[alsa-devel] [PATCH v3 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. At the moment, the driver is not very capable and it is expected to be extended as more features are needed for specifying the configuration in the device tree.
This is most likely temporary glue code to work around limitations in the ASoC v1 framework. When v2 is merged, most of this driver will need to be reworked.
Signed-off-by: Grant Likely grant.likely@secretlab.ca ---
include/sound/soc-of-simple.h | 21 +++++ sound/soc/fsl/Kconfig | 3 + sound/soc/fsl/Makefile | 3 + sound/soc/fsl/soc-of-simple.c | 171 +++++++++++++++++++++++++++++++++++++++++ 4 files changed, 198 insertions(+), 0 deletions(-)
diff --git a/include/sound/soc-of-simple.h b/include/sound/soc-of-simple.h new file mode 100644 index 0000000..696fc51 --- /dev/null +++ b/include/sound/soc-of-simple.h @@ -0,0 +1,21 @@ +/* + * OF helpers for ALSA SoC + * + * Copyright (C) 2008, Secret Lab Technologies Ltd. + */ + +#ifndef _INCLUDE_SOC_OF_H_ +#define _INCLUDE_SOC_OF_H_ + +#include <linux/of.h> +#include <sound/soc.h> + +int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev, + void *codec_data, struct snd_soc_dai *dai, + struct device_node *node); + +int of_snd_soc_register_platform(struct snd_soc_platform *platform, + struct device_node *node, + struct snd_soc_dai *cpu_dai); + +#endif /* _INCLUDE_SOC_OF_H_ */ diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3368ace..398f002 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,3 +1,6 @@ +config SND_SOC_OF_SIMPLE + tristate + config SND_SOC_MPC8610 bool "ALSA SoC support for the MPC8610 SOC" depends on MPC8610_HPCD diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 62f680a..aa2100b 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -1,3 +1,6 @@ +# Simple machine driver that extracts configuration from the OF device tree +obj-$(CONFIG_SND_SOC_OF_SIMPLE) += soc-of-simple.o + # MPC8610 HPCD Machine Support obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o
diff --git a/sound/soc/fsl/soc-of-simple.c b/sound/soc/fsl/soc-of-simple.c new file mode 100644 index 0000000..0382fda --- /dev/null +++ b/sound/soc/fsl/soc-of-simple.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-simple.h> +#include <sound/initval.h> + +MODULE_AUTHOR("Grant Likely grant.likely@secretlab.ca"); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("ALSA SoC OpenFirmware bindings"); + +static DEFINE_MUTEX(of_snd_soc_mutex); +static 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_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 = (char *)node->name; + of_soc->dai_link.stream_name = (char *)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_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 depends on the soc-of helper functions to match a PSC device with a codec based on data in the device tree.
Signed-off-by: Grant Likely grant.likely@secretlab.ca ---
sound/soc/fsl/Kconfig | 7 sound/soc/fsl/Makefile | 2 sound/soc/fsl/mpc5200_psc_i2s.c | 884 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 893 insertions(+), 0 deletions(-)
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 398f002..bba9546 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -17,3 +17,10 @@ config SND_SOC_MPC8610_HPCD default y if MPC8610_HPCD help Say Y if you want to enable audio on the Freescale MPC8610 HPCD. + +config SND_SOC_MPC5200_I2S + tristate "Freescale MPC5200 PSC in I2S mode driver" + select SND_SOC_OF_SIMPLE + depends on SND_SOC && PPC_MPC52xx + help + Say Y here to support the MPC5200 PSCs in I2S mode. diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index aa2100b..035da4a 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -7,3 +7,5 @@ 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..8692329 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -0,0 +1,884 @@ +/* + * 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-simple.h> + +#include <sysdev/bestcomm/bestcomm.h> +#include <sysdev/bestcomm/gen_bd.h> +#include <asm/mpc52xx_psc.h> + +MODULE_AUTHOR("Grant Likely grant.likely@secretlab.ca"); +MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver"); +MODULE_LICENSE("GPL"); + +/** + * 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 + * @dai: the CPU DAI for this device + * @sicr: Base value used in serial interface control register; mode is ORed + * with this value. + * @playback: Playback stream context data + * @capture: Capture stream context data + */ +struct psc_i2s { + char name[32]; + struct mpc52xx_psc __iomem *psc_regs; + struct mpc52xx_psc_fifo __iomem *fifo_regs; + unsigned int irq; + struct device *dev; + struct snd_soc_dai dai; + spinlock_t lock; + u32 sicr; + + /* per-stream data */ + struct psc_i2s_stream playback; + struct psc_i2s_stream capture; + + /* Statistics */ + struct { + int overrun_count; + int underrun_count; + } stats; +}; + +/* + * Interrupt handlers + */ +static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s) +{ + struct psc_i2s *psc_i2s = _psc_i2s; + struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs; + u16 isr; + + isr = in_be16(®s->mpc52xx_psc_isr); + + /* Playback underrun error */ + if (psc_i2s->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP)) + psc_i2s->stats.underrun_count++; + + /* Capture overrun error */ + if (psc_i2s->capture.active && (isr & MPC52xx_PSC_IMR_ORERR)) + psc_i2s->stats.overrun_count++; + + out_8(®s->command, 4 << 4); /* reset the error status */ + + return IRQ_HANDLED; +} + +/** + * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer + * @s: pointer to stream private data structure + * + * Enqueues another audio period buffer into the bestcomm queue. + * + * Note: The routine must only be called when there is space available in + * the queue. Otherwise the enqueue will fail and the audio ring buffer + * will get out of sync + */ +static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s) +{ + struct bcom_bd *bd; + + /* Prepare and enqueue the next buffer descriptor */ + bd = bcom_prepare_next_buffer(s->bcom_task); + bd->status = s->period_bytes; + bd->data[0] = s->period_next_pt; + bcom_submit_next_buffer(s->bcom_task, NULL); + + /* Update for next period */ + s->period_next_pt += s->period_bytes; + if (s->period_next_pt >= s->period_end) + s->period_next_pt = s->period_start; +} + +/* Bestcomm DMA irq handler */ +static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream) +{ + struct psc_i2s_stream *s = _psc_i2s_stream; + + /* For each finished period, dequeue the completed period buffer + * and enqueue a new one in it's place. */ + while (bcom_buffer_done(s->bcom_task)) { + bcom_retrieve_buffer(s->bcom_task, NULL, NULL); + s->period_current_pt += s->period_bytes; + if (s->period_current_pt >= s->period_end) + s->period_current_pt = s->period_start; + psc_i2s_bcom_enqueue_next_buffer(s); + bcom_enable(s->bcom_task); + } + + /* If the stream is active, then also inform the PCM middle layer + * of the period finished event. */ + if (s->active) + snd_pcm_period_elapsed(s->stream); + + return IRQ_HANDLED; +} + +/** + * psc_i2s_startup: create a new substream + * + * This is the first function called when a stream is opened. + * + * If this is the first stream open, then grab the IRQ and program most of + * the PSC registers. + */ +static int psc_i2s_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; + int rc; + + dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream); + + if (!psc_i2s->playback.active && + !psc_i2s->capture.active) { + /* Setup the IRQs */ + rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED, + "psc-i2s-status", psc_i2s); + rc |= request_irq(psc_i2s->capture.irq, + &psc_i2s_bcom_irq, IRQF_SHARED, + "psc-i2s-capture", &psc_i2s->capture); + rc |= request_irq(psc_i2s->playback.irq, + &psc_i2s_bcom_irq, IRQF_SHARED, + "psc-i2s-playback", &psc_i2s->playback); + if (rc) { + free_irq(psc_i2s->irq, psc_i2s); + free_irq(psc_i2s->capture.irq, + &psc_i2s->capture); + free_irq(psc_i2s->playback.irq, + &psc_i2s->playback); + return -ENODEV; + } + } + + return 0; +} + +static int psc_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; + u32 mode; + + dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i" + " periods=%i buffer_size=%i buffer_bytes=%i\n", + __func__, substream, params_period_size(params), + params_period_bytes(params), params_periods(params), + params_buffer_size(params), params_buffer_bytes(params)); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + mode = MPC52xx_PSC_SICR_SIM_CODEC_8; + break; + case SNDRV_PCM_FORMAT_S16_BE: + mode = MPC52xx_PSC_SICR_SIM_CODEC_16; + break; + case SNDRV_PCM_FORMAT_S24_BE: + mode = MPC52xx_PSC_SICR_SIM_CODEC_24; + break; + case SNDRV_PCM_FORMAT_S32_BE: + mode = MPC52xx_PSC_SICR_SIM_CODEC_32; + break; + default: + dev_dbg(psc_i2s->dev, "invalid format\n"); + return -EINVAL; + } + out_be32(&psc_i2s->psc_regs->sicr, psc_i2s->sicr | mode); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static int psc_i2s_hw_free(struct snd_pcm_substream *substream) +{ + 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; + else + s = &psc_i2s->playback; + + dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)" + " stream_id=%i\n", + substream, cmd, substream->pstr->stream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + s->period_bytes = frames_to_bytes(runtime, + runtime->period_size); + s->period_start = virt_to_phys(runtime->dma_area); + s->period_end = s->period_start + + (s->period_bytes * runtime->periods); + s->period_next_pt = s->period_start; + s->period_current_pt = s->period_start; + s->active = 1; + + /* First; reset everything */ + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + out_8(®s->command, MPC52xx_PSC_RST_RX); + out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT); + } else { + out_8(®s->command, MPC52xx_PSC_RST_TX); + out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT); + } + + /* Next, fill up the bestcomm bd queue and enable DMA. + * This will begin filling the PSC's fifo. */ + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + bcom_gen_bd_rx_reset(s->bcom_task); + else + bcom_gen_bd_tx_reset(s->bcom_task); + while (!bcom_queue_full(s->bcom_task)) + psc_i2s_bcom_enqueue_next_buffer(s); + bcom_enable(s->bcom_task); + + /* Due to errata in the i2s mode; need to line up enabling + * the transmitter with a transition on the frame sync + * line */ + + spin_lock_irqsave(&psc_i2s->lock, flags); + /* first make sure it is low */ + while ((in_8(®s->ipcr_acr.ipcr) & 0x80) != 0) + ; + /* then wait for the transition to high */ + while ((in_8(®s->ipcr_acr.ipcr) & 0x80) == 0) + ; + /* Finally, enable the PSC. + * Receiver must always be enabled; even when we only want + * transmit. (see 15.3.2.3 of MPC5200B User's Guide) */ + psc_cmd = MPC52xx_PSC_RX_ENABLE; + if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK) + psc_cmd |= MPC52xx_PSC_TX_ENABLE; + out_8(®s->command, psc_cmd); + spin_unlock_irqrestore(&psc_i2s->lock, flags); + + break; + + case SNDRV_PCM_TRIGGER_STOP: + /* Turn off the PSC */ + s->active = 0; + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (!psc_i2s->playback.active) { + out_8(®s->command, 2 << 4); /* reset rx */ + out_8(®s->command, 3 << 4); /* reset tx */ + out_8(®s->command, 4 << 4); /* reset err */ + } + } else { + out_8(®s->command, 3 << 4); /* reset tx */ + out_8(®s->command, 4 << 4); /* reset err */ + if (!psc_i2s->capture.active) + out_8(®s->command, 2 << 4); /* reset rx */ + } + + bcom_disable(s->bcom_task); + while (!bcom_queue_empty(s->bcom_task)) + bcom_retrieve_buffer(s->bcom_task, NULL, NULL); + + break; + + default: + dev_dbg(psc_i2s->dev, "invalid command\n"); + return -EINVAL; + } + + /* Update interrupt enable settings */ + imr = 0; + if (psc_i2s->playback.active) + imr |= MPC52xx_PSC_IMR_TXEMP; + if (psc_i2s->capture.active) + imr |= MPC52xx_PSC_IMR_ORERR; + out_be16(®s->isr_imr.imr, imr); + + 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.active && + !psc_i2s->capture.active) { + + /* Disable all interrupts and reset the PSC */ + out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0); + out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset tx */ + out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset rx */ + out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */ + out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */ + + /* Release irqs */ + free_irq(psc_i2s->irq, psc_i2s); + free_irq(psc_i2s->capture.irq, &psc_i2s->capture); + free_irq(psc_i2s->playback.irq, &psc_i2s->playback); + } +} + +/** + * psc_i2s_set_sysclk: set the clock frequency and direction + * + * This function is called by the machine driver to tell us what the clock + * 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_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_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_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, + .periods_min = 2, + .periods_max = 256, + .buffer_bytes_max = 2 * 1024 * 1024, + .fifo_size = 0, +}; + +static int psc_i2s_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; + struct psc_i2s_stream *s; + + dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream); + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + s = &psc_i2s->capture; + else + s = &psc_i2s->playback; + + snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware); + + s->stream = substream; + return 0; +} + +static int psc_i2s_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; + struct psc_i2s_stream *s; + + dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream); + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + s = &psc_i2s->capture; + else + s = &psc_i2s->playback; + + s->stream = NULL; + return 0; +} + +static snd_pcm_uframes_t +psc_i2s_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; + struct psc_i2s_stream *s; + dma_addr_t count; + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + s = &psc_i2s->capture; + else + s = &psc_i2s->playback; + + count = s->period_current_pt - s->period_start; + + return bytes_to_frames(substream->runtime, count); +} + +static struct snd_pcm_ops psc_i2s_pcm_ops = { + .open = psc_i2s_pcm_open, + .close = psc_i2s_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .pointer = psc_i2s_pcm_pointer, +}; + +static u64 psc_i2s_pcm_dmamask = 0xffffffff; +static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *rtd = pcm->private_data; + size_t size = psc_i2s_pcm_hardware.buffer_bytes_max; + int rc = 0; + + dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n", + card, dai, pcm); + + if (!card->dev->dma_mask) + card->dev->dma_mask = &psc_i2s_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (pcm->streams[0].substream) { + rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size, + &pcm->streams[0].substream->dma_buffer); + if (rc) + goto playback_alloc_err; + } + + if (pcm->streams[1].substream) { + rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size, + &pcm->streams[1].substream->dma_buffer); + if (rc) + goto capture_alloc_err; + } + + return 0; + + capture_alloc_err: + if (pcm->streams[0].substream) + snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); + playback_alloc_err: + dev_err(card->dev, "Cannot allocate buffer(s)\n"); + return -ENOMEM; +} + +static void psc_i2s_pcm_free(struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *rtd = pcm->private_data; + struct snd_pcm_substream *substream; + int stream; + + dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm); + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (substream) { + snd_dma_free_pages(&substream->dma_buffer); + substream->dma_buffer.area = NULL; + substream->dma_buffer.addr = 0; + } + } +} + +struct snd_soc_platform psc_i2s_pcm_soc_platform = { + .name = "mpc5200-psc-audio", + .pcm_ops = &psc_i2s_pcm_ops, + .pcm_new = &psc_i2s_pcm_new, + .pcm_free = &psc_i2s_pcm_free, +}; + +/* --------------------------------------------------------------------- + * Sysfs attributes for debugging + */ + +static ssize_t psc_i2s_status_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct psc_i2s *psc_i2s = dev_get_drvdata(dev); + + return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x " + "tfnum=%i tfstat=0x%.4x\n", + in_be16(&psc_i2s->psc_regs->sr_csr.status), + in_be32(&psc_i2s->psc_regs->sicr), + in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff, + in_be16(&psc_i2s->fifo_regs->rfstat), + in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff, + in_be16(&psc_i2s->fifo_regs->tfstat)); +} + +static int *psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s, const char *name) +{ + if (strcmp(name, "playback_underrun") == 0) + return &psc_i2s->stats.underrun_count; + if (strcmp(name, "capture_overrun") == 0) + return &psc_i2s->stats.overrun_count; + + return NULL; +} + +static ssize_t psc_i2s_stat_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct psc_i2s *psc_i2s = dev_get_drvdata(dev); + int *attrib; + + attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name); + if (!attrib) + return 0; + + return sprintf(buf, "%i\n", *attrib); +} + +static ssize_t psc_i2s_stat_store(struct device *dev, + struct device_attribute *attr, + const char *buf, + size_t count) +{ + struct psc_i2s *psc_i2s = dev_get_drvdata(dev); + int *attrib; + + attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name); + if (!attrib) + return 0; + + *attrib = simple_strtoul(buf, NULL, 0); + return count; +} + +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.psc_i2s = psc_i2s; + psc_i2s->capture.psc_i2s = psc_i2s; + snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1); + + /* Fill out the CPU DAI structure */ + memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai); + psc_i2s->dai.private_data = psc_i2s; + psc_i2s->dai.name = psc_i2s->name; + psc_i2s->dai.id = psc_id; + + /* Find the address of the fifo data registers and setup the + * DMA tasks */ + fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32); + psc_i2s->capture.bcom_task = + bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512); + psc_i2s->playback.bcom_task = + bcom_psc_gen_bd_tx_init(psc_id, 10, fifo); + if (!psc_i2s->capture.bcom_task || + !psc_i2s->playback.bcom_task) { + dev_err(&op->dev, "Could not allocate bestcomm tasks\n"); + iounmap(regs); + kfree(psc_i2s); + return -ENODEV; + } + + /* Disable all interrupts and reset the PSC */ + out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0); + out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset transmitter */ + out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset receiver */ + out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */ + out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */ + + /* Configure the serial interface mode; defaulting to CODEC8 mode */ + psc_i2s->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S | + MPC52xx_PSC_SICR_CLKPOL; + if (of_get_property(op->node, "fsl,cellslave", NULL)) + psc_i2s->sicr |= MPC52xx_PSC_SICR_CELLSLAVE | + MPC52xx_PSC_SICR_GENCLK; + out_be32(&psc_i2s->psc_regs->sicr, + psc_i2s->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8); + + /* Check for the codec handle. If it is not present then we + * are done */ + if (!of_get_property(op->node, "codec-handle", NULL)) + return 0; + + /* Set up mode register; + * First write: RxRdy (FIFO Alarm) generates rx FIFO irq + * Second write: register Normal mode for non loopback + */ + out_8(&psc_i2s->psc_regs->mode, 0); + out_8(&psc_i2s->psc_regs->mode, 0); + + /* Set the TX and RX fifo alarm thresholds */ + out_be16(&psc_i2s->fifo_regs->rfalarm, 0x100); + out_8(&psc_i2s->fifo_regs->rfcntl, 0x4); + out_be16(&psc_i2s->fifo_regs->tfalarm, 0x100); + out_8(&psc_i2s->fifo_regs->tfcntl, 0x7); + + /* Lookup the IRQ numbers */ + psc_i2s->playback.irq = + bcom_get_task_irq(psc_i2s->playback.bcom_task); + psc_i2s->capture.irq = + bcom_get_task_irq(psc_i2s->capture.bcom_task); + + /* Save what we've done so it can be found again later */ + dev_set_drvdata(&op->dev, psc_i2s); + + /* Register the SYSFS files */ + rc = device_create_file(psc_i2s->dev, &dev_attr_status); + rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun); + rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun); + if (rc) + dev_info(psc_i2s->dev, "error creating sysfs files\n"); + + /* 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.bcom_task); + bcom_gen_bd_tx_release(psc_i2s->playback.bcom_task); + + iounmap(psc_i2s->psc_regs); + iounmap(psc_i2s->fifo_regs); + kfree(psc_i2s); + dev_set_drvdata(&op->dev, NULL); + + return 0; +} + +/* 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, Jul 22, 2008 at 12:53:58AM -0600, Grant Likely wrote:
Signed-off-by: Grant Likely grant.likely@secretlab.ca
Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com
There's a few issues that were raised on the previous review cycle that still need to be addressed but they should be fixable in incremental patches (and easier to review that way):
+static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
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)
;
These loops should really have some sort of time limit on them, otherwise they'll lock hard if the expected events don't happen. Given that in slave mode they're synchronising with an externally generated clock this is something that might happen in practice and should produce better diagnostics.
- default:
dev_dbg(psc_i2s->dev, "invalid command\n");
return -EINVAL;
- }
I'd really expect to see the other possible triggers handled, even if the appropriate action is to silently ignore them, rather than having them generate an error message.
On Tue, Jul 22, 2008 at 11:09:52AM +0100, Mark Brown wrote:
On Tue, Jul 22, 2008 at 12:53:58AM -0600, Grant Likely wrote:
Signed-off-by: Grant Likely grant.likely@secretlab.ca
Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com
There's a few issues that were raised on the previous review cycle that still need to be addressed but they should be fixable in incremental patches (and easier to review that way):
+static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
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)
;
These loops should really have some sort of time limit on them, otherwise they'll lock hard if the expected events don't happen. Given that in slave mode they're synchronising with an externally generated clock this is something that might happen in practice and should produce better diagnostics.
Yes, I hope to rework these two lines entirely. I'm not happy with the current implementation either.
- default:
dev_dbg(psc_i2s->dev, "invalid command\n");
return -EINVAL;
- }
I'd really expect to see the other possible triggers handled, even if the appropriate action is to silently ignore them, rather than having them generate an error message.
Okay, I'll do that.
g.
What about the mpc5200b?
+/* 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); +
I'm just being grumpy because updating to linus/master made me fix over a hundred merge conflicts, now I have to test everything again.
From: Grant Likely grant.likely@secretlab.ca
ASoC Codec driver for the TLV320AIC26 device. As it stands, this driver doesn't support all the modes and clocking options of the AIC16, but it is a start.
Signed-off-by: Grant Likely grant.likely@secretlab.ca ---
sound/soc/codecs/Kconfig | 4 sound/soc/codecs/Makefile | 2 sound/soc/codecs/tlv320aic26.c | 519 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic26.h | 93 +++++++ 4 files changed, 618 insertions(+), 0 deletions(-)
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1db04a2..b399a64 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -47,6 +47,10 @@ 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 I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d7b97ab..dc0357e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -9,6 +9,7 @@ snd-soc-wm8990-objs := wm8990.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 @@ -22,4 +23,5 @@ obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(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..4621fda --- /dev/null +++ b/sound/soc/codecs/tlv320aic26.c @@ -0,0 +1,519 @@ +/* + * 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-simple.h> +#include <sound/initval.h> + +#include "tlv320aic26.h" + +MODULE_DESCRIPTION("ASoC TLV320AIC26 codec driver"); +MODULE_AUTHOR("Grant Likely grant.likely@secretlab.ca"); +MODULE_LICENSE("GPL"); + +/* AIC26 driver private data */ +struct aic26 { + struct spi_device *spi; + struct snd_soc_codec codec; + u16 reg_cache[AIC26_NUM_REGS]; /* 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 >= 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 */ + cache[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 >= AIC26_NUM_REGS) { + WARN_ON_ONCE(1); + return 0; + } + + return cache[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 >= 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 */ + cache[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); + + /* 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_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_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_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 + */ +#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) + +struct snd_soc_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_SINGLE("Keyclick activate", AIC26_REG_AUDIO_CTRL2, 15, 0x1, 0), + SOC_SINGLE("Keyclick amplitude", AIC26_REG_AUDIO_CTRL2, 12, 0x7, 0), + SOC_SINGLE("Keyclick frequency", AIC26_REG_AUDIO_CTRL2, 8, 0x7, 0), + SOC_SINGLE("Keyclick period", AIC26_REG_AUDIO_CTRL2, 4, 0xf, 0), + 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_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 >> 4) & 0xf)); + + return sprintf(buf, "amp=%x freq=%iHz len=%iclks\n", amp, freq, len); +} + +/* Any write to the keyclick attribute will trigger the keyclick event */ +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(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 = AIC26_NUM_REGS; + 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 register cache */ + for (i = 0; i < ARRAY_SIZE(aic26->reg_cache); i++) + aic26_reg_read(&aic26->codec, i); + + /* Register the sysfs files for debugging */ + /* Create SysFS files */ + rc = device_create_file(&spi->dev, &dev_attr_keyclick); + if (rc) + dev_info(&spi->dev, "error creating sysfs files\n"); + +#if defined(CONFIG_SND_SOC_OF_SIMPLE) + /* Tell the of_soc helper about this codec */ + of_snd_soc_register_codec(&aic26_soc_codec_dev, aic26, &aic26_dai, + spi->dev.archdata.of_node); +#endif + + 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); diff --git a/sound/soc/codecs/tlv320aic26.h b/sound/soc/codecs/tlv320aic26.h new file mode 100644 index 0000000..62b1f22 --- /dev/null +++ b/sound/soc/codecs/tlv320aic26.h @@ -0,0 +1,93 @@ +/* + * Texas Instruments TLV320AIC26 low power audio CODEC + * register definitions + * + * Copyright (C) 2008 Secret Lab Technologies Ltd. + */ + +#ifndef _TLV320AIC16_H_ +#define _TLV320AIC16_H_ + +/* 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) + +/* 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) + +/* 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, +}; + +#endif /* _TLV320AIC16_H_ */
On Tue, Jul 22, 2008 at 12:54:03AM -0600, Grant Likely wrote:
Signed-off-by: Grant Likely grant.likely@secretlab.ca
Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com
with the same comments about outstanding issues as applied to the CPU side.
+static int aic26_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
+{
- switch (params_rate(params)) {
- case 8000: fsref = 48000; divisor = AIC26_DIV_6; break;
- /* Configure PLL */
- pval = 1;
- jval = (fsref == 44100) ? 7 : 8;
- dval = (fsref == 44100) ? 5264 : 1920;
Without having looked at the chip datasheet these parameters probably all need to depend on the input clock rate and the PLL configuration should be done in a set_pll() rather than here.
+#if defined(CONFIG_SND_SOC_OF_SIMPLE)
- /* Tell the of_soc helper about this codec */
- of_snd_soc_register_codec(&aic26_soc_codec_dev, aic26, &aic26_dai,
spi->dev.archdata.of_node);
+#endif
This won't work if the OF_SIMPLE stuff is a module. There's also a checkpatch issue which I'll fix up.
On Tue, Jul 22, 2008 at 12:53:53AM -0600, Grant Likely wrote:
This is most likely temporary glue code to work around limitations in the ASoC v1 framework. When v2 is merged, most of this driver will need to be reworked.
Whatever is needed in v2 can probably have the client registration functions integrated into the core registration functions.
Signed-off-by: Grant Likely grant.likely@secretlab.ca
Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com
On 7/22/08, 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. At the moment, the driver is not very capable and it is expected to be extended as more features are needed for specifying the configuration in the device tree.
This is most likely temporary glue code to work around limitations in the ASoC v1 framework. When v2 is merged, most of this driver will need to be reworked.
Signed-off-by: Grant Likely grant.likely@secretlab.ca
include/sound/soc-of-simple.h | 21 +++++ sound/soc/fsl/Kconfig | 3 + sound/soc/fsl/Makefile | 3 + sound/soc/fsl/soc-of-simple.c | 171 +++++++++++++++++++++++++++++++++++++++++ 4 files changed, 198 insertions(+), 0 deletions(-)
diff --git a/include/sound/soc-of-simple.h b/include/sound/soc-of-simple.h new file mode 100644 index 0000000..696fc51 --- /dev/null +++ b/include/sound/soc-of-simple.h @@ -0,0 +1,21 @@ +/*
- OF helpers for ALSA SoC
- Copyright (C) 2008, Secret Lab Technologies Ltd.
- */
+#ifndef _INCLUDE_SOC_OF_H_ +#define _INCLUDE_SOC_OF_H_
+#include <linux/of.h> +#include <sound/soc.h>
+int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev,
void *codec_data, struct snd_soc_dai *dai,
struct device_node *node);
+int of_snd_soc_register_platform(struct snd_soc_platform *platform,
struct device_node *node,
struct snd_soc_dai *cpu_dai);
This doesn't compile for me. Where is struct snd_soc_dai being defined?
It used to be....
+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); + +int of_snd_soc_register_platform(struct snd_soc_platform *platform, + struct device_node *node, + struct snd_soc_cpu_dai *cpu_dai);
+#endif /* _INCLUDE_SOC_OF_H_ */ diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3368ace..398f002 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,3 +1,6 @@ +config SND_SOC_OF_SIMPLE
tristate
config SND_SOC_MPC8610 bool "ALSA SoC support for the MPC8610 SOC" depends on MPC8610_HPCD diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 62f680a..aa2100b 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -1,3 +1,6 @@ +# Simple machine driver that extracts configuration from the OF device tree +obj-$(CONFIG_SND_SOC_OF_SIMPLE) += soc-of-simple.o
# MPC8610 HPCD Machine Support obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o
diff --git a/sound/soc/fsl/soc-of-simple.c b/sound/soc/fsl/soc-of-simple.c new file mode 100644 index 0000000..0382fda --- /dev/null +++ b/sound/soc/fsl/soc-of-simple.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-simple.h> +#include <sound/initval.h>
+MODULE_AUTHOR("Grant Likely grant.likely@secretlab.ca"); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("ALSA SoC OpenFirmware bindings");
+static DEFINE_MUTEX(of_snd_soc_mutex); +static 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_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 = (char *)node->name;
of_soc->dai_link.stream_name = (char *)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_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);
On Tue, Jul 22, 2008 at 12:38:30PM -0400, Jon Smirl wrote:
On 7/22/08, Grant Likely grant.likely@secretlab.ca wrote:
+int of_snd_soc_register_platform(struct snd_soc_platform *platform,
struct device_node *node,
struct snd_soc_dai *cpu_dai);
This doesn't compile for me. Where is struct snd_soc_dai being defined?
It used to be....
In ALSA and ASoC git struct snd_soc_{cpu,codec}_dai have been replaced by a shared snd_soc_dai defined in soc.h as the previous two were.
On Tue, Jul 22, 2008 at 12:38:30PM -0400, Jon Smirl wrote:
On 7/22/08, Grant Likely grant.likely@secretlab.ca wrote:
+int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev,
void *codec_data, struct snd_soc_dai *dai,
struct device_node *node);
+int of_snd_soc_register_platform(struct snd_soc_platform *platform,
struct device_node *node,
struct snd_soc_dai *cpu_dai);
This doesn't compile for me. Where is struct snd_soc_dai being defined?
It used to be....
+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);
+int of_snd_soc_register_platform(struct snd_soc_platform *platform,
struct device_node *node,
struct snd_soc_cpu_dai *cpu_dai);
I had to change it to match what is in Linus' current top of tree. The snd_soc_cpu_dai and snd_soc_codec_dai structures have been merged.
g.
participants (3)
-
Grant Likely
-
Jon Smirl
-
Mark Brown