_bytes=%i\n",__FUNCTION__, substream, params_period_size(params),params_period_bytes(params), params_periods(params),params_buffer_size(params), params_buffer_bytes(params));sicr = MPC52xx_PSC_SICR_DTS1 |MPC52xx_PSC_SICR_I2S | MPC52xx_PSC_SICR_CLKPOL;switch (params_format(params)) {case SNDRV_PCM_FORMAT_S8:sicr |= MPC52xx_PSC_SICR_SIM_CODEC_8;break;case SNDRV_PCM_FORMAT_S16_BE:sicr |= MPC52xx_PSC_SICR_SIM_CODEC_16;break;case SNDRV_PCM_FORMAT_S24_BE:sicr |= MPC52xx_PSC_SICR_SIM_CODEC_24;break;case SNDRV_PCM_FORMAT_S32_BE:sicr |= MPC52xx_PSC_SICR_SIM_CODEC_32;break;default:dev_dbg(psc_i2s->dev, "invalid format\n");return -EINVAL;}out_be32(&psc_i2s->psc_regs->sicr, sicr);//rc = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));//if (rc) {// dev_err(psc_i2s->dev, "could not allocate dma buffer\n");// return rc;//}snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);return 0;+}
+static int psc_i2s_hw_free(struct snd_pcm_substream *substream) +{
//return snd_pcm_lib_free_pages(substream);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_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;long flags;if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)s = &psc_i2s->capture_stream;elses = &psc_i2s->playback_stream;dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)"" stream_id=%i\n",substream, cmd, substream->pstr->stream);switch (cmd) {case SNDRV_PCM_TRIGGER_START:s->period_bytes = frames_to_bytes(runtime,runtime->period_size);s->period_start = virt_to_phys(runtime->dma_area);s->period_end = s->period_start +(s->period_bytes * runtime->periods);s->period_next_pt = s->period_start;s->period_current_pt = s->period_start;s->active = 1;/* First; reset everything */if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {out_8(®s->command, MPC52xx_PSC_RST_RX);out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT);} else {out_8(®s->command, MPC52xx_PSC_RST_TX);out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT);}/* Next, fill up the bestcomm bd queue and enable DMA.* This will begin filling the PSC's fifo. */if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)bcom_gen_bd_rx_reset(s->bcom_task);elsebcom_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);/* Update interrupt enable settings. This must be done* before the PSC is enabled so that TX underrun events* are not missed. */imr = 0;if (psc_i2s->playback_stream.active)imr |= MPC52xx_PSC_IMR_TXEMP;if (psc_i2s->capture_stream.active)imr |= MPC52xx_PSC_IMR_ORERR;out_be16(®s->isr_imr.imr, imr);/* Due to errata in the i2s mode; need to line up enabling* the transmitter with a transition on the frame sync* line */spin_lock_irqsave(&psc_i2s->lock, flags);/* first make sure it is low */while ((in_8(®s->ipcr_acr.ipcr) & 0x80) != 0);/* then wait for the transition to high */while ((in_8(®s->ipcr_acr.ipcr) & 0x80) == 0);/* Finally, enable the PSC.* Receiver must always be enabled; even when we only want* transmit. (see 15.3.2.3 of MPC5200B User's Guide) */psc_cmd = MPC52xx_PSC_RX_ENABLE;if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK)psc_cmd |= MPC52xx_PSC_TX_ENABLE;out_8(®s->command, psc_cmd);spin_unlock_irqrestore(&psc_i2s->lock, flags);break;case SNDRV_PCM_TRIGGER_STOP:/* Turn off the PSC */s->active = 0;if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {if (!psc_i2s->playback_stream.active) {out_8(®s->command, 2 << 4); /* reset rx */out_8(®s->command, 3 << 4); /* reset tx */out_8(®s->command, 4 << 4); /* reset err */}} else {out_8(®s->command, 3 << 4); /* reset tx */out_8(®s->command, 4 << 4); /* reset err */if (!psc_i2s->capture_stream.active)out_8(®s->command, 2 << 4); /* reset rx */}bcom_disable(s->bcom_task);while (!bcom_queue_empty(s->bcom_task))bcom_retrieve_buffer(s->bcom_task, NULL, NULL);break;default:dev_dbg(psc_i2s->dev, "invalid command\n");return -EINVAL;}/* Update interrupt enable settings */imr = 0;if (psc_i2s->playback_stream.active) imr |= MPC52xx_PSC_IMR_TXEMP;if (psc_i2s->capture_stream.active) imr |= MPC52xx_PSC_IMR_ORERR;out_be16(®s->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_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_stream.active &&!psc_i2s->capture_stream.active) {/* TODO: shut off channels */free_irq(psc_i2s->irq, psc_i2s);free_irq(bcom_get_task_irq(psc_i2s->capture_stream.bcom_task),&psc_i2s->capture_stream);free_irq(bcom_get_task_irq(psc_i2s->playback_stream.bcom_task),&psc_i2s->playback_stream);}+}
+/**
- psc_i2s_set_sysclk: set the clock frequency and direction
- This function is called by the machine driver to tell us what the clock
- frequency and direction are.
- Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN),
- and we don't care about the frequency. Return an error if the direction
- is not SND_SOC_CLOCK_IN.
- @clk_id: reserved, should be zero
- @freq: the frequency of the given clock ID, currently ignored
- @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master)
- */
+static int psc_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
int clk_id, unsigned int freq, int dir)+{
struct psc_i2s *psc_i2s = cpu_dai->private_data;dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",cpu_dai, dir);return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;+}
+/**
- psc_i2s_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 I2S mode. Return an error if the format is
- not SND_SOC_DAIFMT_I2S.
- @format: one of SND_SOC_DAIFMT_xxx
- */
+static int psc_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format) +{
struct psc_i2s *psc_i2s = cpu_dai->private_data;dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",cpu_dai, format);return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;+}
+/* ---------------------------------------------------------------------
- ALSA SoC Bindings
- Digital Audio Interface (DAI) template
- create/destroy dai hooks
- */
+/**
- psc_i2s_dai_template: template CPU Digital Audio Interface
- */
+static struct snd_soc_cpu_dai psc_i2s_dai_template = {
.type = SND_SOC_DAI_I2S,.playback = {.channels_min = 2,.channels_max = 2,.rates = PSC_I2S_RATES,.formats = PSC_I2S_FORMATS,},.capture = {.channels_min = 2,.channels_max = 2,.rates = PSC_I2S_RATES,.formats = PSC_I2S_FORMATS,},.ops = {.startup = psc_i2s_startup,.hw_params = psc_i2s_hw_params,.hw_free = psc_i2s_hw_free,.shutdown = psc_i2s_shutdown,.trigger = psc_i2s_trigger,},.dai_ops = {.set_sysclk = psc_i2s_set_sysclk,.set_fmt = psc_i2s_set_fmt,},+};
+/* ---------------------------------------------------------------------
- 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,.period_bytes_max = 1024 * 1024,.periods_min = 2,.periods_max = 256,.buffer_bytes_max = 2 * 1024 * 1024,.fifo_size = 0,+};
+static unsigned int psc_i2s_fixed_rates[] = {
8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,+};
+static struct snd_pcm_hw_constraint_list psc_i2s_constraints_rates = {
.count = ARRAY_SIZE(psc_i2s_fixed_rates),.list = psc_i2s_fixed_rates,.mask = 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;int rc;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_stream;elses = &psc_i2s->playback_stream;snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware);rc = snd_pcm_hw_constraint_integer(substream->runtime,SNDRV_PCM_HW_PARAM_PERIODS);if (rc < 0) {dev_err(psc_i2s->dev, "invalid buffer size\n");return rc;}rc = snd_pcm_hw_constraint_list(substream->runtime, 0,SNDRV_PCM_HW_PARAM_RATE,&psc_i2s_constraints_rates);if (rc < 0) {dev_err(psc_i2s->dev, "invalid rate\n");return rc;}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_stream;elses = &psc_i2s->playback_stream;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_stream;elses = &psc_i2s->playback_stream;/*FIXME: count = s->sdma->bd[s->sdma->outdex].data - s->period_start;*/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_codec_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;rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,&pcm->streams[0].substream->dma_buffer);if (rc) {dev_err(card->dev, "Cannot alloc playback DMA buffer\n");return -ENOMEM;}rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,&pcm->streams[1].substream->dma_buffer);if (rc) {snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);dev_err(card->dev, "Can't allocate capture DMA buffer\n");return -ENOMEM;}return 0;+}
+static void psc_i2s_pcm_free(struct snd_pcm *pcm) +{
struct snd_soc_pcm_runtime *rtd = pcm->private_data;struct snd_pcm_substream *substream;int stream;dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm);for (stream = 0; stream < 2; stream++) {substream = pcm->streams[stream].substream;if (substream) {snd_dma_free_pages(&substream->dma_buffer);substream->dma_buffer.area = NULL;substream->dma_buffer.addr = 0;}}+}
+struct snd_soc_platform psc_i2s_pcm_soc_platform = {
.name = "mpc5200-psc-audio",.pcm_ops = &psc_i2s_pcm_ops,.pcm_new = &psc_i2s_pcm_new,.pcm_free = &psc_i2s_pcm_free,+};
+/* ---------------------------------------------------------------------
- Sysfs attributes for debugging
- */
+static ssize_t psc_i2s_status_show(struct device *dev,
struct device_attribute *attr, char *buf)+{
struct psc_i2s *psc_i2s = dev_get_drvdata(dev);return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x tfnum=%i tfstat=0x%.4x\n",in_be16(&psc_i2s->psc_regs->sr_csr.status),in_be32(&psc_i2s->psc_regs->sicr),in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff,in_be16(&psc_i2s->fifo_regs->rfstat),in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff,in_be16(&psc_i2s->fifo_regs->tfstat));+}
+static int * psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s,
const char *name)+{
if (strcmp(name, "playback_underrun") == 0)return &psc_i2s->stats.underrun_count;if (strcmp(name, "capture_overrun") == 0)return &psc_i2s->stats.overrun_count;return NULL;+}
+static ssize_t psc_i2s_stat_show(struct device *dev,
struct device_attribute *attr, char *buf)+{
struct psc_i2s *psc_i2s = dev_get_drvdata(dev);int *attrib;attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);if (!attrib)return 0;return sprintf(buf, "%i\n", *attrib);+}
+static ssize_t psc_i2s_stat_store(struct device *dev,
struct device_attribute *attr,const char *buf,size_t count)+{
struct psc_i2s *psc_i2s = dev_get_drvdata(dev);int *attrib;attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);if (!attrib)return 0;*attrib = simple_strtoul(buf, NULL, 0);return count;+}
+DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL); +DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show,psc_i2s_stat_store); +DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store);
+/* ---------------------------------------------------------------------
- OF platform bus binding code:
- Probe/remove operations
- OF device match table
- */
+static int __devinit psc_i2s_of_probe(struct of_device *op,
const struct of_device_id *match)+{
phys_addr_t fifo;struct psc_i2s *psc_i2s;struct resource res;int size, psc_id, irq, rc;const __be32 *prop;void __iomem *regs;dev_dbg(&op->dev, "probing psc i2s device\n");/* Get the PSC ID */prop = of_get_property(op->node, "cell-index", &size);if (!prop || size < sizeof *prop)return -ENODEV;psc_id = be32_to_cpu(*prop);/* Fetch the registers and IRQ of the PSC */irq = irq_of_parse_and_map(op->node, 0);if (of_address_to_resource(op->node, 0, &res)) {dev_err(&op->dev, "Missing reg property\n");return -ENODEV;}regs = ioremap(res.start, 1 + res.end - res.start);if (!regs) {dev_err(&op->dev, "Could not map registers\n");return -ENODEV;}/* Allocate and initialize the driver private data */psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL);if (!psc_i2s) {iounmap(regs);return -ENOMEM;}spin_lock_init(&psc_i2s->lock);psc_i2s->irq = irq;psc_i2s->psc_regs = regs;psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs;psc_i2s->dev = &op->dev;psc_i2s->playback_stream.psc_i2s = psc_i2s;psc_i2s->capture_stream.psc_i2s = psc_i2s;snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1);/* Fill out the CPU DAI structure */memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai);psc_i2s->dai.private_data = psc_i2s;psc_i2s->dai.name = psc_i2s->name;psc_i2s->dai.id = psc_id;/* Find the address of the fifo data registers and setup the* DMA tasks */fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);psc_i2s->capture_stream.bcom_task =bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512);psc_i2s->playback_stream.bcom_task =bcom_psc_gen_bd_tx_init(psc_id, 10, fifo);if (!psc_i2s->capture_stream.bcom_task ||!psc_i2s->playback_stream.bcom_task) {dev_err(&op->dev, "Could not allocate bestcomm tasks\n");iounmap(regs);kfree(psc_i2s);return -ENODEV;}/* Save what we've done so it can be found again later */dev_set_drvdata(&op->dev, psc_i2s);/* Register the SYSFS files */rc = device_create_file(psc_i2s->dev, &dev_attr_status);rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);if (rc)dev_info(psc_i2s->dev, "error creating sysfs files\n");/* Tell the ASoC OF helpers about it */of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,&psc_i2s->dai);return 0;+}
+static int __devexit psc_i2s_of_remove(struct of_device *op) +{
struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev);dev_dbg(&op->dev, "psc_i2s_remove()\n");bcom_gen_bd_rx_release(psc_i2s->capture_stream.bcom_task);bcom_gen_bd_tx_release(psc_i2s->playback_stream.bcom_task);iounmap(psc_i2s->psc_regs);iounmap(psc_i2s->fifo_regs);kfree(psc_i2s);dev_set_drvdata(&op->dev, NULL);return 0;+}
+/* Match table for of_platform binding */ +static struct of_device_id psc_i2s_match[] __devinitdata = {
{ .compatible = "fsl,mpc5200-psc-i2s", },{}+}; +MODULE_DEVICE_TABLE(of, psc_i2s_match);
+static struct of_platform_driver psc_i2s_driver = {
.match_table = psc_i2s_match,.probe = psc_i2s_of_probe,.remove = __devexit_p(psc_i2s_of_remove),.driver = {.name = "mpc5200-psc-i2s",.owner = THIS_MODULE,},+};
+/* ---------------------------------------------------------------------
- Module setup and teardown; simply register the of_platform driver
- for the PSC in I2S mode.
- */
+static int __init psc_i2s_init(void) +{
return of_register_platform_driver(&psc_i2s_driver);+} +module_init(psc_i2s_init);
+static void __exit psc_i2s_exit(void) +{
of_unregister_platform_driver(&psc_i2s_driver);+} +module_exit(psc_i2s_exit);
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
On Wed, Jul 02, 2008 at 09:51:59AM -0400, Jon Smirl wrote:
DMA, needs to be split out. Efika is AC97 on the MPC5200 and needs to share DMA code. The new Phytec pcm030 baseboard is AC97 too.
I agree, but I'm not sure the best way to organize a split. I'm not doing anything with the Efika at the moment and I haven't dug into the details of what needs to be done for AC97.
What does the device tree look like?
Here is the relevant excerpt, but it is not documented yet. I haven't made any attempt encode the layout into the device tree (clocking, etc.)
spi@f00 { #address-cells = <1>; #size-cells = <0>; compatible = "fsl,mpc5200b-spi","fsl,mpc5200-spi"; reg = <0xf00 0x20>; num-slaves = <32>; interrupts = <2 13 0 2 14 0>; interrupt-parent = <&mpc5200_pic>;
codec1: codec@2 { compatible = "ti,tlv320aic26"; linux,modalias = "tlv320aic26"; max-speed = <1000000>; reg = <2>; };
codec2: codec@3 { compatible = "ti,tlv320aic26"; linux,modalias = "tlv320aic26"; max-speed = <1000000>; reg = <3>; }; };
i2s@2200 { // PSC2 compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s"; cell-index = <1>; reg = <0x2200 0x100>; interrupts = <2 2 0>; interrupt-parent = <&mpc5200_pic>; codec-handle = <&codec1>; }; i2s@2400 { // PSC3 compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s"; cell-index = <2>; reg = <0x2400 0x100>; interrupts = <2 3 0>; interrupt-parent = <&mpc5200_pic>; codec-handle = <&codec2>; };
On Fri, Jul 04, 2008 at 07:03:25AM -0400, Timur Tabi wrote:
i2s@2200 { // PSC2 compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s"; cell-index = <1>;
cell-index should be zero-based, not one-based.
Umm... this is for PSC #2... cell index is set to '1'. It is zero based.
g.
On Fri, Jul 04, 2008 at 07:03:25AM -0400, Timur Tabi wrote:
i2s@2200 { // PSC2 compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s"; cell-index = <1>;
cell-index should be zero-based, not one-based.
Well...since cell-index is for indexing shared resources, the cell-index values should be whatever the convention is for that shared resource, which can be defined as whatever is convenient for that resource. I think that's been zero-based in every user of cell-index so far, but there's no reason it *has* to be if a different enumeration is convenient for the shared resource in question.
On 7/1/08, Grant Likely grant.likely@secretlab.ca wrote:
From: Grant Likely grant.likely@secretlab.ca
This is an I2S bus driver for the MPC5200 PSC device. It is probably will not be merged as-is because it uses v1 of the ASoC API, but I want to get it out there for comments.
sound/soc/fsl/Kconfig | 6 sound/soc/fsl/Makefile | 2 sound/soc/fsl/mpc5200_psc_i2s.c | 899 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 907 insertions(+), 0 deletions(-)
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 257101f..5daa8d3 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -17,4 +17,10 @@ config SND_SOC_MPC8610_HPCD help Say Y if you want to enable audio on the Freescale MPC8610 HPCD.
+config SND_SOC_MPC5200_I2S
bool "Freescale MPC5200 PSC in I2S mode driver"depends on SND_SOC && PPC_MPC52xxhelpSay Y here to support the MPC5200 PSCs in I2S mode.endmenu diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 62f680a..98729a1 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -4,3 +4,5 @@ obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o # MPC8610 Platform Support obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o
+obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c new file mode 100644 index 0000000..81d0933 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -0,0 +1,899 @@ +/*
- Freescale MPC5200 PSC in I2S mode
- ALSA SoC Digital Audio Interface (DAI) 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.h>
+#include <sysdev/bestcomm/bestcomm.h> +#include <sysdev/bestcomm/gen_bd.h> +#include <asm/mpc52xx_psc.h>
+MODULE_AUTHOR("Grant Likely grant.likely@secretlab.ca"); +MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver"); +MODULE_LICENSE("GPL");
+/**
- PSC_I2S_RATES: sample rates supported by the I2S
- This driver currently only supports the PSC running in I2S slave mode,
- which means the codec determines the sample rate. Therefore, we tell
- ALSA that we support all rates and let the codec driver decide what rates
- are really supported.
- */
+#define PSC_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \
SNDRV_PCM_RATE_CONTINUOUS)+/**
- PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode
- */
+#define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \SNDRV_PCM_FMTBIT_S32_BE)+/**
- psc_i2s_stream - Data specific to a single stream (playback or capture)
- @active: flag indicating if the stream is active
- @psc_i2s: pointer back to parent psc_i2s data structure
- @bcom_task: bestcomm task structure
- @irq: irq number for bestcomm task
- @period_start: physical address of start of DMA region
- @period_end: physical address of end of DMA region
- @period_next_pt: physical address of next DMA buffer to enqueue
- @period_bytes: size of DMA period in bytes
- */
+struct psc_i2s_stream {
int active;struct psc_i2s *psc_i2s;struct bcom_task *bcom_task;int irq;struct snd_pcm_substream *stream;dma_addr_t period_start;dma_addr_t period_end;dma_addr_t period_next_pt;dma_addr_t period_current_pt;int period_bytes;+};
+/**
- psc_i2s - Private driver data
- @name: short name for this device ("PSC0", "PSC1", etc)
- @psc_regs: pointer to the PSC's registers
- @fifo_regs: pointer to the PSC's FIFO registers
- @irq: IRQ of this PSC
- @dev: struct device pointer
- @playback: the number of playback streams opened
- @capture: the number of capture streams opened
- @dai: the CPU DAI for this device
- @playback_stream: Playback stream context data
- @capture_stream: 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_cpu_dai dai;spinlock_t lock;/* per-stream data */struct psc_i2s_stream playback_stream;struct psc_i2s_stream capture_stream;/* 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 imr;u16 isr;isr = in_be16(®s->mpc52xx_psc_isr);imr = in_be16(®s->mpc52xx_psc_imr);/* Playback underrun error */if (isr & imr & MPC52xx_PSC_IMR_TXEMP)psc_i2s->stats.underrun_count++;/* Capture overrun error */if (isr & imr & MPC52xx_PSC_IMR_ORERR)psc_i2s->stats.overrun_count++;out_8(®s->command, 4 << 4); /* reset the error status */return IRQ_HANDLED;+}
+/**
- psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer
- @s: pointer to stream private data structure
- Enqueues another audio period buffer into the bestcomm queue.
- Note: The routine must only be called when there is space available in
- the queue. Otherwise the enqueue will fail and the audio ring buffer
- will get out of sync
- */
+static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s) +{
struct bcom_bd *bd;/* Prepare and enqueue the next buffer descriptor */bd = bcom_prepare_next_buffer(s->bcom_task);bd->status = s->period_bytes;bd->data[0] = s->period_next_pt;bcom_submit_next_buffer(s->bcom_task, NULL);/* Update for next period */s->period_next_pt += s->period_bytes;if (s->period_next_pt >= s->period_end)s->period_next_pt = s->period_start;+}
+/* Bestcomm DMA irq handler */ +static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream) +{
struct psc_i2s_stream *s = _psc_i2s_stream;//spin_lock(&s->psc_i2s->lock);/* For each finished period, dequeue the completed period buffer* and enqueue a new one in it's place. */while (bcom_buffer_done(s->bcom_task)) {bcom_retrieve_buffer(s->bcom_task, NULL, NULL);s->period_current_pt += s->period_bytes;if (s->period_current_pt >= s->period_end)s->period_current_pt = s->period_start;psc_i2s_bcom_enqueue_next_buffer(s);bcom_enable(s->bcom_task);}//spin_unlock(&s->psc_i2s->lock);/* If the stream is active, then also inform the PCM middle layer* of the period finished event. */if (s->active)snd_pcm_period_elapsed(s->stream);return IRQ_HANDLED;+}
+/**
- 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) +{
int playback_irq, capture_irq, rc;struct snd_soc_pcm_runtime *rtd = substream->private_data;struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;struct mpc52xx_psc_fifo __iomem *fiforegs = psc_i2s->fifo_regs;dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream);/* Disable all interrupts and reset the PSC */out_be16(®s->mpc52xx_psc_imr, 0);out_8(®s->command, 3 << 4); /* reset transmitter */out_8(®s->command, 2 << 4); /* reset receiver */out_8(®s->command, 1 << 4); /* reset mode */out_8(®s->command, 4 << 4); /* reset error *//* Default to CODEC8 mode */out_be32(®s->sicr,MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |MPC52xx_PSC_SICR_CLKPOL | MPC52xx_PSC_SICR_SIM_CODEC_8);/* First write: RxRdy (FIFO Alarm) generates receive FIFO interrupt *//* Second write to mode: register Normal mode for non loopback */out_8(®s->mode, 0);out_8(®s->mode, 0);/* Set the TX and RX fifo alarm thresholds */out_be16(&fiforegs->rfalarm, 0x100); /* set RFALARM level */out_8(&fiforegs->rfcntl, 0x4); /* set RFGRAN level (bytes) */out_be16(&fiforegs->tfalarm, 0x100); /* set TFALARM level */out_8(&fiforegs->tfcntl, 0x7); /* set TFGRAN level (bytes*4) *//* Setup the IRQs */playback_irq = bcom_get_task_irq(psc_i2s->playback_stream.bcom_task);capture_irq = bcom_get_task_irq(psc_i2s->capture_stream.bcom_task);rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED,"psc-i2s-status", psc_i2s);rc |= request_irq(capture_irq, &psc_i2s_bcom_irq, IRQF_SHARED,"psc-i2s-capture", &psc_i2s->capture_stream);rc |= request_irq(playback_irq, &psc_i2s_bcom_irq, IRQF_SHARED,"psc-i2s-playback", &psc_i2s->playback_stream);if (rc) {free_irq(psc_i2s->irq, psc_i2s);free_irq(capture_irq, &psc_i2s->capture_stream);free_irq(playback_irq, &psc_i2s->playback_stream);return -ENODEV;}return 0;+}
+static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)+{
struct snd_soc_pcm_runtime *rtd = substream->private_data;struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;u32 sicr;dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i"" periods=%i buffer_size=%i buffer_bytes=%i\n",__FUNCTION__, substream, params_period_size(params),params_period_bytes(params), params_periods(params),params_buffer_size(params), params_buffer_bytes(params));sicr = MPC52xx_PSC_SICR_DTS1 |MPC52xx_PSC_SICR_I2S | MPC52xx_PSC_SICR_CLKPOL;switch (params_format(params)) {case SNDRV_PCM_FORMAT_S8:sicr |= MPC52xx_PSC_SICR_SIM_CODEC_8;break;case SNDRV_PCM_FORMAT_S16_BE:sicr |= MPC52xx_PSC_SICR_SIM_CODEC_16;break;case SNDRV_PCM_FORMAT_S24_BE:sicr |= MPC52xx_PSC_SICR_SIM_CODEC_24;break;case SNDRV_PCM_FORMAT_S32_BE:sicr |= MPC52xx_PSC_SICR_SIM_CODEC_32;break;default:dev_dbg(psc_i2s->dev, "invalid format\n");return -EINVAL;}out_be32(&psc_i2s->psc_regs->sicr, sicr);//rc = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));//if (rc) {// dev_err(psc_i2s->dev, "could not allocate dma buffer\n");// return rc;//}snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);return 0;+}
+static int psc_i2s_hw_free(struct snd_pcm_substream *substream) +{
//return snd_pcm_lib_free_pages(substream);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_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;long flags;if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)s = &psc_i2s->capture_stream;elses = &psc_i2s->playback_stream;dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)"" stream_id=%i\n",substream, cmd, substream->pstr->stream);switch (cmd) {case SNDRV_PCM_TRIGGER_START:s->period_bytes = frames_to_bytes(runtime,runtime->period_size);s->period_start = virt_to_phys(runtime->dma_area);s->period_end = s->period_start +(s->period_bytes * runtime->periods);s->period_next_pt = s->period_start;s->period_current_pt = s->period_start;s->active = 1;/* First; reset everything */if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {out_8(®s->command, MPC52xx_PSC_RST_RX);out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT);} else {out_8(®s->command, MPC52xx_PSC_RST_TX);out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT);}/* Next, fill up the bestcomm bd queue and enable DMA.* This will begin filling the PSC's fifo. */if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)bcom_gen_bd_rx_reset(s->bcom_task);elsebcom_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);/* Update interrupt enable settings. This must be done* before the PSC is enabled so that TX underrun events* are not missed. */imr = 0;if (psc_i2s->playback_stream.active)imr |= MPC52xx_PSC_IMR_TXEMP;if (psc_i2s->capture_stream.active)imr |= MPC52xx_PSC_IMR_ORERR;out_be16(®s->isr_imr.imr, imr);/* Due to errata in the i2s mode; need to line up enabling* the transmitter with a transition on the frame sync* line */spin_lock_irqsave(&psc_i2s->lock, flags);/* first make sure it is low */while ((in_8(®s->ipcr_acr.ipcr) & 0x80) != 0);
Could this be moved to the front of the routine, to increase parallelism?
Once you detect the 0, it will be a fixed interval before the 1 happens. Might as well overlap the computations.
/* then wait for the transition to high */while ((in_8(®s->ipcr_acr.ipcr) & 0x80) == 0);/* Finally, enable the PSC.* Receiver must always be enabled; even when we only want* transmit. (see 15.3.2.3 of MPC5200B User's Guide) */psc_cmd = MPC52xx_PSC_RX_ENABLE;if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK)psc_cmd |= MPC52xx_PSC_TX_ENABLE;out_8(®s->command, psc_cmd);spin_unlock_irqrestore(&psc_i2s->lock, flags);break;case SNDRV_PCM_TRIGGER_STOP:/* Turn off the PSC */s->active = 0;if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {if (!psc_i2s->playback_stream.active) {out_8(®s->command, 2 << 4); /* reset rx */out_8(®s->command, 3 << 4); /* reset tx */out_8(®s->command, 4 << 4); /* reset err */}} else {out_8(®s->command, 3 << 4); /* reset tx */out_8(®s->command, 4 << 4); /* reset err */if (!psc_i2s->capture_stream.active)out_8(®s->command, 2 << 4); /* reset rx */}bcom_disable(s->bcom_task);while (!bcom_queue_empty(s->bcom_task))bcom_retrieve_buffer(s->bcom_task, NULL, NULL);break;default:dev_dbg(psc_i2s->dev, "invalid command\n");return -EINVAL;}/* Update interrupt enable settings */imr = 0;if (psc_i2s->playback_stream.active) imr |= MPC52xx_PSC_IMR_TXEMP;if (psc_i2s->capture_stream.active) imr |= MPC52xx_PSC_IMR_ORERR;out_be16(®s->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_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_stream.active &&!psc_i2s->capture_stream.active) {/* TODO: shut off channels */free_irq(psc_i2s->irq, psc_i2s);free_irq(bcom_get_task_irq(psc_i2s->capture_stream.bcom_task),&psc_i2s->capture_stream);free_irq(bcom_get_task_irq(psc_i2s->playback_stream.bcom_task),&psc_i2s->playback_stream);}+}
+/**
- psc_i2s_set_sysclk: set the clock frequency and direction
- This function is called by the machine driver to tell us what the clock
- frequency and direction are.
- Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN),
- and we don't care about the frequency. Return an error if the direction
- is not SND_SOC_CLOCK_IN.
- @clk_id: reserved, should be zero
- @freq: the frequency of the given clock ID, currently ignored
- @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master)
- */
+static int psc_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
int clk_id, unsigned int freq, int dir)+{
struct psc_i2s *psc_i2s = cpu_dai->private_data;dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",cpu_dai, dir);return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;+}
+/**
- psc_i2s_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 I2S mode. Return an error if the format is
- not SND_SOC_DAIFMT_I2S.
- @format: one of SND_SOC_DAIFMT_xxx
- */
+static int psc_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format) +{
struct psc_i2s *psc_i2s = cpu_dai->private_data;dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",cpu_dai, format);return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;+}
+/* ---------------------------------------------------------------------
- ALSA SoC Bindings
- Digital Audio Interface (DAI) template
- create/destroy dai hooks
- */
+/**
- psc_i2s_dai_template: template CPU Digital Audio Interface
- */
+static struct snd_soc_cpu_dai psc_i2s_dai_template = {
.type = SND_SOC_DAI_I2S,.playback = {.channels_min = 2,.channels_max = 2,.rates = PSC_I2S_RATES,.formats = PSC_I2S_FORMATS,},.capture = {.channels_min = 2,.channels_max = 2,.rates = PSC_I2S_RATES,.formats = PSC_I2S_FORMATS,},.ops = {.startup = psc_i2s_startup,.hw_params = psc_i2s_hw_params,.hw_free = psc_i2s_hw_free,.shutdown = psc_i2s_shutdown,.trigger = psc_i2s_trigger,},.dai_ops = {.set_sysclk = psc_i2s_set_sysclk,.set_fmt = psc_i2s_set_fmt,},+};
+/* ---------------------------------------------------------------------
- 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,.period_bytes_max = 1024 * 1024,.periods_min = 2,.periods_max = 256,.buffer_bytes_max = 2 * 1024 * 1024,.fifo_size = 0,+};
+static unsigned int psc_i2s_fixed_rates[] = {
8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,+};
+static struct snd_pcm_hw_constraint_list psc_i2s_constraints_rates = {
.count = ARRAY_SIZE(psc_i2s_fixed_rates),.list = psc_i2s_fixed_rates,.mask = 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;int rc;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_stream;elses = &psc_i2s->playback_stream;snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware);rc = snd_pcm_hw_constraint_integer(substream->runtime,SNDRV_PCM_HW_PARAM_PERIODS);if (rc < 0) {dev_err(psc_i2s->dev, "invalid buffer size\n");return rc;}rc = snd_pcm_hw_constraint_list(substream->runtime, 0,SNDRV_PCM_HW_PARAM_RATE,&psc_i2s_constraints_rates);if (rc < 0) {dev_err(psc_i2s->dev, "invalid rate\n");return rc;}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_stream;elses = &psc_i2s->playback_stream;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_stream;elses = &psc_i2s->playback_stream;/*FIXME: count = s->sdma->bd[s->sdma->outdex].data - s->period_start;*/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_codec_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;rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,&pcm->streams[0].substream->dma_buffer);if (rc) {dev_err(card->dev, "Cannot alloc playback DMA buffer\n");return -ENOMEM;}rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,&pcm->streams[1].substream->dma_buffer);if (rc) {snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);dev_err(card->dev, "Can't allocate capture DMA buffer\n");return -ENOMEM;}return 0;+}
+static void psc_i2s_pcm_free(struct snd_pcm *pcm) +{
struct snd_soc_pcm_runtime *rtd = pcm->private_data;struct snd_pcm_substream *substream;int stream;dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm);for (stream = 0; stream < 2; stream++) {substream = pcm->streams[stream].substream;if (substream) {snd_dma_free_pages(&substream->dma_buffer);substream->dma_buffer.area = NULL;substream->dma_buffer.addr = 0;}}+}
+struct snd_soc_platform psc_i2s_pcm_soc_platform = {
.name = "mpc5200-psc-audio",.pcm_ops = &psc_i2s_pcm_ops,.pcm_new = &psc_i2s_pcm_new,.pcm_free = &psc_i2s_pcm_free,+};
+/* ---------------------------------------------------------------------
- Sysfs attributes for debugging
- */
+static ssize_t psc_i2s_status_show(struct device *dev,
struct device_attribute *attr, char *buf)+{
struct psc_i2s *psc_i2s = dev_get_drvdata(dev);return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x tfnum=%i tfstat=0x%.4x\n",in_be16(&psc_i2s->psc_regs->sr_csr.status),in_be32(&psc_i2s->psc_regs->sicr),in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff,in_be16(&psc_i2s->fifo_regs->rfstat),in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff,in_be16(&psc_i2s->fifo_regs->tfstat));+}
+static int * psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s,
const char *name)+{
if (strcmp(name, "playback_underrun") == 0)return &psc_i2s->stats.underrun_count;if (strcmp(name, "capture_overrun") == 0)return &psc_i2s->stats.overrun_count;return NULL;+}
+static ssize_t psc_i2s_stat_show(struct device *dev,
struct device_attribute *attr, char *buf)+{
struct psc_i2s *psc_i2s = dev_get_drvdata(dev);int *attrib;attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);if (!attrib)return 0;return sprintf(buf, "%i\n", *attrib);+}
+static ssize_t psc_i2s_stat_store(struct device *dev,
struct device_attribute *attr,const char *buf,size_t count)+{
struct psc_i2s *psc_i2s = dev_get_drvdata(dev);int *attrib;attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);if (!attrib)return 0;*attrib = simple_strtoul(buf, NULL, 0);return count;+}
+DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL); +DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show,psc_i2s_stat_store); +DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store);
+/* ---------------------------------------------------------------------
- OF platform bus binding code:
- Probe/remove operations
- OF device match table
- */
+static int __devinit psc_i2s_of_probe(struct of_device *op,
const struct of_device_id *match)+{
phys_addr_t fifo;struct psc_i2s *psc_i2s;struct resource res;int size, psc_id, irq, rc;const __be32 *prop;void __iomem *regs;dev_dbg(&op->dev, "probing psc i2s device\n");/* Get the PSC ID */prop = of_get_property(op->node, "cell-index", &size);if (!prop || size < sizeof *prop)return -ENODEV;psc_id = be32_to_cpu(*prop);/* Fetch the registers and IRQ of the PSC */irq = irq_of_parse_and_map(op->node, 0);if (of_address_to_resource(op->node, 0, &res)) {dev_err(&op->dev, "Missing reg property\n");return -ENODEV;}regs = ioremap(res.start, 1 + res.end - res.start);if (!regs) {dev_err(&op->dev, "Could not map registers\n");return -ENODEV;}/* Allocate and initialize the driver private data */psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL);if (!psc_i2s) {iounmap(regs);return -ENOMEM;}spin_lock_init(&psc_i2s->lock);psc_i2s->irq = irq;psc_i2s->psc_regs = regs;psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs;psc_i2s->dev = &op->dev;psc_i2s->playback_stream.psc_i2s = psc_i2s;psc_i2s->capture_stream.psc_i2s = psc_i2s;snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1);/* Fill out the CPU DAI structure */memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai);psc_i2s->dai.private_data = psc_i2s;psc_i2s->dai.name = psc_i2s->name;psc_i2s->dai.id = psc_id;/* Find the address of the fifo data registers and setup the* DMA tasks */fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);psc_i2s->capture_stream.bcom_task =bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512);psc_i2s->playback_stream.bcom_task =bcom_psc_gen_bd_tx_init(psc_id, 10, fifo);if (!psc_i2s->capture_stream.bcom_task ||!psc_i2s->playback_stream.bcom_task) {dev_err(&op->dev, "Could not allocate bestcomm tasks\n");iounmap(regs);kfree(psc_i2s);return -ENODEV;}/* Save what we've done so it can be found again later */dev_set_drvdata(&op->dev, psc_i2s);/* Register the SYSFS files */rc = device_create_file(psc_i2s->dev, &dev_attr_status);rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);if (rc)dev_info(psc_i2s->dev, "error creating sysfs files\n");/* Tell the ASoC OF helpers about it */of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,&psc_i2s->dai);return 0;+}
+static int __devexit psc_i2s_of_remove(struct of_device *op) +{
struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev);dev_dbg(&op->dev, "psc_i2s_remove()\n");bcom_gen_bd_rx_release(psc_i2s->capture_stream.bcom_task);bcom_gen_bd_tx_release(psc_i2s->playback_stream.bcom_task);iounmap(psc_i2s->psc_regs);iounmap(psc_i2s->fifo_regs);kfree(psc_i2s);dev_set_drvdata(&op->dev, NULL);return 0;+}
+/* Match table for of_platform binding */ +static struct of_device_id psc_i2s_match[] __devinitdata = {
{ .compatible = "fsl,mpc5200-psc-i2s", },{}+}; +MODULE_DEVICE_TABLE(of, psc_i2s_match);
+static struct of_platform_driver psc_i2s_driver = {
.match_table = psc_i2s_match,.probe = psc_i2s_of_probe,.remove = __devexit_p(psc_i2s_of_remove),.driver = {.name = "mpc5200-psc-i2s",.owner = THIS_MODULE,},+};
+/* ---------------------------------------------------------------------
- Module setup and teardown; simply register the of_platform driver
- for the PSC in I2S mode.
- */
+static int __init psc_i2s_init(void) +{
return of_register_platform_driver(&psc_i2s_driver);+} +module_init(psc_i2s_init);
+static void __exit psc_i2s_exit(void) +{
of_unregister_platform_driver(&psc_i2s_driver);+} +module_exit(psc_i2s_exit);
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
On Wed, Jul 02, 2008 at 11:19:18AM -0400, Jon Smirl wrote:
On 7/1/08, Grant Likely grant.likely@secretlab.ca wrote:
/* Due to errata in the i2s mode; need to line up enabling* the transmitter with a transition on the frame sync* line */spin_lock_irqsave(&psc_i2s->lock, flags);/* first make sure it is low */while ((in_8(®s->ipcr_acr.ipcr) & 0x80) != 0);Could this be moved to the front of the routine, to increase parallelism?
Once you detect the 0, it will be a fixed interval before the 1 happens. Might as well overlap the computations.
Good point. I'll try this out and see if it remains stable. If this gets done wrong, then left and right channels will often get swapped.
g.
On 7/1/08, Grant Likely grant.likely@secretlab.ca wrote:
From: Grant Likely grant.likely@secretlab.ca
This is an I2S bus driver for the MPC5200 PSC device. It is probably will not be merged as-is because it uses v1 of the ASoC API, but I want to get it out there for comments.
The driver is assuming a capture stream exists. My codec is output only.
I'm using external clocking, but the driver should support using the mpc5200 for clocking. That's a little complicated since you have to compute the divisors. For example the Phytec pcm030 board has a 33.3333Mhz xtal and runs at 400Mhz.
In order to reduce options, can the psc-i2s driver always try to use mpc5200 clocking, then let the codec or fabric driver override it?
On Sun, Jul 06, 2008 at 01:56:48PM -0400, Jon Smirl wrote:
The driver is assuming a capture stream exists. My codec is output only.
While the driver declares a capture stream the core doesn't require that both capture and playback be available - it will cope with a capture only or a playback only DAI (this is fairly common due to DAC only and ADC only parts). Unless there's some other issue specific to this driver?
I'm using external clocking, but the driver should support using the mpc5200 for clocking. That's a little complicated since you have to compute the divisors. For example the Phytec pcm030 board has a 33.3333Mhz xtal and runs at 400Mhz.
This is desirable, though it shouldn't be an obstacle for merging if the driver only supports running in slave mode.
In order to reduce options, can the psc-i2s driver always try to use mpc5200 clocking, then let the codec or fabric driver override it?
The clocking should always be under the control of the machine driver with the codec and platform drivers exporting the required dividers and PLLs/FLLs. Neither the platform driver nor the codec driver are really in a position to know how a given board is wired up and what interdependencies or external requirements there are.
On 7/7/08, Mark Brown broonie@opensource.wolfsonmicro.com wrote:
On Sun, Jul 06, 2008 at 01:56:48PM -0400, Jon Smirl wrote:
The driver is assuming a capture stream exists. My codec is output only.
While the driver declares a capture stream the core doesn't require that both capture and playback be available - it will cope with a capture only or a playback only DAI (this is fairly common due to DAC only and ADC only parts). Unless there's some other issue specific to this driver?
Yes, it GPFs allocating a a DMA buffer on the null capture stream pointer.
I'm using external clocking, but the driver should support using the mpc5200 for clocking. That's a little complicated since you have to compute the divisors. For example the Phytec pcm030 board has a 33.3333Mhz xtal and runs at 400Mhz.
This is desirable, though it shouldn't be an obstacle for merging if the driver only supports running in slave mode.
In order to reduce options, can the psc-i2s driver always try to use mpc5200 clocking, then let the codec or fabric driver override it?
The clocking should always be under the control of the machine driver with the codec and platform drivers exporting the required dividers and PLLs/FLLs. Neither the platform driver nor the codec driver are really in a position to know how a given board is wired up and what interdependencies or external requirements there are.
On Mon, Jul 07, 2008 at 09:23:24AM -0400, Jon Smirl wrote:
On 7/7/08, Mark Brown broonie@opensource.wolfsonmicro.com wrote:
On Sun, Jul 06, 2008 at 01:56:48PM -0400, Jon Smirl wrote:
The driver is assuming a capture stream exists. My codec is output only.
While the driver declares a capture stream the core doesn't require that both capture and playback be available - it will cope with a capture only or a playback only DAI (this is fairly common due to DAC only and ADC only parts). Unless there's some other issue specific to this driver?
Yes, it GPFs allocating a a DMA buffer on the null capture stream pointer.
Where does it GPF? When dereferencing pcm->streams[x].substream in psc_i2s_pcm_new?
On 7/1/08, Grant Likely grant.likely@secretlab.ca wrote:
From: Grant Likely grant.likely@secretlab.ca
This is an I2S bus driver for the MPC5200 PSC device. It is probably will not be merged as-is because it uses v1 of the ASoC API, but I want to get it out there for comments.
I need some slight tweaks since we are using PSC1 in cellphone mode to distribute the audio clock.
i2s@2000 { /* PSC1 in i2s mode */ compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s"; cell-index = <0>; reg = <0x2000 0x100>; interrupts = <0x2 0x1 0x0>; interrupt-parent = <&mpc5200_pic>; };
i2s@2200 { /* PSC2 in i2s mode */ compatible = "fsl,mpc5200b-psc-i2s","fsl,mpc5200-psc-i2s"; cell-index = <1>; reg = <0x2200 0x100>; interrupts = <0x2 0x2 0x0>; interrupt-parent = <&mpc5200_pic>; codec-handle = <&tas0>; fsl5200-cellslave; };
Our PSC1 is in master mode, but it doesn't have a codec hooked to it. I needed to modify the driver to initialize the PSC to i2s master mode but then not start all of the ALSA support. You can detect this state since there is no codec node. Putting PSC1 into master mode lets us get our external audio clock inside the mpc5200.
PSC2 is a cellphone slave. It gets its clock from PSC1. Everything is the same as what you are doing except I need to set MPC52xx_PSC_SICR_CELLSLAVE and MPC52xx_PSC_SICR_GENCLK when the fsl5200-cellslave attribute is present.
We need to tie the two PSCs up like this to get the audio clock in via PSC1 and then have PSC2 generate the frame clock when the i2s data is transmitted.
Do you want a diff, or do you have a new version with DMA broken out?
On Mon, Jul 07, 2008 at 12:32:29PM -0400, Jon Smirl wrote:
On 7/1/08, Grant Likely grant.likely@secretlab.ca wrote:
From: Grant Likely grant.likely@secretlab.ca
This is an I2S bus driver for the MPC5200 PSC device. It is probably will not be merged as-is because it uses v1 of the ASoC API, but I want to get it out there for comments.
We need to tie the two PSCs up like this to get the audio clock in via PSC1 and then have PSC2 generate the frame clock when the i2s data is transmitted.
Do you want a diff, or do you have a new version with DMA broken out?
You may as well send me the diff. I can't promise that I'll actually get it merged in, but I'll try.
g.
From: Grant Likely grant.likely@secretlab.ca
ASoC Codec driver for the TLV320AIC26 device. This driver uses the ASoC v1 API, so I don't expect it to get merged as-is, but I want to get it out there for review. ---
sound/soc/codecs/Kconfig | 4 sound/soc/codecs/Makefile | 2 sound/soc/codecs/tlv320aic26.c | 630 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 636 insertions(+), 0 deletions(-)
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3903ab7..96c7bfe 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -41,6 +41,10 @@ config SND_SOC_CS4270_VD33_ERRATA bool depends on SND_SOC_CS4270
+config SND_SOC_TLV320AIC26 + tristate "TI TLB320AIC26 Codec support" + depends on SND_SOC && SPI + config SND_SOC_TLV320AIC3X tristate depends on SND_SOC && I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4e1314c..ec0cd93 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -5,6 +5,7 @@ snd-soc-wm8753-objs := wm8753.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o +snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o
obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o @@ -14,4 +15,5 @@ obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.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/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c new file mode 100644 index 0000000..aee1dbc --- /dev/null +++ b/sound/soc/codecs/tlv320aic26.c @@ -0,0 +1,630 @@ +/* + * Texas Instruments TLV320AIC26 low power audio CODEC + * ALSA SoC CODEC driver + * + * Copyright (C) 2008 Secret Lab Technologies Ltd. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/device.h> +#include <linux/sysfs.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/soc-of.h> +#include <sound/initval.h> + +MODULE_DESCRIPTION("ASoC TLV320AIC26 codec driver"); +MODULE_AUTHOR("Grant Likely grant.likely@secretlab.ca"); +MODULE_LICENSE("GPL"); + +/* AIC26 Registers */ +#define AIC26_READ_COMMAND_WORD(addr) ((1 << 15) | (addr << 5)) +#define AIC26_WRITE_COMMAND_WORD(addr) ((0 << 15) | (addr << 5)) +#define AIC26_PAGE_ADDR(page, offset) ((page << 6) | offset) +#define AIC26_NUM_REGS AIC26_PAGE_ADDR(3, 0) +#define AIC26_REG_CACHE_SIZE (0x20) /* only page 2 cached */ +#define AIC26_REG_IS_CACHED(addr) ((addr & ~0x1f) == (2 << 6)) +#define AIC26_REG_CACHE_ADDR(addr) (addr & 0x1f) + +/* Page 0: Auxillary data registers */ +#define AIC26_REG_BAT1 AIC26_PAGE_ADDR(0, 0x05) +#define AIC26_REG_BAT2 AIC26_PAGE_ADDR(0, 0x06) +#define AIC26_REG_AUX AIC26_PAGE_ADDR(0, 0x07) +#define AIC26_REG_TEMP1 AIC26_PAGE_ADDR(0, 0x09) +#define AIC26_REG_TEMP2 AIC26_PAGE_ADDR(0, 0x0A) + +/* Page 1: Auxillary control registers */ +#define AIC26_REG_AUX_ADC AIC26_PAGE_ADDR(1, 0x00) +#define AIC26_REG_STATUS AIC26_PAGE_ADDR(1, 0x01) +#define AIC26_REG_REFERENCE AIC26_PAGE_ADDR(1, 0x03) +#define AIC26_REG_RESET AIC26_PAGE_ADDR(1, 0x04) + +/* Page 2: Audio control registers */ +#define AIC26_REG_AUDIO_CTRL1 AIC26_PAGE_ADDR(2, 0x00) +#define AIC26_REG_ADC_GAIN AIC26_PAGE_ADDR(2, 0x01) +#define AIC26_REG_DAC_GAIN AIC26_PAGE_ADDR(2, 0x02) +#define AIC26_REG_SIDETONE AIC26_PAGE_ADDR(2, 0x03) +#define AIC26_REG_AUDIO_CTRL2 AIC26_PAGE_ADDR(2, 0x04) +#define AIC26_REG_POWER_CTRL AIC26_PAGE_ADDR(2, 0x05) +#define AIC26_REG_AUDIO_CTRL3 AIC26_PAGE_ADDR(2, 0x06) + +#define AIC26_REG_FILTER_COEFF_L_N0 AIC26_PAGE_ADDR(2, 0x07) +#define AIC26_REG_FILTER_COEFF_L_N1 AIC26_PAGE_ADDR(2, 0x08) +#define AIC26_REG_FILTER_COEFF_L_N2 AIC26_PAGE_ADDR(2, 0x09) +#define AIC26_REG_FILTER_COEFF_L_N3 AIC26_PAGE_ADDR(2, 0x0A) +#define AIC26_REG_FILTER_COEFF_L_N4 AIC26_PAGE_ADDR(2, 0x0B) +#define AIC26_REG_FILTER_COEFF_L_N5 AIC26_PAGE_ADDR(2, 0x0C) +#define AIC26_REG_FILTER_COEFF_L_D1 AIC26_PAGE_ADDR(2, 0x0D) +#define AIC26_REG_FILTER_COEFF_L_D2 AIC26_PAGE_ADDR(2, 0x0E) +#define AIC26_REG_FILTER_COEFF_L_D4 AIC26_PAGE_ADDR(2, 0x0F) +#define AIC26_REG_FILTER_COEFF_L_D5 AIC26_PAGE_ADDR(2, 0x10) +#define AIC26_REG_FILTER_COEFF_R_N0 AIC26_PAGE_ADDR(2, 0x11) +#define AIC26_REG_FILTER_COEFF_R_N1 AIC26_PAGE_ADDR(2, 0x12) +#define AIC26_REG_FILTER_COEFF_R_N2 AIC26_PAGE_ADDR(2, 0x13) +#define AIC26_REG_FILTER_COEFF_R_N3 AIC26_PAGE_ADDR(2, 0x14) +#define AIC26_REG_FILTER_COEFF_R_N4 AIC26_PAGE_ADDR(2, 0x15) +#define AIC26_REG_FILTER_COEFF_R_N5 AIC26_PAGE_ADDR(2, 0x16) +#define AIC26_REG_FILTER_COEFF_R_D1 AIC26_PAGE_ADDR(2, 0x17) +#define AIC26_REG_FILTER_COEFF_R_D2 AIC26_PAGE_ADDR(2, 0x18) +#define AIC26_REG_FILTER_COEFF_R_D4 AIC26_PAGE_ADDR(2, 0x19) +#define AIC26_REG_FILTER_COEFF_R_D5 AIC26_PAGE_ADDR(2, 0x1A) + +#define AIC26_REG_PLL_PROG1 AIC26_PAGE_ADDR(2, 0x1B) +#define AIC26_REG_PLL_PROG2 AIC26_PAGE_ADDR(2, 0x1C) +#define AIC26_REG_AUDIO_CTRL4 AIC26_PAGE_ADDR(2, 0x1D) +#define AIC26_REG_AUDIO_CTRL5 AIC26_PAGE_ADDR(2, 0x1E) + +#define AIC26_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) +#define AIC26_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |\ + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) + +/* fsref dividers; used in register 'Audio Control 1' */ +enum aic26_divisors { + AIC26_DIV_1 = 0, + AIC26_DIV_1_5 = 1, + AIC26_DIV_2 = 2, + AIC26_DIV_3 = 3, + AIC26_DIV_4 = 4, + AIC26_DIV_5 = 5, + AIC26_DIV_5_5 = 6, + AIC26_DIV_6 = 7, +}; + +/* Digital data format */ +enum aic26_datfm { + AIC26_DATFM_I2S = 0 << 8, + AIC26_DATFM_DSP = 1 << 8, + AIC26_DATFM_RIGHTJ = 2 << 8, /* right justified */ + AIC26_DATFM_LEFTJ = 3 << 8, /* left justified */ +}; + +/* Sample word length in bits; used in register 'Audio Control 1' */ +enum aic26_wlen { + AIC26_WLEN_16 = 0 << 10, + AIC26_WLEN_20 = 1 << 10, + AIC26_WLEN_24 = 2 << 10, + AIC26_WLEN_32 = 3 << 10, +}; + +/* AIC26 driver private data */ +struct aic26 { + struct spi_device *spi; + struct snd_soc_codec codec; + u16 reg_cache[AIC26_REG_CACHE_SIZE]; /* shadow registers */ + int master; + int datfm; + int mclk; + + /* Keyclick parameters */ + int keyclick_amplitude; + int keyclick_freq; + int keyclick_len; +}; + +/* --------------------------------------------------------------------- + * Register access routines + */ +static unsigned int aic26_reg_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct aic26 *aic26 = codec->private_data; + u16 *cache = codec->reg_cache; + u16 cmd, value; + u8 buffer[2]; + int rc; + + if ((reg < 0) || (reg >= AIC26_NUM_REGS)) { + WARN_ON_ONCE(1); + return 0; + } + + /* Do SPI transfer; first 16bits are command; remaining is + * register contents */ + cmd = AIC26_READ_COMMAND_WORD(reg); + buffer[0] = (cmd >> 8) & 0xff; + buffer[1] = cmd & 0xff; + rc = spi_write_then_read(aic26->spi, buffer, 2, buffer, 2); + if (rc) { + dev_err(&aic26->spi->dev, "AIC26 reg read error\n"); + return -EIO; + } + value = (buffer[0] << 8) | buffer[1]; + + /* Update the cache before returning with the value */ + if (AIC26_REG_IS_CACHED(reg)) + cache[AIC26_REG_CACHE_ADDR(reg)] = value; + return value; +} + +static unsigned int aic26_reg_read_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + + if ((reg < 0) || (reg >= AIC26_NUM_REGS)) { + WARN_ON_ONCE(1); + return 0; + } + + if (AIC26_REG_IS_CACHED(reg)) + return cache[AIC26_REG_CACHE_ADDR(reg)]; + + return aic26_reg_read(codec, reg); +} + +static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct aic26 *aic26 = codec->private_data; + u16 *cache = codec->reg_cache; + u16 cmd; + u8 buffer[4]; + int rc; + + if ((reg < 0) || (reg >= AIC26_NUM_REGS)) { + WARN_ON_ONCE(1); + return -EINVAL; + } + + /* Do SPI transfer; first 16bits are command; remaining is data + * to write into register */ + cmd = AIC26_WRITE_COMMAND_WORD(reg); + buffer[0] = (cmd >> 8) & 0xff; + buffer[1] = cmd & 0xff; + buffer[2] = value >> 8; + buffer[3] = value; + rc = spi_write(aic26->spi, buffer, 4); + if (rc) { + dev_err(&aic26->spi->dev, "AIC26 reg read error\n"); + return -EIO; + } + + /* update cache before returning */ + if (AIC26_REG_IS_CACHED(reg)) + cache[AIC26_REG_CACHE_ADDR(reg)] = value; + return 0; +} + +/* --------------------------------------------------------------------- + * Digital Audio Interface Operations + */ +static int aic26_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct aic26 *aic26 = codec->private_data; + int fsref, divisor, wlen, pval, jval, dval, qval; + u16 reg; + + dev_dbg(&aic26->spi->dev, "aic26_hw_params(substream=%p, params=%p)\n", + substream, params); + dev_dbg(&aic26->spi->dev, "rate=%i format=%i\n", params_rate(params), + params_format(params)); + + switch (params_rate(params)) { + case 8000: fsref = 48000; divisor = AIC26_DIV_6; break; + case 11025: fsref = 44100; divisor = AIC26_DIV_4; break; + case 12000: fsref = 48000; divisor = AIC26_DIV_4; break; + case 16000: fsref = 48000; divisor = AIC26_DIV_3; break; + case 22050: fsref = 44100; divisor = AIC26_DIV_2; break; + case 24000: fsref = 48000; divisor = AIC26_DIV_2; break; + case 32000: fsref = 48000; divisor = AIC26_DIV_1_5; break; + case 44100: fsref = 44100; divisor = AIC26_DIV_1; break; + case 48000: fsref = 48000; divisor = AIC26_DIV_1; break; + default: dev_dbg(&aic26->spi->dev, "bad rate\n"); return -EINVAL; + } + + /* select data word length */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: wlen = AIC26_WLEN_16; break; + case SNDRV_PCM_FORMAT_S16_BE: wlen = AIC26_WLEN_16; break; + case SNDRV_PCM_FORMAT_S24_BE: wlen = AIC26_WLEN_24; break; + case SNDRV_PCM_FORMAT_S32_BE: wlen = AIC26_WLEN_32; break; + default: dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL; + } + + /* Configure PLL */ + pval = 1; + jval = (fsref == 44100) ? 7 : 8; + dval = (fsref == 44100) ? 5264 : 1920; + qval = 0; + reg = 0x8000 | qval << 11 | pval << 8 | jval << 2; + aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg); + reg = dval << 2; + aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg); + + /* Power up CODEC */ + aic26_reg_write(codec, AIC26_REG_POWER_CTRL, 0); + + /* Audio Control 3 (master mode, fsref rate) */ + reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3); + reg &= ~0xf800; + if (aic26->master) + reg |= 0x0800; + if (fsref == 48000) + reg |= 0x2000; + aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg); + + /* Audio Control 1 (FSref divisor) */ + reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1); + reg &= ~0x0fff; + reg |= wlen | aic26->datfm | (divisor << 3) | divisor; + aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL1, reg); + + return 0; +} + +/** + * aic26_mute - Mute control to reduce noise when changing audio format + */ +static int aic26_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct aic26 *aic26 = codec->private_data; + u16 reg = aic26_reg_read_cache(codec, AIC26_REG_DAC_GAIN); + + dev_dbg(&aic26->spi->dev, "aic26_mute(dai=%p, mute=%i)\n", + dai, mute); + + if (mute) + reg |= 0x8080; + else + reg &= ~0x8080; + aic26_reg_write(codec, AIC26_REG_DAC_GAIN, reg); + + return 0; +} + +static int aic26_set_sysclk(struct snd_soc_codec_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic26 *aic26 = codec->private_data; + + dev_dbg(&aic26->spi->dev, "aic26_set_sysclk(dai=%p, clk_id==%i," + " freq=%i, dir=%i)\n", + codec_dai, clk_id, freq, dir); + + /* MCLK needs to fall between 2MHz and 50 MHz */ + if ((freq < 2000000) || (freq > 50000000)) + return -EINVAL; + + aic26->mclk = freq; + return 0; +} + +static int aic26_set_fmt(struct snd_soc_codec_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic26 *aic26 = codec->private_data; + + dev_dbg(&aic26->spi->dev, "aic26_set_fmt(dai=%p, fmt==%i)\n", + codec_dai, fmt); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: aic26->master = 1; break; + //case SND_SOC_DAIFMT_CBS_CFS: aic26->master = 0; break; + default: dev_dbg(&aic26->spi->dev, "bad master\n"); return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: aic26->datfm = AIC26_DATFM_I2S; break; + case SND_SOC_DAIFMT_DSP_A: aic26->datfm = AIC26_DATFM_DSP; break; + case SND_SOC_DAIFMT_RIGHT_J: aic26->datfm = AIC26_DATFM_RIGHTJ; break; + case SND_SOC_DAIFMT_LEFT_J: aic26->datfm = AIC26_DATFM_LEFTJ; break; + default: dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL; + } + + return 0; +} + +/* --------------------------------------------------------------------- + * Digital Audio Interface Definition + */ +struct snd_soc_codec_dai aic26_dai = { + .name = "tlv320aic26", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AIC26_RATES, + .formats = AIC26_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AIC26_RATES, + .formats = AIC26_FORMATS, + }, + .ops = { + .hw_params = aic26_hw_params, + }, + .dai_ops = { + .digital_mute = aic26_mute, + .set_sysclk = aic26_set_sysclk, + .set_fmt = aic26_set_fmt, + }, +}; +EXPORT_SYMBOL_GPL(aic26_dai); + +/* --------------------------------------------------------------------- + * ALSA controls + */ +static const char *aic26_capture_src_text[] = {"Mic", "Aux"}; +static const struct soc_enum aic26_capture_src_enum = + SOC_ENUM_SINGLE(AIC26_REG_AUDIO_CTRL1, 12,2, aic26_capture_src_text); + +static const struct snd_kcontrol_new aic26_snd_controls[] = { + /* Output */ + SOC_DOUBLE("PCM Playback Volume", AIC26_REG_DAC_GAIN, 8, 0, 0x7f, 1), + SOC_DOUBLE("PCM Playback Switch", AIC26_REG_DAC_GAIN, 15, 7, 1, 1), + SOC_SINGLE("PCM Capture Volume", AIC26_REG_ADC_GAIN, 8, 0x7f, 0), + SOC_SINGLE("PCM Capture Mute", AIC26_REG_ADC_GAIN, 15, 1, 1), + SOC_ENUM("Capture Source", aic26_capture_src_enum), +}; + +/* --------------------------------------------------------------------- + * SoC CODEC portion of driver: probe and release routines + */ +static int aic26_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct snd_kcontrol *kcontrol; + struct aic26 *aic26; + int i, ret, err; + + dev_info(&pdev->dev, "Probing AIC26 SoC CODEC driver\n"); + dev_dbg(&pdev->dev, "socdev=%p\n", socdev); + dev_dbg(&pdev->dev, "codec_data=%p\n", socdev->codec_data); + + /* Fetch the relevant aic26 private data here (it's already been + * stored in the .codec pointer) */ + aic26 = socdev->codec_data; + if (aic26 == NULL) { + dev_err(&pdev->dev, "aic26: missing codec pointer\n"); + return -ENODEV; + } + codec = &aic26->codec; + socdev->codec = codec; + + dev_dbg(&pdev->dev, "Registering PCMs, dev=%p, socdev->dev=%p\n", + &pdev->dev, socdev->dev); + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "aic26: failed to create pcms\n"); + return -ENODEV; + } + + /* register controls */ + dev_dbg(&pdev->dev, "Registering controls\n"); + for (i = 0; i < ARRAY_SIZE(aic26_snd_controls); i++) { + kcontrol = snd_soc_cnew(&aic26_snd_controls[i], codec, NULL); + err = snd_ctl_add(codec->card, kcontrol); + WARN_ON(err < 0); + } + + /* CODEC is setup, we can register the card now */ + dev_dbg(&pdev->dev, "Registering card\n"); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + dev_err(&pdev->dev, "aic26: failed to register card\n"); + goto card_err; + } + return 0; + + card_err: + snd_soc_free_pcms(socdev); + return ret; +} + +static int aic26_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + snd_soc_free_pcms(socdev); + return 0; +} + +struct snd_soc_codec_device aic26_soc_codec_dev = { + .probe = aic26_probe, + .remove = aic26_remove, +}; + +/* --------------------------------------------------------------------- + * SPI device portion of driver: sysfs files for debugging + */ + +static ssize_t aic26_regs_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct aic26 *aic26 = dev_get_drvdata(dev); + char *idx = buf; + int cache_flag, addr, page, i, reg; + + cache_flag = (strcmp(attr->attr.name, "regs_cache") == 0); + + for (page = 0; page < 3; page++) { + for (i = 0; i < 0x20; i++) { + addr = AIC26_PAGE_ADDR(page, i); + if (i % 8 == 0) + idx += sprintf(idx, "%i:%.2i:", page,i); + if (cache_flag) + reg = aic26_reg_read_cache(&aic26->codec, addr); + else + reg = aic26_reg_read(&aic26->codec, addr); + idx += sprintf(idx, " %.4x", reg); + if (i % 8 == 7) + idx += sprintf(idx, "\n"); + } + } + return idx - buf; +} + +static ssize_t aic26_keyclick_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct aic26 *aic26 = dev_get_drvdata(dev); + int val, amp, freq, len; + + val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2); + amp = (val >> 12) & 0x7; + freq = (125 << ((val >> 8) & 0x7)) >> 1; + len = 2 * (1 +((val >> 8) & 0xf)); + + return sprintf(buf, "amp=%x freq=%iHz len=%iclks\n", amp, freq, len); +} + +/* Any write to the keyclick attribute will trigger the keyclick */ +static ssize_t aic26_keyclick_set(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct aic26 *aic26 = dev_get_drvdata(dev); + int val; + + val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2); + val |= 0x8000; + aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL2, val); + + return count; +} + +DEVICE_ATTR(regs, 0644, aic26_regs_show, NULL); +DEVICE_ATTR(regs_cache, 0644, aic26_regs_show, NULL); +DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set); + +/* --------------------------------------------------------------------- + * SPI device portion of driver: probe and release routines and SPI + * driver registration. + */ +static int aic26_spi_probe(struct spi_device *spi) +{ + struct aic26 *aic26; + int rc, i, reg; + + dev_dbg(&spi->dev, "probing tlv320aic26 spi device\n"); + + /* Allocate driver data */ + aic26 = kzalloc(sizeof *aic26, GFP_KERNEL); + if (!aic26) + return -ENOMEM; + + /* Initialize the driver data */ + aic26->spi = spi; + dev_set_drvdata(&spi->dev, aic26); + + /* Setup what we can in the codec structure so that the register + * access functions will work as expected. More will be filled + * out when it is probed by the SoC CODEC part of this driver */ + aic26->codec.private_data = aic26; + aic26->codec.name = "aic26"; + aic26->codec.owner = THIS_MODULE; + aic26->codec.dai = &aic26_dai; + aic26->codec.num_dai = 1; + aic26->codec.read = aic26_reg_read; + aic26->codec.write = aic26_reg_write; + aic26->master = 1; + mutex_init(&aic26->codec.mutex); + INIT_LIST_HEAD(&aic26->codec.dapm_widgets); + INIT_LIST_HEAD(&aic26->codec.dapm_paths); + aic26->codec.reg_cache_size = sizeof(aic26->reg_cache); + aic26->codec.reg_cache = aic26->reg_cache; + + /* Reset the codec to power on defaults */ + aic26_reg_write(&aic26->codec, AIC26_REG_RESET, 0xBB00); + + /* Power up CODEC */ + aic26_reg_write(&aic26->codec, AIC26_REG_POWER_CTRL, 0); + + /* Audio Control 3 (master mode, fsref rate) */ + reg = aic26_reg_read(&aic26->codec, AIC26_REG_AUDIO_CTRL3); + reg &= ~0xf800; + reg |= 0x0800; /* set master mode */ + aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL3, reg); + + /* Fill page 2 register cache */ + for (i = 0; i < ARRAY_SIZE(aic26->reg_cache); i++) + aic26_reg_read(&aic26->codec, AIC26_PAGE_ADDR(2, i)); + + /* Register the sysfs files for debugging */ + /* Create SysFS files */ + rc = device_create_file(&spi->dev, &dev_attr_regs); + rc |= device_create_file(&spi->dev, &dev_attr_regs_cache); + rc |= device_create_file(&spi->dev, &dev_attr_keyclick); + if (rc) + dev_info(&spi->dev, "error creating sysfs files\n"); + + /* Tell the of_soc helper about this codec */ + of_snd_soc_register_codec(&aic26_soc_codec_dev, aic26, &aic26_dai, + spi->dev.archdata.of_node); + + dev_dbg(&spi->dev, "SPI device initialized\n"); + return 0; +} + +static int aic26_spi_remove(struct spi_device *spi) +{ + struct aic26 *aic26 = dev_get_drvdata(&spi->dev); + + kfree(aic26); + + return 0; +} + +static struct spi_driver aic26_spi = { + .driver = { + .name = "tlv320aic26", + .owner = THIS_MODULE, + }, + .probe = aic26_spi_probe, + .remove = aic26_spi_remove, +}; + +static int __init aic26_init(void) +{ + return spi_register_driver(&aic26_spi); +} +module_init(aic26_init); + +static void __exit aic26_exit(void) +{ + spi_unregister_driver(&aic26_spi); +} +module_exit(aic26_exit);
On Tue, 2008-07-01 at 17:53 -0600, Grant Likely wrote:
From: Grant Likely grant.likely@secretlab.ca
ASoC Codec driver for the TLV320AIC26 device. This driver uses the ASoC v1 API, so I don't expect it to get merged as-is, but I want to get it out there for review.
sound/soc/codecs/Kconfig | 4 sound/soc/codecs/Makefile | 2 sound/soc/codecs/tlv320aic26.c | 630 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 636 insertions(+), 0 deletions(-)
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3903ab7..96c7bfe 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -41,6 +41,10 @@ config SND_SOC_CS4270_VD33_ERRATA bool depends on SND_SOC_CS4270
+config SND_SOC_TLV320AIC26
- tristate "TI TLB320AIC26 Codec support"
- depends on SND_SOC && SPI
config SND_SOC_TLV320AIC3X tristate depends on SND_SOC && I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4e1314c..ec0cd93 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -5,6 +5,7 @@ snd-soc-wm8753-objs := wm8753.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o +snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o
obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o @@ -14,4 +15,5 @@ obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o +obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c new file mode 100644 index 0000000..aee1dbc --- /dev/null +++ b/sound/soc/codecs/tlv320aic26.c @@ -0,0 +1,630 @@ +/*
- Texas Instruments TLV320AIC26 low power audio CODEC
- ALSA SoC CODEC driver
- Copyright (C) 2008 Secret Lab Technologies Ltd.
- */
+#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/device.h> +#include <linux/sysfs.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/soc-of.h> +#include <sound/initval.h>
+MODULE_DESCRIPTION("ASoC TLV320AIC26 codec driver"); +MODULE_AUTHOR("Grant Likely grant.likely@secretlab.ca"); +MODULE_LICENSE("GPL");
+/* AIC26 Registers */ +#define AIC26_READ_COMMAND_WORD(addr) ((1 << 15) | (addr << 5)) +#define AIC26_WRITE_COMMAND_WORD(addr) ((0 << 15) | (addr << 5)) +#define AIC26_PAGE_ADDR(page, offset) ((page << 6) | offset) +#define AIC26_NUM_REGS AIC26_PAGE_ADDR(3, 0) +#define AIC26_REG_CACHE_SIZE (0x20) /* only page 2 cached */ +#define AIC26_REG_IS_CACHED(addr) ((addr & ~0x1f) == (2 << 6)) +#define AIC26_REG_CACHE_ADDR(addr) (addr & 0x1f)
+/* Page 0: Auxillary data registers */ +#define AIC26_REG_BAT1 AIC26_PAGE_ADDR(0, 0x05) +#define AIC26_REG_BAT2 AIC26_PAGE_ADDR(0, 0x06) +#define AIC26_REG_AUX AIC26_PAGE_ADDR(0, 0x07) +#define AIC26_REG_TEMP1 AIC26_PAGE_ADDR(0, 0x09) +#define AIC26_REG_TEMP2 AIC26_PAGE_ADDR(0, 0x0A)
+/* Page 1: Auxillary control registers */ +#define AIC26_REG_AUX_ADC AIC26_PAGE_ADDR(1, 0x00) +#define AIC26_REG_STATUS AIC26_PAGE_ADDR(1, 0x01) +#define AIC26_REG_REFERENCE AIC26_PAGE_ADDR(1, 0x03) +#define AIC26_REG_RESET AIC26_PAGE_ADDR(1, 0x04)
+/* Page 2: Audio control registers */ +#define AIC26_REG_AUDIO_CTRL1 AIC26_PAGE_ADDR(2, 0x00) +#define AIC26_REG_ADC_GAIN AIC26_PAGE_ADDR(2, 0x01) +#define AIC26_REG_DAC_GAIN AIC26_PAGE_ADDR(2, 0x02) +#define AIC26_REG_SIDETONE AIC26_PAGE_ADDR(2, 0x03) +#define AIC26_REG_AUDIO_CTRL2 AIC26_PAGE_ADDR(2, 0x04) +#define AIC26_REG_POWER_CTRL AIC26_PAGE_ADDR(2, 0x05) +#define AIC26_REG_AUDIO_CTRL3 AIC26_PAGE_ADDR(2, 0x06)
+#define AIC26_REG_FILTER_COEFF_L_N0 AIC26_PAGE_ADDR(2, 0x07) +#define AIC26_REG_FILTER_COEFF_L_N1 AIC26_PAGE_ADDR(2, 0x08) +#define AIC26_REG_FILTER_COEFF_L_N2 AIC26_PAGE_ADDR(2, 0x09) +#define AIC26_REG_FILTER_COEFF_L_N3 AIC26_PAGE_ADDR(2, 0x0A) +#define AIC26_REG_FILTER_COEFF_L_N4 AIC26_PAGE_ADDR(2, 0x0B) +#define AIC26_REG_FILTER_COEFF_L_N5 AIC26_PAGE_ADDR(2, 0x0C) +#define AIC26_REG_FILTER_COEFF_L_D1 AIC26_PAGE_ADDR(2, 0x0D) +#define AIC26_REG_FILTER_COEFF_L_D2 AIC26_PAGE_ADDR(2, 0x0E) +#define AIC26_REG_FILTER_COEFF_L_D4 AIC26_PAGE_ADDR(2, 0x0F) +#define AIC26_REG_FILTER_COEFF_L_D5 AIC26_PAGE_ADDR(2, 0x10) +#define AIC26_REG_FILTER_COEFF_R_N0 AIC26_PAGE_ADDR(2, 0x11) +#define AIC26_REG_FILTER_COEFF_R_N1 AIC26_PAGE_ADDR(2, 0x12) +#define AIC26_REG_FILTER_COEFF_R_N2 AIC26_PAGE_ADDR(2, 0x13) +#define AIC26_REG_FILTER_COEFF_R_N3 AIC26_PAGE_ADDR(2, 0x14) +#define AIC26_REG_FILTER_COEFF_R_N4 AIC26_PAGE_ADDR(2, 0x15) +#define AIC26_REG_FILTER_COEFF_R_N5 AIC26_PAGE_ADDR(2, 0x16) +#define AIC26_REG_FILTER_COEFF_R_D1 AIC26_PAGE_ADDR(2, 0x17) +#define AIC26_REG_FILTER_COEFF_R_D2 AIC26_PAGE_ADDR(2, 0x18) +#define AIC26_REG_FILTER_COEFF_R_D4 AIC26_PAGE_ADDR(2, 0x19) +#define AIC26_REG_FILTER_COEFF_R_D5 AIC26_PAGE_ADDR(2, 0x1A)
+#define AIC26_REG_PLL_PROG1 AIC26_PAGE_ADDR(2, 0x1B) +#define AIC26_REG_PLL_PROG2 AIC26_PAGE_ADDR(2, 0x1C) +#define AIC26_REG_AUDIO_CTRL4 AIC26_PAGE_ADDR(2, 0x1D) +#define AIC26_REG_AUDIO_CTRL5 AIC26_PAGE_ADDR(2, 0x1E)
+#define AIC26_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\SNDRV_PCM_RATE_48000)+#define AIC26_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |\
SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
Fwiw, I usually put all the codec registers defs in a separate header just in case we need to do any codec stuff in the machine driver.
+/* fsref dividers; used in register 'Audio Control 1' */ +enum aic26_divisors {
- AIC26_DIV_1 = 0,
- AIC26_DIV_1_5 = 1,
- AIC26_DIV_2 = 2,
- AIC26_DIV_3 = 3,
- AIC26_DIV_4 = 4,
- AIC26_DIV_5 = 5,
- AIC26_DIV_5_5 = 6,
- AIC26_DIV_6 = 7,
+};
+/* Digital data format */ +enum aic26_datfm {
- AIC26_DATFM_I2S = 0 << 8,
- AIC26_DATFM_DSP = 1 << 8,
- AIC26_DATFM_RIGHTJ = 2 << 8, /* right justified */
- AIC26_DATFM_LEFTJ = 3 << 8, /* left justified */
+};
+/* Sample word length in bits; used in register 'Audio Control 1' */ +enum aic26_wlen {
- AIC26_WLEN_16 = 0 << 10,
- AIC26_WLEN_20 = 1 << 10,
- AIC26_WLEN_24 = 2 << 10,
- AIC26_WLEN_32 = 3 << 10,
+};
+/* AIC26 driver private data */ +struct aic26 {
- struct spi_device *spi;
- struct snd_soc_codec codec;
- u16 reg_cache[AIC26_REG_CACHE_SIZE]; /* shadow registers */
- int master;
- int datfm;
- int mclk;
- /* Keyclick parameters */
- int keyclick_amplitude;
- int keyclick_freq;
- int keyclick_len;
+};
+/* ---------------------------------------------------------------------
- Register access routines
- */
+static unsigned int aic26_reg_read(struct snd_soc_codec *codec,
unsigned int reg)+{
- struct aic26 *aic26 = codec->private_data;
- u16 *cache = codec->reg_cache;
- u16 cmd, value;
- u8 buffer[2];
- int rc;
- if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {
WARN_ON_ONCE(1);return 0;- }
- /* Do SPI transfer; first 16bits are command; remaining is
* register contents */- cmd = AIC26_READ_COMMAND_WORD(reg);
- buffer[0] = (cmd >> 8) & 0xff;
- buffer[1] = cmd & 0xff;
- rc = spi_write_then_read(aic26->spi, buffer, 2, buffer, 2);
- if (rc) {
dev_err(&aic26->spi->dev, "AIC26 reg read error\n");return -EIO;- }
- value = (buffer[0] << 8) | buffer[1];
- /* Update the cache before returning with the value */
- if (AIC26_REG_IS_CACHED(reg))
cache[AIC26_REG_CACHE_ADDR(reg)] = value;- return value;
+}
+static unsigned int aic26_reg_read_cache(struct snd_soc_codec *codec,
unsigned int reg)+{
- u16 *cache = codec->reg_cache;
- if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {
WARN_ON_ONCE(1);return 0;- }
- if (AIC26_REG_IS_CACHED(reg))
return cache[AIC26_REG_CACHE_ADDR(reg)];- return aic26_reg_read(codec, reg);
+}
+static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)+{
- struct aic26 *aic26 = codec->private_data;
- u16 *cache = codec->reg_cache;
- u16 cmd;
- u8 buffer[4];
- int rc;
- if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {
WARN_ON_ONCE(1);return -EINVAL;- }
- /* Do SPI transfer; first 16bits are command; remaining is data
* to write into register */- cmd = AIC26_WRITE_COMMAND_WORD(reg);
- buffer[0] = (cmd >> 8) & 0xff;
- buffer[1] = cmd & 0xff;
- buffer[2] = value >> 8;
- buffer[3] = value;
- rc = spi_write(aic26->spi, buffer, 4);
- if (rc) {
dev_err(&aic26->spi->dev, "AIC26 reg read error\n");return -EIO;- }
- /* update cache before returning */
- if (AIC26_REG_IS_CACHED(reg))
cache[AIC26_REG_CACHE_ADDR(reg)] = value;- return 0;
+}
+/* ---------------------------------------------------------------------
- Digital Audio Interface Operations
- */
+static int aic26_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)+{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_device *socdev = rtd->socdev;
- struct snd_soc_codec *codec = socdev->codec;
- struct aic26 *aic26 = codec->private_data;
- int fsref, divisor, wlen, pval, jval, dval, qval;
- u16 reg;
- dev_dbg(&aic26->spi->dev, "aic26_hw_params(substream=%p, params=%p)\n",
substream, params);- dev_dbg(&aic26->spi->dev, "rate=%i format=%i\n", params_rate(params),
params_format(params));- switch (params_rate(params)) {
case 8000: fsref = 48000; divisor = AIC26_DIV_6; break;case 11025: fsref = 44100; divisor = AIC26_DIV_4; break;case 12000: fsref = 48000; divisor = AIC26_DIV_4; break;case 16000: fsref = 48000; divisor = AIC26_DIV_3; break;case 22050: fsref = 44100; divisor = AIC26_DIV_2; break;case 24000: fsref = 48000; divisor = AIC26_DIV_2; break;case 32000: fsref = 48000; divisor = AIC26_DIV_1_5; break;case 44100: fsref = 44100; divisor = AIC26_DIV_1; break;case 48000: fsref = 48000; divisor = AIC26_DIV_1; break;default: dev_dbg(&aic26->spi->dev, "bad rate\n"); return -EINVAL;- }
Indentation.
- /* select data word length */
- switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8: wlen = AIC26_WLEN_16; break;case SNDRV_PCM_FORMAT_S16_BE: wlen = AIC26_WLEN_16; break;case SNDRV_PCM_FORMAT_S24_BE: wlen = AIC26_WLEN_24; break;case SNDRV_PCM_FORMAT_S32_BE: wlen = AIC26_WLEN_32; break;default: dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;- }
ditto.
- /* Configure PLL */
- pval = 1;
- jval = (fsref == 44100) ? 7 : 8;
- dval = (fsref == 44100) ? 5264 : 1920;
- qval = 0;
- reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;
- aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);
- reg = dval << 2;
- aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg);
PLL/FLL/clock config is usually done in a separate function (codec_set_pll(), callable by machine driver) so that we can change clocks depending on the available machine clocks and srate.
- /* Power up CODEC */
- aic26_reg_write(codec, AIC26_REG_POWER_CTRL, 0);
Codec domain (i.e Bias power) PM stuff should be done in codec_dapm_event(). This allows us to power the codec on when we do things like sidetone (with no active playback or capture stream).
- /* Audio Control 3 (master mode, fsref rate) */
- reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3);
- reg &= ~0xf800;
- if (aic26->master)
reg |= 0x0800;- if (fsref == 48000)
reg |= 0x2000;- aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg);
- /* Audio Control 1 (FSref divisor) */
- reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1);
- reg &= ~0x0fff;
- reg |= wlen | aic26->datfm | (divisor << 3) | divisor;
- aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL1, reg);
- return 0;
+}
+/**
- aic26_mute - Mute control to reduce noise when changing audio format
- */
+static int aic26_mute(struct snd_soc_codec_dai *dai, int mute) +{
- struct snd_soc_codec *codec = dai->codec;
- struct aic26 *aic26 = codec->private_data;
- u16 reg = aic26_reg_read_cache(codec, AIC26_REG_DAC_GAIN);
- dev_dbg(&aic26->spi->dev, "aic26_mute(dai=%p, mute=%i)\n",
dai, mute);- if (mute)
reg |= 0x8080;- else
reg &= ~0x8080;- aic26_reg_write(codec, AIC26_REG_DAC_GAIN, reg);
- return 0;
+}
+static int aic26_set_sysclk(struct snd_soc_codec_dai *codec_dai,
int clk_id, unsigned int freq, int dir)+{
- struct snd_soc_codec *codec = codec_dai->codec;
- struct aic26 *aic26 = codec->private_data;
- dev_dbg(&aic26->spi->dev, "aic26_set_sysclk(dai=%p, clk_id==%i,"
" freq=%i, dir=%i)\n",codec_dai, clk_id, freq, dir);- /* MCLK needs to fall between 2MHz and 50 MHz */
- if ((freq < 2000000) || (freq > 50000000))
return -EINVAL;- aic26->mclk = freq;
- return 0;
+}
+static int aic26_set_fmt(struct snd_soc_codec_dai *codec_dai, unsigned int fmt) +{
- struct snd_soc_codec *codec = codec_dai->codec;
- struct aic26 *aic26 = codec->private_data;
- dev_dbg(&aic26->spi->dev, "aic26_set_fmt(dai=%p, fmt==%i)\n",
codec_dai, fmt);- /* set master/slave audio interface */
- switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM: aic26->master = 1; break;//case SND_SOC_DAIFMT_CBS_CFS: aic26->master = 0; break;default: dev_dbg(&aic26->spi->dev, "bad master\n"); return -EINVAL;- }
- /* interface format */
- switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S: aic26->datfm = AIC26_DATFM_I2S; break;case SND_SOC_DAIFMT_DSP_A: aic26->datfm = AIC26_DATFM_DSP; break;case SND_SOC_DAIFMT_RIGHT_J: aic26->datfm = AIC26_DATFM_RIGHTJ; break;case SND_SOC_DAIFMT_LEFT_J: aic26->datfm = AIC26_DATFM_LEFTJ; break;default: dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;- }
- return 0;
+}
+/* ---------------------------------------------------------------------
- Digital Audio Interface Definition
- */
+struct snd_soc_codec_dai aic26_dai = {
- .name = "tlv320aic26",
- .playback = {
.stream_name = "Playback",.channels_min = 2,.channels_max = 2,.rates = AIC26_RATES,.formats = AIC26_FORMATS,- },
- .capture = {
.stream_name = "Capture",.channels_min = 2,.channels_max = 2,.rates = AIC26_RATES,.formats = AIC26_FORMATS,- },
- .ops = {
.hw_params = aic26_hw_params,- },
- .dai_ops = {
.digital_mute = aic26_mute,.set_sysclk = aic26_set_sysclk,.set_fmt = aic26_set_fmt,- },
+}; +EXPORT_SYMBOL_GPL(aic26_dai);
+/* ---------------------------------------------------------------------
- ALSA controls
- */
+static const char *aic26_capture_src_text[] = {"Mic", "Aux"}; +static const struct soc_enum aic26_capture_src_enum =
- SOC_ENUM_SINGLE(AIC26_REG_AUDIO_CTRL1, 12,2, aic26_capture_src_text);
+static const struct snd_kcontrol_new aic26_snd_controls[] = {
- /* Output */
- SOC_DOUBLE("PCM Playback Volume", AIC26_REG_DAC_GAIN, 8, 0, 0x7f, 1),
- SOC_DOUBLE("PCM Playback Switch", AIC26_REG_DAC_GAIN, 15, 7, 1, 1),
- SOC_SINGLE("PCM Capture Volume", AIC26_REG_ADC_GAIN, 8, 0x7f, 0),
- SOC_SINGLE("PCM Capture Mute", AIC26_REG_ADC_GAIN, 15, 1, 1),
- SOC_ENUM("Capture Source", aic26_capture_src_enum),
+};
+/* ---------------------------------------------------------------------
- SoC CODEC portion of driver: probe and release routines
- */
+static int aic26_probe(struct platform_device *pdev) +{
- struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- struct snd_soc_codec *codec;
- struct snd_kcontrol *kcontrol;
- struct aic26 *aic26;
- int i, ret, err;
- dev_info(&pdev->dev, "Probing AIC26 SoC CODEC driver\n");
- dev_dbg(&pdev->dev, "socdev=%p\n", socdev);
- dev_dbg(&pdev->dev, "codec_data=%p\n", socdev->codec_data);
- /* Fetch the relevant aic26 private data here (it's already been
* stored in the .codec pointer) */- aic26 = socdev->codec_data;
- if (aic26 == NULL) {
dev_err(&pdev->dev, "aic26: missing codec pointer\n");return -ENODEV;- }
- codec = &aic26->codec;
- socdev->codec = codec;
- dev_dbg(&pdev->dev, "Registering PCMs, dev=%p, socdev->dev=%p\n",
&pdev->dev, socdev->dev);- /* register pcms */
- ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
- if (ret < 0) {
dev_err(&pdev->dev, "aic26: failed to create pcms\n");return -ENODEV;- }
- /* register controls */
- dev_dbg(&pdev->dev, "Registering controls\n");
- for (i = 0; i < ARRAY_SIZE(aic26_snd_controls); i++) {
kcontrol = snd_soc_cnew(&aic26_snd_controls[i], codec, NULL);err = snd_ctl_add(codec->card, kcontrol);WARN_ON(err < 0);- }
- /* CODEC is setup, we can register the card now */
- dev_dbg(&pdev->dev, "Registering card\n");
- ret = snd_soc_register_card(socdev);
- if (ret < 0) {
dev_err(&pdev->dev, "aic26: failed to register card\n");goto card_err;- }
- return 0;
- card_err:
- snd_soc_free_pcms(socdev);
- return ret;
+}
+static int aic26_remove(struct platform_device *pdev) +{
- struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- snd_soc_free_pcms(socdev);
- return 0;
+}
+struct snd_soc_codec_device aic26_soc_codec_dev = {
- .probe = aic26_probe,
- .remove = aic26_remove,
+};
+/* ---------------------------------------------------------------------
- SPI device portion of driver: sysfs files for debugging
- */
+static ssize_t aic26_regs_show(struct device *dev,
struct device_attribute *attr, char *buf)+{
- struct aic26 *aic26 = dev_get_drvdata(dev);
- char *idx = buf;
- int cache_flag, addr, page, i, reg;
- cache_flag = (strcmp(attr->attr.name, "regs_cache") == 0);
- for (page = 0; page < 3; page++) {
for (i = 0; i < 0x20; i++) {addr = AIC26_PAGE_ADDR(page, i);if (i % 8 == 0)idx += sprintf(idx, "%i:%.2i:", page,i);if (cache_flag)reg = aic26_reg_read_cache(&aic26->codec, addr);elsereg = aic26_reg_read(&aic26->codec, addr);idx += sprintf(idx, " %.4x", reg);if (i % 8 == 7)idx += sprintf(idx, "\n");}- }
- return idx - buf;
+}
The soc_core already has a codec reg dump sysfs file, so we don't need this.
+static ssize_t aic26_keyclick_show(struct device *dev,
struct device_attribute *attr, char *buf)+{
- struct aic26 *aic26 = dev_get_drvdata(dev);
- int val, amp, freq, len;
- val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
- amp = (val >> 12) & 0x7;
- freq = (125 << ((val >> 8) & 0x7)) >> 1;
- len = 2 * (1 +((val >> 8) & 0xf));
- return sprintf(buf, "amp=%x freq=%iHz len=%iclks\n", amp, freq, len);
+}
+/* Any write to the keyclick attribute will trigger the keyclick */ +static ssize_t aic26_keyclick_set(struct device *dev,
struct device_attribute *attr,const char *buf, size_t count)+{
- struct aic26 *aic26 = dev_get_drvdata(dev);
- int val;
- val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
- val |= 0x8000;
- aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL2, val);
- return count;
+}
+DEVICE_ATTR(regs, 0644, aic26_regs_show, NULL); +DEVICE_ATTR(regs_cache, 0644, aic26_regs_show, NULL); +DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set);
+/* ---------------------------------------------------------------------
- SPI device portion of driver: probe and release routines and SPI
driver registration.- */
+static int aic26_spi_probe(struct spi_device *spi) +{
- struct aic26 *aic26;
- int rc, i, reg;
- dev_dbg(&spi->dev, "probing tlv320aic26 spi device\n");
- /* Allocate driver data */
- aic26 = kzalloc(sizeof *aic26, GFP_KERNEL);
- if (!aic26)
return -ENOMEM;- /* Initialize the driver data */
- aic26->spi = spi;
- dev_set_drvdata(&spi->dev, aic26);
- /* Setup what we can in the codec structure so that the register
* access functions will work as expected. More will be filled* out when it is probed by the SoC CODEC part of this driver */- aic26->codec.private_data = aic26;
- aic26->codec.name = "aic26";
- aic26->codec.owner = THIS_MODULE;
- aic26->codec.dai = &aic26_dai;
- aic26->codec.num_dai = 1;
- aic26->codec.read = aic26_reg_read;
- aic26->codec.write = aic26_reg_write;
- aic26->master = 1;
- mutex_init(&aic26->codec.mutex);
- INIT_LIST_HEAD(&aic26->codec.dapm_widgets);
- INIT_LIST_HEAD(&aic26->codec.dapm_paths);
- aic26->codec.reg_cache_size = sizeof(aic26->reg_cache);
- aic26->codec.reg_cache = aic26->reg_cache;
- /* Reset the codec to power on defaults */
- aic26_reg_write(&aic26->codec, AIC26_REG_RESET, 0xBB00);
- /* Power up CODEC */
- aic26_reg_write(&aic26->codec, AIC26_REG_POWER_CTRL, 0);
- /* Audio Control 3 (master mode, fsref rate) */
- reg = aic26_reg_read(&aic26->codec, AIC26_REG_AUDIO_CTRL3);
- reg &= ~0xf800;
- reg |= 0x0800; /* set master mode */
- aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL3, reg);
- /* Fill page 2 register cache */
- for (i = 0; i < ARRAY_SIZE(aic26->reg_cache); i++)
aic26_reg_read(&aic26->codec, AIC26_PAGE_ADDR(2, i));- /* Register the sysfs files for debugging */
- /* Create SysFS files */
- rc = device_create_file(&spi->dev, &dev_attr_regs);
- rc |= device_create_file(&spi->dev, &dev_attr_regs_cache);
- rc |= device_create_file(&spi->dev, &dev_attr_keyclick);
- if (rc)
dev_info(&spi->dev, "error creating sysfs files\n");- /* Tell the of_soc helper about this codec */
- of_snd_soc_register_codec(&aic26_soc_codec_dev, aic26, &aic26_dai,
spi->dev.archdata.of_node);- dev_dbg(&spi->dev, "SPI device initialized\n");
- return 0;
+}
+static int aic26_spi_remove(struct spi_device *spi) +{
- struct aic26 *aic26 = dev_get_drvdata(&spi->dev);
- kfree(aic26);
- return 0;
+}
+static struct spi_driver aic26_spi = {
- .driver = {
.name = "tlv320aic26",.owner = THIS_MODULE,- },
- .probe = aic26_spi_probe,
- .remove = aic26_spi_remove,
+};
+static int __init aic26_init(void) +{
- return spi_register_driver(&aic26_spi);
+} +module_init(aic26_init);
+static void __exit aic26_exit(void) +{
- spi_unregister_driver(&aic26_spi);
+} +module_exit(aic26_exit);
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On Wed, Jul 02, 2008 at 11:48:33AM +0100, Liam Girdwood wrote:
On Tue, 2008-07-01 at 17:53 -0600, Grant Likely wrote:
From: Grant Likely grant.likely@secretlab.ca
ASoC Codec driver for the TLV320AIC26 device. This driver uses the ASoC v1 API, so I don't expect it to get merged as-is, but I want to get it out there for review.
Fwiw, I usually put all the codec registers defs in a separate header just in case we need to do any codec stuff in the machine driver.
okay, done
- switch (params_rate(params)) {
case 8000: fsref = 48000; divisor = AIC26_DIV_6; break;case 11025: fsref = 44100; divisor = AIC26_DIV_4; break;case 12000: fsref = 48000; divisor = AIC26_DIV_4; break;case 16000: fsref = 48000; divisor = AIC26_DIV_3; break;case 22050: fsref = 44100; divisor = AIC26_DIV_2; break;case 24000: fsref = 48000; divisor = AIC26_DIV_2; break;case 32000: fsref = 48000; divisor = AIC26_DIV_1_5; break;case 44100: fsref = 44100; divisor = AIC26_DIV_1; break;case 48000: fsref = 48000; divisor = AIC26_DIV_1; break;default: dev_dbg(&aic26->spi->dev, "bad rate\n"); return -EINVAL;- }
Indentation.
done.
- /* Configure PLL */
- pval = 1;
- jval = (fsref == 44100) ? 7 : 8;
- dval = (fsref == 44100) ? 5264 : 1920;
- qval = 0;
- reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;
- aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);
- reg = dval << 2;
- aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg);
PLL/FLL/clock config is usually done in a separate function (codec_set_pll(), callable by machine driver) so that we can change clocks depending on the available machine clocks and srate.
- /* Power up CODEC */
- aic26_reg_write(codec, AIC26_REG_POWER_CTRL, 0);
Codec domain (i.e Bias power) PM stuff should be done in codec_dapm_event(). This allows us to power the codec on when we do things like sidetone (with no active playback or capture stream).
Ugh, I'm going to have to leave these two for now. I don't understand enough about the ASoC structure yet to understand what it should look like. I'll probably need help, but I don't think I can get it sorted out before the merge window.
Do these two comments need to be addressed before the driver is merged?
+static ssize_t aic26_regs_show(struct device *dev,
struct device_attribute *attr, char *buf)+{
- struct aic26 *aic26 = dev_get_drvdata(dev);
- char *idx = buf;
- int cache_flag, addr, page, i, reg;
- cache_flag = (strcmp(attr->attr.name, "regs_cache") == 0);
- for (page = 0; page < 3; page++) {
for (i = 0; i < 0x20; i++) {addr = AIC26_PAGE_ADDR(page, i);if (i % 8 == 0)idx += sprintf(idx, "%i:%.2i:", page,i);if (cache_flag)reg = aic26_reg_read_cache(&aic26->codec, addr);elsereg = aic26_reg_read(&aic26->codec, addr);idx += sprintf(idx, " %.4x", reg);if (i % 8 == 7)idx += sprintf(idx, "\n");}- }
- return idx - buf;
+}
The soc_core already has a codec reg dump sysfs file, so we don't need this.
Hmmm, I haven't been able to find this; either in the code or on a live running system. Where is the common reg dump implemented.
Thanks for the comments. g.
On Sat, Jul 12, 2008 at 12:00:18AM -0600, Grant Likely wrote:
On Wed, Jul 02, 2008 at 11:48:33AM +0100, Liam Girdwood wrote:
PLL/FLL/clock config is usually done in a separate function (codec_set_pll(), callable by machine driver) so that we can change clocks depending on the available machine clocks and srate.
...
Codec domain (i.e Bias power) PM stuff should be done in codec_dapm_event(). This allows us to power the codec on when we do things like sidetone (with no active playback or capture stream).
Ugh, I'm going to have to leave these two for now. I don't understand enough about the ASoC structure yet to understand what it should look like. I'll probably need help, but I don't think I can get it sorted out before the merge window.
Do these two comments need to be addressed before the driver is merged?
It wouldn't be the only driver not to implement PLL configuration in this way so that's probably be OK for an initial merge. What's expected for PLL configuration is that you implement the DAI set_pll() operation in the codec driver, allowing machine drivers to configure the PLL when they wish.
The power configuration should be fixed, though. Normally drivers either fully implement DAPM (including set_bias_level()) or power everything in the codec up when the driver is loaded. At the minute what the driver is doing appears to be powering the codec up in both _hw_params() and _probe() but never powering anything down - if that is the case then probably all you need to do is remove the extra power up from hw_params(), giving you the simple option.
Hmmm, I haven't been able to find this; either in the code or on a live running system. Where is the common reg dump implemented.
/sys/bus/platform/devices/soc-audio/codec_reg
On Sat, Jul 12, 2008 at 11:36 AM, Mark Brown broonie@opensource.wolfsonmicro.com wrote:
The power configuration should be fixed, though. Normally drivers either fully implement DAPM (including set_bias_level()) or power everything in the codec up when the driver is loaded. At the minute what the driver is doing appears to be powering the codec up in both _hw_params() and _probe() but never powering anything down - if that is the case then probably all you need to do is remove the extra power up from hw_params(), giving you the simple option.
Okay, cool. I'll do this for the time being then.
Thanks, g.
On Sat, Jul 12, 2008 at 06:36:10PM +0100, Mark Brown wrote:
On Sat, Jul 12, 2008 at 12:00:18AM -0600, Grant Likely wrote:
It wouldn't be the only driver not to implement PLL configuration in this way so that's probably be OK for an initial merge. What's expected for PLL configuration is that you implement the DAI set_pll() operation in the codec driver, allowing machine drivers to configure the PLL when they wish.
okay
The power configuration should be fixed, though. Normally drivers either fully implement DAPM (including set_bias_level()) or power everything in the codec up when the driver is loaded. At the minute what the driver is doing appears to be powering the codec up in both _hw_params() and _probe() but never powering anything down - if that is the case then probably all you need to do is remove the extra power up from hw_params(), giving you the simple option.
done
Hmmm, I haven't been able to find this; either in the code or on a live running system. Where is the common reg dump implemented.
/sys/bus/platform/devices/soc-audio/codec_reg
Yikes. The AIC26 has registers all over the place and most of them are empty. The codec_reg attribute handling means I need to maintain a cache of the entire register file; not just the part that is actually used. Oh well; I can work around it.
Thanks, g.
On Thu, Jul 17, 2008 at 05:31:53PM -0600, Grant Likely wrote:
On Sat, Jul 12, 2008 at 06:36:10PM +0100, Mark Brown wrote:
/sys/bus/platform/devices/soc-audio/codec_reg
Yikes. The AIC26 has registers all over the place and most of them are empty. The codec_reg attribute handling means I need to maintain a cache of the entire register file; not just the part that is actually used. Oh well; I can work around it.
Sparse (or sparsely used) register maps are pretty common - in practice it's not been a problem to just have a cache that covers everything and only gets read, the amount of data involved is rarely that large in the context of what you need to store the audio you're working with.
On Fri, Jul 18, 2008 at 10:58:25AM +0100, Mark Brown wrote:
Sparse (or sparsely used) register maps are pretty common - in practice it's not been a problem to just have a cache that covers everything and only gets read, the amount of data involved is rarely that large in the context of what you need to store the audio you're working with.
okay.
g.
On 7/1/08, Grant Likely grant.likely@secretlab.ca wrote:
From: Grant Likely grant.likely@secretlab.ca
ASoC Codec driver for the TLV320AIC26 device. This driver uses the ASoC v1 API, so I don't expect it to get merged as-is, but I want to get it out there for review.
sound/soc/codecs/Kconfig | 4 sound/soc/codecs/Makefile | 2 sound/soc/codecs/tlv320aic26.c | 630 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 636 insertions(+), 0 deletions(-)
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3903ab7..96c7bfe 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -41,6 +41,10 @@ config SND_SOC_CS4270_VD33_ERRATA bool depends on SND_SOC_CS4270
+config SND_SOC_TLV320AIC26
tristate "TI TLB320AIC26 Codec support"depends on SND_SOC && SPIconfig SND_SOC_TLV320AIC3X tristate depends on SND_SOC && I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4e1314c..ec0cd93 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -5,6 +5,7 @@ snd-soc-wm8753-objs := wm8753.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o +snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o
obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o @@ -14,4 +15,5 @@ obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.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/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c new file mode 100644 index 0000000..aee1dbc --- /dev/null +++ b/sound/soc/codecs/tlv320aic26.c @@ -0,0 +1,630 @@ +/*
- Texas Instruments TLV320AIC26 low power audio CODEC
- ALSA SoC CODEC driver
- Copyright (C) 2008 Secret Lab Technologies Ltd.
- */
+#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/device.h> +#include <linux/sysfs.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/soc-of.h> +#include <sound/initval.h>
+MODULE_DESCRIPTION("ASoC TLV320AIC26 codec driver"); +MODULE_AUTHOR("Grant Likely grant.likely@secretlab.ca"); +MODULE_LICENSE("GPL");
+/* AIC26 Registers */ +#define AIC26_READ_COMMAND_WORD(addr) ((1 << 15) | (addr << 5)) +#define AIC26_WRITE_COMMAND_WORD(addr) ((0 << 15) | (addr << 5)) +#define AIC26_PAGE_ADDR(page, offset) ((page << 6) | offset) +#define AIC26_NUM_REGS AIC26_PAGE_ADDR(3, 0) +#define AIC26_REG_CACHE_SIZE (0x20) /* only page 2 cached */ +#define AIC26_REG_IS_CACHED(addr) ((addr & ~0x1f) == (2 << 6)) +#define AIC26_REG_CACHE_ADDR(addr) (addr & 0x1f)
+/* Page 0: Auxillary data registers */ +#define AIC26_REG_BAT1 AIC26_PAGE_ADDR(0, 0x05) +#define AIC26_REG_BAT2 AIC26_PAGE_ADDR(0, 0x06) +#define AIC26_REG_AUX AIC26_PAGE_ADDR(0, 0x07) +#define AIC26_REG_TEMP1 AIC26_PAGE_ADDR(0, 0x09) +#define AIC26_REG_TEMP2 AIC26_PAGE_ADDR(0, 0x0A)
+/* Page 1: Auxillary control registers */ +#define AIC26_REG_AUX_ADC AIC26_PAGE_ADDR(1, 0x00) +#define AIC26_REG_STATUS AIC26_PAGE_ADDR(1, 0x01) +#define AIC26_REG_REFERENCE AIC26_PAGE_ADDR(1, 0x03) +#define AIC26_REG_RESET AIC26_PAGE_ADDR(1, 0x04)
+/* Page 2: Audio control registers */ +#define AIC26_REG_AUDIO_CTRL1 AIC26_PAGE_ADDR(2, 0x00) +#define AIC26_REG_ADC_GAIN AIC26_PAGE_ADDR(2, 0x01) +#define AIC26_REG_DAC_GAIN AIC26_PAGE_ADDR(2, 0x02) +#define AIC26_REG_SIDETONE AIC26_PAGE_ADDR(2, 0x03) +#define AIC26_REG_AUDIO_CTRL2 AIC26_PAGE_ADDR(2, 0x04) +#define AIC26_REG_POWER_CTRL AIC26_PAGE_ADDR(2, 0x05) +#define AIC26_REG_AUDIO_CTRL3 AIC26_PAGE_ADDR(2, 0x06)
+#define AIC26_REG_FILTER_COEFF_L_N0 AIC26_PAGE_ADDR(2, 0x07) +#define AIC26_REG_FILTER_COEFF_L_N1 AIC26_PAGE_ADDR(2, 0x08) +#define AIC26_REG_FILTER_COEFF_L_N2 AIC26_PAGE_ADDR(2, 0x09) +#define AIC26_REG_FILTER_COEFF_L_N3 AIC26_PAGE_ADDR(2, 0x0A) +#define AIC26_REG_FILTER_COEFF_L_N4 AIC26_PAGE_ADDR(2, 0x0B) +#define AIC26_REG_FILTER_COEFF_L_N5 AIC26_PAGE_ADDR(2, 0x0C) +#define AIC26_REG_FILTER_COEFF_L_D1 AIC26_PAGE_ADDR(2, 0x0D) +#define AIC26_REG_FILTER_COEFF_L_D2 AIC26_PAGE_ADDR(2, 0x0E) +#define AIC26_REG_FILTER_COEFF_L_D4 AIC26_PAGE_ADDR(2, 0x0F) +#define AIC26_REG_FILTER_COEFF_L_D5 AIC26_PAGE_ADDR(2, 0x10) +#define AIC26_REG_FILTER_COEFF_R_N0 AIC26_PAGE_ADDR(2, 0x11) +#define AIC26_REG_FILTER_COEFF_R_N1 AIC26_PAGE_ADDR(2, 0x12) +#define AIC26_REG_FILTER_COEFF_R_N2 AIC26_PAGE_ADDR(2, 0x13) +#define AIC26_REG_FILTER_COEFF_R_N3 AIC26_PAGE_ADDR(2, 0x14) +#define AIC26_REG_FILTER_COEFF_R_N4 AIC26_PAGE_ADDR(2, 0x15) +#define AIC26_REG_FILTER_COEFF_R_N5 AIC26_PAGE_ADDR(2, 0x16) +#define AIC26_REG_FILTER_COEFF_R_D1 AIC26_PAGE_ADDR(2, 0x17) +#define AIC26_REG_FILTER_COEFF_R_D2 AIC26_PAGE_ADDR(2, 0x18) +#define AIC26_REG_FILTER_COEFF_R_D4 AIC26_PAGE_ADDR(2, 0x19) +#define AIC26_REG_FILTER_COEFF_R_D5 AIC26_PAGE_ADDR(2, 0x1A)
+#define AIC26_REG_PLL_PROG1 AIC26_PAGE_ADDR(2, 0x1B) +#define AIC26_REG_PLL_PROG2 AIC26_PAGE_ADDR(2, 0x1C) +#define AIC26_REG_AUDIO_CTRL4 AIC26_PAGE_ADDR(2, 0x1D) +#define AIC26_REG_AUDIO_CTRL5 AIC26_PAGE_ADDR(2, 0x1E)
+#define AIC26_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\SNDRV_PCM_RATE_48000)+#define AIC26_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |\
SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)+/* fsref dividers; used in register 'Audio Control 1' */ +enum aic26_divisors {
AIC26_DIV_1 = 0,AIC26_DIV_1_5 = 1,AIC26_DIV_2 = 2,AIC26_DIV_3 = 3,AIC26_DIV_4 = 4,AIC26_DIV_5 = 5,AIC26_DIV_5_5 = 6,AIC26_DIV_6 = 7,+};
+/* Digital data format */ +enum aic26_datfm {
AIC26_DATFM_I2S = 0 << 8,AIC26_DATFM_DSP = 1 << 8,AIC26_DATFM_RIGHTJ = 2 << 8, /* right justified */AIC26_DATFM_LEFTJ = 3 << 8, /* left justified */+};
+/* Sample word length in bits; used in register 'Audio Control 1' */ +enum aic26_wlen {
AIC26_WLEN_16 = 0 << 10,AIC26_WLEN_20 = 1 << 10,AIC26_WLEN_24 = 2 << 10,AIC26_WLEN_32 = 3 << 10,+};
+/* AIC26 driver private data */ +struct aic26 {
struct spi_device *spi;struct snd_soc_codec codec;u16 reg_cache[AIC26_REG_CACHE_SIZE]; /* shadow registers */int master;int datfm;int mclk;/* Keyclick parameters */int keyclick_amplitude;int keyclick_freq;int keyclick_len;+};
+/* ---------------------------------------------------------------------
- Register access routines
- */
+static unsigned int aic26_reg_read(struct snd_soc_codec *codec,
unsigned int reg)+{
struct aic26 *aic26 = codec->private_data;u16 *cache = codec->reg_cache;u16 cmd, value;u8 buffer[2];int rc;if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {WARN_ON_ONCE(1);return 0;}/* Do SPI transfer; first 16bits are command; remaining is* register contents */cmd = AIC26_READ_COMMAND_WORD(reg);buffer[0] = (cmd >> 8) & 0xff;buffer[1] = cmd & 0xff;rc = spi_write_then_read(aic26->spi, buffer, 2, buffer, 2);if (rc) {dev_err(&aic26->spi->dev, "AIC26 reg read error\n");return -EIO;}value = (buffer[0] << 8) | buffer[1];/* Update the cache before returning with the value */if (AIC26_REG_IS_CACHED(reg))cache[AIC26_REG_CACHE_ADDR(reg)] = value;return value;+}
+static unsigned int aic26_reg_read_cache(struct snd_soc_codec *codec,
unsigned int reg)+{
u16 *cache = codec->reg_cache;if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {WARN_ON_ONCE(1);return 0;}if (AIC26_REG_IS_CACHED(reg))return cache[AIC26_REG_CACHE_ADDR(reg)];return aic26_reg_read(codec, reg);+}
+static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)+{
struct aic26 *aic26 = codec->private_data;u16 *cache = codec->reg_cache;u16 cmd;u8 buffer[4];int rc;if ((reg < 0) || (reg >= AIC26_NUM_REGS)) {WARN_ON_ONCE(1);return -EINVAL;}/* Do SPI transfer; first 16bits are command; remaining is data* to write into register */cmd = AIC26_WRITE_COMMAND_WORD(reg);buffer[0] = (cmd >> 8) & 0xff;buffer[1] = cmd & 0xff;buffer[2] = value >> 8;buffer[3] = value;rc = spi_write(aic26->spi, buffer, 4);if (rc) {dev_err(&aic26->spi->dev, "AIC26 reg read error\n");return -EIO;}/* update cache before returning */if (AIC26_REG_IS_CACHED(reg))cache[AIC26_REG_CACHE_ADDR(reg)] = value;return 0;+}
+/* ---------------------------------------------------------------------
- Digital Audio Interface Operations
- */
+static int aic26_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)+{
struct snd_soc_pcm_runtime *rtd = substream->private_data;struct snd_soc_device *socdev = rtd->socdev;struct snd_soc_codec *codec = socdev->codec;struct aic26 *aic26 = codec->private_data;int fsref, divisor, wlen, pval, jval, dval, qval;u16 reg;dev_dbg(&aic26->spi->dev, "aic26_hw_params(substream=%p, params=%p)\n",substream, params);dev_dbg(&aic26->spi->dev, "rate=%i format=%i\n", params_rate(params),params_format(params));switch (params_rate(params)) {case 8000: fsref = 48000; divisor = AIC26_DIV_6; break;case 11025: fsref = 44100; divisor = AIC26_DIV_4; break;case 12000: fsref = 48000; divisor = AIC26_DIV_4; break;case 16000: fsref = 48000; divisor = AIC26_DIV_3; break;case 22050: fsref = 44100; divisor = AIC26_DIV_2; break;case 24000: fsref = 48000; divisor = AIC26_DIV_2; break;case 32000: fsref = 48000; divisor = AIC26_DIV_1_5; break;case 44100: fsref = 44100; divisor = AIC26_DIV_1; break;case 48000: fsref = 48000; divisor = AIC26_DIV_1; break;default: dev_dbg(&aic26->spi->dev, "bad rate\n"); return -EINVAL;}/* select data word length */switch (params_format(params)) {case SNDRV_PCM_FORMAT_S8: wlen = AIC26_WLEN_16; break;case SNDRV_PCM_FORMAT_S16_BE: wlen = AIC26_WLEN_16; break;case SNDRV_PCM_FORMAT_S24_BE: wlen = AIC26_WLEN_24; break;case SNDRV_PCM_FORMAT_S32_BE: wlen = AIC26_WLEN_32; break;default: dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;}/* Configure PLL */pval = 1;jval = (fsref == 44100) ? 7 : 8;dval = (fsref == 44100) ? 5264 : 1920;qval = 0;reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);reg = dval << 2;aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg);/* Power up CODEC */aic26_reg_write(codec, AIC26_REG_POWER_CTRL, 0);/* Audio Control 3 (master mode, fsref rate) */reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3);reg &= ~0xf800;if (aic26->master)reg |= 0x0800;if (fsref == 48000)reg |= 0x2000;aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg);/* Audio Control 1 (FSref divisor) */reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1);reg &= ~0x0fff;reg |= wlen | aic26->datfm | (divisor << 3) | divisor;aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL1, reg);return 0;+}
+/**
- aic26_mute - Mute control to reduce noise when changing audio format
- */
+static int aic26_mute(struct snd_soc_codec_dai *dai, int mute) +{
struct snd_soc_codec *codec = dai->codec;struct aic26 *aic26 = codec->private_data;u16 reg = aic26_reg_read_cache(codec, AIC26_REG_DAC_GAIN);dev_dbg(&aic26->spi->dev, "aic26_mute(dai=%p, mute=%i)\n",dai, mute);if (mute)reg |= 0x8080;elsereg &= ~0x8080;aic26_reg_write(codec, AIC26_REG_DAC_GAIN, reg);return 0;+}
+static int aic26_set_sysclk(struct snd_soc_codec_dai *codec_dai,
int clk_id, unsigned int freq, int dir)+{
struct snd_soc_codec *codec = codec_dai->codec;struct aic26 *aic26 = codec->private_data;dev_dbg(&aic26->spi->dev, "aic26_set_sysclk(dai=%p, clk_id==%i,"" freq=%i, dir=%i)\n",codec_dai, clk_id, freq, dir);/* MCLK needs to fall between 2MHz and 50 MHz */if ((freq < 2000000) || (freq > 50000000))return -EINVAL;aic26->mclk = freq;return 0;+}
+static int aic26_set_fmt(struct snd_soc_codec_dai *codec_dai, unsigned int fmt) +{
struct snd_soc_codec *codec = codec_dai->codec;struct aic26 *aic26 = codec->private_data;dev_dbg(&aic26->spi->dev, "aic26_set_fmt(dai=%p, fmt==%i)\n",codec_dai, fmt);/* set master/slave audio interface */switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {case SND_SOC_DAIFMT_CBM_CFM: aic26->master = 1; break;//case SND_SOC_DAIFMT_CBS_CFS: aic26->master = 0; break;default: dev_dbg(&aic26->spi->dev, "bad master\n"); return -EINVAL;}/* interface format */switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {case SND_SOC_DAIFMT_I2S: aic26->datfm = AIC26_DATFM_I2S; break;case SND_SOC_DAIFMT_DSP_A: aic26->datfm = AIC26_DATFM_DSP; break;case SND_SOC_DAIFMT_RIGHT_J: aic26->datfm = AIC26_DATFM_RIGHTJ; break;case SND_SOC_DAIFMT_LEFT_J: aic26->datfm = AIC26_DATFM_LEFTJ; break;default: dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;}return 0;+}
+/* ---------------------------------------------------------------------
- Digital Audio Interface Definition
- */
+struct snd_soc_codec_dai aic26_dai = {
.name = "tlv320aic26",.playback = {.stream_name = "Playback",.channels_min = 2,.channels_max = 2,.rates = AIC26_RATES,.formats = AIC26_FORMATS,},.capture = {.stream_name = "Capture",.channels_min = 2,.channels_max = 2,.rates = AIC26_RATES,.formats = AIC26_FORMATS,},.ops = {.hw_params = aic26_hw_params,},.dai_ops = {.digital_mute = aic26_mute,.set_sysclk = aic26_set_sysclk,.set_fmt = aic26_set_fmt,},+}; +EXPORT_SYMBOL_GPL(aic26_dai);
+/* ---------------------------------------------------------------------
- ALSA controls
- */
+static const char *aic26_capture_src_text[] = {"Mic", "Aux"}; +static const struct soc_enum aic26_capture_src_enum =
SOC_ENUM_SINGLE(AIC26_REG_AUDIO_CTRL1, 12,2, aic26_capture_src_text);+static const struct snd_kcontrol_new aic26_snd_controls[] = {
/* Output */SOC_DOUBLE("PCM Playback Volume", AIC26_REG_DAC_GAIN, 8, 0, 0x7f, 1),SOC_DOUBLE("PCM Playback Switch", AIC26_REG_DAC_GAIN, 15, 7, 1, 1),SOC_SINGLE("PCM Capture Volume", AIC26_REG_ADC_GAIN, 8, 0x7f, 0),SOC_SINGLE("PCM Capture Mute", AIC26_REG_ADC_GAIN, 15, 1, 1),SOC_ENUM("Capture Source", aic26_capture_src_enum),+};
+/* ---------------------------------------------------------------------
- SoC CODEC portion of driver: probe and release routines
- */
+static int aic26_probe(struct platform_device *pdev) +{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);struct snd_soc_codec *codec;struct snd_kcontrol *kcontrol;struct aic26 *aic26;int i, ret, err;dev_info(&pdev->dev, "Probing AIC26 SoC CODEC driver\n");dev_dbg(&pdev->dev, "socdev=%p\n", socdev);dev_dbg(&pdev->dev, "codec_data=%p\n", socdev->codec_data);/* Fetch the relevant aic26 private data here (it's already been* stored in the .codec pointer) */aic26 = socdev->codec_data;if (aic26 == NULL) {dev_err(&pdev->dev, "aic26: missing codec pointer\n");return -ENODEV;}codec = &aic26->codec;socdev->codec = codec;dev_dbg(&pdev->dev, "Registering PCMs, dev=%p, socdev->dev=%p\n",&pdev->dev, socdev->dev);/* register pcms */ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);if (ret < 0) {dev_err(&pdev->dev, "aic26: failed to create pcms\n");return -ENODEV;}/* register controls */dev_dbg(&pdev->dev, "Registering controls\n");for (i = 0; i < ARRAY_SIZE(aic26_snd_controls); i++) {kcontrol = snd_soc_cnew(&aic26_snd_controls[i], codec, NULL);err = snd_ctl_add(codec->card, kcontrol);WARN_ON(err < 0);}/* CODEC is setup, we can register the card now */dev_dbg(&pdev->dev, "Registering card\n");ret = snd_soc_register_card(socdev);if (ret < 0) {dev_err(&pdev->dev, "aic26: failed to register card\n");goto card_err;}return 0;- card_err:
snd_soc_free_pcms(socdev);return ret;+}
+static int aic26_remove(struct platform_device *pdev) +{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);snd_soc_free_pcms(socdev);return 0;+}
+struct snd_soc_codec_device aic26_soc_codec_dev = {
.probe = aic26_probe,.remove = aic26_remove,+};
+/* ---------------------------------------------------------------------
- SPI device portion of driver: sysfs files for debugging
- */
+static ssize_t aic26_regs_show(struct device *dev,
struct device_attribute *attr, char *buf)+{
struct aic26 *aic26 = dev_get_drvdata(dev);char *idx = buf;int cache_flag, addr, page, i, reg;cache_flag = (strcmp(attr->attr.name, "regs_cache") == 0);for (page = 0; page < 3; page++) {for (i = 0; i < 0x20; i++) {addr = AIC26_PAGE_ADDR(page, i);if (i % 8 == 0)idx += sprintf(idx, "%i:%.2i:", page,i);if (cache_flag)reg = aic26_reg_read_cache(&aic26->codec, addr);elsereg = aic26_reg_read(&aic26->codec, addr);idx += sprintf(idx, " %.4x", reg);if (i % 8 == 7)idx += sprintf(idx, "\n");}}return idx - buf;+}
+static ssize_t aic26_keyclick_show(struct device *dev,
struct device_attribute *attr, char *buf)+{
struct aic26 *aic26 = dev_get_drvdata(dev);int val, amp, freq, len;val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);amp = (val >> 12) & 0x7;freq = (125 << ((val >> 8) & 0x7)) >> 1;len = 2 * (1 +((val >> 8) & 0xf));return sprintf(buf, "amp=%x freq=%iHz len=%iclks\n", amp, freq, len);+}
+/* Any write to the keyclick attribute will trigger the keyclick */ +static ssize_t aic26_keyclick_set(struct device *dev,
struct device_attribute *attr,const char *buf, size_t count)+{
struct aic26 *aic26 = dev_get_drvdata(dev);int val;val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);val |= 0x8000;aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL2, val);return count;+}
+DEVICE_ATTR(regs, 0644, aic26_regs_show, NULL); +DEVICE_ATTR(regs_cache, 0644, aic26_regs_show, NULL); +DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set);
+/* ---------------------------------------------------------------------
- SPI device portion of driver: probe and release routines and SPI
driver registration.- */
+static int aic26_spi_probe(struct spi_device *spi) +{
struct aic26 *aic26;int rc, i, reg;dev_dbg(&spi->dev, "probing tlv320aic26 spi device\n");/* Allocate driver data */aic26 = kzalloc(sizeof *aic26, GFP_KERNEL);if (!aic26)return -ENOMEM;/* Initialize the driver data */aic26->spi = spi;dev_set_drvdata(&spi->dev, aic26);/* Setup what we can in the codec structure so that the register* access functions will work as expected. More will be filled* out when it is probed by the SoC CODEC part of this driver */aic26->codec.private_data = aic26;aic26->codec.name = "aic26";aic26->codec.owner = THIS_MODULE;aic26->codec.dai = &aic26_dai;aic26->codec.num_dai = 1;aic26->codec.read = aic26_reg_read;aic26->codec.write = aic26_reg_write;aic26->master = 1;mutex_init(&aic26->codec.mutex);INIT_LIST_HEAD(&aic26->codec.dapm_widgets);INIT_LIST_HEAD(&aic26->codec.dapm_paths);aic26->codec.reg_cache_size = sizeof(aic26->reg_cache);aic26->codec.reg_cache = aic26->reg_cache;/* Reset the codec to power on defaults */aic26_reg_write(&aic26->codec, AIC26_REG_RESET, 0xBB00);/* Power up CODEC */aic26_reg_write(&aic26->codec, AIC26_REG_POWER_CTRL, 0);/* Audio Control 3 (master mode, fsref rate) */reg = aic26_reg_read(&aic26->codec, AIC26_REG_AUDIO_CTRL3);reg &= ~0xf800;reg |= 0x0800; /* set master mode */aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL3, reg);/* Fill page 2 register cache */for (i = 0; i < ARRAY_SIZE(aic26->reg_cache); i++)aic26_reg_read(&aic26->codec, AIC26_PAGE_ADDR(2, i));/* Register the sysfs files for debugging *//* Create SysFS files */rc = device_create_file(&spi->dev, &dev_attr_regs);rc |= device_create_file(&spi->dev, &dev_attr_regs_cache);rc |= device_create_file(&spi->dev, &dev_attr_keyclick);if (rc)dev_info(&spi->dev, "error creating sysfs files\n");/* Tell the of_soc helper about this codec */of_snd_soc_register_codec(&aic26_soc_codec_dev, aic26, &aic26_dai,spi->dev.archdata.of_node);
I've been using asoc-v2 so it may not be possible to make this arch independent in asoc-v1. Maybe try porting this to asoc-v2 and see if a bunch of the registration complexity disappears. Most of soc-of.c should be unnecessary.
Liam, when is asoc-v2 going into mainline?
dev_dbg(&spi->dev, "SPI device initialized\n");return 0;+}
+static int aic26_spi_remove(struct spi_device *spi) +{
struct aic26 *aic26 = dev_get_drvdata(&spi->dev);kfree(aic26);return 0;+}
+static struct spi_driver aic26_spi = {
.driver = {.name = "tlv320aic26",.owner = THIS_MODULE,},.probe = aic26_spi_probe,.remove = aic26_spi_remove,+};
+static int __init aic26_init(void) +{
return spi_register_driver(&aic26_spi);+} +module_init(aic26_init);
+static void __exit aic26_exit(void) +{
spi_unregister_driver(&aic26_spi);+} +module_exit(aic26_exit);
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
On Wed, 2008-07-02 at 09:52 -0400, Jon Smirl wrote:
I've been using asoc-v2 so it may not be possible to make this arch independent in asoc-v1. Maybe try porting this to asoc-v2 and see if a bunch of the registration complexity disappears. Most of soc-of.c should be unnecessary.
Liam, when is asoc-v2 going into mainline?
Hopefully over the next 2 merge windows. I'm about to start queueing patches....
This also assumes Mark and I remain at our current work load levels.
Liam
On Tue, Jul 01, 2008 at 05:53:40PM -0600, Grant Likely wrote:
+static ssize_t aic26_keyclick_show(struct device *dev,
struct device_attribute *attr, char *buf)+{
- struct aic26 *aic26 = dev_get_drvdata(dev);
- int val, amp, freq, len;
- val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
- amp = (val >> 12) & 0x7;
- freq = (125 << ((val >> 8) & 0x7)) >> 1;
- len = 2 * (1 +((val >> 8) & 0xf));
- return sprintf(buf, "amp=%x freq=%iHz len=%iclks\n", amp, freq, len);
+}
It might be nice if these parameters were exposed as controls if they can be configured, though obviously that's not essential.
- /* Tell the of_soc helper about this codec */
- of_snd_soc_register_codec(&aic26_soc_codec_dev, aic26, &aic26_dai,
spi->dev.archdata.of_node);
Does this need to be ifdefed out if of_snd_soc_register_codec() is not in use or is the function stubbed out if it's not in use? Stubbing it out would be cleaner.
On Fri, Jul 4, 2008 at 2:49 PM, Mark Brown broonie@opensource.wolfsonmicro.com wrote:
On Tue, Jul 01, 2008 at 05:53:40PM -0600, Grant Likely wrote:
/* Tell the of_soc helper about this codec */of_snd_soc_register_codec(&aic26_soc_codec_dev, aic26, &aic26_dai,spi->dev.archdata.of_node);Does this need to be ifdefed out if of_snd_soc_register_codec() is not in use or is the function stubbed out if it's not in use? Stubbing it out would be cleaner.
yeah, probably. This is a stop-gap measure anyway until ASoC v2 is merged.
g.
At Tue, 01 Jul 2008 17:53:30 -0600, Grant Likely wrote:
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 18f28ac..c5736e5 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -23,6 +23,12 @@ config SND_SOC This ASoC audio support can also be built as a module. If so, the module will be called snd-soc-core.
+config SND_SOC_OF
- tristate "OF helpers for SoC audio support"
- depends on SND_SOC
- ---help---
Add support for OpenFirmware device tree descriptions of sound device
This is a helper module and not necessarily manually selectable. Better to make the other driver selecting this.
diff --git a/sound/soc/soc-of.c b/sound/soc/soc-of.c new file mode 100644 index 0000000..9694979 --- /dev/null +++ b/sound/soc/soc-of.c
(snip)
+DEFINE_MUTEX(of_snd_soc_mutex); +LIST_HEAD(of_snd_soc_device_list);
Missing static.
Takashi
On Wed, Jul 02, 2008 at 11:50:43AM +0200, Takashi Iwai wrote:
At Tue, 01 Jul 2008 17:53:30 -0600, Grant Likely wrote:
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 18f28ac..c5736e5 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -23,6 +23,12 @@ config SND_SOC This ASoC audio support can also be built as a module. If so, the module will be called snd-soc-core.
+config SND_SOC_OF
- tristate "OF helpers for SoC audio support"
- depends on SND_SOC
- ---help---
Add support for OpenFirmware device tree descriptions of sound deviceThis is a helper module and not necessarily manually selectable. Better to make the other driver selecting this.
Yes, you're right. I hadn't put too much thought into the Kconfig stuff for this patch because, at the moment, I'm viewing this as a temporary solution until ASoCv2 is merged.
Which raises another question: Liam and Mark, what is your opinion on merging this driver? Is it something that you would merge with the v1 API and then rework it when v2 is merged? Or would you rather it be kept out until v2 is ready? I haven't even looked at the v2 API yet, so I don't know how much rework is involved.
Cheers, g.
On Wed, 2008-07-02 at 09:48 -0600, Grant Likely wrote:
On Wed, Jul 02, 2008 at 11:50:43AM +0200, Takashi Iwai wrote:
This is a helper module and not necessarily manually selectable. Better to make the other driver selecting this.
Yes, you're right. I hadn't put too much thought into the Kconfig stuff for this patch because, at the moment, I'm viewing this as a temporary solution until ASoCv2 is merged.
Which raises another question: Liam and Mark, what is your opinion on merging this driver? Is it something that you would merge with the v1 API and then rework it when v2 is merged? Or would you rather it be kept out until v2 is ready? I haven't even looked at the v2 API yet, so I don't know how much rework is involved.
I'm happy with this atm for v1. We can then rework for V2.
Liam
On 7/1/08, Grant Likely grant.likely@secretlab.ca wrote:
From: Grant Likely grant.likely@secretlab.ca
Simple utility layer for creating ASoC machine instances based on data in the OpenFirmware device tree. OF aware platform drivers and codec drivers register themselves with this framework and the framework automatically instantiates a machine driver.
This is most likely temporary glue code to work around limitations in the ASoC v1 framework. I expect ASoC v2 won't need this.
sound/soc/Kconfig | 6 ++ sound/soc/Makefile | 1 sound/soc/soc-of.c | 171 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 178 insertions(+), 0 deletions(-)
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 18f28ac..c5736e5 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -23,6 +23,12 @@ config SND_SOC This ASoC audio support can also be built as a module. If so, the module will be called snd-soc-core.
+config SND_SOC_OF
tristate "OF helpers for SoC audio support"depends on SND_SOC---help---Add support for OpenFirmware device tree descriptions of sound device# All the supported Soc's source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 782db21..191c2e5 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -2,3 +2,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o
obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/ +obj-$(CONFIG_SND_SOC_OF) += soc-of.o diff --git a/sound/soc/soc-of.c b/sound/soc/soc-of.c new file mode 100644 index 0000000..9694979 --- /dev/null +++ b/sound/soc/soc-of.c @@ -0,0 +1,171 @@ +/*
- OF helpers for ALSA SoC Layer
- Copyright (C) 2008, Secret Lab Technologies Ltd.
- */
+#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/bitops.h> +#include <linux/platform_device.h> +#include <linux/of.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-of.h> +#include <sound/initval.h>
+MODULE_AUTHOR("Grant Likely grant.likely@secretlab.ca"); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("ALSA SoC OpenFirmware bindings");
+DEFINE_MUTEX(of_snd_soc_mutex); +LIST_HEAD(of_snd_soc_device_list); +static int of_snd_soc_next_index;
+struct of_snd_soc_device {
int id;struct list_head list;struct snd_soc_device device;struct snd_soc_machine machine;struct snd_soc_dai_link dai_link;struct platform_device *pdev;struct device_node *platform_node;struct device_node *codec_node;+};
+static struct snd_soc_ops of_snd_soc_ops = { +};
+static struct of_snd_soc_device * +of_snd_soc_get_device(struct device_node *codec_node) +{
struct of_snd_soc_device *of_soc;list_for_each_entry(of_soc, &of_snd_soc_device_list, list) {if (of_soc->codec_node == codec_node)return of_soc;}of_soc = kzalloc(sizeof(struct of_snd_soc_device), GFP_KERNEL);if (!of_soc)return NULL;/* 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->machine.dai_link = &of_soc->dai_link;of_soc->machine.num_links = 1;of_soc->device.machine = &of_soc->machine;of_soc->dai_link.ops = &of_snd_soc_ops;list_add(&of_soc->list, &of_snd_soc_device_list);return of_soc;+}
+static void of_snd_soc_register_device(struct of_snd_soc_device *of_soc) +{
struct platform_device *pdev;int rc;/* Only register the device if both the codec and platform have* been registered */if ((!of_soc->device.codec_data) || (!of_soc->platform_node))return;pr_info("platform<-->codec match achieved; registering machine\n");pdev = platform_device_alloc("soc-audio", of_soc->id);if (!pdev) {pr_err("of_soc: platform_device_alloc() failed\n");return;}pdev->dev.platform_data = of_soc;platform_set_drvdata(pdev, &of_soc->device);of_soc->device.dev = &pdev->dev;/* The ASoC device is complete; register it */rc = platform_device_add(pdev);if (rc) {pr_err("of_soc: platform_device_add() failed\n");return;}
Is there a driver for this device? Is another file missing? sof_of.h is missing too.
+}
+int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev,
void *codec_data, struct snd_soc_codec_dai *dai,struct device_node *node)+{
struct of_snd_soc_device *of_soc;int rc = 0;pr_info("registering ASoC codec driver: %s\n", node->full_name);mutex_lock(&of_snd_soc_mutex);of_soc = of_snd_soc_get_device(node);if (!of_soc) {rc = -ENOMEM;goto out;}/* Store the codec data */of_soc->device.codec_data = codec_data;of_soc->device.codec_dev = codec_dev;of_soc->dai_link.name = node->name;of_soc->dai_link.stream_name = node->name;of_soc->dai_link.codec_dai = dai;/* Now try to register the SoC device */of_snd_soc_register_device(of_soc);- out:
mutex_unlock(&of_snd_soc_mutex);return rc;+} +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_cpu_dai *cpu_dai)+{
struct of_snd_soc_device *of_soc;struct device_node *codec_node;const phandle *handle;int len, rc = 0;pr_info("registering ASoC platform 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 (!codec_node)return -ENODEV;pr_info("looking for codec: %s\n", codec_node->full_name);mutex_lock(&of_snd_soc_mutex);of_soc = of_snd_soc_get_device(codec_node);if (!of_soc) {rc = -ENOMEM;goto out;}of_soc->platform_node = node;of_soc->dai_link.cpu_dai = cpu_dai;of_soc->device.platform = platform;of_soc->machine.name = of_soc->dai_link.cpu_dai->name;/* Now try to register the SoC device */of_snd_soc_register_device(of_soc);- out:
mutex_unlock(&of_snd_soc_mutex);return rc;+} +EXPORT_SYMBOL_GPL(of_snd_soc_register_platform);
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
On 7/1/08, Grant Likely grant.likely@secretlab.ca wrote:
From: Grant Likely grant.likely@secretlab.ca
Simple utility layer for creating ASoC machine instances based on data in the OpenFirmware device tree. OF aware platform drivers and codec drivers register themselves with this framework and the framework automatically instantiates a machine driver.
This is most likely temporary glue code to work around limitations in the ASoC v1 framework. I expect ASoC v2 won't need this.
sound/soc/Kconfig | 6 ++ sound/soc/Makefile | 1 sound/soc/soc-of.c | 171 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 178 insertions(+), 0 deletions(-)
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 18f28ac..c5736e5 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -23,6 +23,12 @@ config SND_SOC This ASoC audio support can also be built as a module. If so, the module will be called snd-soc-core.
+config SND_SOC_OF
tristate "OF helpers for SoC audio support"depends on SND_SOC---help---Add support for OpenFirmware device tree descriptions of sound device# All the supported Soc's source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 782db21..191c2e5 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -2,3 +2,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o
obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/ +obj-$(CONFIG_SND_SOC_OF) += soc-of.o diff --git a/sound/soc/soc-of.c b/sound/soc/soc-of.c new file mode 100644 index 0000000..9694979 --- /dev/null +++ b/sound/soc/soc-of.c @@ -0,0 +1,171 @@ +/*
- OF helpers for ALSA SoC Layer
- Copyright (C) 2008, Secret Lab Technologies Ltd.
- */
+#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/bitops.h> +#include <linux/platform_device.h> +#include <linux/of.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-of.h> +#include <sound/initval.h>
+MODULE_AUTHOR("Grant Likely grant.likely@secretlab.ca"); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("ALSA SoC OpenFirmware bindings");
+DEFINE_MUTEX(of_snd_soc_mutex); +LIST_HEAD(of_snd_soc_device_list); +static int of_snd_soc_next_index;
+struct of_snd_soc_device {
int id;struct list_head list;struct snd_soc_device device;struct snd_soc_machine machine;struct snd_soc_dai_link dai_link;struct platform_device *pdev;struct device_node *platform_node;struct device_node *codec_node;+};
+static struct snd_soc_ops of_snd_soc_ops = { +};
+static struct of_snd_soc_device * +of_snd_soc_get_device(struct device_node *codec_node) +{
struct of_snd_soc_device *of_soc;list_for_each_entry(of_soc, &of_snd_soc_device_list, list) {if (of_soc->codec_node == codec_node)return of_soc;}of_soc = kzalloc(sizeof(struct of_snd_soc_device), GFP_KERNEL);if (!of_soc)return NULL;/* 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->machine.dai_link = &of_soc->dai_link;of_soc->machine.num_links = 1;of_soc->device.machine = &of_soc->machine;of_soc->dai_link.ops = &of_snd_soc_ops;list_add(&of_soc->list, &of_snd_soc_device_list);return of_soc;+}
Isn't this performing the same basic function as this code (except for spi)? Should this list be maintained in alsa or should there be an equivalent for searching the spi bus? If you follow the link to the codec node and get it's parent you know which bus to search. But it might be simpler to just search the buses sequentially for the node.
static int of_dev_node_match(struct device *dev, void *data) { return dev->archdata.of_node == data; }
struct i2c_client *of_find_i2c_device_by_node(struct device_node *node) { struct device *dev; dev = bus_find_device(&i2c_bus_type, NULL, node, of_dev_node_match); if (!dev) return NULL; return to_i2c_client(dev); } EXPORT_SYMBOL(of_find_i2c_device_by_node);
+static void of_snd_soc_register_device(struct of_snd_soc_device *of_soc) +{
struct platform_device *pdev;int rc;/* Only register the device if both the codec and platform have* been registered */if ((!of_soc->device.codec_data) || (!of_soc->platform_node))return;pr_info("platform<-->codec match achieved; registering machine\n");pdev = platform_device_alloc("soc-audio", of_soc->id);if (!pdev) {pr_err("of_soc: platform_device_alloc() failed\n");return;}pdev->dev.platform_data = of_soc;platform_set_drvdata(pdev, &of_soc->device);of_soc->device.dev = &pdev->dev;/* The ASoC device is complete; register it */rc = platform_device_add(pdev);if (rc) {pr_err("of_soc: platform_device_add() failed\n");return;}+}
+int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev,
void *codec_data, struct snd_soc_codec_dai *dai,struct device_node *node)+{
struct of_snd_soc_device *of_soc;int rc = 0;pr_info("registering ASoC codec driver: %s\n", node->full_name);mutex_lock(&of_snd_soc_mutex);of_soc = of_snd_soc_get_device(node);if (!of_soc) {rc = -ENOMEM;goto out;}/* Store the codec data */of_soc->device.codec_data = codec_data;of_soc->device.codec_dev = codec_dev;of_soc->dai_link.name = node->name;of_soc->dai_link.stream_name = node->name;of_soc->dai_link.codec_dai = dai;/* Now try to register the SoC device */of_snd_soc_register_device(of_soc);- out:
mutex_unlock(&of_snd_soc_mutex);return rc;+} +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_cpu_dai *cpu_dai)+{
struct of_snd_soc_device *of_soc;struct device_node *codec_node;const phandle *handle;int len, rc = 0;pr_info("registering ASoC platform 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 (!codec_node)return -ENODEV;pr_info("looking for codec: %s\n", codec_node->full_name);mutex_lock(&of_snd_soc_mutex);of_soc = of_snd_soc_get_device(codec_node);if (!of_soc) {rc = -ENOMEM;goto out;}of_soc->platform_node = node;of_soc->dai_link.cpu_dai = cpu_dai;of_soc->device.platform = platform;of_soc->machine.name = of_soc->dai_link.cpu_dai->name;/* Now try to register the SoC device */of_snd_soc_register_device(of_soc);- out:
mutex_unlock(&of_snd_soc_mutex);return rc;+} +EXPORT_SYMBOL_GPL(of_snd_soc_register_platform);
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
On Wed, Jul 02, 2008 at 11:27:17AM -0400, Jon Smirl wrote:
On 7/1/08, Grant Likely grant.likely@secretlab.ca wrote:
+static struct of_snd_soc_device * +of_snd_soc_get_device(struct device_node *codec_node) +{
struct of_snd_soc_device *of_soc;list_for_each_entry(of_soc, &of_snd_soc_device_list, list) {if (of_soc->codec_node == codec_node)return of_soc;}of_soc = kzalloc(sizeof(struct of_snd_soc_device), GFP_KERNEL);if (!of_soc)return NULL;/* 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->machine.dai_link = &of_soc->dai_link;of_soc->machine.num_links = 1;of_soc->device.machine = &of_soc->machine;of_soc->dai_link.ops = &of_snd_soc_ops;list_add(&of_soc->list, &of_snd_soc_device_list);return of_soc;+}
Isn't this performing the same basic function as this code (except for spi)? Should this list be maintained in alsa or should there be an equivalent for searching the spi bus? If you follow the link to the codec node and get it's parent you know which bus to search. But it might be simpler to just search the buses sequentially for the node.
static int of_dev_node_match(struct device *dev, void *data) { return dev->archdata.of_node == data; }
struct i2c_client *of_find_i2c_device_by_node(struct device_node *node) { struct device *dev;
dev = bus_find_device(&i2c_bus_type, NULL, node, of_dev_node_match); if (!dev) return NULL; return to_i2c_client(dev); } EXPORT_SYMBOL(of_find_i2c_device_by_node);
Yes, but you hadn't written these functions when I wrote this helper. :-P. I believe ASoC v2 makes all this stuff unnecessary, but I'm waiting for v2 to hit mainline before I port forward.
g.
On Jul 3, 2008, at 12:33 PM, Grant Likely wrote:
Yes, but you hadn't written these functions when I wrote this helper. :-P. I believe ASoC v2 makes all this stuff unnecessary, but I'm waiting for v2 to hit mainline before I port forward.
FYI, it took me about a month to port my drivers to ASoC V2. I had more work than most people because I addressed issues that others drivers didn't. For instance, my CS4270 driver is the only codec driver in V2 that supports multiple instances of itself.
participants (8)
-
David Gibson -
Grant Likely -
Jon Smirl -
Liam Girdwood -
Liam Girdwood -
Mark Brown -
Takashi Iwai -
Timur Tabi