[alsa-devel] [PATCH 2/9] ARM: DaVinci: ASoC: Adds ALSA SoC McASP Audio Layer for TI DM646X processor
Adds ALSA SoC McASP Audio Layer for TI DM646X processor.
Signed-off-by: Naresh Medisetty naresh@ti.com --- This patch applies on the ASoC tree available at http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;a....
sound/soc/davinci/davinci-i2s-mcasp.c | 641 +++++++++++++++++++++++++++++++++ 1 files changed, 641 insertions(+), 0 deletions(-) create mode 100644 sound/soc/davinci/davinci-i2s-mcasp.c
diff --git a/sound/soc/davinci/davinci-i2s-mcasp.c b/sound/soc/davinci/davinci-i2s-mcasp.c new file mode 100644 index 0000000..eb632cd --- /dev/null +++ b/sound/soc/davinci/davinci-i2s-mcasp.c @@ -0,0 +1,641 @@ +/* + * ALSA SoC I2S (McASP) Audio Layer for TI DAVINCI processor + * + * Author: Nirmal Pandey n-pandey@ti.com, + * Suresh Rajashekara suresh.r@ti.com + * Steve Chen, <schen@.mvista.com> + * + * Copyright: (C) 2008 MontaVista Software, Inc., source@mvista.com + * Copyright: (C) 2008 Texas Instruments, India + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-i2s-mcasp.h" + +static inline void mcasp_set_bits(void __iomem *reg, u32 val) +{ + __raw_writel(__raw_readl(reg) | val, reg); +} + +static inline void mcasp_clr_bits(void __iomem *reg, u32 val) +{ + __raw_writel((__raw_readl(reg) & ~(val)), reg); +} + +static inline void mcasp_mod_bits(void __iomem *reg, u32 val, u32 mask) +{ + __raw_writel((__raw_readl(reg) & ~mask) | val, reg); +} + +static inline void mcasp_set_reg(void __iomem *reg, u32 val) +{ + __raw_writel(val, reg); +} + +static inline u32 mcasp_get_reg(void __iomem *reg) +{ + return (unsigned int)__raw_readl(reg); +} + +static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val) +{ + mcasp_set_bits(regs, val); + while ((mcasp_get_reg(regs) & val) != val) + ; +} +int davinci_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + cpu_dai->dma_data = dev->dma_params[substream->stream]; + + return 0; +} +EXPORT_SYMBOL(davinci_i2s_startup); + +int davinci_i2s_mcasp_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + int tmp, link_cnt; + int count = 0; + int backup_count = 0; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; + struct snd_soc_dai *cpu_dai; + struct davinci_audio_dev *dev; + struct resource *mem, *ioarea; + struct evm_snd_platform_data *parray = pdev->dev.platform_data; + struct evm_snd_platform_data *pdata; + struct davinci_pcm_dma_params *dma_data; + int ret = 0; + + dev = kzalloc(sizeof(struct davinci_audio_dev) * card->num_links, + GFP_KERNEL); + if (!dev) + return -ENOMEM; + + dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * + (card->num_links << 1), GFP_KERNEL); + if (!dma_data) + goto err_release_dev; + + for (link_cnt = 0; link_cnt < card->num_links; link_cnt++) { + mem = platform_get_resource(pdev, IORESOURCE_MEM, link_cnt); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + ret = -ENODEV; + backup_count = 0; + goto err_release_data; + } + + ioarea = request_mem_region(mem->start, + (mem->end - mem->start) + 1, pdev->name); + if (!ioarea) { + dev_err(&pdev->dev, "Audio region already claimed\n"); + ret = -EBUSY; + backup_count = 1; + goto err_release_data; + } + + + cpu_dai = card->dai_link[link_cnt].cpu_dai; + cpu_dai->private_data = &dev[link_cnt]; + pdata = &parray[link_cnt]; + dev[link_cnt].clk = clk_get(&pdev->dev, NULL); + + if (IS_ERR(dev[link_cnt].clk)) { + ret = -ENODEV; + backup_count = 2; + goto err_release_data; + } + clk_enable(dev[link_cnt].clk); + + dev[link_cnt].base = (void __iomem *)IO_ADDRESS(mem->start); + dev[link_cnt].op_mode = pdata->op_mode; + dev[link_cnt].tdm_slots = pdata->tdm_slots; + dev[link_cnt].num_serializer = pdata->num_serializer; + dev[link_cnt].serial_dir = pdata->serial_dir; + dev[link_cnt].codec_fmt = pdata->codec_fmt; + + + dma_data[count].name = "I2S PCM Stereo out"; + dma_data[count].channel = pdata->tx_dma_ch; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset + + io_v2p(dev[link_cnt].base)); + dev[link_cnt].dma_params[SNDRV_PCM_STREAM_PLAYBACK] = + &dma_data[count]; + + + count++; + dma_data[count].name = "I2S PCM Stereo in"; + dma_data[count].channel = pdata->rx_dma_ch; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset + + io_v2p(dev[link_cnt].base)); + dev[link_cnt].dma_params[SNDRV_PCM_STREAM_CAPTURE] = + &dma_data[count]; + count++; + + } + return 0; + +err_release_data: + for (tmp = link_cnt; tmp >= 0; tmp--) { + if (backup_count > 2) + clk_disable(dev[tmp].clk); + + if (backup_count > 1) { + mem = platform_get_resource(pdev, IORESOURCE_MEM, tmp); + release_mem_region(mem->start, + (mem->end - mem->start) + 1); + } + backup_count = 3; + } + kfree(dma_data); + +err_release_dev: + kfree(dev); + + return ret; +} +EXPORT_SYMBOL(davinci_i2s_mcasp_probe); + +void davinci_i2s_mcasp_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + int i; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; + struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; + struct davinci_audio_dev *dev_list = cpu_dai->private_data; + struct davinci_audio_dev *dev; + struct resource *mem; + struct davinci_pcm_dma_params *ptr; + + for (i = 0; i < card->num_links; i++) { + dev = &dev_list[i]; + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, i); + release_mem_region(mem->start, (mem->end - mem->start) + 1); + } + ptr = dev_list->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + kfree(ptr); + kfree(dev_list); +} +EXPORT_SYMBOL(davinci_i2s_mcasp_remove); + + +void mcasp_start_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); +} + +void mcasp_start_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + + /* wait for TX ready */ + while (!(mcasp_get_reg(dev->base + DAVINCI_MCASP_XRSRCTL_REG(0)) + & TXSTATE)) + ; + + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); +} + +static void davinci_mcasp_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_start_tx(dev); + else + mcasp_start_rx(dev); +} + +void mcasp_stop_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); +} + +void mcasp_stop_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); +} + +static void davinci_mcasp_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_stop_tx(dev); + else + mcasp_stop_rx(dev); +} + +static int davinci_i2s_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + void __iomem *base = dev->base; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* codec is clock and frame slave */ + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26)); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* codec is clock and frame master */ + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, (0x3f << 26)); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + case SND_SOC_DAIFMT_NB_IF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + case SND_SOC_DAIFMT_IB_IF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + case SND_SOC_DAIFMT_NB_NF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int davinci_config_channel_size(struct snd_pcm_substream *substream, + int channel_size) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + u32 fmt = 0; + + switch (channel_size) { + case DAVINCI_AUDIO_WORD_8: + fmt = 0x03 << 4; + break; + + case DAVINCI_AUDIO_WORD_12: + fmt = 0x05 << 4; + break; + + case DAVINCI_AUDIO_WORD_16: + fmt = 0x07 << 4; + break; + + case DAVINCI_AUDIO_WORD_20: + fmt = 0x09 << 4; + break; + + case DAVINCI_AUDIO_WORD_24: + fmt = 0x0B << 4; + break; + + case DAVINCI_AUDIO_WORD_28: + fmt = 0x0D << 4; + break; + + case DAVINCI_AUDIO_WORD_32: + fmt = 0x0F << 4; + break; + + default: + return -1; + } + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, fmt, (0x0F << 4)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, fmt, (0x0F << 4)); + return 0; +} + +static void davinci_hw_common_param(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + int i; + + /* Default configuration */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, SOFT); + + /* All PINS as McASP */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, + TXDATADMADIS); + } else { + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_REVTCTL_REG, + RXDATADMADIS); + } + + for (i = 0; i < dev->num_serializer; i++) { + mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + dev->serial_dir[i]); + if (dev->serial_dir[i] == TX_MODE) + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + else if (dev->serial_dir[i] == RX_MODE) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + } +} +static void davinci_hw_iis_param(struct snd_pcm_substream *substream) +{ + int i, active_slots; + u32 mask = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + active_slots = (dev->tdm_slots > 31) ? 32 : dev->tdm_slots; + for (i = 0; i < active_slots; i++) + mask |= (1 << i); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* bit stream is MSB first */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(dev->tdm_slots), FSXMOD(0x1FF)); + else + printk(KERN_ERR "playback tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + + } else { + /* bit stream is MSB first with no delay */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + FSRDLY(1) | RXORD); + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, + AHCLKRE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(dev->tdm_slots), FSRMOD(0x1FF)); + else + printk(KERN_ERR "capture tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + } +} + +/* S/PDIF */ +static void davinci_hw_dit_param(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + /* Set the PDIR for Serialiser as output */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); + + /* TXMASK for 24 bits */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); + + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 + and LSB first */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXROT(6) | TXSSZ(15)); + + /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + AFSXE | FSXMOD(0x180)); + + /* Set the TX tdm : for all the slots */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); + + /* Set the TX clock controls : div = 1 and internal */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE | TX_ASYNC); + + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); + + /* Only 44100 and 48000 are valid, both have the same setting */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXDIV(3)); + + /* Enable the DIT */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN); +} + +static int davinci_i2s_mcasp_hw_params( + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + struct davinci_pcm_dma_params *dma_params = + dev->dma_params[substream->stream]; + int word_length; + + davinci_hw_common_param(substream); + + if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) + davinci_hw_dit_param(substream); + else + davinci_hw_iis_param(substream); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + word_length = DAVINCI_AUDIO_WORD_8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + word_length = DAVINCI_AUDIO_WORD_16; + break; + case SNDRV_PCM_FORMAT_S32_LE: + dma_params->data_type = 4; + word_length = DAVINCI_AUDIO_WORD_32; + break; + default: + printk(KERN_WARNING "davinci-i2s: unsupported PCM format"); + return -EINVAL; + } + davinci_config_channel_size(substream, word_length); + + return 0; +} + +static int davinci_i2s_mcasp_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_mcasp_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_mcasp_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +struct snd_soc_dai davinci_iis_mcasp_dai[] = { + { + .name = "davinci-i2s", + .id = 0, + .probe = davinci_i2s_mcasp_probe, + .remove = davinci_i2s_mcasp_remove, + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_mcasp_trigger, + .hw_params = davinci_i2s_mcasp_hw_params, + .set_fmt = davinci_i2s_mcasp_set_dai_fmt, + }, + }, + { + .name = "davinci-dit", + .id = 1, + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_mcasp_trigger, + .hw_params = davinci_i2s_mcasp_hw_params, + .set_fmt = davinci_i2s_mcasp_set_dai_fmt, + }, + }, + +}; +EXPORT_SYMBOL_GPL(davinci_iis_mcasp_dai); + +static int __init davinci_i2s_mcasp_init(void) +{ + return snd_soc_register_dais(davinci_iis_mcasp_dai, + ARRAY_SIZE(davinci_iis_mcasp_dai)); + +} +module_init(davinci_i2s_mcasp_init); + +static void __exit davinci_i2s_mcasp_exit(void) +{ + return snd_soc_unregister_dais(davinci_iis_mcasp_dai, + ARRAY_SIZE(davinci_iis_mcasp_dai)); + +} +module_exit(davinci_i2s_mcasp_exit); + + + +MODULE_AUTHOR("Steve Chen"); +MODULE_DESCRIPTION("TI DAVINCI I2S (McASP) SoC Interface"); +MODULE_LICENSE("GPL"); +
On Mon, Mar 16, 2009 at 08:02:57AM -0400, Naresh Medisetty wrote:
Adds ALSA SoC McASP Audio Layer for TI DM646X processor.
This patch should come before the machine support that uses it in your patch series.
Signed-off-by: Naresh Medisetty naresh@ti.com
This patch applies on the ASoC tree available at http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;a....
Please submit ASoC patches against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git (specifying which branch you're working against in that form rather than a gitweb URL is more common, BTW).
+#include "davinci-pcm.h" +#include "davinci-i2s-mcasp.h"
This file isn't present in the tree? You should try to submit patches in an order where all the bits required for the patch are in the tree when it's applied.
+static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val) +{
- mcasp_set_bits(regs, val);
- while ((mcasp_get_reg(regs) & val) != val)
;+}
This will lock up hard if the write fails... Also, there should be a blank line between this and the following function.
+int davinci_i2s_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
Indentation.
+{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
- struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data;
- cpu_dai->dma_data = dev->dma_params[substream->stream];
- return 0;
+} +EXPORT_SYMBOL(davinci_i2s_startup);
Why is this being exported?
Please also note that all ASoC core APIs are EXPORT_SYMBOL_GPL.
cpu_dai = card->dai_link[link_cnt].cpu_dai;cpu_dai->private_data = &dev[link_cnt];pdata = &parray[link_cnt];dev[link_cnt].clk = clk_get(&pdev->dev, NULL);
I'd really expect to have a name for the clock here.
if (IS_ERR(dev[link_cnt].clk)) {ret = -ENODEV;backup_count = 2;
This backup count stuff is very obscure - it really either needs some comments to explain what it's doing or a restructuring of the code. I suspect that having a probe_one() fuction which you call in a loop and which does its own unwinding is the most sensible approach, it looks like what you're trying to simulate here.
+EXPORT_SYMBOL(davinci_i2s_mcasp_remove);
Again, why is this being exported?
- /* wait for TX ready */
- while (!(mcasp_get_reg(dev->base + DAVINCI_MCASP_XRSRCTL_REG(0))
& TXSTATE));
This is also going to lock up on error...
+struct snd_soc_dai davinci_iis_mcasp_dai[] = {
- {
.name = "davinci-i2s",.id = 0,.probe = davinci_i2s_mcasp_probe,.remove = davinci_i2s_mcasp_remove,.playback = {.channels_min = 1,.channels_max = 384,.rates = DAVINCI_I2S_RATES,.formats = SNDRV_PCM_FMTBIT_S16_LE,},
The code above suggested that you supported rather more sample sizes than this?
.ops = {.startup = davinci_i2s_startup,.trigger = davinci_i2s_mcasp_trigger,.hw_params = davinci_i2s_mcasp_hw_params,.set_fmt = davinci_i2s_mcasp_set_dai_fmt,},
This needs to be updated for current ASoC.
participants (2)
-
Mark Brown -
Naresh Medisetty