diff options
Diffstat (limited to 'ANDROID_3.4.5/sound/soc/fsl')
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/fsl/Kconfig | 67 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/fsl/Makefile | 22 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/fsl/efika-audio-fabric.c | 91 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/fsl/fsl_dma.c | 999 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/fsl/fsl_dma.h | 129 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/fsl/fsl_ssi.c | 800 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/fsl/fsl_ssi.h | 200 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/fsl/mpc5200_dma.c | 534 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/fsl/mpc5200_dma.h | 84 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/fsl/mpc5200_psc_ac97.c | 333 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/fsl/mpc5200_psc_ac97.h | 13 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/fsl/mpc5200_psc_i2s.c | 230 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/fsl/mpc8610_hpcd.c | 595 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/fsl/p1022_ds.c | 601 | ||||
| -rw-r--r-- | ANDROID_3.4.5/sound/soc/fsl/pcm030-audio-fabric.c | 91 |
15 files changed, 0 insertions, 4789 deletions
diff --git a/ANDROID_3.4.5/sound/soc/fsl/Kconfig b/ANDROID_3.4.5/sound/soc/fsl/Kconfig deleted file mode 100644 index d754d34d..00000000 --- a/ANDROID_3.4.5/sound/soc/fsl/Kconfig +++ /dev/null @@ -1,67 +0,0 @@ -config SND_MPC52xx_DMA - tristate - -# ASoC platform support for the Freescale PowerPC SOCs that have an SSI and -# an Elo DMA controller, such as the MPC8610 and P1022. You will still need to -# select a platform driver and a codec driver. -config SND_SOC_POWERPC_SSI - tristate - depends on FSL_SOC - -config SND_SOC_MPC8610_HPCD - tristate "ALSA SoC support for the Freescale MPC8610 HPCD board" - # I2C is necessary for the CS4270 driver - depends on MPC8610_HPCD && I2C - select SND_SOC_POWERPC_SSI - select SND_SOC_CS4270 - select SND_SOC_CS4270_VD33_ERRATA - default y if MPC8610_HPCD - help - Say Y if you want to enable audio on the Freescale MPC8610 HPCD. - -config SND_SOC_P1022_DS - tristate "ALSA SoC support for the Freescale P1022 DS board" - # I2C is necessary for the WM8776 driver - depends on P1022_DS && I2C - select SND_SOC_POWERPC_SSI - select SND_SOC_WM8776 - default y if P1022_DS - help - Say Y if you want to enable audio on the Freescale P1022 DS board. - This will also include the Wolfson Microelectronics WM8776 codec - driver. - -config SND_SOC_MPC5200_I2S - tristate "Freescale MPC5200 PSC in I2S mode driver" - depends on PPC_MPC52xx && PPC_BESTCOMM - select SND_MPC52xx_DMA - select PPC_BESTCOMM_GEN_BD - help - Say Y here to support the MPC5200 PSCs in I2S mode. - -config SND_SOC_MPC5200_AC97 - tristate "Freescale MPC5200 PSC in AC97 mode driver" - depends on PPC_MPC52xx && PPC_BESTCOMM - select SND_SOC_AC97_BUS - select SND_MPC52xx_DMA - select PPC_BESTCOMM_GEN_BD - help - Say Y here to support the MPC5200 PSCs in AC97 mode. - -config SND_MPC52xx_SOC_PCM030 - tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712" - depends on PPC_MPC5200_SIMPLE - select SND_SOC_MPC5200_AC97 - select SND_SOC_WM9712 - help - Say Y if you want to add support for sound on the Phytec pcm030 - baseboard. - -config SND_MPC52xx_SOC_EFIKA - tristate "SoC AC97 Audio support for bbplan Efika and STAC9766" - depends on PPC_EFIKA - select SND_SOC_MPC5200_AC97 - select SND_SOC_STAC9766 - help - Say Y if you want to add support for sound on the Efika. - diff --git a/ANDROID_3.4.5/sound/soc/fsl/Makefile b/ANDROID_3.4.5/sound/soc/fsl/Makefile deleted file mode 100644 index b4a38c0a..00000000 --- a/ANDROID_3.4.5/sound/soc/fsl/Makefile +++ /dev/null @@ -1,22 +0,0 @@ -# MPC8610 HPCD Machine Support -snd-soc-mpc8610-hpcd-objs := mpc8610_hpcd.o -obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += snd-soc-mpc8610-hpcd.o - -# P1022 DS Machine Support -snd-soc-p1022-ds-objs := p1022_ds.o -obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o - -# Freescale PowerPC SSI/DMA Platform Support -snd-soc-fsl-ssi-objs := fsl_ssi.o -snd-soc-fsl-dma-objs := fsl_dma.o -obj-$(CONFIG_SND_SOC_POWERPC_SSI) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o - -# MPC5200 Platform Support -obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o -obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o -obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o - -# MPC5200 Machine Support -obj-$(CONFIG_SND_MPC52xx_SOC_PCM030) += pcm030-audio-fabric.o -obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o - diff --git a/ANDROID_3.4.5/sound/soc/fsl/efika-audio-fabric.c b/ANDROID_3.4.5/sound/soc/fsl/efika-audio-fabric.c deleted file mode 100644 index b2acd329..00000000 --- a/ANDROID_3.4.5/sound/soc/fsl/efika-audio-fabric.c +++ /dev/null @@ -1,91 +0,0 @@ -/* - * Efika driver for the PSC of the Freescale MPC52xx - * configured as AC97 interface - * - * Copyright 2008 Jon Smirl, Digispeaker - * Author: Jon Smirl <jonsmirl@gmail.com> - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/of_device.h> -#include <linux/of_platform.h> -#include <linux/dma-mapping.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include "mpc5200_dma.h" -#include "mpc5200_psc_ac97.h" -#include "../codecs/stac9766.h" - -#define DRV_NAME "efika-audio-fabric" - -static struct snd_soc_dai_link efika_fabric_dai[] = { -{ - .name = "AC97", - .stream_name = "AC97 Analog", - .codec_dai_name = "stac9766-hifi-analog", - .cpu_dai_name = "mpc5200-psc-ac97.0", - .platform_name = "mpc5200-pcm-audio", - .codec_name = "stac9766-codec", -}, -{ - .name = "AC97", - .stream_name = "AC97 IEC958", - .codec_dai_name = "stac9766-hifi-IEC958", - .cpu_dai_name = "mpc5200-psc-ac97.1", - .platform_name = "mpc5200-pcm-audio", - .codec_name = "stac9766-codec", -}, -}; - -static struct snd_soc_card card = { - .name = "Efika", - .owner = THIS_MODULE, - .dai_link = efika_fabric_dai, - .num_links = ARRAY_SIZE(efika_fabric_dai), -}; - -static __init int efika_fabric_init(void) -{ - struct platform_device *pdev; - int rc; - - if (!of_machine_is_compatible("bplan,efika")) - return -ENODEV; - - pdev = platform_device_alloc("soc-audio", 1); - if (!pdev) { - pr_err("efika_fabric_init: platform_device_alloc() failed\n"); - return -ENODEV; - } - - platform_set_drvdata(pdev, &card); - - rc = platform_device_add(pdev); - if (rc) { - pr_err("efika_fabric_init: platform_device_add() failed\n"); - platform_device_put(pdev); - return -ENODEV; - } - return 0; -} - -module_init(efika_fabric_init); - - -MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>"); -MODULE_DESCRIPTION(DRV_NAME ": mpc5200 Efika fabric driver"); -MODULE_LICENSE("GPL"); - diff --git a/ANDROID_3.4.5/sound/soc/fsl/fsl_dma.c b/ANDROID_3.4.5/sound/soc/fsl/fsl_dma.c deleted file mode 100644 index 96bb92dd..00000000 --- a/ANDROID_3.4.5/sound/soc/fsl/fsl_dma.c +++ /dev/null @@ -1,999 +0,0 @@ -/* - * Freescale DMA ALSA SoC PCM driver - * - * Author: Timur Tabi <timur@freescale.com> - * - * Copyright 2007-2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - * - * This driver implements ASoC support for the Elo DMA controller, which is - * the DMA controller on Freescale 83xx, 85xx, and 86xx SOCs. In ALSA terms, - * the PCM driver is what handles the DMA buffer. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/dma-mapping.h> -#include <linux/interrupt.h> -#include <linux/delay.h> -#include <linux/gfp.h> -#include <linux/of_platform.h> -#include <linux/list.h> -#include <linux/slab.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/io.h> - -#include "fsl_dma.h" -#include "fsl_ssi.h" /* For the offset of stx0 and srx0 */ - -/* - * The formats that the DMA controller supports, which is anything - * that is 8, 16, or 32 bits. - */ -#define FSLDMA_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ - SNDRV_PCM_FMTBIT_U8 | \ - SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S16_BE | \ - SNDRV_PCM_FMTBIT_U16_LE | \ - SNDRV_PCM_FMTBIT_U16_BE | \ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S24_BE | \ - SNDRV_PCM_FMTBIT_U24_LE | \ - SNDRV_PCM_FMTBIT_U24_BE | \ - SNDRV_PCM_FMTBIT_S32_LE | \ - SNDRV_PCM_FMTBIT_S32_BE | \ - SNDRV_PCM_FMTBIT_U32_LE | \ - SNDRV_PCM_FMTBIT_U32_BE) - -#define FSLDMA_PCM_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \ - SNDRV_PCM_RATE_CONTINUOUS) - -struct dma_object { - struct snd_soc_platform_driver dai; - dma_addr_t ssi_stx_phys; - dma_addr_t ssi_srx_phys; - unsigned int ssi_fifo_depth; - struct ccsr_dma_channel __iomem *channel; - unsigned int irq; - bool assigned; - char path[1]; -}; - -/* - * The number of DMA links to use. Two is the bare minimum, but if you - * have really small links you might need more. - */ -#define NUM_DMA_LINKS 2 - -/** fsl_dma_private: p-substream DMA data - * - * Each substream has a 1-to-1 association with a DMA channel. - * - * The link[] array is first because it needs to be aligned on a 32-byte - * boundary, so putting it first will ensure alignment without padding the - * structure. - * - * @link[]: array of link descriptors - * @dma_channel: pointer to the DMA channel's registers - * @irq: IRQ for this DMA channel - * @substream: pointer to the substream object, needed by the ISR - * @ssi_sxx_phys: bus address of the STX or SRX register to use - * @ld_buf_phys: physical address of the LD buffer - * @current_link: index into link[] of the link currently being processed - * @dma_buf_phys: physical address of the DMA buffer - * @dma_buf_next: physical address of the next period to process - * @dma_buf_end: physical address of the byte after the end of the DMA - * @buffer period_size: the size of a single period - * @num_periods: the number of periods in the DMA buffer - */ -struct fsl_dma_private { - struct fsl_dma_link_descriptor link[NUM_DMA_LINKS]; - struct ccsr_dma_channel __iomem *dma_channel; - unsigned int irq; - struct snd_pcm_substream *substream; - dma_addr_t ssi_sxx_phys; - unsigned int ssi_fifo_depth; - dma_addr_t ld_buf_phys; - unsigned int current_link; - dma_addr_t dma_buf_phys; - dma_addr_t dma_buf_next; - dma_addr_t dma_buf_end; - size_t period_size; - unsigned int num_periods; -}; - -/** - * fsl_dma_hardare: define characteristics of the PCM hardware. - * - * The PCM hardware is the Freescale DMA controller. This structure defines - * the capabilities of that hardware. - * - * Since the sampling rate and data format are not controlled by the DMA - * controller, we specify no limits for those values. The only exception is - * period_bytes_min, which is set to a reasonably low value to prevent the - * DMA controller from generating too many interrupts per second. - * - * Since each link descriptor has a 32-bit byte count field, we set - * period_bytes_max to the largest 32-bit number. We also have no maximum - * number of periods. - * - * Note that we specify SNDRV_PCM_INFO_JOINT_DUPLEX here, but only because a - * limitation in the SSI driver requires the sample rates for playback and - * capture to be the same. - */ -static const struct snd_pcm_hardware fsl_dma_hardware = { - - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_JOINT_DUPLEX | - SNDRV_PCM_INFO_PAUSE, - .formats = FSLDMA_PCM_FORMATS, - .rates = FSLDMA_PCM_RATES, - .rate_min = 5512, - .rate_max = 192000, - .period_bytes_min = 512, /* A reasonable limit */ - .period_bytes_max = (u32) -1, - .periods_min = NUM_DMA_LINKS, - .periods_max = (unsigned int) -1, - .buffer_bytes_max = 128 * 1024, /* A reasonable limit */ -}; - -/** - * fsl_dma_abort_stream: tell ALSA that the DMA transfer has aborted - * - * This function should be called by the ISR whenever the DMA controller - * halts data transfer. - */ -static void fsl_dma_abort_stream(struct snd_pcm_substream *substream) -{ - unsigned long flags; - - snd_pcm_stream_lock_irqsave(substream, flags); - - if (snd_pcm_running(substream)) - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - - snd_pcm_stream_unlock_irqrestore(substream, flags); -} - -/** - * fsl_dma_update_pointers - update LD pointers to point to the next period - * - * As each period is completed, this function changes the the link - * descriptor pointers for that period to point to the next period. - */ -static void fsl_dma_update_pointers(struct fsl_dma_private *dma_private) -{ - struct fsl_dma_link_descriptor *link = - &dma_private->link[dma_private->current_link]; - - /* Update our link descriptors to point to the next period. On a 36-bit - * system, we also need to update the ESAD bits. We also set (keep) the - * snoop bits. See the comments in fsl_dma_hw_params() about snooping. - */ - if (dma_private->substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - link->source_addr = cpu_to_be32(dma_private->dma_buf_next); -#ifdef CONFIG_PHYS_64BIT - link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP | - upper_32_bits(dma_private->dma_buf_next)); -#endif - } else { - link->dest_addr = cpu_to_be32(dma_private->dma_buf_next); -#ifdef CONFIG_PHYS_64BIT - link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP | - upper_32_bits(dma_private->dma_buf_next)); -#endif - } - - /* Update our variables for next time */ - dma_private->dma_buf_next += dma_private->period_size; - - if (dma_private->dma_buf_next >= dma_private->dma_buf_end) - dma_private->dma_buf_next = dma_private->dma_buf_phys; - - if (++dma_private->current_link >= NUM_DMA_LINKS) - dma_private->current_link = 0; -} - -/** - * fsl_dma_isr: interrupt handler for the DMA controller - * - * @irq: IRQ of the DMA channel - * @dev_id: pointer to the dma_private structure for this DMA channel - */ -static irqreturn_t fsl_dma_isr(int irq, void *dev_id) -{ - struct fsl_dma_private *dma_private = dev_id; - struct snd_pcm_substream *substream = dma_private->substream; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct device *dev = rtd->platform->dev; - struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; - irqreturn_t ret = IRQ_NONE; - u32 sr, sr2 = 0; - - /* We got an interrupt, so read the status register to see what we - were interrupted for. - */ - sr = in_be32(&dma_channel->sr); - - if (sr & CCSR_DMA_SR_TE) { - dev_err(dev, "dma transmit error\n"); - fsl_dma_abort_stream(substream); - sr2 |= CCSR_DMA_SR_TE; - ret = IRQ_HANDLED; - } - - if (sr & CCSR_DMA_SR_CH) - ret = IRQ_HANDLED; - - if (sr & CCSR_DMA_SR_PE) { - dev_err(dev, "dma programming error\n"); - fsl_dma_abort_stream(substream); - sr2 |= CCSR_DMA_SR_PE; - ret = IRQ_HANDLED; - } - - if (sr & CCSR_DMA_SR_EOLNI) { - sr2 |= CCSR_DMA_SR_EOLNI; - ret = IRQ_HANDLED; - } - - if (sr & CCSR_DMA_SR_CB) - ret = IRQ_HANDLED; - - if (sr & CCSR_DMA_SR_EOSI) { - /* Tell ALSA we completed a period. */ - snd_pcm_period_elapsed(substream); - - /* - * Update our link descriptors to point to the next period. We - * only need to do this if the number of periods is not equal to - * the number of links. - */ - if (dma_private->num_periods != NUM_DMA_LINKS) - fsl_dma_update_pointers(dma_private); - - sr2 |= CCSR_DMA_SR_EOSI; - ret = IRQ_HANDLED; - } - - if (sr & CCSR_DMA_SR_EOLSI) { - sr2 |= CCSR_DMA_SR_EOLSI; - ret = IRQ_HANDLED; - } - - /* Clear the bits that we set */ - if (sr2) - out_be32(&dma_channel->sr, sr2); - - return ret; -} - -/** - * fsl_dma_new: initialize this PCM driver. - * - * This function is called when the codec driver calls snd_soc_new_pcms(), - * once for each .dai_link in the machine driver's snd_soc_card - * structure. - * - * snd_dma_alloc_pages() is just a front-end to dma_alloc_coherent(), which - * (currently) always allocates the DMA buffer in lowmem, even if GFP_HIGHMEM - * is specified. Therefore, any DMA buffers we allocate will always be in low - * memory, but we support for 36-bit physical addresses anyway. - * - * Regardless of where the memory is actually allocated, since the device can - * technically DMA to any 36-bit address, we do need to set the DMA mask to 36. - */ -static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - static u64 fsl_dma_dmamask = DMA_BIT_MASK(36); - int ret; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &fsl_dma_dmamask; - - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = fsl_dma_dmamask; - - /* Some codecs have separate DAIs for playback and capture, so we - * should allocate a DMA buffer only for the streams that are valid. - */ - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, - fsl_dma_hardware.buffer_bytes_max, - &pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->dma_buffer); - if (ret) { - dev_err(card->dev, "can't alloc playback dma buffer\n"); - return ret; - } - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, - fsl_dma_hardware.buffer_bytes_max, - &pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->dma_buffer); - if (ret) { - dev_err(card->dev, "can't alloc capture dma buffer\n"); - snd_dma_free_pages(&pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->dma_buffer); - return ret; - } - } - - return 0; -} - -/** - * fsl_dma_open: open a new substream. - * - * Each substream has its own DMA buffer. - * - * ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link - * descriptors that ping-pong from one period to the next. For example, if - * there are six periods and two link descriptors, this is how they look - * before playback starts: - * - * The last link descriptor - * ____________ points back to the first - * | | - * V | - * ___ ___ | - * | |->| |->| - * |___| |___| - * | | - * | | - * V V - * _________________________________________ - * | | | | | | | The DMA buffer is - * | | | | | | | divided into 6 parts - * |______|______|______|______|______|______| - * - * and here's how they look after the first period is finished playing: - * - * ____________ - * | | - * V | - * ___ ___ | - * | |->| |->| - * |___| |___| - * | | - * |______________ - * | | - * V V - * _________________________________________ - * | | | | | | | - * | | | | | | | - * |______|______|______|______|______|______| - * - * The first link descriptor now points to the third period. The DMA - * controller is currently playing the second period. When it finishes, it - * will jump back to the first descriptor and play the third period. - * - * There are four reasons we do this: - * - * 1. The only way to get the DMA controller to automatically restart the - * transfer when it gets to the end of the buffer is to use chaining - * mode. Basic direct mode doesn't offer that feature. - * 2. We need to receive an interrupt at the end of every period. The DMA - * controller can generate an interrupt at the end of every link transfer - * (aka segment). Making each period into a DMA segment will give us the - * interrupts we need. - * 3. By creating only two link descriptors, regardless of the number of - * periods, we do not need to reallocate the link descriptors if the - * number of periods changes. - * 4. All of the audio data is still stored in a single, contiguous DMA - * buffer, which is what ALSA expects. We're just dividing it into - * contiguous parts, and creating a link descriptor for each one. - */ -static int fsl_dma_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct device *dev = rtd->platform->dev; - struct dma_object *dma = - container_of(rtd->platform->driver, struct dma_object, dai); - struct fsl_dma_private *dma_private; - struct ccsr_dma_channel __iomem *dma_channel; - dma_addr_t ld_buf_phys; - u64 temp_link; /* Pointer to next link descriptor */ - u32 mr; - unsigned int channel; - int ret = 0; - unsigned int i; - - /* - * Reject any DMA buffer whose size is not a multiple of the period - * size. We need to make sure that the DMA buffer can be evenly divided - * into periods. - */ - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) { - dev_err(dev, "invalid buffer size\n"); - return ret; - } - - channel = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - - if (dma->assigned) { - dev_err(dev, "dma channel already assigned\n"); - return -EBUSY; - } - - dma_private = dma_alloc_coherent(dev, sizeof(struct fsl_dma_private), - &ld_buf_phys, GFP_KERNEL); - if (!dma_private) { - dev_err(dev, "can't allocate dma private data\n"); - return -ENOMEM; - } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_private->ssi_sxx_phys = dma->ssi_stx_phys; - else - dma_private->ssi_sxx_phys = dma->ssi_srx_phys; - - dma_private->ssi_fifo_depth = dma->ssi_fifo_depth; - dma_private->dma_channel = dma->channel; - dma_private->irq = dma->irq; - dma_private->substream = substream; - dma_private->ld_buf_phys = ld_buf_phys; - dma_private->dma_buf_phys = substream->dma_buffer.addr; - - ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "fsldma-audio", - dma_private); - if (ret) { - dev_err(dev, "can't register ISR for IRQ %u (ret=%i)\n", - dma_private->irq, ret); - dma_free_coherent(dev, sizeof(struct fsl_dma_private), - dma_private, dma_private->ld_buf_phys); - return ret; - } - - dma->assigned = 1; - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware); - runtime->private_data = dma_private; - - /* Program the fixed DMA controller parameters */ - - dma_channel = dma_private->dma_channel; - - temp_link = dma_private->ld_buf_phys + - sizeof(struct fsl_dma_link_descriptor); - - for (i = 0; i < NUM_DMA_LINKS; i++) { - dma_private->link[i].next = cpu_to_be64(temp_link); - - temp_link += sizeof(struct fsl_dma_link_descriptor); - } - /* The last link descriptor points to the first */ - dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys); - - /* Tell the DMA controller where the first link descriptor is */ - out_be32(&dma_channel->clndar, - CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys)); - out_be32(&dma_channel->eclndar, - CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys)); - - /* The manual says the BCR must be clear before enabling EMP */ - out_be32(&dma_channel->bcr, 0); - - /* - * Program the mode register for interrupts, external master control, - * and source/destination hold. Also clear the Channel Abort bit. - */ - mr = in_be32(&dma_channel->mr) & - ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE); - - /* - * We want External Master Start and External Master Pause enabled, - * because the SSI is controlling the DMA controller. We want the DMA - * controller to be set up in advance, and then we signal only the SSI - * to start transferring. - * - * We want End-Of-Segment Interrupts enabled, because this will generate - * an interrupt at the end of each segment (each link descriptor - * represents one segment). Each DMA segment is the same thing as an - * ALSA period, so this is how we get an interrupt at the end of every - * period. - * - * We want Error Interrupt enabled, so that we can get an error if - * the DMA controller is mis-programmed somehow. - */ - mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN | - CCSR_DMA_MR_EMS_EN; - - /* For playback, we want the destination address to be held. For - capture, set the source address to be held. */ - mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE; - - out_be32(&dma_channel->mr, mr); - - return 0; -} - -/** - * fsl_dma_hw_params: continue initializing the DMA links - * - * This function obtains hardware parameters about the opened stream and - * programs the DMA controller accordingly. - * - * One drawback of big-endian is that when copying integers of different - * sizes to a fixed-sized register, the address to which the integer must be - * copied is dependent on the size of the integer. - * - * For example, if P is the address of a 32-bit register, and X is a 32-bit - * integer, then X should be copied to address P. However, if X is a 16-bit - * integer, then it should be copied to P+2. If X is an 8-bit register, - * then it should be copied to P+3. - * - * So for playback of 8-bit samples, the DMA controller must transfer single - * bytes from the DMA buffer to the last byte of the STX0 register, i.e. - * offset by 3 bytes. For 16-bit samples, the offset is two bytes. - * - * For 24-bit samples, the offset is 1 byte. However, the DMA controller - * does not support 3-byte copies (the DAHTS register supports only 1, 2, 4, - * and 8 bytes at a time). So we do not support packed 24-bit samples. - * 24-bit data must be padded to 32 bits. - */ -static int fsl_dma_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct fsl_dma_private *dma_private = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct device *dev = rtd->platform->dev; - - /* Number of bits per sample */ - unsigned int sample_bits = - snd_pcm_format_physical_width(params_format(hw_params)); - - /* Number of bytes per frame */ - unsigned int sample_bytes = sample_bits / 8; - - /* Bus address of SSI STX register */ - dma_addr_t ssi_sxx_phys = dma_private->ssi_sxx_phys; - - /* Size of the DMA buffer, in bytes */ - size_t buffer_size = params_buffer_bytes(hw_params); - - /* Number of bytes per period */ - size_t period_size = params_period_bytes(hw_params); - - /* Pointer to next period */ - dma_addr_t temp_addr = substream->dma_buffer.addr; - - /* Pointer to DMA controller */ - struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; - - u32 mr; /* DMA Mode Register */ - - unsigned int i; - - /* Initialize our DMA tracking variables */ - dma_private->period_size = period_size; - dma_private->num_periods = params_periods(hw_params); - dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size; - dma_private->dma_buf_next = dma_private->dma_buf_phys + - (NUM_DMA_LINKS * period_size); - - if (dma_private->dma_buf_next >= dma_private->dma_buf_end) - /* This happens if the number of periods == NUM_DMA_LINKS */ - dma_private->dma_buf_next = dma_private->dma_buf_phys; - - mr = in_be32(&dma_channel->mr) & ~(CCSR_DMA_MR_BWC_MASK | - CCSR_DMA_MR_SAHTS_MASK | CCSR_DMA_MR_DAHTS_MASK); - - /* Due to a quirk of the SSI's STX register, the target address - * for the DMA operations depends on the sample size. So we calculate - * that offset here. While we're at it, also tell the DMA controller - * how much data to transfer per sample. - */ - switch (sample_bits) { - case 8: - mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1; - ssi_sxx_phys += 3; - break; - case 16: - mr |= CCSR_DMA_MR_DAHTS_2 | CCSR_DMA_MR_SAHTS_2; - ssi_sxx_phys += 2; - break; - case 32: - mr |= CCSR_DMA_MR_DAHTS_4 | CCSR_DMA_MR_SAHTS_4; - break; - default: - /* We should never get here */ - dev_err(dev, "unsupported sample size %u\n", sample_bits); - return -EINVAL; - } - - /* - * BWC determines how many bytes are sent/received before the DMA - * controller checks the SSI to see if it needs to stop. BWC should - * always be a multiple of the frame size, so that we always transmit - * whole frames. Each frame occupies two slots in the FIFO. The - * parameter for CCSR_DMA_MR_BWC() is rounded down the next power of two - * (MR[BWC] can only represent even powers of two). - * - * To simplify the process, we set BWC to the largest value that is - * less than or equal to the FIFO watermark. For playback, this ensures - * that we transfer the maximum amount without overrunning the FIFO. - * For capture, this ensures that we transfer the maximum amount without - * underrunning the FIFO. - * - * f = SSI FIFO depth - * w = SSI watermark value (which equals f - 2) - * b = DMA bandwidth count (in bytes) - * s = sample size (in bytes, which equals frame_size * 2) - * - * For playback, we never transmit more than the transmit FIFO - * watermark, otherwise we might write more data than the FIFO can hold. - * The watermark is equal to the FIFO depth minus two. - * - * For capture, two equations must hold: - * w > f - (b / s) - * w >= b / s - * - * So, b > 2 * s, but b must also be <= s * w. To simplify, we set - * b = s * w, which is equal to - * (dma_private->ssi_fifo_depth - 2) * sample_bytes. - */ - mr |= CCSR_DMA_MR_BWC((dma_private->ssi_fifo_depth - 2) * sample_bytes); - - out_be32(&dma_channel->mr, mr); - - for (i = 0; i < NUM_DMA_LINKS; i++) { - struct fsl_dma_link_descriptor *link = &dma_private->link[i]; - - link->count = cpu_to_be32(period_size); - - /* The snoop bit tells the DMA controller whether it should tell - * the ECM to snoop during a read or write to an address. For - * audio, we use DMA to transfer data between memory and an I/O - * device (the SSI's STX0 or SRX0 register). Snooping is only - * needed if there is a cache, so we need to snoop memory - * addresses only. For playback, that means we snoop the source - * but not the destination. For capture, we snoop the - * destination but not the source. - * - * Note that failing to snoop properly is unlikely to cause - * cache incoherency if the period size is larger than the - * size of L1 cache. This is because filling in one period will - * flush out the data for the previous period. So if you - * increased period_bytes_min to a large enough size, you might - * get more performance by not snooping, and you'll still be - * okay. You'll need to update fsl_dma_update_pointers() also. - */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - link->source_addr = cpu_to_be32(temp_addr); - link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP | - upper_32_bits(temp_addr)); - - link->dest_addr = cpu_to_be32(ssi_sxx_phys); - link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP | - upper_32_bits(ssi_sxx_phys)); - } else { - link->source_addr = cpu_to_be32(ssi_sxx_phys); - link->source_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP | - upper_32_bits(ssi_sxx_phys)); - - link->dest_addr = cpu_to_be32(temp_addr); - link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP | - upper_32_bits(temp_addr)); - } - - temp_addr += period_size; - } - - return 0; -} - -/** - * fsl_dma_pointer: determine the current position of the DMA transfer - * - * This function is called by ALSA when ALSA wants to know where in the - * stream buffer the hardware currently is. - * - * For playback, the SAR register contains the physical address of the most - * recent DMA transfer. For capture, the value is in the DAR register. - * - * The base address of the buffer is stored in the source_addr field of the - * first link descriptor. - */ -static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct fsl_dma_private *dma_private = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct device *dev = rtd->platform->dev; - struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; - dma_addr_t position; - snd_pcm_uframes_t frames; - - /* Obtain the current DMA pointer, but don't read the ESAD bits if we - * only have 32-bit DMA addresses. This function is typically called - * in interrupt context, so we need to optimize it. - */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - position = in_be32(&dma_channel->sar); -#ifdef CONFIG_PHYS_64BIT - position |= (u64)(in_be32(&dma_channel->satr) & - CCSR_DMA_ATR_ESAD_MASK) << 32; -#endif - } else { - position = in_be32(&dma_channel->dar); -#ifdef CONFIG_PHYS_64BIT - position |= (u64)(in_be32(&dma_channel->datr) & - CCSR_DMA_ATR_ESAD_MASK) << 32; -#endif - } - - /* - * When capture is started, the SSI immediately starts to fill its FIFO. - * This means that the DMA controller is not started until the FIFO is - * full. However, ALSA calls this function before that happens, when - * MR.DAR is still zero. In this case, just return zero to indicate - * that nothing has been received yet. - */ - if (!position) - return 0; - - if ((position < dma_private->dma_buf_phys) || - (position > dma_private->dma_buf_end)) { - dev_err(dev, "dma pointer is out of range, halting stream\n"); - return SNDRV_PCM_POS_XRUN; - } - - frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys); - - /* - * If the current address is just past the end of the buffer, wrap it - * around. - */ - if (frames == runtime->buffer_size) - frames = 0; - - return frames; -} - -/** - * fsl_dma_hw_free: release resources allocated in fsl_dma_hw_params() - * - * Release the resources allocated in fsl_dma_hw_params() and de-program the - * registers. - * - * This function can be called multiple times. - */ -static int fsl_dma_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct fsl_dma_private *dma_private = runtime->private_data; - - if (dma_private) { - struct ccsr_dma_channel __iomem *dma_channel; - - dma_channel = dma_private->dma_channel; - - /* Stop the DMA */ - out_be32(&dma_channel->mr, CCSR_DMA_MR_CA); - out_be32(&dma_channel->mr, 0); - - /* Reset all the other registers */ - out_be32(&dma_channel->sr, -1); - out_be32(&dma_channel->clndar, 0); - out_be32(&dma_channel->eclndar, 0); - out_be32(&dma_channel->satr, 0); - out_be32(&dma_channel->sar, 0); - out_be32(&dma_channel->datr, 0); - out_be32(&dma_channel->dar, 0); - out_be32(&dma_channel->bcr, 0); - out_be32(&dma_channel->nlndar, 0); - out_be32(&dma_channel->enlndar, 0); - } - - return 0; -} - -/** - * fsl_dma_close: close the stream. - */ -static int fsl_dma_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct fsl_dma_private *dma_private = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct device *dev = rtd->platform->dev; - struct dma_object *dma = - container_of(rtd->platform->driver, struct dma_object, dai); - - if (dma_private) { - if (dma_private->irq) - free_irq(dma_private->irq, dma_private); - - if (dma_private->ld_buf_phys) { - dma_unmap_single(dev, dma_private->ld_buf_phys, - sizeof(dma_private->link), - DMA_TO_DEVICE); - } - - /* Deallocate the fsl_dma_private structure */ - dma_free_coherent(dev, sizeof(struct fsl_dma_private), - dma_private, dma_private->ld_buf_phys); - substream->runtime->private_data = NULL; - } - - dma->assigned = 0; - - return 0; -} - -/* - * Remove this PCM driver. - */ -static void fsl_dma_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - unsigned int i; - - for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) { - substream = pcm->streams[i].substream; - if (substream) { - snd_dma_free_pages(&substream->dma_buffer); - substream->dma_buffer.area = NULL; - substream->dma_buffer.addr = 0; - } - } -} - -/** - * find_ssi_node -- returns the SSI node that points to his DMA channel node - * - * Although this DMA driver attempts to operate independently of the other - * devices, it still needs to determine some information about the SSI device - * that it's working with. Unfortunately, the device tree does not contain - * a pointer from the DMA channel node to the SSI node -- the pointer goes the - * other way. So we need to scan the device tree for SSI nodes until we find - * the one that points to the given DMA channel node. It's ugly, but at least - * it's contained in this one function. - */ -static struct device_node *find_ssi_node(struct device_node *dma_channel_np) -{ - struct device_node *ssi_np, *np; - - for_each_compatible_node(ssi_np, NULL, "fsl,mpc8610-ssi") { - /* Check each DMA phandle to see if it points to us. We - * assume that device_node pointers are a valid comparison. - */ - np = of_parse_phandle(ssi_np, "fsl,playback-dma", 0); - of_node_put(np); - if (np == dma_channel_np) - return ssi_np; - - np = of_parse_phandle(ssi_np, "fsl,capture-dma", 0); - of_node_put(np); - if (np == dma_channel_np) - return ssi_np; - } - - return NULL; -} - -static struct snd_pcm_ops fsl_dma_ops = { - .open = fsl_dma_open, - .close = fsl_dma_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = fsl_dma_hw_params, - .hw_free = fsl_dma_hw_free, - .pointer = fsl_dma_pointer, -}; - -static int __devinit fsl_soc_dma_probe(struct platform_device *pdev) - { - struct dma_object *dma; - struct device_node *np = pdev->dev.of_node; - struct device_node *ssi_np; - struct resource res; - const uint32_t *iprop; - int ret; - - /* Find the SSI node that points to us. */ - ssi_np = find_ssi_node(np); - if (!ssi_np) { - dev_err(&pdev->dev, "cannot find parent SSI node\n"); - return -ENODEV; - } - - ret = of_address_to_resource(ssi_np, 0, &res); - if (ret) { - dev_err(&pdev->dev, "could not determine resources for %s\n", - ssi_np->full_name); - of_node_put(ssi_np); - return ret; - } - - dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL); - if (!dma) { - dev_err(&pdev->dev, "could not allocate dma object\n"); - of_node_put(ssi_np); - return -ENOMEM; - } - - strcpy(dma->path, np->full_name); - dma->dai.ops = &fsl_dma_ops; - dma->dai.pcm_new = fsl_dma_new; - dma->dai.pcm_free = fsl_dma_free_dma_buffers; - - /* Store the SSI-specific information that we need */ - dma->ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0); - dma->ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0); - - iprop = of_get_property(ssi_np, "fsl,fifo-depth", NULL); - if (iprop) - dma->ssi_fifo_depth = be32_to_cpup(iprop); - else - /* Older 8610 DTs didn't have the fifo-depth property */ - dma->ssi_fifo_depth = 8; - - of_node_put(ssi_np); - - ret = snd_soc_register_platform(&pdev->dev, &dma->dai); - if (ret) { - dev_err(&pdev->dev, "could not register platform\n"); - kfree(dma); - return ret; - } - - dma->channel = of_iomap(np, 0); - dma->irq = irq_of_parse_and_map(np, 0); - - dev_set_drvdata(&pdev->dev, dma); - - return 0; -} - -static int __devexit fsl_soc_dma_remove(struct platform_device *pdev) -{ - struct dma_object *dma = dev_get_drvdata(&pdev->dev); - - snd_soc_unregister_platform(&pdev->dev); - iounmap(dma->channel); - irq_dispose_mapping(dma->irq); - kfree(dma); - - return 0; -} - -static const struct of_device_id fsl_soc_dma_ids[] = { - { .compatible = "fsl,ssi-dma-channel", }, - {} -}; -MODULE_DEVICE_TABLE(of, fsl_soc_dma_ids); - -static struct platform_driver fsl_soc_dma_driver = { - .driver = { - .name = "fsl-pcm-audio", - .owner = THIS_MODULE, - .of_match_table = fsl_soc_dma_ids, - }, - .probe = fsl_soc_dma_probe, - .remove = __devexit_p(fsl_soc_dma_remove), -}; - -module_platform_driver(fsl_soc_dma_driver); - -MODULE_AUTHOR("Timur Tabi <timur@freescale.com>"); -MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM Driver"); -MODULE_LICENSE("GPL v2"); diff --git a/ANDROID_3.4.5/sound/soc/fsl/fsl_dma.h b/ANDROID_3.4.5/sound/soc/fsl/fsl_dma.h deleted file mode 100644 index 78fee97e..00000000 --- a/ANDROID_3.4.5/sound/soc/fsl/fsl_dma.h +++ /dev/null @@ -1,129 +0,0 @@ -/* - * mpc8610-pcm.h - ALSA PCM interface for the Freescale MPC8610 SoC - * - * 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. - */ - -#ifndef _MPC8610_PCM_H -#define _MPC8610_PCM_H - -struct ccsr_dma { - u8 res0[0x100]; - struct ccsr_dma_channel { - __be32 mr; /* Mode register */ - __be32 sr; /* Status register */ - __be32 eclndar; /* Current link descriptor extended addr reg */ - __be32 clndar; /* Current link descriptor address register */ - __be32 satr; /* Source attributes register */ - __be32 sar; /* Source address register */ - __be32 datr; /* Destination attributes register */ - __be32 dar; /* Destination address register */ - __be32 bcr; /* Byte count register */ - __be32 enlndar; /* Next link descriptor extended address reg */ - __be32 nlndar; /* Next link descriptor address register */ - u8 res1[4]; - __be32 eclsdar; /* Current list descriptor extended addr reg */ - __be32 clsdar; /* Current list descriptor address register */ - __be32 enlsdar; /* Next list descriptor extended address reg */ - __be32 nlsdar; /* Next list descriptor address register */ - __be32 ssr; /* Source stride register */ - __be32 dsr; /* Destination stride register */ - u8 res2[0x38]; - } channel[4]; - __be32 dgsr; -}; - -#define CCSR_DMA_MR_BWC_DISABLED 0x0F000000 -#define CCSR_DMA_MR_BWC_SHIFT 24 -#define CCSR_DMA_MR_BWC_MASK 0x0F000000 -#define CCSR_DMA_MR_BWC(x) \ - ((ilog2(x) << CCSR_DMA_MR_BWC_SHIFT) & CCSR_DMA_MR_BWC_MASK) -#define CCSR_DMA_MR_EMP_EN 0x00200000 -#define CCSR_DMA_MR_EMS_EN 0x00040000 -#define CCSR_DMA_MR_DAHTS_MASK 0x00030000 -#define CCSR_DMA_MR_DAHTS_1 0x00000000 -#define CCSR_DMA_MR_DAHTS_2 0x00010000 -#define CCSR_DMA_MR_DAHTS_4 0x00020000 -#define CCSR_DMA_MR_DAHTS_8 0x00030000 -#define CCSR_DMA_MR_SAHTS_MASK 0x0000C000 -#define CCSR_DMA_MR_SAHTS_1 0x00000000 -#define CCSR_DMA_MR_SAHTS_2 0x00004000 -#define CCSR_DMA_MR_SAHTS_4 0x00008000 -#define CCSR_DMA_MR_SAHTS_8 0x0000C000 -#define CCSR_DMA_MR_DAHE 0x00002000 -#define CCSR_DMA_MR_SAHE 0x00001000 -#define CCSR_DMA_MR_SRW 0x00000400 -#define CCSR_DMA_MR_EOSIE 0x00000200 -#define CCSR_DMA_MR_EOLNIE 0x00000100 -#define CCSR_DMA_MR_EOLSIE 0x00000080 -#define CCSR_DMA_MR_EIE 0x00000040 -#define CCSR_DMA_MR_XFE 0x00000020 -#define CCSR_DMA_MR_CDSM_SWSM 0x00000010 -#define CCSR_DMA_MR_CA 0x00000008 -#define CCSR_DMA_MR_CTM 0x00000004 -#define CCSR_DMA_MR_CC 0x00000002 -#define CCSR_DMA_MR_CS 0x00000001 - -#define CCSR_DMA_SR_TE 0x00000080 -#define CCSR_DMA_SR_CH 0x00000020 -#define CCSR_DMA_SR_PE 0x00000010 -#define CCSR_DMA_SR_EOLNI 0x00000008 -#define CCSR_DMA_SR_CB 0x00000004 -#define CCSR_DMA_SR_EOSI 0x00000002 -#define CCSR_DMA_SR_EOLSI 0x00000001 - -/* ECLNDAR takes bits 32-36 of the CLNDAR register */ -static inline u32 CCSR_DMA_ECLNDAR_ADDR(u64 x) -{ - return (x >> 32) & 0xf; -} - -#define CCSR_DMA_CLNDAR_ADDR(x) ((x) & 0xFFFFFFFE) -#define CCSR_DMA_CLNDAR_EOSIE 0x00000008 - -/* SATR and DATR, combined */ -#define CCSR_DMA_ATR_PBATMU 0x20000000 -#define CCSR_DMA_ATR_TFLOWLVL_0 0x00000000 -#define CCSR_DMA_ATR_TFLOWLVL_1 0x06000000 -#define CCSR_DMA_ATR_TFLOWLVL_2 0x08000000 -#define CCSR_DMA_ATR_TFLOWLVL_3 0x0C000000 -#define CCSR_DMA_ATR_PCIORDER 0x02000000 -#define CCSR_DMA_ATR_SME 0x01000000 -#define CCSR_DMA_ATR_NOSNOOP 0x00040000 -#define CCSR_DMA_ATR_SNOOP 0x00050000 -#define CCSR_DMA_ATR_ESAD_MASK 0x0000000F - -/** - * List Descriptor for extended chaining mode DMA operations. - * - * The CLSDAR register points to the first (in a linked-list) List - * Descriptor. Each object must be aligned on a 32-byte boundary. Each - * list descriptor points to a linked-list of link Descriptors. - */ -struct fsl_dma_list_descriptor { - __be64 next; /* Address of next list descriptor */ - __be64 first_link; /* Address of first link descriptor */ - __be32 source; /* Source stride */ - __be32 dest; /* Destination stride */ - u8 res[8]; /* Reserved */ -} __attribute__ ((aligned(32), packed)); - -/** - * Link Descriptor for basic and extended chaining mode DMA operations. - * - * A Link Descriptor points to a single DMA buffer. Each link descriptor - * must be aligned on a 32-byte boundary. - */ -struct fsl_dma_link_descriptor { - __be32 source_attr; /* Programmed into SATR register */ - __be32 source_addr; /* Programmed into SAR register */ - __be32 dest_attr; /* Programmed into DATR register */ - __be32 dest_addr; /* Programmed into DAR register */ - __be64 next; /* Address of next link descriptor */ - __be32 count; /* Byte count */ - u8 res[4]; /* Reserved */ -} __attribute__ ((aligned(32), packed)); - -#endif diff --git a/ANDROID_3.4.5/sound/soc/fsl/fsl_ssi.c b/ANDROID_3.4.5/sound/soc/fsl/fsl_ssi.c deleted file mode 100644 index 2eb407fa..00000000 --- a/ANDROID_3.4.5/sound/soc/fsl/fsl_ssi.c +++ /dev/null @@ -1,800 +0,0 @@ -/* - * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver - * - * Author: Timur Tabi <timur@freescale.com> - * - * Copyright 2007-2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/slab.h> -#include <linux/of_platform.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include "fsl_ssi.h" - -/** - * FSLSSI_I2S_RATES: sample rates supported by the I2S - * - * This driver currently only supports the SSI 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 FSLSSI_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \ - SNDRV_PCM_RATE_CONTINUOUS) - -/** - * FSLSSI_I2S_FORMATS: audio formats supported by the SSI - * - * This driver currently only supports the SSI running in I2S slave mode. - * - * The SSI has a limitation in that the samples must be in the same byte - * order as the host CPU. This is because when multiple bytes are written - * to the STX register, the bytes and bits must be written in the same - * order. The STX is a shift register, so all the bits need to be aligned - * (bit-endianness must match byte-endianness). Processors typically write - * the bits within a byte in the same order that the bytes of a word are - * written in. So if the host CPU is big-endian, then only big-endian - * samples will be written to STX properly. - */ -#ifdef __BIG_ENDIAN -#define FSLSSI_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \ - SNDRV_PCM_FMTBIT_S18_3BE | SNDRV_PCM_FMTBIT_S20_3BE | \ - SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_S24_BE) -#else -#define FSLSSI_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE | \ - SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE) -#endif - -/* SIER bitflag of interrupts to enable */ -#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \ - CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \ - CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \ - CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \ - CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN) - -/** - * fsl_ssi_private: per-SSI private data - * - * @ssi: pointer to the SSI's registers - * @ssi_phys: physical address of the SSI registers - * @irq: IRQ of this SSI - * @first_stream: pointer to the stream that was opened first - * @second_stream: pointer to second stream - * @playback: the number of playback streams opened - * @capture: the number of capture streams opened - * @cpu_dai: the CPU DAI for this device - * @dev_attr: the sysfs device attribute structure - * @stats: SSI statistics - * @name: name for this device - */ -struct fsl_ssi_private { - struct ccsr_ssi __iomem *ssi; - dma_addr_t ssi_phys; - unsigned int irq; - struct snd_pcm_substream *first_stream; - struct snd_pcm_substream *second_stream; - unsigned int fifo_depth; - struct snd_soc_dai_driver cpu_dai_drv; - struct device_attribute dev_attr; - struct platform_device *pdev; - - struct { - unsigned int rfrc; - unsigned int tfrc; - unsigned int cmdau; - unsigned int cmddu; - unsigned int rxt; - unsigned int rdr1; - unsigned int rdr0; - unsigned int tde1; - unsigned int tde0; - unsigned int roe1; - unsigned int roe0; - unsigned int tue1; - unsigned int tue0; - unsigned int tfs; - unsigned int rfs; - unsigned int tls; - unsigned int rls; - unsigned int rff1; - unsigned int rff0; - unsigned int tfe1; - unsigned int tfe0; - } stats; - - char name[1]; -}; - -/** - * fsl_ssi_isr: SSI interrupt handler - * - * Although it's possible to use the interrupt handler to send and receive - * data to/from the SSI, we use the DMA instead. Programming is more - * complicated, but the performance is much better. - * - * This interrupt handler is used only to gather statistics. - * - * @irq: IRQ of the SSI device - * @dev_id: pointer to the ssi_private structure for this SSI device - */ -static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) -{ - struct fsl_ssi_private *ssi_private = dev_id; - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - irqreturn_t ret = IRQ_NONE; - __be32 sisr; - __be32 sisr2 = 0; - - /* We got an interrupt, so read the status register to see what we - were interrupted for. We mask it with the Interrupt Enable register - so that we only check for events that we're interested in. - */ - sisr = in_be32(&ssi->sisr) & SIER_FLAGS; - - if (sisr & CCSR_SSI_SISR_RFRC) { - ssi_private->stats.rfrc++; - sisr2 |= CCSR_SSI_SISR_RFRC; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_TFRC) { - ssi_private->stats.tfrc++; - sisr2 |= CCSR_SSI_SISR_TFRC; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_CMDAU) { - ssi_private->stats.cmdau++; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_CMDDU) { - ssi_private->stats.cmddu++; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_RXT) { - ssi_private->stats.rxt++; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_RDR1) { - ssi_private->stats.rdr1++; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_RDR0) { - ssi_private->stats.rdr0++; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_TDE1) { - ssi_private->stats.tde1++; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_TDE0) { - ssi_private->stats.tde0++; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_ROE1) { - ssi_private->stats.roe1++; - sisr2 |= CCSR_SSI_SISR_ROE1; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_ROE0) { - ssi_private->stats.roe0++; - sisr2 |= CCSR_SSI_SISR_ROE0; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_TUE1) { - ssi_private->stats.tue1++; - sisr2 |= CCSR_SSI_SISR_TUE1; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_TUE0) { - ssi_private->stats.tue0++; - sisr2 |= CCSR_SSI_SISR_TUE0; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_TFS) { - ssi_private->stats.tfs++; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_RFS) { - ssi_private->stats.rfs++; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_TLS) { - ssi_private->stats.tls++; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_RLS) { - ssi_private->stats.rls++; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_RFF1) { - ssi_private->stats.rff1++; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_RFF0) { - ssi_private->stats.rff0++; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_TFE1) { - ssi_private->stats.tfe1++; - ret = IRQ_HANDLED; - } - - if (sisr & CCSR_SSI_SISR_TFE0) { - ssi_private->stats.tfe0++; - ret = IRQ_HANDLED; - } - - /* Clear the bits that we set */ - if (sisr2) - out_be32(&ssi->sisr, sisr2); - - return ret; -} - -/** - * fsl_ssi_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 SSI registers. - */ -static int fsl_ssi_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = - snd_soc_dai_get_drvdata(rtd->cpu_dai); - int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; - - /* - * If this is the first stream opened, then request the IRQ - * and initialize the SSI registers. - */ - if (!ssi_private->first_stream) { - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - - ssi_private->first_stream = substream; - - /* - * Section 16.5 of the MPC8610 reference manual says that the - * SSI needs to be disabled before updating the registers we set - * here. - */ - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); - - /* - * Program the SSI into I2S Slave Non-Network Synchronous mode. - * Also enable the transmit and receive FIFO. - * - * FIXME: Little-endian samples require a different shift dir - */ - clrsetbits_be32(&ssi->scr, - CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, - CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE - | (synchronous ? CCSR_SSI_SCR_SYN : 0)); - - out_be32(&ssi->stcr, - CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | - CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS | - CCSR_SSI_STCR_TSCKP); - - out_be32(&ssi->srcr, - CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 | - CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS | - CCSR_SSI_SRCR_RSCKP); - - /* - * The DC and PM bits are only used if the SSI is the clock - * master. - */ - - /* Enable the interrupts and DMA requests */ - out_be32(&ssi->sier, SIER_FLAGS); - - /* - * Set the watermark for transmit FIFI 0 and receive FIFO 0. We - * don't use FIFO 1. We program the transmit water to signal a - * DMA transfer if there are only two (or fewer) elements left - * in the FIFO. Two elements equals one frame (left channel, - * right channel). This value, however, depends on the depth of - * the transmit buffer. - * - * We program the receive FIFO to notify us if at least two - * elements (one frame) have been written to the FIFO. We could - * make this value larger (and maybe we should), but this way - * data will be written to memory as soon as it's available. - */ - out_be32(&ssi->sfcsr, - CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) | - CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2)); - - /* - * We keep the SSI disabled because if we enable it, then the - * DMA controller will start. It's not supposed to start until - * the SCR.TE (or SCR.RE) bit is set, but it does anyway. The - * DMA controller will transfer one "BWC" of data (i.e. the - * amount of data that the MR.BWC bits are set to). The reason - * this is bad is because at this point, the PCM driver has not - * finished initializing the DMA controller. - */ - } else { - if (synchronous) { - struct snd_pcm_runtime *first_runtime = - ssi_private->first_stream->runtime; - /* - * This is the second stream open, and we're in - * synchronous mode, so we need to impose sample - * sample size constraints. This is because STCCR is - * used for playback and capture in synchronous mode, - * so there's no way to specify different word - * lengths. - * - * Note that this can cause a race condition if the - * second stream is opened before the first stream is - * fully initialized. We provide some protection by - * checking to make sure the first stream is - * initialized, but it's not perfect. ALSA sometimes - * re-initializes the driver with a different sample - * rate or size. If the second stream is opened - * before the first stream has received its final - * parameters, then the second stream may be - * constrained to the wrong sample rate or size. - */ - if (!first_runtime->sample_bits) { - dev_err(substream->pcm->card->dev, - "set sample size in %s stream first\n", - substream->stream == - SNDRV_PCM_STREAM_PLAYBACK - ? "capture" : "playback"); - return -EAGAIN; - } - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - first_runtime->sample_bits, - first_runtime->sample_bits); - } - - ssi_private->second_stream = substream; - } - - return 0; -} - -/** - * fsl_ssi_hw_params - program the sample size - * - * Most of the SSI registers have been programmed in the startup function, - * but the word length must be programmed here. Unfortunately, programming - * the SxCCR.WL bits requires the SSI to be temporarily disabled. This can - * cause a problem with supporting simultaneous playback and capture. If - * the SSI is already playing a stream, then that stream may be temporarily - * stopped when you start capture. - * - * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the - * clock master. - */ -static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *cpu_dai) -{ - struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - unsigned int sample_size = - snd_pcm_format_width(params_format(hw_params)); - u32 wl = CCSR_SSI_SxCCR_WL(sample_size); - int enabled = in_be32(&ssi->scr) & CCSR_SSI_SCR_SSIEN; - - /* - * If we're in synchronous mode, and the SSI is already enabled, - * then STCCR is already set properly. - */ - if (enabled && ssi_private->cpu_dai_drv.symmetric_rates) - return 0; - - /* - * FIXME: The documentation says that SxCCR[WL] should not be - * modified while the SSI is enabled. The only time this can - * happen is if we're trying to do simultaneous playback and - * capture in asynchronous mode. Unfortunately, I have been enable - * to get that to work at all on the P1022DS. Therefore, we don't - * bother to disable/enable the SSI when setting SxCCR[WL], because - * the SSI will stop anyway. Maybe one day, this will get fixed. - */ - - /* In synchronous mode, the SSI uses STCCR for capture */ - if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) || - ssi_private->cpu_dai_drv.symmetric_rates) - clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); - else - clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); - - return 0; -} - -/** - * fsl_ssi_trigger: start and stop the DMA transfer. - * - * This function is called by ALSA to start, stop, pause, and resume the DMA - * transfer of data. - * - * The DMA channel is in external master start and pause mode, which - * means the SSI completely controls the flow of data. - */ -static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - setbits32(&ssi->scr, - CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); - else - setbits32(&ssi->scr, - CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - clrbits32(&ssi->scr, CCSR_SSI_SCR_TE); - else - clrbits32(&ssi->scr, CCSR_SSI_SCR_RE); - break; - - default: - return -EINVAL; - } - - return 0; -} - -/** - * fsl_ssi_shutdown: shutdown the SSI - * - * Shutdown the SSI if there are no other substreams open. - */ -static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); - - if (ssi_private->first_stream == substream) - ssi_private->first_stream = ssi_private->second_stream; - - ssi_private->second_stream = NULL; - - /* - * If this is the last active substream, disable the SSI. - */ - if (!ssi_private->first_stream) { - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); - } -} - -static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { - .startup = fsl_ssi_startup, - .hw_params = fsl_ssi_hw_params, - .shutdown = fsl_ssi_shutdown, - .trigger = fsl_ssi_trigger, -}; - -/* Template for the CPU dai driver structure */ -static struct snd_soc_dai_driver fsl_ssi_dai_template = { - .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, - .channels_max = 2, - .rates = FSLSSI_I2S_RATES, - .formats = FSLSSI_I2S_FORMATS, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = FSLSSI_I2S_RATES, - .formats = FSLSSI_I2S_FORMATS, - }, - .ops = &fsl_ssi_dai_ops, -}; - -/* Show the statistics of a flag only if its interrupt is enabled. The - * compiler will optimze this code to a no-op if the interrupt is not - * enabled. - */ -#define SIER_SHOW(flag, name) \ - do { \ - if (SIER_FLAGS & CCSR_SSI_SIER_##flag) \ - length += sprintf(buf + length, #name "=%u\n", \ - ssi_private->stats.name); \ - } while (0) - - -/** - * fsl_sysfs_ssi_show: display SSI statistics - * - * Display the statistics for the current SSI device. To avoid confusion, - * we only show those counts that are enabled. - */ -static ssize_t fsl_sysfs_ssi_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - struct fsl_ssi_private *ssi_private = - container_of(attr, struct fsl_ssi_private, dev_attr); - ssize_t length = 0; - - SIER_SHOW(RFRC_EN, rfrc); - SIER_SHOW(TFRC_EN, tfrc); - SIER_SHOW(CMDAU_EN, cmdau); - SIER_SHOW(CMDDU_EN, cmddu); - SIER_SHOW(RXT_EN, rxt); - SIER_SHOW(RDR1_EN, rdr1); - SIER_SHOW(RDR0_EN, rdr0); - SIER_SHOW(TDE1_EN, tde1); - SIER_SHOW(TDE0_EN, tde0); - SIER_SHOW(ROE1_EN, roe1); - SIER_SHOW(ROE0_EN, roe0); - SIER_SHOW(TUE1_EN, tue1); - SIER_SHOW(TUE0_EN, tue0); - SIER_SHOW(TFS_EN, tfs); - SIER_SHOW(RFS_EN, rfs); - SIER_SHOW(TLS_EN, tls); - SIER_SHOW(RLS_EN, rls); - SIER_SHOW(RFF1_EN, rff1); - SIER_SHOW(RFF0_EN, rff0); - SIER_SHOW(TFE1_EN, tfe1); - SIER_SHOW(TFE0_EN, tfe0); - - return length; -} - -/** - * Make every character in a string lower-case - */ -static void make_lowercase(char *s) -{ - char *p = s; - char c; - - while ((c = *p)) { - if ((c >= 'A') && (c <= 'Z')) - *p = c + ('a' - 'A'); - p++; - } -} - -static int __devinit fsl_ssi_probe(struct platform_device *pdev) -{ - struct fsl_ssi_private *ssi_private; - int ret = 0; - struct device_attribute *dev_attr = NULL; - struct device_node *np = pdev->dev.of_node; - const char *p, *sprop; - const uint32_t *iprop; - struct resource res; - char name[64]; - - /* SSIs that are not connected on the board should have a - * status = "disabled" - * property in their device tree nodes. - */ - if (!of_device_is_available(np)) - return -ENODEV; - - /* Check for a codec-handle property. */ - if (!of_get_property(np, "codec-handle", NULL)) { - dev_err(&pdev->dev, "missing codec-handle property\n"); - return -ENODEV; - } - - /* We only support the SSI in "I2S Slave" mode */ - sprop = of_get_property(np, "fsl,mode", NULL); - if (!sprop || strcmp(sprop, "i2s-slave")) { - dev_notice(&pdev->dev, "mode %s is unsupported\n", sprop); - return -ENODEV; - } - - /* The DAI name is the last part of the full name of the node. */ - p = strrchr(np->full_name, '/') + 1; - ssi_private = kzalloc(sizeof(struct fsl_ssi_private) + strlen(p), - GFP_KERNEL); - if (!ssi_private) { - dev_err(&pdev->dev, "could not allocate DAI object\n"); - return -ENOMEM; - } - - strcpy(ssi_private->name, p); - - /* Initialize this copy of the CPU DAI driver structure */ - memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, - sizeof(fsl_ssi_dai_template)); - ssi_private->cpu_dai_drv.name = ssi_private->name; - - /* Get the addresses and IRQ */ - ret = of_address_to_resource(np, 0, &res); - if (ret) { - dev_err(&pdev->dev, "could not determine device resources\n"); - goto error_kmalloc; - } - ssi_private->ssi = of_iomap(np, 0); - if (!ssi_private->ssi) { - dev_err(&pdev->dev, "could not map device resources\n"); - ret = -ENOMEM; - goto error_kmalloc; - } - ssi_private->ssi_phys = res.start; - - ssi_private->irq = irq_of_parse_and_map(np, 0); - if (ssi_private->irq == NO_IRQ) { - dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); - ret = -ENXIO; - goto error_iomap; - } - - /* The 'name' should not have any slashes in it. */ - ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, ssi_private->name, - ssi_private); - if (ret < 0) { - dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq); - goto error_irqmap; - } - - /* Are the RX and the TX clocks locked? */ - if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) - ssi_private->cpu_dai_drv.symmetric_rates = 1; - - /* Determine the FIFO depth. */ - iprop = of_get_property(np, "fsl,fifo-depth", NULL); - if (iprop) - ssi_private->fifo_depth = be32_to_cpup(iprop); - else - /* Older 8610 DTs didn't have the fifo-depth property */ - ssi_private->fifo_depth = 8; - - /* Initialize the the device_attribute structure */ - dev_attr = &ssi_private->dev_attr; - sysfs_attr_init(&dev_attr->attr); - dev_attr->attr.name = "statistics"; - dev_attr->attr.mode = S_IRUGO; - dev_attr->show = fsl_sysfs_ssi_show; - - ret = device_create_file(&pdev->dev, dev_attr); - if (ret) { - dev_err(&pdev->dev, "could not create sysfs %s file\n", - ssi_private->dev_attr.attr.name); - goto error_irq; - } - - /* Register with ASoC */ - dev_set_drvdata(&pdev->dev, ssi_private); - - ret = snd_soc_register_dai(&pdev->dev, &ssi_private->cpu_dai_drv); - if (ret) { - dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); - goto error_dev; - } - - /* Trigger the machine driver's probe function. The platform driver - * name of the machine driver is taken from /compatible property of the - * device tree. We also pass the address of the CPU DAI driver - * structure. - */ - sprop = of_get_property(of_find_node_by_path("/"), "compatible", NULL); - /* Sometimes the compatible name has a "fsl," prefix, so we strip it. */ - p = strrchr(sprop, ','); - if (p) - sprop = p + 1; - snprintf(name, sizeof(name), "snd-soc-%s", sprop); - make_lowercase(name); - - ssi_private->pdev = - platform_device_register_data(&pdev->dev, name, 0, NULL, 0); - if (IS_ERR(ssi_private->pdev)) { - ret = PTR_ERR(ssi_private->pdev); - dev_err(&pdev->dev, "failed to register platform: %d\n", ret); - goto error_dai; - } - - return 0; - -error_dai: - snd_soc_unregister_dai(&pdev->dev); - -error_dev: - dev_set_drvdata(&pdev->dev, NULL); - device_remove_file(&pdev->dev, dev_attr); - -error_irq: - free_irq(ssi_private->irq, ssi_private); - -error_irqmap: - irq_dispose_mapping(ssi_private->irq); - -error_iomap: - iounmap(ssi_private->ssi); - -error_kmalloc: - kfree(ssi_private); - - return ret; -} - -static int fsl_ssi_remove(struct platform_device *pdev) -{ - struct fsl_ssi_private *ssi_private = dev_get_drvdata(&pdev->dev); - - platform_device_unregister(ssi_private->pdev); - snd_soc_unregister_dai(&pdev->dev); - device_remove_file(&pdev->dev, &ssi_private->dev_attr); - - free_irq(ssi_private->irq, ssi_private); - irq_dispose_mapping(ssi_private->irq); - - kfree(ssi_private); - dev_set_drvdata(&pdev->dev, NULL); - - return 0; -} - -static const struct of_device_id fsl_ssi_ids[] = { - { .compatible = "fsl,mpc8610-ssi", }, - {} -}; -MODULE_DEVICE_TABLE(of, fsl_ssi_ids); - -static struct platform_driver fsl_ssi_driver = { - .driver = { - .name = "fsl-ssi-dai", - .owner = THIS_MODULE, - .of_match_table = fsl_ssi_ids, - }, - .probe = fsl_ssi_probe, - .remove = fsl_ssi_remove, -}; - -module_platform_driver(fsl_ssi_driver); - -MODULE_AUTHOR("Timur Tabi <timur@freescale.com>"); -MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver"); -MODULE_LICENSE("GPL v2"); diff --git a/ANDROID_3.4.5/sound/soc/fsl/fsl_ssi.h b/ANDROID_3.4.5/sound/soc/fsl/fsl_ssi.h deleted file mode 100644 index 21730002..00000000 --- a/ANDROID_3.4.5/sound/soc/fsl/fsl_ssi.h +++ /dev/null @@ -1,200 +0,0 @@ -/* - * fsl_ssi.h - ALSA SSI interface for the Freescale MPC8610 SoC - * - * Author: Timur Tabi <timur@freescale.com> - * - * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed - * under the terms of the GNU General Public License version 2. This - * program is licensed "as is" without any warranty of any kind, whether - * express or implied. - */ - -#ifndef _MPC8610_I2S_H -#define _MPC8610_I2S_H - -/* SSI Register Map */ -struct ccsr_ssi { - __be32 stx0; /* 0x.0000 - SSI Transmit Data Register 0 */ - __be32 stx1; /* 0x.0004 - SSI Transmit Data Register 1 */ - __be32 srx0; /* 0x.0008 - SSI Receive Data Register 0 */ - __be32 srx1; /* 0x.000C - SSI Receive Data Register 1 */ - __be32 scr; /* 0x.0010 - SSI Control Register */ - __be32 sisr; /* 0x.0014 - SSI Interrupt Status Register Mixed */ - __be32 sier; /* 0x.0018 - SSI Interrupt Enable Register */ - __be32 stcr; /* 0x.001C - SSI Transmit Configuration Register */ - __be32 srcr; /* 0x.0020 - SSI Receive Configuration Register */ - __be32 stccr; /* 0x.0024 - SSI Transmit Clock Control Register */ - __be32 srccr; /* 0x.0028 - SSI Receive Clock Control Register */ - __be32 sfcsr; /* 0x.002C - SSI FIFO Control/Status Register */ - __be32 str; /* 0x.0030 - SSI Test Register */ - __be32 sor; /* 0x.0034 - SSI Option Register */ - __be32 sacnt; /* 0x.0038 - SSI AC97 Control Register */ - __be32 sacadd; /* 0x.003C - SSI AC97 Command Address Register */ - __be32 sacdat; /* 0x.0040 - SSI AC97 Command Data Register */ - __be32 satag; /* 0x.0044 - SSI AC97 Tag Register */ - __be32 stmsk; /* 0x.0048 - SSI Transmit Time Slot Mask Register */ - __be32 srmsk; /* 0x.004C - SSI Receive Time Slot Mask Register */ - __be32 saccst; /* 0x.0050 - SSI AC97 Channel Status Register */ - __be32 saccen; /* 0x.0054 - SSI AC97 Channel Enable Register */ - __be32 saccdis; /* 0x.0058 - SSI AC97 Channel Disable Register */ -}; - -#define CCSR_SSI_SCR_RFR_CLK_DIS 0x00000800 -#define CCSR_SSI_SCR_TFR_CLK_DIS 0x00000400 -#define CCSR_SSI_SCR_TCH_EN 0x00000100 -#define CCSR_SSI_SCR_SYS_CLK_EN 0x00000080 -#define CCSR_SSI_SCR_I2S_MODE_MASK 0x00000060 -#define CCSR_SSI_SCR_I2S_MODE_NORMAL 0x00000000 -#define CCSR_SSI_SCR_I2S_MODE_MASTER 0x00000020 -#define CCSR_SSI_SCR_I2S_MODE_SLAVE 0x00000040 -#define CCSR_SSI_SCR_SYN 0x00000010 -#define CCSR_SSI_SCR_NET 0x00000008 -#define CCSR_SSI_SCR_RE 0x00000004 -#define CCSR_SSI_SCR_TE 0x00000002 -#define CCSR_SSI_SCR_SSIEN 0x00000001 - -#define CCSR_SSI_SISR_RFRC 0x01000000 -#define CCSR_SSI_SISR_TFRC 0x00800000 -#define CCSR_SSI_SISR_CMDAU 0x00040000 -#define CCSR_SSI_SISR_CMDDU 0x00020000 -#define CCSR_SSI_SISR_RXT 0x00010000 -#define CCSR_SSI_SISR_RDR1 0x00008000 -#define CCSR_SSI_SISR_RDR0 0x00004000 -#define CCSR_SSI_SISR_TDE1 0x00002000 -#define CCSR_SSI_SISR_TDE0 0x00001000 -#define CCSR_SSI_SISR_ROE1 0x00000800 -#define CCSR_SSI_SISR_ROE0 0x00000400 -#define CCSR_SSI_SISR_TUE1 0x00000200 -#define CCSR_SSI_SISR_TUE0 0x00000100 -#define CCSR_SSI_SISR_TFS 0x00000080 -#define CCSR_SSI_SISR_RFS 0x00000040 -#define CCSR_SSI_SISR_TLS 0x00000020 -#define CCSR_SSI_SISR_RLS 0x00000010 -#define CCSR_SSI_SISR_RFF1 0x00000008 -#define CCSR_SSI_SISR_RFF0 0x00000004 -#define CCSR_SSI_SISR_TFE1 0x00000002 -#define CCSR_SSI_SISR_TFE0 0x00000001 - -#define CCSR_SSI_SIER_RFRC_EN 0x01000000 -#define CCSR_SSI_SIER_TFRC_EN 0x00800000 -#define CCSR_SSI_SIER_RDMAE 0x00400000 -#define CCSR_SSI_SIER_RIE 0x00200000 -#define CCSR_SSI_SIER_TDMAE 0x00100000 -#define CCSR_SSI_SIER_TIE 0x00080000 -#define CCSR_SSI_SIER_CMDAU_EN 0x00040000 -#define CCSR_SSI_SIER_CMDDU_EN 0x00020000 -#define CCSR_SSI_SIER_RXT_EN 0x00010000 -#define CCSR_SSI_SIER_RDR1_EN 0x00008000 -#define CCSR_SSI_SIER_RDR0_EN 0x00004000 -#define CCSR_SSI_SIER_TDE1_EN 0x00002000 -#define CCSR_SSI_SIER_TDE0_EN 0x00001000 -#define CCSR_SSI_SIER_ROE1_EN 0x00000800 -#define CCSR_SSI_SIER_ROE0_EN 0x00000400 -#define CCSR_SSI_SIER_TUE1_EN 0x00000200 -#define CCSR_SSI_SIER_TUE0_EN 0x00000100 -#define CCSR_SSI_SIER_TFS_EN 0x00000080 -#define CCSR_SSI_SIER_RFS_EN 0x00000040 -#define CCSR_SSI_SIER_TLS_EN 0x00000020 -#define CCSR_SSI_SIER_RLS_EN 0x00000010 -#define CCSR_SSI_SIER_RFF1_EN 0x00000008 -#define CCSR_SSI_SIER_RFF0_EN 0x00000004 -#define CCSR_SSI_SIER_TFE1_EN 0x00000002 -#define CCSR_SSI_SIER_TFE0_EN 0x00000001 - -#define CCSR_SSI_STCR_TXBIT0 0x00000200 -#define CCSR_SSI_STCR_TFEN1 0x00000100 -#define CCSR_SSI_STCR_TFEN0 0x00000080 -#define CCSR_SSI_STCR_TFDIR 0x00000040 -#define CCSR_SSI_STCR_TXDIR 0x00000020 -#define CCSR_SSI_STCR_TSHFD 0x00000010 -#define CCSR_SSI_STCR_TSCKP 0x00000008 -#define CCSR_SSI_STCR_TFSI 0x00000004 -#define CCSR_SSI_STCR_TFSL 0x00000002 -#define CCSR_SSI_STCR_TEFS 0x00000001 - -#define CCSR_SSI_SRCR_RXEXT 0x00000400 -#define CCSR_SSI_SRCR_RXBIT0 0x00000200 -#define CCSR_SSI_SRCR_RFEN1 0x00000100 -#define CCSR_SSI_SRCR_RFEN0 0x00000080 -#define CCSR_SSI_SRCR_RFDIR 0x00000040 -#define CCSR_SSI_SRCR_RXDIR 0x00000020 -#define CCSR_SSI_SRCR_RSHFD 0x00000010 -#define CCSR_SSI_SRCR_RSCKP 0x00000008 -#define CCSR_SSI_SRCR_RFSI 0x00000004 -#define CCSR_SSI_SRCR_RFSL 0x00000002 -#define CCSR_SSI_SRCR_REFS 0x00000001 - -/* STCCR and SRCCR */ -#define CCSR_SSI_SxCCR_DIV2 0x00040000 -#define CCSR_SSI_SxCCR_PSR 0x00020000 -#define CCSR_SSI_SxCCR_WL_SHIFT 13 -#define CCSR_SSI_SxCCR_WL_MASK 0x0001E000 -#define CCSR_SSI_SxCCR_WL(x) \ - (((((x) / 2) - 1) << CCSR_SSI_SxCCR_WL_SHIFT) & CCSR_SSI_SxCCR_WL_MASK) -#define CCSR_SSI_SxCCR_DC_SHIFT 8 -#define CCSR_SSI_SxCCR_DC_MASK 0x00001F00 -#define CCSR_SSI_SxCCR_DC(x) \ - ((((x) - 1) << CCSR_SSI_SxCCR_DC_SHIFT) & CCSR_SSI_SxCCR_DC_MASK) -#define CCSR_SSI_SxCCR_PM_SHIFT 0 -#define CCSR_SSI_SxCCR_PM_MASK 0x000000FF -#define CCSR_SSI_SxCCR_PM(x) \ - ((((x) - 1) << CCSR_SSI_SxCCR_PM_SHIFT) & CCSR_SSI_SxCCR_PM_MASK) - -/* - * The xFCNT bits are read-only, and the xFWM bits are read/write. Use the - * CCSR_SSI_SFCSR_xFCNTy() macros to read the FIFO counters, and use the - * CCSR_SSI_SFCSR_xFWMy() macros to set the watermarks. - */ -#define CCSR_SSI_SFCSR_RFCNT1_SHIFT 28 -#define CCSR_SSI_SFCSR_RFCNT1_MASK 0xF0000000 -#define CCSR_SSI_SFCSR_RFCNT1(x) \ - (((x) & CCSR_SSI_SFCSR_RFCNT1_MASK) >> CCSR_SSI_SFCSR_RFCNT1_SHIFT) -#define CCSR_SSI_SFCSR_TFCNT1_SHIFT 24 -#define CCSR_SSI_SFCSR_TFCNT1_MASK 0x0F000000 -#define CCSR_SSI_SFCSR_TFCNT1(x) \ - (((x) & CCSR_SSI_SFCSR_TFCNT1_MASK) >> CCSR_SSI_SFCSR_TFCNT1_SHIFT) -#define CCSR_SSI_SFCSR_RFWM1_SHIFT 20 -#define CCSR_SSI_SFCSR_RFWM1_MASK 0x00F00000 -#define CCSR_SSI_SFCSR_RFWM1(x) \ - (((x) << CCSR_SSI_SFCSR_RFWM1_SHIFT) & CCSR_SSI_SFCSR_RFWM1_MASK) -#define CCSR_SSI_SFCSR_TFWM1_SHIFT 16 -#define CCSR_SSI_SFCSR_TFWM1_MASK 0x000F0000 -#define CCSR_SSI_SFCSR_TFWM1(x) \ - (((x) << CCSR_SSI_SFCSR_TFWM1_SHIFT) & CCSR_SSI_SFCSR_TFWM1_MASK) -#define CCSR_SSI_SFCSR_RFCNT0_SHIFT 12 -#define CCSR_SSI_SFCSR_RFCNT0_MASK 0x0000F000 -#define CCSR_SSI_SFCSR_RFCNT0(x) \ - (((x) & CCSR_SSI_SFCSR_RFCNT0_MASK) >> CCSR_SSI_SFCSR_RFCNT0_SHIFT) -#define CCSR_SSI_SFCSR_TFCNT0_SHIFT 8 -#define CCSR_SSI_SFCSR_TFCNT0_MASK 0x00000F00 -#define CCSR_SSI_SFCSR_TFCNT0(x) \ - (((x) & CCSR_SSI_SFCSR_TFCNT0_MASK) >> CCSR_SSI_SFCSR_TFCNT0_SHIFT) -#define CCSR_SSI_SFCSR_RFWM0_SHIFT 4 -#define CCSR_SSI_SFCSR_RFWM0_MASK 0x000000F0 -#define CCSR_SSI_SFCSR_RFWM0(x) \ - (((x) << CCSR_SSI_SFCSR_RFWM0_SHIFT) & CCSR_SSI_SFCSR_RFWM0_MASK) -#define CCSR_SSI_SFCSR_TFWM0_SHIFT 0 -#define CCSR_SSI_SFCSR_TFWM0_MASK 0x0000000F -#define CCSR_SSI_SFCSR_TFWM0(x) \ - (((x) << CCSR_SSI_SFCSR_TFWM0_SHIFT) & CCSR_SSI_SFCSR_TFWM0_MASK) - -#define CCSR_SSI_STR_TEST 0x00008000 -#define CCSR_SSI_STR_RCK2TCK 0x00004000 -#define CCSR_SSI_STR_RFS2TFS 0x00002000 -#define CCSR_SSI_STR_RXSTATE(x) (((x) >> 8) & 0x1F) -#define CCSR_SSI_STR_TXD2RXD 0x00000080 -#define CCSR_SSI_STR_TCK2RCK 0x00000040 -#define CCSR_SSI_STR_TFS2RFS 0x00000020 -#define CCSR_SSI_STR_TXSTATE(x) ((x) & 0x1F) - -#define CCSR_SSI_SOR_CLKOFF 0x00000040 -#define CCSR_SSI_SOR_RX_CLR 0x00000020 -#define CCSR_SSI_SOR_TX_CLR 0x00000010 -#define CCSR_SSI_SOR_INIT 0x00000008 -#define CCSR_SSI_SOR_WAIT_SHIFT 1 -#define CCSR_SSI_SOR_WAIT_MASK 0x00000006 -#define CCSR_SSI_SOR_WAIT(x) (((x) & 3) << CCSR_SSI_SOR_WAIT_SHIFT) -#define CCSR_SSI_SOR_SYNRST 0x00000001 - -#endif - diff --git a/ANDROID_3.4.5/sound/soc/fsl/mpc5200_dma.c b/ANDROID_3.4.5/sound/soc/fsl/mpc5200_dma.c deleted file mode 100644 index 9a3f7c5a..00000000 --- a/ANDROID_3.4.5/sound/soc/fsl/mpc5200_dma.c +++ /dev/null @@ -1,534 +0,0 @@ -/* - * Freescale MPC5200 PSC DMA - * ALSA SoC Platform driver - * - * Copyright (C) 2008 Secret Lab Technologies Ltd. - * Copyright (C) 2009 Jon Smirl, Digispeaker - */ - -#include <linux/module.h> -#include <linux/of_device.h> -#include <linux/dma-mapping.h> -#include <linux/slab.h> -#include <linux/of_platform.h> - -#include <sound/soc.h> - -#include <sysdev/bestcomm/bestcomm.h> -#include <sysdev/bestcomm/gen_bd.h> -#include <asm/mpc52xx_psc.h> - -#include "mpc5200_dma.h" - -/* - * Interrupt handlers - */ -static irqreturn_t psc_dma_status_irq(int irq, void *_psc_dma) -{ - struct psc_dma *psc_dma = _psc_dma; - struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; - u16 isr; - - isr = in_be16(®s->mpc52xx_psc_isr); - - /* Playback underrun error */ - if (psc_dma->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP)) - psc_dma->stats.underrun_count++; - - /* Capture overrun error */ - if (psc_dma->capture.active && (isr & MPC52xx_PSC_IMR_ORERR)) - psc_dma->stats.overrun_count++; - - out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT); - - return IRQ_HANDLED; -} - -/** - * psc_dma_bcom_enqueue_next_buffer - Enqueue another audio buffer - * @s: pointer to stream private data structure - * - * Enqueues another audio period buffer into the bestcomm queue. - * - * Note: The routine must only be called when there is space available in - * the queue. Otherwise the enqueue will fail and the audio ring buffer - * will get out of sync - */ -static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) -{ - struct bcom_bd *bd; - - /* Prepare and enqueue the next buffer descriptor */ - bd = bcom_prepare_next_buffer(s->bcom_task); - bd->status = s->period_bytes; - bd->data[0] = s->runtime->dma_addr + (s->period_next * s->period_bytes); - bcom_submit_next_buffer(s->bcom_task, NULL); - - /* Update for next period */ - s->period_next = (s->period_next + 1) % s->runtime->periods; -} - -/* Bestcomm DMA irq handler */ -static irqreturn_t psc_dma_bcom_irq(int irq, void *_psc_dma_stream) -{ - struct psc_dma_stream *s = _psc_dma_stream; - - spin_lock(&s->psc_dma->lock); - /* For each finished period, dequeue the completed period buffer - * and enqueue a new one in it's place. */ - while (bcom_buffer_done(s->bcom_task)) { - bcom_retrieve_buffer(s->bcom_task, NULL, NULL); - - s->period_current = (s->period_current+1) % s->runtime->periods; - s->period_count++; - - psc_dma_bcom_enqueue_next_buffer(s); - } - spin_unlock(&s->psc_dma->lock); - - /* If the stream is active, then also inform the PCM middle layer - * of the period finished event. */ - if (s->active) - snd_pcm_period_elapsed(s->stream); - - return IRQ_HANDLED; -} - -static int psc_dma_hw_free(struct snd_pcm_substream *substream) -{ - snd_pcm_set_runtime_buffer(substream, NULL); - return 0; -} - -/** - * psc_dma_trigger: start and stop the DMA transfer. - * - * This function is called by ALSA to start, stop, pause, and resume the DMA - * transfer of data. - */ -static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct snd_pcm_runtime *runtime = substream->runtime; - struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma); - struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; - u16 imr; - unsigned long flags; - int i; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - dev_dbg(psc_dma->dev, "START: stream=%i fbits=%u ps=%u #p=%u\n", - substream->pstr->stream, runtime->frame_bits, - (int)runtime->period_size, runtime->periods); - s->period_bytes = frames_to_bytes(runtime, - runtime->period_size); - s->period_next = 0; - s->period_current = 0; - s->active = 1; - s->period_count = 0; - s->runtime = runtime; - - /* Fill up the bestcomm bd queue and enable DMA. - * This will begin filling the PSC's fifo. - */ - spin_lock_irqsave(&psc_dma->lock, flags); - - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - bcom_gen_bd_rx_reset(s->bcom_task); - else - bcom_gen_bd_tx_reset(s->bcom_task); - - for (i = 0; i < runtime->periods; i++) - if (!bcom_queue_full(s->bcom_task)) - psc_dma_bcom_enqueue_next_buffer(s); - - bcom_enable(s->bcom_task); - spin_unlock_irqrestore(&psc_dma->lock, flags); - - out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT); - - break; - - case SNDRV_PCM_TRIGGER_STOP: - dev_dbg(psc_dma->dev, "STOP: stream=%i periods_count=%i\n", - substream->pstr->stream, s->period_count); - s->active = 0; - - spin_lock_irqsave(&psc_dma->lock, flags); - bcom_disable(s->bcom_task); - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - bcom_gen_bd_rx_reset(s->bcom_task); - else - bcom_gen_bd_tx_reset(s->bcom_task); - spin_unlock_irqrestore(&psc_dma->lock, flags); - - break; - - default: - dev_dbg(psc_dma->dev, "unhandled trigger: stream=%i cmd=%i\n", - substream->pstr->stream, cmd); - return -EINVAL; - } - - /* Update interrupt enable settings */ - imr = 0; - if (psc_dma->playback.active) - imr |= MPC52xx_PSC_IMR_TXEMP; - if (psc_dma->capture.active) - imr |= MPC52xx_PSC_IMR_ORERR; - out_be16(®s->isr_imr.imr, psc_dma->imr | imr); - - return 0; -} - - -/* --------------------------------------------------------------------- - * The PSC DMA 'ASoC platform' driver - * - * Can be referenced by an 'ASoC machine' driver - * This driver only deals with the audio bus; it doesn't have any - * interaction with the attached codec - */ - -static const struct snd_pcm_hardware psc_dma_hardware = { - .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_BATCH, - .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | - SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 1, - .channels_max = 2, - .period_bytes_max = 1024 * 1024, - .period_bytes_min = 32, - .periods_min = 2, - .periods_max = 256, - .buffer_bytes_max = 2 * 1024 * 1024, - .fifo_size = 512, -}; - -static int psc_dma_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct psc_dma_stream *s; - int rc; - - dev_dbg(psc_dma->dev, "psc_dma_open(substream=%p)\n", substream); - - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - s = &psc_dma->capture; - else - s = &psc_dma->playback; - - snd_soc_set_runtime_hwparams(substream, &psc_dma_hardware); - - rc = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (rc < 0) { - dev_err(substream->pcm->card->dev, "invalid buffer size\n"); - return rc; - } - - s->stream = substream; - return 0; -} - -static int psc_dma_close(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct psc_dma_stream *s; - - dev_dbg(psc_dma->dev, "psc_dma_close(substream=%p)\n", substream); - - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - s = &psc_dma->capture; - else - s = &psc_dma->playback; - - if (!psc_dma->playback.active && - !psc_dma->capture.active) { - - /* Disable all interrupts and reset the PSC */ - out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); - out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */ - } - s->stream = NULL; - return 0; -} - -static snd_pcm_uframes_t -psc_dma_pointer(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct psc_dma_stream *s; - dma_addr_t count; - - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - s = &psc_dma->capture; - else - s = &psc_dma->playback; - - count = s->period_current * s->period_bytes; - - return bytes_to_frames(substream->runtime, count); -} - -static int -psc_dma_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - return 0; -} - -static struct snd_pcm_ops psc_dma_ops = { - .open = psc_dma_open, - .close = psc_dma_close, - .hw_free = psc_dma_hw_free, - .ioctl = snd_pcm_lib_ioctl, - .pointer = psc_dma_pointer, - .trigger = psc_dma_trigger, - .hw_params = psc_dma_hw_params, -}; - -static u64 psc_dma_dmamask = DMA_BIT_MASK(32); -static int psc_dma_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; - struct snd_pcm *pcm = rtd->pcm; - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); - size_t size = psc_dma_hardware.buffer_bytes_max; - int rc = 0; - - dev_dbg(rtd->platform->dev, "psc_dma_new(card=%p, dai=%p, pcm=%p)\n", - card, dai, pcm); - - if (!card->dev->dma_mask) - card->dev->dma_mask = &psc_dma_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev, - size, &pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->dma_buffer); - if (rc) - goto playback_alloc_err; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev, - size, &pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->dma_buffer); - if (rc) - goto capture_alloc_err; - } - - if (rtd->codec->ac97) - rtd->codec->ac97->private_data = psc_dma; - - return 0; - - capture_alloc_err: - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) - snd_dma_free_pages(&pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->dma_buffer); - - playback_alloc_err: - dev_err(card->dev, "Cannot allocate buffer(s)\n"); - - return -ENOMEM; -} - -static void psc_dma_free(struct snd_pcm *pcm) -{ - struct snd_soc_pcm_runtime *rtd = pcm->private_data; - struct snd_pcm_substream *substream; - int stream; - - dev_dbg(rtd->platform->dev, "psc_dma_free(pcm=%p)\n", pcm); - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (substream) { - snd_dma_free_pages(&substream->dma_buffer); - substream->dma_buffer.area = NULL; - substream->dma_buffer.addr = 0; - } - } -} - -static struct snd_soc_platform_driver mpc5200_audio_dma_platform = { - .ops = &psc_dma_ops, - .pcm_new = &psc_dma_new, - .pcm_free = &psc_dma_free, -}; - -static int mpc5200_hpcd_probe(struct platform_device *op) -{ - phys_addr_t fifo; - struct psc_dma *psc_dma; - struct resource res; - int size, irq, rc; - const __be32 *prop; - void __iomem *regs; - int ret; - - /* Fetch the registers and IRQ of the PSC */ - irq = irq_of_parse_and_map(op->dev.of_node, 0); - if (of_address_to_resource(op->dev.of_node, 0, &res)) { - dev_err(&op->dev, "Missing reg property\n"); - return -ENODEV; - } - regs = ioremap(res.start, resource_size(&res)); - if (!regs) { - dev_err(&op->dev, "Could not map registers\n"); - return -ENODEV; - } - - /* Allocate and initialize the driver private data */ - psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL); - if (!psc_dma) { - ret = -ENOMEM; - goto out_unmap; - } - - /* Get the PSC ID */ - prop = of_get_property(op->dev.of_node, "cell-index", &size); - if (!prop || size < sizeof *prop) { - ret = -ENODEV; - goto out_free; - } - - spin_lock_init(&psc_dma->lock); - mutex_init(&psc_dma->mutex); - psc_dma->id = be32_to_cpu(*prop); - psc_dma->irq = irq; - psc_dma->psc_regs = regs; - psc_dma->fifo_regs = regs + sizeof *psc_dma->psc_regs; - psc_dma->dev = &op->dev; - psc_dma->playback.psc_dma = psc_dma; - psc_dma->capture.psc_dma = psc_dma; - snprintf(psc_dma->name, sizeof psc_dma->name, "PSC%u", psc_dma->id); - - /* Find the address of the fifo data registers and setup the - * DMA tasks */ - fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32); - psc_dma->capture.bcom_task = - bcom_psc_gen_bd_rx_init(psc_dma->id, 10, fifo, 512); - psc_dma->playback.bcom_task = - bcom_psc_gen_bd_tx_init(psc_dma->id, 10, fifo); - if (!psc_dma->capture.bcom_task || - !psc_dma->playback.bcom_task) { - dev_err(&op->dev, "Could not allocate bestcomm tasks\n"); - ret = -ENODEV; - goto out_free; - } - - /* Disable all interrupts and reset the PSC */ - out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); - /* reset receiver */ - out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_RX); - /* reset transmitter */ - out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_TX); - /* reset error */ - out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_ERR_STAT); - /* reset mode */ - out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_SEL_MODE_REG_1); - - /* Set up mode register; - * First write: RxRdy (FIFO Alarm) generates rx FIFO irq - * Second write: register Normal mode for non loopback - */ - out_8(&psc_dma->psc_regs->mode, 0); - out_8(&psc_dma->psc_regs->mode, 0); - - /* Set the TX and RX fifo alarm thresholds */ - out_be16(&psc_dma->fifo_regs->rfalarm, 0x100); - out_8(&psc_dma->fifo_regs->rfcntl, 0x4); - out_be16(&psc_dma->fifo_regs->tfalarm, 0x100); - out_8(&psc_dma->fifo_regs->tfcntl, 0x7); - - /* Lookup the IRQ numbers */ - psc_dma->playback.irq = - bcom_get_task_irq(psc_dma->playback.bcom_task); - psc_dma->capture.irq = - bcom_get_task_irq(psc_dma->capture.bcom_task); - - rc = request_irq(psc_dma->irq, &psc_dma_status_irq, IRQF_SHARED, - "psc-dma-status", psc_dma); - rc |= request_irq(psc_dma->capture.irq, &psc_dma_bcom_irq, IRQF_SHARED, - "psc-dma-capture", &psc_dma->capture); - rc |= request_irq(psc_dma->playback.irq, &psc_dma_bcom_irq, IRQF_SHARED, - "psc-dma-playback", &psc_dma->playback); - if (rc) { - ret = -ENODEV; - goto out_irq; - } - - /* Save what we've done so it can be found again later */ - dev_set_drvdata(&op->dev, psc_dma); - - /* Tell the ASoC OF helpers about it */ - return snd_soc_register_platform(&op->dev, &mpc5200_audio_dma_platform); -out_irq: - free_irq(psc_dma->irq, psc_dma); - free_irq(psc_dma->capture.irq, &psc_dma->capture); - free_irq(psc_dma->playback.irq, &psc_dma->playback); -out_free: - kfree(psc_dma); -out_unmap: - iounmap(regs); - return ret; -} - -static int mpc5200_hpcd_remove(struct platform_device *op) -{ - struct psc_dma *psc_dma = dev_get_drvdata(&op->dev); - - dev_dbg(&op->dev, "mpc5200_audio_dma_destroy()\n"); - - snd_soc_unregister_platform(&op->dev); - - bcom_gen_bd_rx_release(psc_dma->capture.bcom_task); - bcom_gen_bd_tx_release(psc_dma->playback.bcom_task); - - /* Release irqs */ - free_irq(psc_dma->irq, psc_dma); - free_irq(psc_dma->capture.irq, &psc_dma->capture); - free_irq(psc_dma->playback.irq, &psc_dma->playback); - - iounmap(psc_dma->psc_regs); - kfree(psc_dma); - dev_set_drvdata(&op->dev, NULL); - - return 0; -} - -static struct of_device_id mpc5200_hpcd_match[] = { - { .compatible = "fsl,mpc5200-pcm", }, - {} -}; -MODULE_DEVICE_TABLE(of, mpc5200_hpcd_match); - -static struct platform_driver mpc5200_hpcd_of_driver = { - .probe = mpc5200_hpcd_probe, - .remove = mpc5200_hpcd_remove, - .driver = { - .owner = THIS_MODULE, - .name = "mpc5200-pcm-audio", - .of_match_table = mpc5200_hpcd_match, - } -}; - -module_platform_driver(mpc5200_hpcd_of_driver); - -MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>"); -MODULE_DESCRIPTION("Freescale MPC5200 PSC in DMA mode ASoC Driver"); -MODULE_LICENSE("GPL"); diff --git a/ANDROID_3.4.5/sound/soc/fsl/mpc5200_dma.h b/ANDROID_3.4.5/sound/soc/fsl/mpc5200_dma.h deleted file mode 100644 index a3c0cd53..00000000 --- a/ANDROID_3.4.5/sound/soc/fsl/mpc5200_dma.h +++ /dev/null @@ -1,84 +0,0 @@ -/* - * Freescale MPC5200 Audio DMA driver - */ - -#ifndef __SOUND_SOC_FSL_MPC5200_DMA_H__ -#define __SOUND_SOC_FSL_MPC5200_DMA_H__ - -#define PSC_STREAM_NAME_LEN 32 - -/** - * psc_ac97_stream - Data specific to a single stream (playback or capture) - * @active: flag indicating if the stream is active - * @psc_dma: pointer back to parent psc_dma data structure - * @bcom_task: bestcomm task structure - * @irq: irq number for bestcomm task - * @period_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 - * @ac97_slot_bits: Enable bits for turning on the correct AC97 slot - */ -struct psc_dma_stream { - struct snd_pcm_runtime *runtime; - int active; - struct psc_dma *psc_dma; - struct bcom_task *bcom_task; - int irq; - struct snd_pcm_substream *stream; - int period_next; - int period_current; - int period_bytes; - int period_count; - - /* AC97 state */ - u32 ac97_slot_bits; -}; - -/** - * psc_dma - Private driver data - * @name: short name for this device ("PSC0", "PSC1", etc) - * @psc_regs: pointer to the PSC's registers - * @fifo_regs: pointer to the PSC's FIFO registers - * @irq: IRQ of this PSC - * @dev: struct device pointer - * @dai: the CPU DAI for this device - * @sicr: Base value used in serial interface control register; mode is ORed - * with this value. - * @playback: Playback stream context data - * @capture: Capture stream context data - */ -struct psc_dma { - char name[32]; - struct mpc52xx_psc __iomem *psc_regs; - struct mpc52xx_psc_fifo __iomem *fifo_regs; - unsigned int irq; - struct device *dev; - spinlock_t lock; - struct mutex mutex; - u32 sicr; - uint sysclk; - int imr; - int id; - unsigned int slots; - - /* per-stream data */ - struct psc_dma_stream playback; - struct psc_dma_stream capture; - - /* Statistics */ - struct { - unsigned long overrun_count; - unsigned long underrun_count; - } stats; -}; - -/* Utility for retrieving psc_dma_stream structure from a substream */ -static inline struct psc_dma_stream * -to_psc_dma_stream(struct snd_pcm_substream *substream, struct psc_dma *psc_dma) -{ - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - return &psc_dma->capture; - return &psc_dma->playback; -} - -#endif /* __SOUND_SOC_FSL_MPC5200_DMA_H__ */ diff --git a/ANDROID_3.4.5/sound/soc/fsl/mpc5200_psc_ac97.c b/ANDROID_3.4.5/sound/soc/fsl/mpc5200_psc_ac97.c deleted file mode 100644 index ffa00a2e..00000000 --- a/ANDROID_3.4.5/sound/soc/fsl/mpc5200_psc_ac97.c +++ /dev/null @@ -1,333 +0,0 @@ -/* - * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip. - * - * Copyright (C) 2009 Jon Smirl, Digispeaker - * Author: Jon Smirl <jonsmirl@gmail.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/module.h> -#include <linux/of_device.h> -#include <linux/of_platform.h> -#include <linux/delay.h> - -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/time.h> -#include <asm/delay.h> -#include <asm/mpc52xx.h> -#include <asm/mpc52xx_psc.h> - -#include "mpc5200_dma.h" -#include "mpc5200_psc_ac97.h" - -#define DRV_NAME "mpc5200-psc-ac97" - -/* ALSA only supports a single AC97 device so static is recommend here */ -static struct psc_dma *psc_dma; - -static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) -{ - int status; - unsigned int val; - - mutex_lock(&psc_dma->mutex); - - /* Wait for command send status zero = ready */ - status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & - MPC52xx_PSC_SR_CMDSEND), 100, 0); - if (status == 0) { - pr_err("timeout on ac97 bus (rdy)\n"); - mutex_unlock(&psc_dma->mutex); - return -ENODEV; - } - - /* Force clear the data valid bit */ - in_be32(&psc_dma->psc_regs->ac97_data); - - /* Send the read */ - out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); - - /* Wait for the answer */ - status = spin_event_timeout((in_be16(&psc_dma->psc_regs->sr_csr.status) & - MPC52xx_PSC_SR_DATA_VAL), 100, 0); - if (status == 0) { - pr_err("timeout on ac97 read (val) %x\n", - in_be16(&psc_dma->psc_regs->sr_csr.status)); - mutex_unlock(&psc_dma->mutex); - return -ENODEV; - } - /* Get the data */ - val = in_be32(&psc_dma->psc_regs->ac97_data); - if (((val >> 24) & 0x7f) != reg) { - pr_err("reg echo error on ac97 read\n"); - mutex_unlock(&psc_dma->mutex); - return -ENODEV; - } - val = (val >> 8) & 0xffff; - - mutex_unlock(&psc_dma->mutex); - return (unsigned short) val; -} - -static void psc_ac97_write(struct snd_ac97 *ac97, - unsigned short reg, unsigned short val) -{ - int status; - - mutex_lock(&psc_dma->mutex); - - /* Wait for command status zero = ready */ - status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & - MPC52xx_PSC_SR_CMDSEND), 100, 0); - if (status == 0) { - pr_err("timeout on ac97 bus (write)\n"); - goto out; - } - /* Write data */ - out_be32(&psc_dma->psc_regs->ac97_cmd, - ((reg & 0x7f) << 24) | (val << 8)); - - out: - mutex_unlock(&psc_dma->mutex); -} - -static void psc_ac97_warm_reset(struct snd_ac97 *ac97) -{ - struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; - - mutex_lock(&psc_dma->mutex); - - out_be32(®s->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR); - udelay(3); - out_be32(®s->sicr, psc_dma->sicr); - - mutex_unlock(&psc_dma->mutex); -} - -static void psc_ac97_cold_reset(struct snd_ac97 *ac97) -{ - struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; - - mutex_lock(&psc_dma->mutex); - dev_dbg(psc_dma->dev, "cold reset\n"); - - mpc5200_psc_ac97_gpio_reset(psc_dma->id); - - /* Notify the PSC that a reset has occurred */ - out_be32(®s->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_ACRB); - - /* Re-enable RX and TX */ - out_8(®s->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE); - - mutex_unlock(&psc_dma->mutex); - - msleep(1); - psc_ac97_warm_reset(ac97); -} - -struct snd_ac97_bus_ops soc_ac97_ops = { - .read = psc_ac97_read, - .write = psc_ac97_write, - .reset = psc_ac97_cold_reset, - .warm_reset = psc_ac97_warm_reset, -}; -EXPORT_SYMBOL_GPL(soc_ac97_ops); - -static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) -{ - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(cpu_dai); - struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma); - - dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" - " periods=%i buffer_size=%i buffer_bytes=%i channels=%i" - " rate=%i format=%i\n", - __func__, substream, params_period_size(params), - params_period_bytes(params), params_periods(params), - params_buffer_size(params), params_buffer_bytes(params), - params_channels(params), params_rate(params), - params_format(params)); - - /* Determine the set of enable bits to turn on */ - s->ac97_slot_bits = (params_channels(params) == 1) ? 0x100 : 0x300; - if (substream->pstr->stream != SNDRV_PCM_STREAM_CAPTURE) - s->ac97_slot_bits <<= 16; - return 0; -} - -static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) -{ - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(cpu_dai); - - dev_dbg(psc_dma->dev, "%s(substream=%p)\n", __func__, substream); - - if (params_channels(params) == 1) - out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000); - else - out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000); - - return 0; -} - -static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(dai); - struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - dev_dbg(psc_dma->dev, "AC97 START: stream=%i\n", - substream->pstr->stream); - - /* Set the slot enable bits */ - psc_dma->slots |= s->ac97_slot_bits; - out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); - break; - - case SNDRV_PCM_TRIGGER_STOP: - dev_dbg(psc_dma->dev, "AC97 STOP: stream=%i\n", - substream->pstr->stream); - - /* Clear the slot enable bits */ - psc_dma->slots &= ~(s->ac97_slot_bits); - out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); - break; - } - return 0; -} - -static int psc_ac97_probe(struct snd_soc_dai *cpu_dai) -{ - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(cpu_dai); - struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; - - /* Go */ - out_8(®s->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE); - return 0; -} - -/* --------------------------------------------------------------------- - * ALSA SoC Bindings - * - * - Digital Audio Interface (DAI) template - * - create/destroy dai hooks - */ - -/** - * psc_ac97_dai_template: template CPU Digital Audio Interface - */ -static const struct snd_soc_dai_ops psc_ac97_analog_ops = { - .hw_params = psc_ac97_hw_analog_params, - .trigger = psc_ac97_trigger, -}; - -static const struct snd_soc_dai_ops psc_ac97_digital_ops = { - .hw_params = psc_ac97_hw_digital_params, -}; - -static struct snd_soc_dai_driver psc_ac97_dai[] = { -{ - .ac97_control = 1, - .probe = psc_ac97_probe, - .playback = { - .channels_min = 1, - .channels_max = 6, - .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S32_BE, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S32_BE, - }, - .ops = &psc_ac97_analog_ops, -}, -{ - .ac97_control = 1, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_32000 | \ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE, - }, - .ops = &psc_ac97_digital_ops, -} }; - - - -/* --------------------------------------------------------------------- - * OF platform bus binding code: - * - Probe/remove operations - * - OF device match table - */ -static int __devinit psc_ac97_of_probe(struct platform_device *op) -{ - int rc; - struct snd_ac97 ac97; - struct mpc52xx_psc __iomem *regs; - - rc = snd_soc_register_dais(&op->dev, psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai)); - if (rc != 0) { - dev_err(&op->dev, "Failed to register DAI\n"); - return rc; - } - - psc_dma = dev_get_drvdata(&op->dev); - regs = psc_dma->psc_regs; - ac97.private_data = psc_dma; - - psc_dma->imr = 0; - out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); - - /* Configure the serial interface mode to AC97 */ - psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97; - out_be32(®s->sicr, psc_dma->sicr); - - /* No slots active */ - out_be32(®s->ac97_slots, 0x00000000); - - return 0; -} - -static int __devexit psc_ac97_of_remove(struct platform_device *op) -{ - snd_soc_unregister_dais(&op->dev, ARRAY_SIZE(psc_ac97_dai)); - return 0; -} - -/* Match table for of_platform binding */ -static struct of_device_id psc_ac97_match[] __devinitdata = { - { .compatible = "fsl,mpc5200-psc-ac97", }, - { .compatible = "fsl,mpc5200b-psc-ac97", }, - {} -}; -MODULE_DEVICE_TABLE(of, psc_ac97_match); - -static struct platform_driver psc_ac97_driver = { - .probe = psc_ac97_of_probe, - .remove = __devexit_p(psc_ac97_of_remove), - .driver = { - .name = "mpc5200-psc-ac97", - .owner = THIS_MODULE, - .of_match_table = psc_ac97_match, - }, -}; - -module_platform_driver(psc_ac97_driver); - -MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>"); -MODULE_DESCRIPTION("mpc5200 AC97 module"); -MODULE_LICENSE("GPL"); - diff --git a/ANDROID_3.4.5/sound/soc/fsl/mpc5200_psc_ac97.h b/ANDROID_3.4.5/sound/soc/fsl/mpc5200_psc_ac97.h deleted file mode 100644 index e881e784..00000000 --- a/ANDROID_3.4.5/sound/soc/fsl/mpc5200_psc_ac97.h +++ /dev/null @@ -1,13 +0,0 @@ -/* - * Freescale MPC5200 PSC in AC97 mode - * ALSA SoC Digital Audio Interface (DAI) driver - * - */ - -#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ -#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ - -#define MPC5200_AC97_NORMAL 0 -#define MPC5200_AC97_SPDIF 1 - -#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */ diff --git a/ANDROID_3.4.5/sound/soc/fsl/mpc5200_psc_i2s.c b/ANDROID_3.4.5/sound/soc/fsl/mpc5200_psc_i2s.c deleted file mode 100644 index 7b530327..00000000 --- a/ANDROID_3.4.5/sound/soc/fsl/mpc5200_psc_i2s.c +++ /dev/null @@ -1,230 +0,0 @@ -/* - * Freescale MPC5200 PSC in I2S mode - * ALSA SoC Digital Audio Interface (DAI) driver - * - * Copyright (C) 2008 Secret Lab Technologies Ltd. - * Copyright (C) 2009 Jon Smirl, Digispeaker - */ - -#include <linux/module.h> -#include <linux/of_device.h> -#include <linux/of_platform.h> - -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/mpc52xx_psc.h> - -#include "mpc5200_dma.h" - -/** - * 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_S32_BE) - -static int psc_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); - u32 mode; - - dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" - " periods=%i buffer_size=%i buffer_bytes=%i\n", - __func__, substream, params_period_size(params), - params_period_bytes(params), params_periods(params), - 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_dma->dev, "invalid format\n"); - return -EINVAL; - } - out_be32(&psc_dma->psc_regs->sicr, psc_dma->sicr | mode); - - return 0; -} - -/** - * 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_dma *psc_dma = snd_soc_dai_get_drvdata(cpu_dai); - dev_dbg(psc_dma->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n", - cpu_dai, dir); - return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL; -} - -/** - * 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_dma *psc_dma = snd_soc_dai_get_drvdata(cpu_dai); - dev_dbg(psc_dma->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n", - cpu_dai, format); - return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL; -} - -/* --------------------------------------------------------------------- - * ALSA SoC Bindings - * - * - Digital Audio Interface (DAI) template - * - create/destroy dai hooks - */ - -/** - * psc_i2s_dai_template: template CPU Digital Audio Interface - */ -static const struct snd_soc_dai_ops psc_i2s_dai_ops = { - .hw_params = psc_i2s_hw_params, - .set_sysclk = psc_i2s_set_sysclk, - .set_fmt = psc_i2s_set_fmt, -}; - -static struct snd_soc_dai_driver psc_i2s_dai[] = {{ - .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 = &psc_i2s_dai_ops, -} }; - -/* --------------------------------------------------------------------- - * OF platform bus binding code: - * - Probe/remove operations - * - OF device match table - */ -static int __devinit psc_i2s_of_probe(struct platform_device *op) -{ - int rc; - struct psc_dma *psc_dma; - struct mpc52xx_psc __iomem *regs; - - rc = snd_soc_register_dais(&op->dev, psc_i2s_dai, ARRAY_SIZE(psc_i2s_dai)); - if (rc != 0) { - pr_err("Failed to register DAI\n"); - return rc; - } - - psc_dma = dev_get_drvdata(&op->dev); - regs = psc_dma->psc_regs; - - /* Configure the serial interface mode; defaulting to CODEC8 mode */ - psc_dma->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S | - MPC52xx_PSC_SICR_CLKPOL; - out_be32(&psc_dma->psc_regs->sicr, - psc_dma->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8); - - /* Check for the codec handle. If it is not present then we - * are done */ - if (!of_get_property(op->dev.of_node, "codec-handle", NULL)) - return 0; - - /* Due to errata in the dma mode; need to line up enabling - * the transmitter with a transition on the frame sync - * line */ - - /* first make sure it is low */ - while ((in_8(®s->ipcr_acr.ipcr) & 0x80) != 0) - ; - /* then wait for the transition to high */ - while ((in_8(®s->ipcr_acr.ipcr) & 0x80) == 0) - ; - /* Finally, enable the PSC. - * Receiver must always be enabled; even when we only want - * transmit. (see 15.3.2.3 of MPC5200B User's Guide) */ - - /* Go */ - out_8(&psc_dma->psc_regs->command, - MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE); - - return 0; - -} - -static int __devexit psc_i2s_of_remove(struct platform_device *op) -{ - snd_soc_unregister_dais(&op->dev, ARRAY_SIZE(psc_i2s_dai)); - return 0; -} - -/* Match table for of_platform binding */ -static struct of_device_id psc_i2s_match[] __devinitdata = { - { .compatible = "fsl,mpc5200-psc-i2s", }, - { .compatible = "fsl,mpc5200b-psc-i2s", }, - {} -}; -MODULE_DEVICE_TABLE(of, psc_i2s_match); - -static struct platform_driver psc_i2s_driver = { - .probe = psc_i2s_of_probe, - .remove = __devexit_p(psc_i2s_of_remove), - .driver = { - .name = "mpc5200-psc-i2s", - .owner = THIS_MODULE, - .of_match_table = psc_i2s_match, - }, -}; - -module_platform_driver(psc_i2s_driver); - -MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>"); -MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver"); -MODULE_LICENSE("GPL"); - diff --git a/ANDROID_3.4.5/sound/soc/fsl/mpc8610_hpcd.c b/ANDROID_3.4.5/sound/soc/fsl/mpc8610_hpcd.c deleted file mode 100644 index 3fea5a15..00000000 --- a/ANDROID_3.4.5/sound/soc/fsl/mpc8610_hpcd.c +++ /dev/null @@ -1,595 +0,0 @@ -/** - * Freescale MPC8610HPCD ALSA SoC Machine driver - * - * Author: Timur Tabi <timur@freescale.com> - * - * Copyright 2007-2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ - -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/of_device.h> -#include <linux/slab.h> -#include <linux/of_i2c.h> -#include <sound/soc.h> -#include <asm/fsl_guts.h> - -#include "fsl_dma.h" -#include "fsl_ssi.h" - -/* There's only one global utilities register */ -static phys_addr_t guts_phys; - -#define DAI_NAME_SIZE 32 - -/** - * mpc8610_hpcd_data: machine-specific ASoC device data - * - * This structure contains data for a single sound platform device on an - * MPC8610 HPCD. Some of the data is taken from the device tree. - */ -struct mpc8610_hpcd_data { - struct snd_soc_dai_link dai[2]; - struct snd_soc_card card; - unsigned int dai_format; - unsigned int codec_clk_direction; - unsigned int cpu_clk_direction; - unsigned int clk_frequency; - unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ - unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */ - unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ - char codec_dai_name[DAI_NAME_SIZE]; - char codec_name[DAI_NAME_SIZE]; - char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */ -}; - -/** - * mpc8610_hpcd_machine_probe: initialize the board - * - * This function is used to initialize the board-specific hardware. - * - * Here we program the DMACR and PMUXCR registers. - */ -static int mpc8610_hpcd_machine_probe(struct snd_soc_card *card) -{ - struct mpc8610_hpcd_data *machine_data = - container_of(card, struct mpc8610_hpcd_data, card); - struct ccsr_guts __iomem *guts; - - guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); - if (!guts) { - dev_err(card->dev, "could not map global utilities\n"); - return -ENOMEM; - } - - /* Program the signal routing between the SSI and the DMA */ - guts_set_dmacr(guts, machine_data->dma_id[0], - machine_data->dma_channel_id[0], - CCSR_GUTS_DMACR_DEV_SSI); - guts_set_dmacr(guts, machine_data->dma_id[1], - machine_data->dma_channel_id[1], - CCSR_GUTS_DMACR_DEV_SSI); - - guts_set_pmuxcr_dma(guts, machine_data->dma_id[0], - machine_data->dma_channel_id[0], 0); - guts_set_pmuxcr_dma(guts, machine_data->dma_id[1], - machine_data->dma_channel_id[1], 0); - - switch (machine_data->ssi_id) { - case 0: - clrsetbits_be32(&guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_SSI); - break; - case 1: - clrsetbits_be32(&guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_SSI); - break; - } - - iounmap(guts); - - return 0; -} - -/** - * mpc8610_hpcd_startup: program the board with various hardware parameters - * - * This function takes board-specific information, like clock frequencies - * and serial data formats, and passes that information to the codec and - * transport drivers. - */ -static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct mpc8610_hpcd_data *machine_data = - container_of(rtd->card, struct mpc8610_hpcd_data, card); - struct device *dev = rtd->card->dev; - int ret = 0; - - /* Tell the codec driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(rtd->codec_dai, machine_data->dai_format); - if (ret < 0) { - dev_err(dev, "could not set codec driver audio format\n"); - return ret; - } - - /* - * Tell the codec driver what the MCLK frequency is, and whether it's - * a slave or master. - */ - ret = snd_soc_dai_set_sysclk(rtd->codec_dai, 0, - machine_data->clk_frequency, - machine_data->codec_clk_direction); - if (ret < 0) { - dev_err(dev, "could not set codec driver clock params\n"); - return ret; - } - - return 0; -} - -/** - * mpc8610_hpcd_machine_remove: Remove the sound device - * - * This function is called to remove the sound device for one SSI. We - * de-program the DMACR and PMUXCR register. - */ -static int mpc8610_hpcd_machine_remove(struct snd_soc_card *card) -{ - struct mpc8610_hpcd_data *machine_data = - container_of(card, struct mpc8610_hpcd_data, card); - struct ccsr_guts __iomem *guts; - - guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); - if (!guts) { - dev_err(card->dev, "could not map global utilities\n"); - return -ENOMEM; - } - - /* Restore the signal routing */ - - guts_set_dmacr(guts, machine_data->dma_id[0], - machine_data->dma_channel_id[0], 0); - guts_set_dmacr(guts, machine_data->dma_id[1], - machine_data->dma_channel_id[1], 0); - - switch (machine_data->ssi_id) { - case 0: - clrsetbits_be32(&guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); - break; - case 1: - clrsetbits_be32(&guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_LA); - break; - } - - iounmap(guts); - - return 0; -} - -/** - * mpc8610_hpcd_ops: ASoC machine driver operations - */ -static struct snd_soc_ops mpc8610_hpcd_ops = { - .startup = mpc8610_hpcd_startup, -}; - -/** - * get_node_by_phandle_name - get a node by its phandle name - * - * This function takes a node, the name of a property in that node, and a - * compatible string. Assuming the property is a phandle to another node, - * it returns that node, (optionally) if that node is compatible. - * - * If the property is not a phandle, or the node it points to is not compatible - * with the specific string, then NULL is returned. - */ -static struct device_node *get_node_by_phandle_name(struct device_node *np, - const char *name, - const char *compatible) -{ - const phandle *ph; - int len; - - ph = of_get_property(np, name, &len); - if (!ph || (len != sizeof(phandle))) - return NULL; - - np = of_find_node_by_phandle(*ph); - if (!np) - return NULL; - - if (compatible && !of_device_is_compatible(np, compatible)) { - of_node_put(np); - return NULL; - } - - return np; -} - -/** - * get_parent_cell_index -- return the cell-index of the parent of a node - * - * Return the value of the cell-index property of the parent of the given - * node. This is used for DMA channel nodes that need to know the DMA ID - * of the controller they are on. - */ -static int get_parent_cell_index(struct device_node *np) -{ - struct device_node *parent = of_get_parent(np); - const u32 *iprop; - - if (!parent) - return -1; - - iprop = of_get_property(parent, "cell-index", NULL); - of_node_put(parent); - - if (!iprop) - return -1; - - return be32_to_cpup(iprop); -} - -/** - * codec_node_dev_name - determine the dev_name for a codec node - * - * This function determines the dev_name for an I2C node. This is the name - * that would be returned by dev_name() if this device_node were part of a - * 'struct device' It's ugly and hackish, but it works. - * - * The dev_name for such devices include the bus number and I2C address. For - * example, "cs4270.0-004f". - */ -static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) -{ - const u32 *iprop; - int addr; - char temp[DAI_NAME_SIZE]; - struct i2c_client *i2c; - - of_modalias_node(np, temp, DAI_NAME_SIZE); - - iprop = of_get_property(np, "reg", NULL); - if (!iprop) - return -EINVAL; - - addr = be32_to_cpup(iprop); - - /* We need the adapter number */ - i2c = of_find_i2c_device_by_node(np); - if (!i2c) - return -ENODEV; - - snprintf(buf, len, "%s.%u-%04x", temp, i2c->adapter->nr, addr); - - return 0; -} - -static int get_dma_channel(struct device_node *ssi_np, - const char *name, - struct snd_soc_dai_link *dai, - unsigned int *dma_channel_id, - unsigned int *dma_id) -{ - struct resource res; - struct device_node *dma_channel_np; - const u32 *iprop; - int ret; - - dma_channel_np = get_node_by_phandle_name(ssi_np, name, - "fsl,ssi-dma-channel"); - if (!dma_channel_np) - return -EINVAL; - - /* Determine the dev_name for the device_node. This code mimics the - * behavior of of_device_make_bus_id(). We need this because ASoC uses - * the dev_name() of the device to match the platform (DMA) device with - * the CPU (SSI) device. It's all ugly and hackish, but it works (for - * now). - * - * dai->platform name should already point to an allocated buffer. - */ - ret = of_address_to_resource(dma_channel_np, 0, &res); - if (ret) - return ret; - snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", - (unsigned long long) res.start, dma_channel_np->name); - - iprop = of_get_property(dma_channel_np, "cell-index", NULL); - if (!iprop) { - of_node_put(dma_channel_np); - return -EINVAL; - } - - *dma_channel_id = be32_to_cpup(iprop); - *dma_id = get_parent_cell_index(dma_channel_np); - of_node_put(dma_channel_np); - - return 0; -} - -/** - * mpc8610_hpcd_probe: platform probe function for the machine driver - * - * Although this is a machine driver, the SSI node is the "master" node with - * respect to audio hardware connections. Therefore, we create a new ASoC - * device for each new SSI node that has a codec attached. - */ -static int mpc8610_hpcd_probe(struct platform_device *pdev) -{ - struct device *dev = pdev->dev.parent; - /* ssi_pdev is the platform device for the SSI node that probed us */ - struct platform_device *ssi_pdev = - container_of(dev, struct platform_device, dev); - struct device_node *np = ssi_pdev->dev.of_node; - struct device_node *codec_np = NULL; - struct platform_device *sound_device = NULL; - struct mpc8610_hpcd_data *machine_data; - int ret = -ENODEV; - const char *sprop; - const u32 *iprop; - - /* Find the codec node for this SSI. */ - codec_np = of_parse_phandle(np, "codec-handle", 0); - if (!codec_np) { - dev_err(dev, "invalid codec node\n"); - return -EINVAL; - } - - machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); - if (!machine_data) { - ret = -ENOMEM; - goto error_alloc; - } - - machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev); - machine_data->dai[0].ops = &mpc8610_hpcd_ops; - - /* Determine the codec name, it will be used as the codec DAI name */ - ret = codec_node_dev_name(codec_np, machine_data->codec_name, - DAI_NAME_SIZE); - if (ret) { - dev_err(&pdev->dev, "invalid codec node %s\n", - codec_np->full_name); - ret = -EINVAL; - goto error; - } - machine_data->dai[0].codec_name = machine_data->codec_name; - - /* The DAI name from the codec (snd_soc_dai_driver.name) */ - machine_data->dai[0].codec_dai_name = "cs4270-hifi"; - - /* We register two DAIs per SSI, one for playback and the other for - * capture. Currently, we only support codecs that have one DAI for - * both playback and capture. - */ - memcpy(&machine_data->dai[1], &machine_data->dai[0], - sizeof(struct snd_soc_dai_link)); - - /* Get the device ID */ - iprop = of_get_property(np, "cell-index", NULL); - if (!iprop) { - dev_err(&pdev->dev, "cell-index property not found\n"); - ret = -EINVAL; - goto error; - } - machine_data->ssi_id = be32_to_cpup(iprop); - - /* Get the serial format and clock direction. */ - sprop = of_get_property(np, "fsl,mode", NULL); - if (!sprop) { - dev_err(&pdev->dev, "fsl,mode property not found\n"); - ret = -EINVAL; - goto error; - } - - if (strcasecmp(sprop, "i2s-slave") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - - /* In i2s-slave mode, the codec has its own clock source, so we - * need to get the frequency from the device tree and pass it to - * the codec driver. - */ - iprop = of_get_property(codec_np, "clock-frequency", NULL); - if (!iprop || !*iprop) { - dev_err(&pdev->dev, "codec bus-frequency " - "property is missing or invalid\n"); - ret = -EINVAL; - goto error; - } - machine_data->clk_frequency = be32_to_cpup(iprop); - } else if (strcasecmp(sprop, "i2s-master") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "lj-slave") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "lj-master") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "rj-slave") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBM_CFM; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "rj-master") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBS_CFS; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "ac97-slave") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBM_CFM; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "ac97-master") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBS_CFS; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else { - dev_err(&pdev->dev, - "unrecognized fsl,mode property '%s'\n", sprop); - ret = -EINVAL; - goto error; - } - - if (!machine_data->clk_frequency) { - dev_err(&pdev->dev, "unknown clock frequency\n"); - ret = -EINVAL; - goto error; - } - - /* Find the playback DMA channel to use. */ - machine_data->dai[0].platform_name = machine_data->platform_name[0]; - ret = get_dma_channel(np, "fsl,playback-dma", &machine_data->dai[0], - &machine_data->dma_channel_id[0], - &machine_data->dma_id[0]); - if (ret) { - dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n"); - goto error; - } - - /* Find the capture DMA channel to use. */ - machine_data->dai[1].platform_name = machine_data->platform_name[1]; - ret = get_dma_channel(np, "fsl,capture-dma", &machine_data->dai[1], - &machine_data->dma_channel_id[1], - &machine_data->dma_id[1]); - if (ret) { - dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n"); - goto error; - } - - /* Initialize our DAI data structure. */ - machine_data->dai[0].stream_name = "playback"; - machine_data->dai[1].stream_name = "capture"; - machine_data->dai[0].name = machine_data->dai[0].stream_name; - machine_data->dai[1].name = machine_data->dai[1].stream_name; - - machine_data->card.probe = mpc8610_hpcd_machine_probe; - machine_data->card.remove = mpc8610_hpcd_machine_remove; - machine_data->card.name = pdev->name; /* The platform driver name */ - machine_data->card.num_links = 2; - machine_data->card.dai_link = machine_data->dai; - - /* Allocate a new audio platform device structure */ - sound_device = platform_device_alloc("soc-audio", -1); - if (!sound_device) { - dev_err(&pdev->dev, "platform device alloc failed\n"); - ret = -ENOMEM; - goto error; - } - - /* Associate the card data with the sound device */ - platform_set_drvdata(sound_device, &machine_data->card); - - /* Register with ASoC */ - ret = platform_device_add(sound_device); - if (ret) { - dev_err(&pdev->dev, "platform device add failed\n"); - goto error_sound; - } - dev_set_drvdata(&pdev->dev, sound_device); - - of_node_put(codec_np); - - return 0; - -error_sound: - platform_device_put(sound_device); -error: - kfree(machine_data); -error_alloc: - of_node_put(codec_np); - return ret; -} - -/** - * mpc8610_hpcd_remove: remove the platform device - * - * This function is called when the platform device is removed. - */ -static int __devexit mpc8610_hpcd_remove(struct platform_device *pdev) -{ - struct platform_device *sound_device = dev_get_drvdata(&pdev->dev); - struct snd_soc_card *card = platform_get_drvdata(sound_device); - struct mpc8610_hpcd_data *machine_data = - container_of(card, struct mpc8610_hpcd_data, card); - - platform_device_unregister(sound_device); - - kfree(machine_data); - sound_device->dev.platform_data = NULL; - - dev_set_drvdata(&pdev->dev, NULL); - - return 0; -} - -static struct platform_driver mpc8610_hpcd_driver = { - .probe = mpc8610_hpcd_probe, - .remove = __devexit_p(mpc8610_hpcd_remove), - .driver = { - /* The name must match 'compatible' property in the device tree, - * in lowercase letters. - */ - .name = "snd-soc-mpc8610hpcd", - .owner = THIS_MODULE, - }, -}; - -/** - * mpc8610_hpcd_init: machine driver initialization. - * - * This function is called when this module is loaded. - */ -static int __init mpc8610_hpcd_init(void) -{ - struct device_node *guts_np; - struct resource res; - - pr_info("Freescale MPC8610 HPCD ALSA SoC machine driver\n"); - - /* Get the physical address of the global utilities registers */ - guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); - if (of_address_to_resource(guts_np, 0, &res)) { - pr_err("mpc8610-hpcd: missing/invalid global utilities node\n"); - return -EINVAL; - } - guts_phys = res.start; - - return platform_driver_register(&mpc8610_hpcd_driver); -} - -/** - * mpc8610_hpcd_exit: machine driver exit - * - * This function is called when this driver is unloaded. - */ -static void __exit mpc8610_hpcd_exit(void) -{ - platform_driver_unregister(&mpc8610_hpcd_driver); -} - -module_init(mpc8610_hpcd_init); -module_exit(mpc8610_hpcd_exit); - -MODULE_AUTHOR("Timur Tabi <timur@freescale.com>"); -MODULE_DESCRIPTION("Freescale MPC8610 HPCD ALSA SoC machine driver"); -MODULE_LICENSE("GPL v2"); diff --git a/ANDROID_3.4.5/sound/soc/fsl/p1022_ds.c b/ANDROID_3.4.5/sound/soc/fsl/p1022_ds.c deleted file mode 100644 index 982a1c94..00000000 --- a/ANDROID_3.4.5/sound/soc/fsl/p1022_ds.c +++ /dev/null @@ -1,601 +0,0 @@ -/** - * Freescale P1022DS ALSA SoC Machine driver - * - * Author: Timur Tabi <timur@freescale.com> - * - * Copyright 2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ - -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/of_device.h> -#include <linux/slab.h> -#include <linux/of_i2c.h> -#include <sound/soc.h> -#include <asm/fsl_guts.h> - -#include "fsl_dma.h" -#include "fsl_ssi.h" - -/* P1022-specific PMUXCR and DMUXCR bit definitions */ - -#define CCSR_GUTS_PMUXCR_UART0_I2C1_MASK 0x0001c000 -#define CCSR_GUTS_PMUXCR_UART0_I2C1_UART0_SSI 0x00010000 -#define CCSR_GUTS_PMUXCR_UART0_I2C1_SSI 0x00018000 - -#define CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK 0x00000c00 -#define CCSR_GUTS_PMUXCR_SSI_DMA_TDM_SSI 0x00000000 - -#define CCSR_GUTS_DMUXCR_PAD 1 /* DMA controller/channel set to pad */ -#define CCSR_GUTS_DMUXCR_SSI 2 /* DMA controller/channel set to SSI */ - -/* - * Set the DMACR register in the GUTS - * - * The DMACR register determines the source of initiated transfers for each - * channel on each DMA controller. Rather than have a bunch of repetitive - * macros for the bit patterns, we just have a function that calculates - * them. - * - * guts: Pointer to GUTS structure - * co: The DMA controller (0 or 1) - * ch: The channel on the DMA controller (0, 1, 2, or 3) - * device: The device to set as the target (CCSR_GUTS_DMUXCR_xxx) - */ -static inline void guts_set_dmuxcr(struct ccsr_guts __iomem *guts, - unsigned int co, unsigned int ch, unsigned int device) -{ - unsigned int shift = 16 + (8 * (1 - co) + 2 * (3 - ch)); - - clrsetbits_be32(&guts->dmuxcr, 3 << shift, device << shift); -} - -/* There's only one global utilities register */ -static phys_addr_t guts_phys; - -#define DAI_NAME_SIZE 32 - -/** - * machine_data: machine-specific ASoC device data - * - * This structure contains data for a single sound platform device on an - * P1022 DS. Some of the data is taken from the device tree. - */ -struct machine_data { - struct snd_soc_dai_link dai[2]; - struct snd_soc_card card; - unsigned int dai_format; - unsigned int codec_clk_direction; - unsigned int cpu_clk_direction; - unsigned int clk_frequency; - unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ - unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */ - unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ - char codec_name[DAI_NAME_SIZE]; - char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */ -}; - -/** - * p1022_ds_machine_probe: initialize the board - * - * This function is used to initialize the board-specific hardware. - * - * Here we program the DMACR and PMUXCR registers. - */ -static int p1022_ds_machine_probe(struct snd_soc_card *card) -{ - struct machine_data *mdata = - container_of(card, struct machine_data, card); - struct ccsr_guts __iomem *guts; - - guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); - if (!guts) { - dev_err(card->dev, "could not map global utilities\n"); - return -ENOMEM; - } - - /* Enable SSI Tx signal */ - clrsetbits_be32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_UART0_I2C1_MASK, - CCSR_GUTS_PMUXCR_UART0_I2C1_UART0_SSI); - - /* Enable SSI Rx signal */ - clrsetbits_be32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK, - CCSR_GUTS_PMUXCR_SSI_DMA_TDM_SSI); - - /* Enable DMA Channel for SSI */ - guts_set_dmuxcr(guts, mdata->dma_id[0], mdata->dma_channel_id[0], - CCSR_GUTS_DMUXCR_SSI); - - guts_set_dmuxcr(guts, mdata->dma_id[1], mdata->dma_channel_id[1], - CCSR_GUTS_DMUXCR_SSI); - - iounmap(guts); - - return 0; -} - -/** - * p1022_ds_startup: program the board with various hardware parameters - * - * This function takes board-specific information, like clock frequencies - * and serial data formats, and passes that information to the codec and - * transport drivers. - */ -static int p1022_ds_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct machine_data *mdata = - container_of(rtd->card, struct machine_data, card); - struct device *dev = rtd->card->dev; - int ret = 0; - - /* Tell the codec driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(rtd->codec_dai, mdata->dai_format); - if (ret < 0) { - dev_err(dev, "could not set codec driver audio format\n"); - return ret; - } - - /* - * Tell the codec driver what the MCLK frequency is, and whether it's - * a slave or master. - */ - ret = snd_soc_dai_set_sysclk(rtd->codec_dai, 0, mdata->clk_frequency, - mdata->codec_clk_direction); - if (ret < 0) { - dev_err(dev, "could not set codec driver clock params\n"); - return ret; - } - - return 0; -} - -/** - * p1022_ds_machine_remove: Remove the sound device - * - * This function is called to remove the sound device for one SSI. We - * de-program the DMACR and PMUXCR register. - */ -static int p1022_ds_machine_remove(struct snd_soc_card *card) -{ - struct machine_data *mdata = - container_of(card, struct machine_data, card); - struct ccsr_guts __iomem *guts; - - guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); - if (!guts) { - dev_err(card->dev, "could not map global utilities\n"); - return -ENOMEM; - } - - /* Restore the signal routing */ - clrbits32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_UART0_I2C1_MASK); - clrbits32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK); - guts_set_dmuxcr(guts, mdata->dma_id[0], mdata->dma_channel_id[0], 0); - guts_set_dmuxcr(guts, mdata->dma_id[1], mdata->dma_channel_id[1], 0); - - iounmap(guts); - - return 0; -} - -/** - * p1022_ds_ops: ASoC machine driver operations - */ -static struct snd_soc_ops p1022_ds_ops = { - .startup = p1022_ds_startup, -}; - -/** - * get_node_by_phandle_name - get a node by its phandle name - * - * This function takes a node, the name of a property in that node, and a - * compatible string. Assuming the property is a phandle to another node, - * it returns that node, (optionally) if that node is compatible. - * - * If the property is not a phandle, or the node it points to is not compatible - * with the specific string, then NULL is returned. - */ -static struct device_node *get_node_by_phandle_name(struct device_node *np, - const char *name, const char *compatible) -{ - np = of_parse_phandle(np, name, 0); - if (!np) - return NULL; - - if (!of_device_is_compatible(np, compatible)) { - of_node_put(np); - return NULL; - } - - return np; -} - -/** - * get_parent_cell_index -- return the cell-index of the parent of a node - * - * Return the value of the cell-index property of the parent of the given - * node. This is used for DMA channel nodes that need to know the DMA ID - * of the controller they are on. - */ -static int get_parent_cell_index(struct device_node *np) -{ - struct device_node *parent = of_get_parent(np); - const u32 *iprop; - int ret = -1; - - if (!parent) - return -1; - - iprop = of_get_property(parent, "cell-index", NULL); - if (iprop) - ret = be32_to_cpup(iprop); - - of_node_put(parent); - - return ret; -} - -/** - * codec_node_dev_name - determine the dev_name for a codec node - * - * This function determines the dev_name for an I2C node. This is the name - * that would be returned by dev_name() if this device_node were part of a - * 'struct device' It's ugly and hackish, but it works. - * - * The dev_name for such devices include the bus number and I2C address. For - * example, "cs4270-codec.0-004f". - */ -static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) -{ - const u32 *iprop; - int addr; - char temp[DAI_NAME_SIZE]; - struct i2c_client *i2c; - - of_modalias_node(np, temp, DAI_NAME_SIZE); - - iprop = of_get_property(np, "reg", NULL); - if (!iprop) - return -EINVAL; - - addr = be32_to_cpup(iprop); - - /* We need the adapter number */ - i2c = of_find_i2c_device_by_node(np); - if (!i2c) - return -ENODEV; - - snprintf(buf, len, "%s.%u-%04x", temp, i2c->adapter->nr, addr); - - return 0; -} - -static int get_dma_channel(struct device_node *ssi_np, - const char *name, - struct snd_soc_dai_link *dai, - unsigned int *dma_channel_id, - unsigned int *dma_id) -{ - struct resource res; - struct device_node *dma_channel_np; - const u32 *iprop; - int ret; - - dma_channel_np = get_node_by_phandle_name(ssi_np, name, - "fsl,ssi-dma-channel"); - if (!dma_channel_np) - return -EINVAL; - - /* Determine the dev_name for the device_node. This code mimics the - * behavior of of_device_make_bus_id(). We need this because ASoC uses - * the dev_name() of the device to match the platform (DMA) device with - * the CPU (SSI) device. It's all ugly and hackish, but it works (for - * now). - * - * dai->platform name should already point to an allocated buffer. - */ - ret = of_address_to_resource(dma_channel_np, 0, &res); - if (ret) { - of_node_put(dma_channel_np); - return ret; - } - snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", - (unsigned long long) res.start, dma_channel_np->name); - - iprop = of_get_property(dma_channel_np, "cell-index", NULL); - if (!iprop) { - of_node_put(dma_channel_np); - return -EINVAL; - } - - *dma_channel_id = be32_to_cpup(iprop); - *dma_id = get_parent_cell_index(dma_channel_np); - of_node_put(dma_channel_np); - - return 0; -} - -/** - * p1022_ds_probe: platform probe function for the machine driver - * - * Although this is a machine driver, the SSI node is the "master" node with - * respect to audio hardware connections. Therefore, we create a new ASoC - * device for each new SSI node that has a codec attached. - */ -static int p1022_ds_probe(struct platform_device *pdev) -{ - struct device *dev = pdev->dev.parent; - /* ssi_pdev is the platform device for the SSI node that probed us */ - struct platform_device *ssi_pdev = - container_of(dev, struct platform_device, dev); - struct device_node *np = ssi_pdev->dev.of_node; - struct device_node *codec_np = NULL; - struct platform_device *sound_device = NULL; - struct machine_data *mdata; - int ret = -ENODEV; - const char *sprop; - const u32 *iprop; - - /* Find the codec node for this SSI. */ - codec_np = of_parse_phandle(np, "codec-handle", 0); - if (!codec_np) { - dev_err(dev, "could not find codec node\n"); - return -EINVAL; - } - - mdata = kzalloc(sizeof(struct machine_data), GFP_KERNEL); - if (!mdata) { - ret = -ENOMEM; - goto error_put; - } - - mdata->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev); - mdata->dai[0].ops = &p1022_ds_ops; - - /* Determine the codec name, it will be used as the codec DAI name */ - ret = codec_node_dev_name(codec_np, mdata->codec_name, DAI_NAME_SIZE); - if (ret) { - dev_err(&pdev->dev, "invalid codec node %s\n", - codec_np->full_name); - ret = -EINVAL; - goto error; - } - mdata->dai[0].codec_name = mdata->codec_name; - - /* We register two DAIs per SSI, one for playback and the other for - * capture. We support codecs that have separate DAIs for both playback - * and capture. - */ - memcpy(&mdata->dai[1], &mdata->dai[0], sizeof(struct snd_soc_dai_link)); - - /* The DAI names from the codec (snd_soc_dai_driver.name) */ - mdata->dai[0].codec_dai_name = "wm8776-hifi-playback"; - mdata->dai[1].codec_dai_name = "wm8776-hifi-capture"; - - /* Get the device ID */ - iprop = of_get_property(np, "cell-index", NULL); - if (!iprop) { - dev_err(&pdev->dev, "cell-index property not found\n"); - ret = -EINVAL; - goto error; - } - mdata->ssi_id = be32_to_cpup(iprop); - - /* Get the serial format and clock direction. */ - sprop = of_get_property(np, "fsl,mode", NULL); - if (!sprop) { - dev_err(&pdev->dev, "fsl,mode property not found\n"); - ret = -EINVAL; - goto error; - } - - if (strcasecmp(sprop, "i2s-slave") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM; - mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; - mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; - - /* In i2s-slave mode, the codec has its own clock source, so we - * need to get the frequency from the device tree and pass it to - * the codec driver. - */ - iprop = of_get_property(codec_np, "clock-frequency", NULL); - if (!iprop || !*iprop) { - dev_err(&pdev->dev, "codec bus-frequency " - "property is missing or invalid\n"); - ret = -EINVAL; - goto error; - } - mdata->clk_frequency = be32_to_cpup(iprop); - } else if (strcasecmp(sprop, "i2s-master") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS; - mdata->codec_clk_direction = SND_SOC_CLOCK_IN; - mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "lj-slave") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM; - mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; - mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "lj-master") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS; - mdata->codec_clk_direction = SND_SOC_CLOCK_IN; - mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "rj-slave") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBM_CFM; - mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; - mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "rj-master") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBS_CFS; - mdata->codec_clk_direction = SND_SOC_CLOCK_IN; - mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "ac97-slave") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBM_CFM; - mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; - mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "ac97-master") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBS_CFS; - mdata->codec_clk_direction = SND_SOC_CLOCK_IN; - mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else { - dev_err(&pdev->dev, - "unrecognized fsl,mode property '%s'\n", sprop); - ret = -EINVAL; - goto error; - } - - if (!mdata->clk_frequency) { - dev_err(&pdev->dev, "unknown clock frequency\n"); - ret = -EINVAL; - goto error; - } - - /* Find the playback DMA channel to use. */ - mdata->dai[0].platform_name = mdata->platform_name[0]; - ret = get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0], - &mdata->dma_channel_id[0], - &mdata->dma_id[0]); - if (ret) { - dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n"); - goto error; - } - - /* Find the capture DMA channel to use. */ - mdata->dai[1].platform_name = mdata->platform_name[1]; - ret = get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1], - &mdata->dma_channel_id[1], - &mdata->dma_id[1]); - if (ret) { - dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n"); - goto error; - } - - /* Initialize our DAI data structure. */ - mdata->dai[0].stream_name = "playback"; - mdata->dai[1].stream_name = "capture"; - mdata->dai[0].name = mdata->dai[0].stream_name; - mdata->dai[1].name = mdata->dai[1].stream_name; - - mdata->card.probe = p1022_ds_machine_probe; - mdata->card.remove = p1022_ds_machine_remove; - mdata->card.name = pdev->name; /* The platform driver name */ - mdata->card.num_links = 2; - mdata->card.dai_link = mdata->dai; - - /* Allocate a new audio platform device structure */ - sound_device = platform_device_alloc("soc-audio", -1); - if (!sound_device) { - dev_err(&pdev->dev, "platform device alloc failed\n"); - ret = -ENOMEM; - goto error; - } - - /* Associate the card data with the sound device */ - platform_set_drvdata(sound_device, &mdata->card); - - /* Register with ASoC */ - ret = platform_device_add(sound_device); - if (ret) { - dev_err(&pdev->dev, "platform device add failed\n"); - goto error; - } - dev_set_drvdata(&pdev->dev, sound_device); - - of_node_put(codec_np); - - return 0; - -error: - if (sound_device) - platform_device_put(sound_device); - - kfree(mdata); -error_put: - of_node_put(codec_np); - return ret; -} - -/** - * p1022_ds_remove: remove the platform device - * - * This function is called when the platform device is removed. - */ -static int __devexit p1022_ds_remove(struct platform_device *pdev) -{ - struct platform_device *sound_device = dev_get_drvdata(&pdev->dev); - struct snd_soc_card *card = platform_get_drvdata(sound_device); - struct machine_data *mdata = - container_of(card, struct machine_data, card); - - platform_device_unregister(sound_device); - - kfree(mdata); - sound_device->dev.platform_data = NULL; - - dev_set_drvdata(&pdev->dev, NULL); - - return 0; -} - -static struct platform_driver p1022_ds_driver = { - .probe = p1022_ds_probe, - .remove = __devexit_p(p1022_ds_remove), - .driver = { - /* - * The name must match 'compatible' property in the device tree, - * in lowercase letters. - */ - .name = "snd-soc-p1022ds", - .owner = THIS_MODULE, - }, -}; - -/** - * p1022_ds_init: machine driver initialization. - * - * This function is called when this module is loaded. - */ -static int __init p1022_ds_init(void) -{ - struct device_node *guts_np; - struct resource res; - - /* Get the physical address of the global utilities registers */ - guts_np = of_find_compatible_node(NULL, NULL, "fsl,p1022-guts"); - if (of_address_to_resource(guts_np, 0, &res)) { - pr_err("snd-soc-p1022ds: missing/invalid global utils node\n"); - of_node_put(guts_np); - return -EINVAL; - } - guts_phys = res.start; - of_node_put(guts_np); - - return platform_driver_register(&p1022_ds_driver); -} - -/** - * p1022_ds_exit: machine driver exit - * - * This function is called when this driver is unloaded. - */ -static void __exit p1022_ds_exit(void) -{ - platform_driver_unregister(&p1022_ds_driver); -} - -module_init(p1022_ds_init); -module_exit(p1022_ds_exit); - -MODULE_AUTHOR("Timur Tabi <timur@freescale.com>"); -MODULE_DESCRIPTION("Freescale P1022 DS ALSA SoC machine driver"); -MODULE_LICENSE("GPL v2"); diff --git a/ANDROID_3.4.5/sound/soc/fsl/pcm030-audio-fabric.c b/ANDROID_3.4.5/sound/soc/fsl/pcm030-audio-fabric.c deleted file mode 100644 index b3af55dc..00000000 --- a/ANDROID_3.4.5/sound/soc/fsl/pcm030-audio-fabric.c +++ /dev/null @@ -1,91 +0,0 @@ -/* - * Phytec pcm030 driver for the PSC of the Freescale MPC52xx - * configured as AC97 interface - * - * Copyright 2008 Jon Smirl, Digispeaker - * Author: Jon Smirl <jonsmirl@gmail.com> - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/of_device.h> -#include <linux/of_platform.h> -#include <linux/dma-mapping.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include "mpc5200_dma.h" -#include "mpc5200_psc_ac97.h" -#include "../codecs/wm9712.h" - -#define DRV_NAME "pcm030-audio-fabric" - -static struct snd_soc_dai_link pcm030_fabric_dai[] = { -{ - .name = "AC97", - .stream_name = "AC97 Analog", - .codec_dai_name = "wm9712-hifi", - .cpu_dai_name = "mpc5200-psc-ac97.0", - .platform_name = "mpc5200-pcm-audio", - .codec_name = "wm9712-codec", -}, -{ - .name = "AC97", - .stream_name = "AC97 IEC958", - .codec_dai_name = "wm9712-aux", - .cpu_dai_name = "mpc5200-psc-ac97.1", - .platform_name = "mpc5200-pcm-audio", - .codec_name = "wm9712-codec", -}, -}; - -static struct snd_soc_card card = { - .name = "pcm030", - .owner = THIS_MODULE, - .dai_link = pcm030_fabric_dai, - .num_links = ARRAY_SIZE(pcm030_fabric_dai), -}; - -static __init int pcm030_fabric_init(void) -{ - struct platform_device *pdev; - int rc; - - if (!of_machine_is_compatible("phytec,pcm030")) - return -ENODEV; - - pdev = platform_device_alloc("soc-audio", 1); - if (!pdev) { - pr_err("pcm030_fabric_init: platform_device_alloc() failed\n"); - return -ENODEV; - } - - platform_set_drvdata(pdev, &card); - - rc = platform_device_add(pdev); - if (rc) { - pr_err("pcm030_fabric_init: platform_device_add() failed\n"); - platform_device_put(pdev); - return -ENODEV; - } - return 0; -} - -module_init(pcm030_fabric_init); - - -MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>"); -MODULE_DESCRIPTION(DRV_NAME ": mpc5200 pcm030 fabric driver"); -MODULE_LICENSE("GPL"); - |