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1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298 299 300 301 302 303 304 305 306 307 308 309 310 311 312 313 314 315 316 317 318 319 320 321 322 323 324 325 326 327 328 329 330 331 332 333 334 335 336 337 338 339 340 341 342 343 344 345 346 347 348 349 350 351 352 353 354 355 356 357 358 359 360 361 362 363 364 365 366 367 368 369 370 371 372 373 374 375 376 377 378 379 380 381 382 383 384 385 386 387 388 389 390 391 392 393 394 395 396 397 398 399 400 401 402 403 404 405 406 407 408 409 410 411 412 413 414 415 416 417 418 419 420 421 422 | /* * 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 | SNDRV_PCM_INFO_BATCH, .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; } static 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 snd_soc_register_platform(&au1xpsc_soc_platform); } static void __exit au1xpsc_audio_dbdma_exit(void) { snd_soc_unregister_platform(&au1xpsc_soc_platform); } 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>"); |