ALSA: ASoC: Au12x0/Au1550 PSC Audio support

Audio for Au12x0/Au1550 PSCs in AC97 and I2S mode, for ASoC v1 framework.

- DBDMA, AC97 and I2S drivers
- sample AC97 machine code (Db1200)

Signed-off-by: Manuel Lauss <mano@roarinelk.homelinux.net>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

1 files changed, 421 insertions, 0 deletions

diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
new file mode 100644
index 000000000000..1466d9328800
--- /dev/null
+++ b/sound/soc/au1x/dbdma2.c
@@ -0,0 +1,421 @@
+/*
+ * Au12x0/Au1550 PSC ALSA ASoC audio support.
+ *
+ * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
+ *      Manuel Lauss <mano@roarinelk.homelinux.net>
+ *
+ * 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.
+ *
+ * DMA glue for Au1x-PSC audio.
+ *
+ * NOTE: all of these drivers can only work with a SINGLE instance
+ *       of a PSC. Multiple independent audio devices are impossible
+ *       with ASoC v1.
+ */
+
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1xxx_dbdma.h>
+#include <asm/mach-au1x00/au1xxx_psc.h>
+
+#include "psc.h"
+
+/*#define PCM_DEBUG*/
+
+#define MSG(x...)       printk(KERN_INFO "au1xpsc_pcm: " x)
+#ifdef PCM_DEBUG
+#define DBG             MSG
+#else
+#define DBG(x...)       do {} while (0)
+#endif
+
+struct au1xpsc_audio_dmadata {
+        /* DDMA control data */
+        unsigned int ddma_id;           /* DDMA direction ID for this PSC */
+        u32 ddma_chan;                  /* DDMA context */
+
+        /* PCM context (for irq handlers) */
+        struct snd_pcm_substream *substream;
+        unsigned long curr_period;      /* current segment DDMA is working on */
+        unsigned long q_period;         /* queue period(s) */
+        unsigned long dma_area;         /* address of queued DMA area */
+        unsigned long dma_area_s;       /* start address of DMA area */
+        unsigned long pos;              /* current byte position being played */
+        unsigned long periods;          /* number of SG segments in total */
+        unsigned long period_bytes;     /* size in bytes of one SG segment */
+
+        /* runtime data */
+        int msbits;
+};
+
+/* instance data. There can be only one, MacLeod!!!! */
+static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2];
+
+/*
+ * These settings are somewhat okay, at least on my machine audio plays
+ * almost skip-free. Especially the 64kB buffer seems to help a LOT.
+ */
+#define AU1XPSC_PERIOD_MIN_BYTES        1024
+#define AU1XPSC_BUFFER_MIN_BYTES        65536
+
+#define AU1XPSC_PCM_FMTS                                        \
+        (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_S32_LE | SNDRV_PCM_FMTBIT_S32_BE |    \
+         SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE |    \
+         0)
+
+/* PCM hardware DMA capabilities - platform specific */
+static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
+        .info             = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+                            SNDRV_PCM_INFO_INTERLEAVED,
+        .formats          = AU1XPSC_PCM_FMTS,
+        .period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES,
+        .period_bytes_max = 4096 * 1024 - 1,
+        .periods_min      = 2,
+        .periods_max      = 4096,       /* 2 to as-much-as-you-like */
+        .buffer_bytes_max = 4096 * 1024 - 1,
+        .fifo_size        = 16,         /* fifo entries of AC97/I2S PSC */
+};
+
+static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd)
+{
+        au1xxx_dbdma_put_source_flags(cd->ddma_chan,
+                                (void *)phys_to_virt(cd->dma_area),
+                                cd->period_bytes, DDMA_FLAGS_IE);
+
+        /* update next-to-queue period */
+        ++cd->q_period;
+        cd->dma_area += cd->period_bytes;
+        if (cd->q_period >= cd->periods) {
+                cd->q_period = 0;
+                cd->dma_area = cd->dma_area_s;
+        }
+}
+
+static void au1x_pcm_queue_rx(struct au1xpsc_audio_dmadata *cd)
+{
+        au1xxx_dbdma_put_dest_flags(cd->ddma_chan,
+                                (void *)phys_to_virt(cd->dma_area),
+                                cd->period_bytes, DDMA_FLAGS_IE);
+
+        /* update next-to-queue period */
+        ++cd->q_period;
+        cd->dma_area += cd->period_bytes;
+        if (cd->q_period >= cd->periods) {
+                cd->q_period = 0;
+                cd->dma_area = cd->dma_area_s;
+        }
+}
+
+static void au1x_pcm_dmatx_cb(int irq, void *dev_id)
+{
+        struct au1xpsc_audio_dmadata *cd = dev_id;
+
+        cd->pos += cd->period_bytes;
+        if (++cd->curr_period >= cd->periods) {
+                cd->pos = 0;
+                cd->curr_period = 0;
+        }
+        snd_pcm_period_elapsed(cd->substream);
+        au1x_pcm_queue_tx(cd);
+}
+
+static void au1x_pcm_dmarx_cb(int irq, void *dev_id)
+{
+        struct au1xpsc_audio_dmadata *cd = dev_id;
+
+        cd->pos += cd->period_bytes;
+        if (++cd->curr_period >= cd->periods) {
+                cd->pos = 0;
+                cd->curr_period = 0;
+        }
+        snd_pcm_period_elapsed(cd->substream);
+        au1x_pcm_queue_rx(cd);
+}
+
+static void au1x_pcm_dbdma_free(struct au1xpsc_audio_dmadata *pcd)
+{
+        if (pcd->ddma_chan) {
+                au1xxx_dbdma_stop(pcd->ddma_chan);
+                au1xxx_dbdma_reset(pcd->ddma_chan);
+                au1xxx_dbdma_chan_free(pcd->ddma_chan);
+                pcd->ddma_chan = 0;
+                pcd->msbits = 0;
+        }
+}
+
+/* in case of missing DMA ring or changed TX-source / RX-dest bit widths,
+ * allocate (or reallocate) a 2-descriptor DMA ring with bit depth according
+ * to ALSA-supplied sample depth.  This is due to limitations in the dbdma api
+ * (cannot adjust source/dest widths of already allocated descriptor ring).
+ */
+static int au1x_pcm_dbdma_realloc(struct au1xpsc_audio_dmadata *pcd,
+                                 int stype, int msbits)
+{
+        /* DMA only in 8/16/32 bit widths */
+        if (msbits == 24)
+                msbits = 32;
+
+        /* check current config: correct bits and descriptors allocated? */
+        if ((pcd->ddma_chan) && (msbits == pcd->msbits))
+                goto out;       /* all ok! */
+
+        au1x_pcm_dbdma_free(pcd);
+
+        if (stype == PCM_RX)
+                pcd->ddma_chan = au1xxx_dbdma_chan_alloc(pcd->ddma_id,
+                                        DSCR_CMD0_ALWAYS,
+                                        au1x_pcm_dmarx_cb, (void *)pcd);
+        else
+                pcd->ddma_chan = au1xxx_dbdma_chan_alloc(DSCR_CMD0_ALWAYS,
+                                        pcd->ddma_id,
+                                        au1x_pcm_dmatx_cb, (void *)pcd);
+
+        if (!pcd->ddma_chan)
+                return -ENOMEM;;
+
+        au1xxx_dbdma_set_devwidth(pcd->ddma_chan, msbits);
+        au1xxx_dbdma_ring_alloc(pcd->ddma_chan, 2);
+
+        pcd->msbits = msbits;
+
+        au1xxx_dbdma_stop(pcd->ddma_chan);
+        au1xxx_dbdma_reset(pcd->ddma_chan);
+
+out:
+        return 0;
+}
+
+static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
+                                 struct snd_pcm_hw_params *params)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct au1xpsc_audio_dmadata *pcd;
+        int stype, ret;
+
+        ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+        if (ret < 0)
+                goto out;
+
+        stype = SUBSTREAM_TYPE(substream);
+        pcd = au1xpsc_audio_pcmdma[stype];
+
+        DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d "
+            "runtime->min_align %d\n",
+                (unsigned long)runtime->dma_area,
+                (unsigned long)runtime->dma_addr, runtime->dma_bytes,
+                runtime->min_align);
+
+        DBG("bits %d  frags %d  frag_bytes %d  is_rx %d\n", params->msbits,
+                params_periods(params), params_period_bytes(params), stype);
+
+        ret = au1x_pcm_dbdma_realloc(pcd, stype, params->msbits);
+        if (ret) {
+                MSG("DDMA channel (re)alloc failed!\n");
+                goto out;
+        }
+
+        pcd->substream = substream;
+        pcd->period_bytes = params_period_bytes(params);
+        pcd->periods = params_periods(params);
+        pcd->dma_area_s = pcd->dma_area = (unsigned long)runtime->dma_addr;
+        pcd->q_period = 0;
+        pcd->curr_period = 0;
+        pcd->pos = 0;
+
+        ret = 0;
+out:
+        return ret;
+}
+
+static int au1xpsc_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+        snd_pcm_lib_free_pages(substream);
+        return 0;
+}
+
+static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream)
+{
+        struct au1xpsc_audio_dmadata *pcd =
+                        au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)];
+
+        au1xxx_dbdma_reset(pcd->ddma_chan);
+
+        if (SUBSTREAM_TYPE(substream) == PCM_RX) {
+                au1x_pcm_queue_rx(pcd);
+                au1x_pcm_queue_rx(pcd);
+        } else {
+                au1x_pcm_queue_tx(pcd);
+                au1x_pcm_queue_tx(pcd);
+        }
+
+        return 0;
+}
+
+static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+        u32 c = au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->ddma_chan;
+
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_START:
+        case SNDRV_PCM_TRIGGER_RESUME:
+                au1xxx_dbdma_start(c);
+                break;
+        case SNDRV_PCM_TRIGGER_STOP:
+        case SNDRV_PCM_TRIGGER_SUSPEND:
+                au1xxx_dbdma_stop(c);
+                break;
+        default:
+                return -EINVAL;
+        }
+        return 0;
+}
+
+static snd_pcm_uframes_t
+au1xpsc_pcm_pointer(struct snd_pcm_substream *substream)
+{
+        return bytes_to_frames(substream->runtime,
+                au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->pos);
+}
+
+static int au1xpsc_pcm_open(struct snd_pcm_substream *substream)
+{
+        snd_soc_set_runtime_hwparams(substream, &au1xpsc_pcm_hardware);
+        return 0;
+}
+
+static int au1xpsc_pcm_close(struct snd_pcm_substream *substream)
+{
+        au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]);
+        return 0;
+}
+
+struct snd_pcm_ops au1xpsc_pcm_ops = {
+        .open           = au1xpsc_pcm_open,
+        .close          = au1xpsc_pcm_close,
+        .ioctl          = snd_pcm_lib_ioctl,
+        .hw_params      = au1xpsc_pcm_hw_params,
+        .hw_free        = au1xpsc_pcm_hw_free,
+        .prepare        = au1xpsc_pcm_prepare,
+        .trigger        = au1xpsc_pcm_trigger,
+        .pointer        = au1xpsc_pcm_pointer,
+};
+
+static void au1xpsc_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+        snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int au1xpsc_pcm_new(struct snd_card *card,
+                           struct snd_soc_dai *dai,
+                           struct snd_pcm *pcm)
+{
+        snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+                card->dev, AU1XPSC_BUFFER_MIN_BYTES, (4096 * 1024) - 1);
+
+        return 0;
+}
+
+static int au1xpsc_pcm_probe(struct platform_device *pdev)
+{
+        struct resource *r;
+        int ret;
+
+        if (au1xpsc_audio_pcmdma[PCM_TX] || au1xpsc_audio_pcmdma[PCM_RX])
+                return -EBUSY;
+
+        /* TX DMA */
+        au1xpsc_audio_pcmdma[PCM_TX]
+                = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL);
+        if (!au1xpsc_audio_pcmdma[PCM_TX])
+                return -ENOMEM;
+
+        r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+        if (!r) {
+                ret = -ENODEV;
+                goto out1;
+        }
+        (au1xpsc_audio_pcmdma[PCM_TX])->ddma_id = r->start;
+
+        /* RX DMA */
+        au1xpsc_audio_pcmdma[PCM_RX]
+                = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL);
+        if (!au1xpsc_audio_pcmdma[PCM_RX])
+                return -ENOMEM;
+
+        r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+        if (!r) {
+                ret = -ENODEV;
+                goto out2;
+        }
+        (au1xpsc_audio_pcmdma[PCM_RX])->ddma_id = r->start;
+
+        return 0;
+
+out2:
+        kfree(au1xpsc_audio_pcmdma[PCM_RX]);
+        au1xpsc_audio_pcmdma[PCM_RX] = NULL;
+out1:
+        kfree(au1xpsc_audio_pcmdma[PCM_TX]);
+        au1xpsc_audio_pcmdma[PCM_TX] = NULL;
+        return ret;
+}
+
+static int au1xpsc_pcm_remove(struct platform_device *pdev)
+{
+        int i;
+
+        for (i = 0; i < 2; i++) {
+                if (au1xpsc_audio_pcmdma[i]) {
+                        au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[i]);
+                        kfree(au1xpsc_audio_pcmdma[i]);
+                        au1xpsc_audio_pcmdma[i] = NULL;
+                }
+        }
+
+        return 0;
+}
+
+/* au1xpsc audio platform */
+struct snd_soc_platform au1xpsc_soc_platform = {
+        .name           = "au1xpsc-pcm-dbdma",
+        .probe          = au1xpsc_pcm_probe,
+        .remove         = au1xpsc_pcm_remove,
+        .pcm_ops        = &au1xpsc_pcm_ops,
+        .pcm_new        = au1xpsc_pcm_new,
+        .pcm_free       = au1xpsc_pcm_free_dma_buffers,
+};
+EXPORT_SYMBOL_GPL(au1xpsc_soc_platform);
+
+static int __init au1xpsc_audio_dbdma_init(void)
+{
+        au1xpsc_audio_pcmdma[PCM_TX] = NULL;
+        au1xpsc_audio_pcmdma[PCM_RX] = NULL;
+        return 0;
+}
+
+static void __exit au1xpsc_audio_dbdma_exit(void)
+{
+}
+
+module_init(au1xpsc_audio_dbdma_init);
+module_exit(au1xpsc_audio_dbdma_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver");
+MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");