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TOMOYO Linux Cross Reference
Linux/sound/soc/qcom/qdsp6/q6asm-dai.c

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Architecture: ~ [ i386 ] ~ [ alpha ] ~ [ m68k ] ~ [ mips ] ~ [ ppc ] ~ [ sparc ] ~ [ sparc64 ] ~

  1 // SPDX-License-Identifier: GPL-2.0
  2 // Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
  3 // Copyright (c) 2018, Linaro Limited
  4 
  5 #include <dt-bindings/sound/qcom,q6asm.h>
  6 #include <linux/init.h>
  7 #include <linux/err.h>
  8 #include <linux/module.h>
  9 #include <linux/of.h>
 10 #include <linux/platform_device.h>
 11 #include <linux/slab.h>
 12 #include <sound/soc.h>
 13 #include <sound/soc-dapm.h>
 14 #include <sound/pcm.h>
 15 #include <linux/spinlock.h>
 16 #include <sound/compress_driver.h>
 17 #include <asm/dma.h>
 18 #include <linux/dma-mapping.h>
 19 #include <sound/pcm_params.h>
 20 #include "q6asm.h"
 21 #include "q6routing.h"
 22 #include "q6dsp-errno.h"
 23 
 24 #define DRV_NAME        "q6asm-fe-dai"
 25 
 26 #define PLAYBACK_MIN_NUM_PERIODS    2
 27 #define PLAYBACK_MAX_NUM_PERIODS   8
 28 #define PLAYBACK_MAX_PERIOD_SIZE    65536
 29 #define PLAYBACK_MIN_PERIOD_SIZE    128
 30 #define CAPTURE_MIN_NUM_PERIODS     2
 31 #define CAPTURE_MAX_NUM_PERIODS     8
 32 #define CAPTURE_MAX_PERIOD_SIZE     4096
 33 #define CAPTURE_MIN_PERIOD_SIZE     320
 34 #define SID_MASK_DEFAULT        0xF
 35 
 36 /* Default values used if user space does not set */
 37 #define COMPR_PLAYBACK_MIN_FRAGMENT_SIZE (8 * 1024)
 38 #define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024)
 39 #define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4)
 40 #define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4)
 41 
 42 #define ALAC_CH_LAYOUT_MONO   ((101 << 16) | 1)
 43 #define ALAC_CH_LAYOUT_STEREO ((101 << 16) | 2)
 44 
 45 enum stream_state {
 46         Q6ASM_STREAM_IDLE = 0,
 47         Q6ASM_STREAM_STOPPED,
 48         Q6ASM_STREAM_RUNNING,
 49 };
 50 
 51 struct q6asm_dai_rtd {
 52         struct snd_pcm_substream *substream;
 53         struct snd_compr_stream *cstream;
 54         struct snd_codec codec;
 55         struct snd_dma_buffer dma_buffer;
 56         spinlock_t lock;
 57         phys_addr_t phys;
 58         unsigned int pcm_size;
 59         unsigned int pcm_count;
 60         unsigned int pcm_irq_pos;       /* IRQ position */
 61         unsigned int periods;
 62         unsigned int bytes_sent;
 63         unsigned int bytes_received;
 64         unsigned int copied_total;
 65         uint16_t bits_per_sample;
 66         uint16_t source; /* Encoding source bit mask */
 67         struct audio_client *audio_client;
 68         uint32_t next_track_stream_id;
 69         bool next_track;
 70         uint32_t stream_id;
 71         uint16_t session_id;
 72         enum stream_state state;
 73         uint32_t initial_samples_drop;
 74         uint32_t trailing_samples_drop;
 75         bool notify_on_drain;
 76 };
 77 
 78 struct q6asm_dai_data {
 79         struct snd_soc_dai_driver *dais;
 80         int num_dais;
 81         long long int sid;
 82 };
 83 
 84 static const struct snd_pcm_hardware q6asm_dai_hardware_capture = {
 85         .info =                 (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH |
 86                                 SNDRV_PCM_INFO_BLOCK_TRANSFER |
 87                                 SNDRV_PCM_INFO_MMAP_VALID |
 88                                 SNDRV_PCM_INFO_INTERLEAVED |
 89                                 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
 90         .formats =              (SNDRV_PCM_FMTBIT_S16_LE |
 91                                 SNDRV_PCM_FMTBIT_S24_LE),
 92         .rates =                SNDRV_PCM_RATE_8000_48000,
 93         .rate_min =             8000,
 94         .rate_max =             48000,
 95         .channels_min =         1,
 96         .channels_max =         4,
 97         .buffer_bytes_max =     CAPTURE_MAX_NUM_PERIODS *
 98                                 CAPTURE_MAX_PERIOD_SIZE,
 99         .period_bytes_min =     CAPTURE_MIN_PERIOD_SIZE,
100         .period_bytes_max =     CAPTURE_MAX_PERIOD_SIZE,
101         .periods_min =          CAPTURE_MIN_NUM_PERIODS,
102         .periods_max =          CAPTURE_MAX_NUM_PERIODS,
103         .fifo_size =            0,
104 };
105 
106 static const struct snd_pcm_hardware q6asm_dai_hardware_playback = {
107         .info =                 (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH |
108                                 SNDRV_PCM_INFO_BLOCK_TRANSFER |
109                                 SNDRV_PCM_INFO_MMAP_VALID |
110                                 SNDRV_PCM_INFO_INTERLEAVED |
111                                 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
112         .formats =              (SNDRV_PCM_FMTBIT_S16_LE |
113                                 SNDRV_PCM_FMTBIT_S24_LE),
114         .rates =                SNDRV_PCM_RATE_8000_192000,
115         .rate_min =             8000,
116         .rate_max =             192000,
117         .channels_min =         1,
118         .channels_max =         8,
119         .buffer_bytes_max =     (PLAYBACK_MAX_NUM_PERIODS *
120                                 PLAYBACK_MAX_PERIOD_SIZE),
121         .period_bytes_min =     PLAYBACK_MIN_PERIOD_SIZE,
122         .period_bytes_max =     PLAYBACK_MAX_PERIOD_SIZE,
123         .periods_min =          PLAYBACK_MIN_NUM_PERIODS,
124         .periods_max =          PLAYBACK_MAX_NUM_PERIODS,
125         .fifo_size =            0,
126 };
127 
128 #define Q6ASM_FEDAI_DRIVER(num) { \
129                 .playback = {                                           \
130                         .stream_name = "MultiMedia"#num" Playback",     \
131                         .rates = (SNDRV_PCM_RATE_8000_192000|           \
132                                         SNDRV_PCM_RATE_KNOT),           \
133                         .formats = (SNDRV_PCM_FMTBIT_S16_LE |           \
134                                         SNDRV_PCM_FMTBIT_S24_LE),       \
135                         .channels_min = 1,                              \
136                         .channels_max = 8,                              \
137                         .rate_min =     8000,                           \
138                         .rate_max =     192000,                         \
139                 },                                                      \
140                 .capture = {                                            \
141                         .stream_name = "MultiMedia"#num" Capture",      \
142                         .rates = (SNDRV_PCM_RATE_8000_48000|            \
143                                         SNDRV_PCM_RATE_KNOT),           \
144                         .formats = (SNDRV_PCM_FMTBIT_S16_LE |           \
145                                     SNDRV_PCM_FMTBIT_S24_LE),           \
146                         .channels_min = 1,                              \
147                         .channels_max = 4,                              \
148                         .rate_min =     8000,                           \
149                         .rate_max =     48000,                          \
150                 },                                                      \
151                 .name = "MultiMedia"#num,                               \
152                 .id = MSM_FRONTEND_DAI_MULTIMEDIA##num,                 \
153         }
154 
155 /* Conventional and unconventional sample rate supported */
156 static unsigned int supported_sample_rates[] = {
157         8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
158         88200, 96000, 176400, 192000
159 };
160 
161 static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
162         .count = ARRAY_SIZE(supported_sample_rates),
163         .list = supported_sample_rates,
164         .mask = 0,
165 };
166 
167 static const struct snd_compr_codec_caps q6asm_compr_caps = {
168         .num_descriptors = 1,
169         .descriptor[0].max_ch = 2,
170         .descriptor[0].sample_rates = { 8000, 11025, 12000, 16000, 22050,
171                                         24000, 32000, 44100, 48000, 88200,
172                                         96000, 176400, 192000 },
173         .descriptor[0].num_sample_rates = 13,
174         .descriptor[0].bit_rate[0] = 320,
175         .descriptor[0].bit_rate[1] = 128,
176         .descriptor[0].num_bitrates = 2,
177         .descriptor[0].profiles = 0,
178         .descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO,
179         .descriptor[0].formats = 0,
180 };
181 
182 static void event_handler(uint32_t opcode, uint32_t token,
183                           void *payload, void *priv)
184 {
185         struct q6asm_dai_rtd *prtd = priv;
186         struct snd_pcm_substream *substream = prtd->substream;
187 
188         switch (opcode) {
189         case ASM_CLIENT_EVENT_CMD_RUN_DONE:
190                 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
191                         q6asm_write_async(prtd->audio_client, prtd->stream_id,
192                                    prtd->pcm_count, 0, 0, 0);
193                 break;
194         case ASM_CLIENT_EVENT_CMD_EOS_DONE:
195                 prtd->state = Q6ASM_STREAM_STOPPED;
196                 break;
197         case ASM_CLIENT_EVENT_DATA_WRITE_DONE: {
198                 prtd->pcm_irq_pos += prtd->pcm_count;
199                 snd_pcm_period_elapsed(substream);
200                 if (prtd->state == Q6ASM_STREAM_RUNNING)
201                         q6asm_write_async(prtd->audio_client, prtd->stream_id,
202                                            prtd->pcm_count, 0, 0, 0);
203 
204                 break;
205                 }
206         case ASM_CLIENT_EVENT_DATA_READ_DONE:
207                 prtd->pcm_irq_pos += prtd->pcm_count;
208                 snd_pcm_period_elapsed(substream);
209                 if (prtd->state == Q6ASM_STREAM_RUNNING)
210                         q6asm_read(prtd->audio_client, prtd->stream_id);
211 
212                 break;
213         default:
214                 break;
215         }
216 }
217 
218 static int q6asm_dai_prepare(struct snd_soc_component *component,
219                              struct snd_pcm_substream *substream)
220 {
221         struct snd_pcm_runtime *runtime = substream->runtime;
222         struct snd_soc_pcm_runtime *soc_prtd = snd_soc_substream_to_rtd(substream);
223         struct q6asm_dai_rtd *prtd = runtime->private_data;
224         struct q6asm_dai_data *pdata;
225         struct device *dev = component->dev;
226         int ret, i;
227 
228         pdata = snd_soc_component_get_drvdata(component);
229         if (!pdata)
230                 return -EINVAL;
231 
232         if (!prtd || !prtd->audio_client) {
233                 dev_err(dev, "%s: private data null or audio client freed\n",
234                         __func__);
235                 return -EINVAL;
236         }
237 
238         prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
239         prtd->pcm_irq_pos = 0;
240         /* rate and channels are sent to audio driver */
241         if (prtd->state) {
242                 /* clear the previous setup if any  */
243                 q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE);
244                 q6asm_unmap_memory_regions(substream->stream,
245                                            prtd->audio_client);
246                 q6routing_stream_close(soc_prtd->dai_link->id,
247                                          substream->stream);
248         }
249 
250         ret = q6asm_map_memory_regions(substream->stream, prtd->audio_client,
251                                        prtd->phys,
252                                        (prtd->pcm_size / prtd->periods),
253                                        prtd->periods);
254 
255         if (ret < 0) {
256                 dev_err(dev, "Audio Start: Buffer Allocation failed rc = %d\n",
257                                                         ret);
258                 return -ENOMEM;
259         }
260 
261         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
262                 ret = q6asm_open_write(prtd->audio_client, prtd->stream_id,
263                                        FORMAT_LINEAR_PCM,
264                                        0, prtd->bits_per_sample, false);
265         } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
266                 ret = q6asm_open_read(prtd->audio_client, prtd->stream_id,
267                                       FORMAT_LINEAR_PCM,
268                                       prtd->bits_per_sample);
269         }
270 
271         if (ret < 0) {
272                 dev_err(dev, "%s: q6asm_open_write failed\n", __func__);
273                 goto open_err;
274         }
275 
276         prtd->session_id = q6asm_get_session_id(prtd->audio_client);
277         ret = q6routing_stream_open(soc_prtd->dai_link->id, LEGACY_PCM_MODE,
278                               prtd->session_id, substream->stream);
279         if (ret) {
280                 dev_err(dev, "%s: stream reg failed ret:%d\n", __func__, ret);
281                 goto routing_err;
282         }
283 
284         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
285                 ret = q6asm_media_format_block_multi_ch_pcm(
286                                 prtd->audio_client, prtd->stream_id,
287                                 runtime->rate, runtime->channels, NULL,
288                                 prtd->bits_per_sample);
289         } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
290                 ret = q6asm_enc_cfg_blk_pcm_format_support(prtd->audio_client,
291                                                            prtd->stream_id,
292                                                            runtime->rate,
293                                                            runtime->channels,
294                                                            prtd->bits_per_sample);
295 
296                 /* Queue the buffers */
297                 for (i = 0; i < runtime->periods; i++)
298                         q6asm_read(prtd->audio_client, prtd->stream_id);
299 
300         }
301         if (ret < 0)
302                 dev_info(dev, "%s: CMD Format block failed\n", __func__);
303         else
304                 prtd->state = Q6ASM_STREAM_RUNNING;
305 
306         return ret;
307 
308 routing_err:
309         q6asm_cmd(prtd->audio_client, prtd->stream_id,  CMD_CLOSE);
310 open_err:
311         q6asm_unmap_memory_regions(substream->stream, prtd->audio_client);
312         q6asm_audio_client_free(prtd->audio_client);
313         prtd->audio_client = NULL;
314 
315         return ret;
316 }
317 
318 static int q6asm_dai_trigger(struct snd_soc_component *component,
319                              struct snd_pcm_substream *substream, int cmd)
320 {
321         int ret = 0;
322         struct snd_pcm_runtime *runtime = substream->runtime;
323         struct q6asm_dai_rtd *prtd = runtime->private_data;
324 
325         switch (cmd) {
326         case SNDRV_PCM_TRIGGER_START:
327         case SNDRV_PCM_TRIGGER_RESUME:
328         case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
329                 ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id,
330                                        0, 0, 0);
331                 break;
332         case SNDRV_PCM_TRIGGER_STOP:
333                 prtd->state = Q6ASM_STREAM_STOPPED;
334                 ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
335                                        CMD_EOS);
336                 break;
337         case SNDRV_PCM_TRIGGER_SUSPEND:
338         case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
339                 ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
340                                        CMD_PAUSE);
341                 break;
342         default:
343                 ret = -EINVAL;
344                 break;
345         }
346 
347         return ret;
348 }
349 
350 static int q6asm_dai_open(struct snd_soc_component *component,
351                           struct snd_pcm_substream *substream)
352 {
353         struct snd_pcm_runtime *runtime = substream->runtime;
354         struct snd_soc_pcm_runtime *soc_prtd = snd_soc_substream_to_rtd(substream);
355         struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(soc_prtd, 0);
356         struct q6asm_dai_rtd *prtd;
357         struct q6asm_dai_data *pdata;
358         struct device *dev = component->dev;
359         int ret = 0;
360         int stream_id;
361 
362         stream_id = cpu_dai->driver->id;
363 
364         pdata = snd_soc_component_get_drvdata(component);
365         if (!pdata) {
366                 dev_err(dev, "Drv data not found ..\n");
367                 return -EINVAL;
368         }
369 
370         prtd = kzalloc(sizeof(struct q6asm_dai_rtd), GFP_KERNEL);
371         if (prtd == NULL)
372                 return -ENOMEM;
373 
374         prtd->substream = substream;
375         prtd->audio_client = q6asm_audio_client_alloc(dev,
376                                 (q6asm_cb)event_handler, prtd, stream_id,
377                                 LEGACY_PCM_MODE);
378         if (IS_ERR(prtd->audio_client)) {
379                 dev_info(dev, "%s: Could not allocate memory\n", __func__);
380                 ret = PTR_ERR(prtd->audio_client);
381                 kfree(prtd);
382                 return ret;
383         }
384 
385         /* DSP expects stream id from 1 */
386         prtd->stream_id = 1;
387 
388         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
389                 runtime->hw = q6asm_dai_hardware_playback;
390         else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
391                 runtime->hw = q6asm_dai_hardware_capture;
392 
393         ret = snd_pcm_hw_constraint_list(runtime, 0,
394                                 SNDRV_PCM_HW_PARAM_RATE,
395                                 &constraints_sample_rates);
396         if (ret < 0)
397                 dev_info(dev, "snd_pcm_hw_constraint_list failed\n");
398         /* Ensure that buffer size is a multiple of period size */
399         ret = snd_pcm_hw_constraint_integer(runtime,
400                                             SNDRV_PCM_HW_PARAM_PERIODS);
401         if (ret < 0)
402                 dev_info(dev, "snd_pcm_hw_constraint_integer failed\n");
403 
404         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
405                 ret = snd_pcm_hw_constraint_minmax(runtime,
406                         SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
407                         PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE,
408                         PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE);
409                 if (ret < 0) {
410                         dev_err(dev, "constraint for buffer bytes min max ret = %d\n",
411                                 ret);
412                 }
413         }
414 
415         ret = snd_pcm_hw_constraint_step(runtime, 0,
416                 SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
417         if (ret < 0) {
418                 dev_err(dev, "constraint for period bytes step ret = %d\n",
419                                                                 ret);
420         }
421         ret = snd_pcm_hw_constraint_step(runtime, 0,
422                 SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
423         if (ret < 0) {
424                 dev_err(dev, "constraint for buffer bytes step ret = %d\n",
425                                                                 ret);
426         }
427 
428         runtime->private_data = prtd;
429 
430         snd_soc_set_runtime_hwparams(substream, &q6asm_dai_hardware_playback);
431 
432         runtime->dma_bytes = q6asm_dai_hardware_playback.buffer_bytes_max;
433 
434 
435         if (pdata->sid < 0)
436                 prtd->phys = substream->dma_buffer.addr;
437         else
438                 prtd->phys = substream->dma_buffer.addr | (pdata->sid << 32);
439 
440         return 0;
441 }
442 
443 static int q6asm_dai_close(struct snd_soc_component *component,
444                            struct snd_pcm_substream *substream)
445 {
446         struct snd_pcm_runtime *runtime = substream->runtime;
447         struct snd_soc_pcm_runtime *soc_prtd = snd_soc_substream_to_rtd(substream);
448         struct q6asm_dai_rtd *prtd = runtime->private_data;
449 
450         if (prtd->audio_client) {
451                 if (prtd->state)
452                         q6asm_cmd(prtd->audio_client, prtd->stream_id,
453                                   CMD_CLOSE);
454 
455                 q6asm_unmap_memory_regions(substream->stream,
456                                            prtd->audio_client);
457                 q6asm_audio_client_free(prtd->audio_client);
458                 prtd->audio_client = NULL;
459         }
460         q6routing_stream_close(soc_prtd->dai_link->id,
461                                                 substream->stream);
462         kfree(prtd);
463         return 0;
464 }
465 
466 static snd_pcm_uframes_t q6asm_dai_pointer(struct snd_soc_component *component,
467                                            struct snd_pcm_substream *substream)
468 {
469 
470         struct snd_pcm_runtime *runtime = substream->runtime;
471         struct q6asm_dai_rtd *prtd = runtime->private_data;
472 
473         if (prtd->pcm_irq_pos >= prtd->pcm_size)
474                 prtd->pcm_irq_pos = 0;
475 
476         return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
477 }
478 
479 static int q6asm_dai_hw_params(struct snd_soc_component *component,
480                                struct snd_pcm_substream *substream,
481                                struct snd_pcm_hw_params *params)
482 {
483         struct snd_pcm_runtime *runtime = substream->runtime;
484         struct q6asm_dai_rtd *prtd = runtime->private_data;
485 
486         prtd->pcm_size = params_buffer_bytes(params);
487         prtd->periods = params_periods(params);
488 
489         switch (params_format(params)) {
490         case SNDRV_PCM_FORMAT_S16_LE:
491                 prtd->bits_per_sample = 16;
492                 break;
493         case SNDRV_PCM_FORMAT_S24_LE:
494                 prtd->bits_per_sample = 24;
495                 break;
496         }
497 
498         return 0;
499 }
500 
501 static void compress_event_handler(uint32_t opcode, uint32_t token,
502                                    void *payload, void *priv)
503 {
504         struct q6asm_dai_rtd *prtd = priv;
505         struct snd_compr_stream *substream = prtd->cstream;
506         unsigned long flags;
507         u32 wflags = 0;
508         uint64_t avail;
509         uint32_t bytes_written, bytes_to_write;
510         bool is_last_buffer = false;
511 
512         switch (opcode) {
513         case ASM_CLIENT_EVENT_CMD_RUN_DONE:
514                 spin_lock_irqsave(&prtd->lock, flags);
515                 if (!prtd->bytes_sent) {
516                         q6asm_stream_remove_initial_silence(prtd->audio_client,
517                                                     prtd->stream_id,
518                                                     prtd->initial_samples_drop);
519 
520                         q6asm_write_async(prtd->audio_client, prtd->stream_id,
521                                           prtd->pcm_count, 0, 0, 0);
522                         prtd->bytes_sent += prtd->pcm_count;
523                 }
524 
525                 spin_unlock_irqrestore(&prtd->lock, flags);
526                 break;
527 
528         case ASM_CLIENT_EVENT_CMD_EOS_DONE:
529                 spin_lock_irqsave(&prtd->lock, flags);
530                 if (prtd->notify_on_drain) {
531                         if (substream->partial_drain) {
532                                 /*
533                                  * Close old stream and make it stale, switch
534                                  * the active stream now!
535                                  */
536                                 q6asm_cmd_nowait(prtd->audio_client,
537                                                  prtd->stream_id,
538                                                  CMD_CLOSE);
539                                 /*
540                                  * vaild stream ids start from 1, So we are
541                                  * toggling this between 1 and 2.
542                                  */
543                                 prtd->stream_id = (prtd->stream_id == 1 ? 2 : 1);
544                         }
545 
546                         snd_compr_drain_notify(prtd->cstream);
547                         prtd->notify_on_drain = false;
548 
549                 } else {
550                         prtd->state = Q6ASM_STREAM_STOPPED;
551                 }
552                 spin_unlock_irqrestore(&prtd->lock, flags);
553                 break;
554 
555         case ASM_CLIENT_EVENT_DATA_WRITE_DONE:
556                 spin_lock_irqsave(&prtd->lock, flags);
557 
558                 bytes_written = token >> ASM_WRITE_TOKEN_LEN_SHIFT;
559                 prtd->copied_total += bytes_written;
560                 snd_compr_fragment_elapsed(substream);
561 
562                 if (prtd->state != Q6ASM_STREAM_RUNNING) {
563                         spin_unlock_irqrestore(&prtd->lock, flags);
564                         break;
565                 }
566 
567                 avail = prtd->bytes_received - prtd->bytes_sent;
568                 if (avail > prtd->pcm_count) {
569                         bytes_to_write = prtd->pcm_count;
570                 } else {
571                         if (substream->partial_drain || prtd->notify_on_drain)
572                                 is_last_buffer = true;
573                         bytes_to_write = avail;
574                 }
575 
576                 if (bytes_to_write) {
577                         if (substream->partial_drain && is_last_buffer) {
578                                 wflags |= ASM_LAST_BUFFER_FLAG;
579                                 q6asm_stream_remove_trailing_silence(prtd->audio_client,
580                                                      prtd->stream_id,
581                                                      prtd->trailing_samples_drop);
582                         }
583 
584                         q6asm_write_async(prtd->audio_client, prtd->stream_id,
585                                           bytes_to_write, 0, 0, wflags);
586 
587                         prtd->bytes_sent += bytes_to_write;
588                 }
589 
590                 if (prtd->notify_on_drain && is_last_buffer)
591                         q6asm_cmd_nowait(prtd->audio_client,
592                                          prtd->stream_id, CMD_EOS);
593 
594                 spin_unlock_irqrestore(&prtd->lock, flags);
595                 break;
596 
597         default:
598                 break;
599         }
600 }
601 
602 static int q6asm_dai_compr_open(struct snd_soc_component *component,
603                                 struct snd_compr_stream *stream)
604 {
605         struct snd_soc_pcm_runtime *rtd = stream->private_data;
606         struct snd_compr_runtime *runtime = stream->runtime;
607         struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0);
608         struct q6asm_dai_data *pdata;
609         struct device *dev = component->dev;
610         struct q6asm_dai_rtd *prtd;
611         int stream_id, size, ret;
612 
613         stream_id = cpu_dai->driver->id;
614         pdata = snd_soc_component_get_drvdata(component);
615         if (!pdata) {
616                 dev_err(dev, "Drv data not found ..\n");
617                 return -EINVAL;
618         }
619 
620         prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
621         if (!prtd)
622                 return -ENOMEM;
623 
624         /* DSP expects stream id from 1 */
625         prtd->stream_id = 1;
626 
627         prtd->cstream = stream;
628         prtd->audio_client = q6asm_audio_client_alloc(dev,
629                                         (q6asm_cb)compress_event_handler,
630                                         prtd, stream_id, LEGACY_PCM_MODE);
631         if (IS_ERR(prtd->audio_client)) {
632                 dev_err(dev, "Could not allocate memory\n");
633                 ret = PTR_ERR(prtd->audio_client);
634                 goto free_prtd;
635         }
636 
637         size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE *
638                         COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
639         ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size,
640                                   &prtd->dma_buffer);
641         if (ret) {
642                 dev_err(dev, "Cannot allocate buffer(s)\n");
643                 goto free_client;
644         }
645 
646         if (pdata->sid < 0)
647                 prtd->phys = prtd->dma_buffer.addr;
648         else
649                 prtd->phys = prtd->dma_buffer.addr | (pdata->sid << 32);
650 
651         snd_compr_set_runtime_buffer(stream, &prtd->dma_buffer);
652         spin_lock_init(&prtd->lock);
653         runtime->private_data = prtd;
654 
655         return 0;
656 
657 free_client:
658         q6asm_audio_client_free(prtd->audio_client);
659 free_prtd:
660         kfree(prtd);
661 
662         return ret;
663 }
664 
665 static int q6asm_dai_compr_free(struct snd_soc_component *component,
666                                 struct snd_compr_stream *stream)
667 {
668         struct snd_compr_runtime *runtime = stream->runtime;
669         struct q6asm_dai_rtd *prtd = runtime->private_data;
670         struct snd_soc_pcm_runtime *rtd = stream->private_data;
671 
672         if (prtd->audio_client) {
673                 if (prtd->state) {
674                         q6asm_cmd(prtd->audio_client, prtd->stream_id,
675                                   CMD_CLOSE);
676                         if (prtd->next_track_stream_id) {
677                                 q6asm_cmd(prtd->audio_client,
678                                           prtd->next_track_stream_id,
679                                           CMD_CLOSE);
680                         }
681                 }
682 
683                 snd_dma_free_pages(&prtd->dma_buffer);
684                 q6asm_unmap_memory_regions(stream->direction,
685                                            prtd->audio_client);
686                 q6asm_audio_client_free(prtd->audio_client);
687                 prtd->audio_client = NULL;
688         }
689         q6routing_stream_close(rtd->dai_link->id, stream->direction);
690         kfree(prtd);
691 
692         return 0;
693 }
694 
695 static int __q6asm_dai_compr_set_codec_params(struct snd_soc_component *component,
696                                               struct snd_compr_stream *stream,
697                                               struct snd_codec *codec,
698                                               int stream_id)
699 {
700         struct snd_compr_runtime *runtime = stream->runtime;
701         struct q6asm_dai_rtd *prtd = runtime->private_data;
702         struct q6asm_flac_cfg flac_cfg;
703         struct q6asm_wma_cfg wma_cfg;
704         struct q6asm_alac_cfg alac_cfg;
705         struct q6asm_ape_cfg ape_cfg;
706         unsigned int wma_v9 = 0;
707         struct device *dev = component->dev;
708         int ret;
709         union snd_codec_options *codec_options;
710         struct snd_dec_flac *flac;
711         struct snd_dec_wma *wma;
712         struct snd_dec_alac *alac;
713         struct snd_dec_ape *ape;
714 
715         codec_options = &(prtd->codec.options);
716 
717         memcpy(&prtd->codec, codec, sizeof(*codec));
718 
719         switch (codec->id) {
720         case SND_AUDIOCODEC_FLAC:
721 
722                 memset(&flac_cfg, 0x0, sizeof(struct q6asm_flac_cfg));
723                 flac = &codec_options->flac_d;
724 
725                 flac_cfg.ch_cfg = codec->ch_in;
726                 flac_cfg.sample_rate = codec->sample_rate;
727                 flac_cfg.stream_info_present = 1;
728                 flac_cfg.sample_size = flac->sample_size;
729                 flac_cfg.min_blk_size = flac->min_blk_size;
730                 flac_cfg.max_blk_size = flac->max_blk_size;
731                 flac_cfg.max_frame_size = flac->max_frame_size;
732                 flac_cfg.min_frame_size = flac->min_frame_size;
733 
734                 ret = q6asm_stream_media_format_block_flac(prtd->audio_client,
735                                                            stream_id,
736                                                            &flac_cfg);
737                 if (ret < 0) {
738                         dev_err(dev, "FLAC CMD Format block failed:%d\n", ret);
739                         return -EIO;
740                 }
741                 break;
742 
743         case SND_AUDIOCODEC_WMA:
744                 wma = &codec_options->wma_d;
745 
746                 memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg));
747 
748                 wma_cfg.sample_rate =  codec->sample_rate;
749                 wma_cfg.num_channels = codec->ch_in;
750                 wma_cfg.bytes_per_sec = codec->bit_rate / 8;
751                 wma_cfg.block_align = codec->align;
752                 wma_cfg.bits_per_sample = prtd->bits_per_sample;
753                 wma_cfg.enc_options = wma->encoder_option;
754                 wma_cfg.adv_enc_options = wma->adv_encoder_option;
755                 wma_cfg.adv_enc_options2 = wma->adv_encoder_option2;
756 
757                 if (wma_cfg.num_channels == 1)
758                         wma_cfg.channel_mask = 4; /* Mono Center */
759                 else if (wma_cfg.num_channels == 2)
760                         wma_cfg.channel_mask = 3; /* Stereo FL/FR */
761                 else
762                         return -EINVAL;
763 
764                 /* check the codec profile */
765                 switch (codec->profile) {
766                 case SND_AUDIOPROFILE_WMA9:
767                         wma_cfg.fmtag = 0x161;
768                         wma_v9 = 1;
769                         break;
770 
771                 case SND_AUDIOPROFILE_WMA10:
772                         wma_cfg.fmtag = 0x166;
773                         break;
774 
775                 case SND_AUDIOPROFILE_WMA9_PRO:
776                         wma_cfg.fmtag = 0x162;
777                         break;
778 
779                 case SND_AUDIOPROFILE_WMA9_LOSSLESS:
780                         wma_cfg.fmtag = 0x163;
781                         break;
782 
783                 case SND_AUDIOPROFILE_WMA10_LOSSLESS:
784                         wma_cfg.fmtag = 0x167;
785                         break;
786 
787                 default:
788                         dev_err(dev, "Unknown WMA profile:%x\n",
789                                 codec->profile);
790                         return -EIO;
791                 }
792 
793                 if (wma_v9)
794                         ret = q6asm_stream_media_format_block_wma_v9(
795                                         prtd->audio_client, stream_id,
796                                         &wma_cfg);
797                 else
798                         ret = q6asm_stream_media_format_block_wma_v10(
799                                         prtd->audio_client, stream_id,
800                                         &wma_cfg);
801                 if (ret < 0) {
802                         dev_err(dev, "WMA9 CMD failed:%d\n", ret);
803                         return -EIO;
804                 }
805                 break;
806 
807         case SND_AUDIOCODEC_ALAC:
808                 memset(&alac_cfg, 0x0, sizeof(alac_cfg));
809                 alac = &codec_options->alac_d;
810 
811                 alac_cfg.sample_rate = codec->sample_rate;
812                 alac_cfg.avg_bit_rate = codec->bit_rate;
813                 alac_cfg.bit_depth = prtd->bits_per_sample;
814                 alac_cfg.num_channels = codec->ch_in;
815 
816                 alac_cfg.frame_length = alac->frame_length;
817                 alac_cfg.pb = alac->pb;
818                 alac_cfg.mb = alac->mb;
819                 alac_cfg.kb = alac->kb;
820                 alac_cfg.max_run = alac->max_run;
821                 alac_cfg.compatible_version = alac->compatible_version;
822                 alac_cfg.max_frame_bytes = alac->max_frame_bytes;
823 
824                 switch (codec->ch_in) {
825                 case 1:
826                         alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_MONO;
827                         break;
828                 case 2:
829                         alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_STEREO;
830                         break;
831                 }
832                 ret = q6asm_stream_media_format_block_alac(prtd->audio_client,
833                                                            stream_id,
834                                                            &alac_cfg);
835                 if (ret < 0) {
836                         dev_err(dev, "ALAC CMD Format block failed:%d\n", ret);
837                         return -EIO;
838                 }
839                 break;
840 
841         case SND_AUDIOCODEC_APE:
842                 memset(&ape_cfg, 0x0, sizeof(ape_cfg));
843                 ape = &codec_options->ape_d;
844 
845                 ape_cfg.sample_rate = codec->sample_rate;
846                 ape_cfg.num_channels = codec->ch_in;
847                 ape_cfg.bits_per_sample = prtd->bits_per_sample;
848 
849                 ape_cfg.compatible_version = ape->compatible_version;
850                 ape_cfg.compression_level = ape->compression_level;
851                 ape_cfg.format_flags = ape->format_flags;
852                 ape_cfg.blocks_per_frame = ape->blocks_per_frame;
853                 ape_cfg.final_frame_blocks = ape->final_frame_blocks;
854                 ape_cfg.total_frames = ape->total_frames;
855                 ape_cfg.seek_table_present = ape->seek_table_present;
856 
857                 ret = q6asm_stream_media_format_block_ape(prtd->audio_client,
858                                                           stream_id,
859                                                           &ape_cfg);
860                 if (ret < 0) {
861                         dev_err(dev, "APE CMD Format block failed:%d\n", ret);
862                         return -EIO;
863                 }
864                 break;
865 
866         default:
867                 break;
868         }
869 
870         return 0;
871 }
872 
873 static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
874                                       struct snd_compr_stream *stream,
875                                       struct snd_compr_params *params)
876 {
877         struct snd_compr_runtime *runtime = stream->runtime;
878         struct q6asm_dai_rtd *prtd = runtime->private_data;
879         struct snd_soc_pcm_runtime *rtd = stream->private_data;
880         int dir = stream->direction;
881         struct q6asm_dai_data *pdata;
882         struct device *dev = component->dev;
883         int ret;
884 
885         pdata = snd_soc_component_get_drvdata(component);
886         if (!pdata)
887                 return -EINVAL;
888 
889         if (!prtd || !prtd->audio_client) {
890                 dev_err(dev, "private data null or audio client freed\n");
891                 return -EINVAL;
892         }
893 
894         prtd->periods = runtime->fragments;
895         prtd->pcm_count = runtime->fragment_size;
896         prtd->pcm_size = runtime->fragments * runtime->fragment_size;
897         prtd->bits_per_sample = 16;
898 
899         if (dir == SND_COMPRESS_PLAYBACK) {
900                 ret = q6asm_open_write(prtd->audio_client, prtd->stream_id, params->codec.id,
901                                 params->codec.profile, prtd->bits_per_sample,
902                                 true);
903 
904                 if (ret < 0) {
905                         dev_err(dev, "q6asm_open_write failed\n");
906                         q6asm_audio_client_free(prtd->audio_client);
907                         prtd->audio_client = NULL;
908                         return ret;
909                 }
910         }
911 
912         prtd->session_id = q6asm_get_session_id(prtd->audio_client);
913         ret = q6routing_stream_open(rtd->dai_link->id, LEGACY_PCM_MODE,
914                               prtd->session_id, dir);
915         if (ret) {
916                 dev_err(dev, "Stream reg failed ret:%d\n", ret);
917                 return ret;
918         }
919 
920         ret = __q6asm_dai_compr_set_codec_params(component, stream,
921                                                  &params->codec,
922                                                  prtd->stream_id);
923         if (ret) {
924                 dev_err(dev, "codec param setup failed ret:%d\n", ret);
925                 return ret;
926         }
927 
928         ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys,
929                                        (prtd->pcm_size / prtd->periods),
930                                        prtd->periods);
931 
932         if (ret < 0) {
933                 dev_err(dev, "Buffer Mapping failed ret:%d\n", ret);
934                 return -ENOMEM;
935         }
936 
937         prtd->state = Q6ASM_STREAM_RUNNING;
938 
939         return 0;
940 }
941 
942 static int q6asm_dai_compr_set_metadata(struct snd_soc_component *component,
943                                         struct snd_compr_stream *stream,
944                                         struct snd_compr_metadata *metadata)
945 {
946         struct snd_compr_runtime *runtime = stream->runtime;
947         struct q6asm_dai_rtd *prtd = runtime->private_data;
948         int ret = 0;
949 
950         switch (metadata->key) {
951         case SNDRV_COMPRESS_ENCODER_PADDING:
952                 prtd->trailing_samples_drop = metadata->value[0];
953                 break;
954         case SNDRV_COMPRESS_ENCODER_DELAY:
955                 prtd->initial_samples_drop = metadata->value[0];
956                 if (prtd->next_track_stream_id) {
957                         ret = q6asm_open_write(prtd->audio_client,
958                                                prtd->next_track_stream_id,
959                                                prtd->codec.id,
960                                                prtd->codec.profile,
961                                                prtd->bits_per_sample,
962                                        true);
963                         if (ret < 0) {
964                                 dev_err(component->dev, "q6asm_open_write failed\n");
965                                 return ret;
966                         }
967                         ret = __q6asm_dai_compr_set_codec_params(component, stream,
968                                                                  &prtd->codec,
969                                                                  prtd->next_track_stream_id);
970                         if (ret < 0) {
971                                 dev_err(component->dev, "q6asm_open_write failed\n");
972                                 return ret;
973                         }
974 
975                         ret = q6asm_stream_remove_initial_silence(prtd->audio_client,
976                                                     prtd->next_track_stream_id,
977                                                     prtd->initial_samples_drop);
978                         prtd->next_track_stream_id = 0;
979 
980                 }
981 
982                 break;
983         default:
984                 ret = -EINVAL;
985                 break;
986         }
987 
988         return ret;
989 }
990 
991 static int q6asm_dai_compr_trigger(struct snd_soc_component *component,
992                                    struct snd_compr_stream *stream, int cmd)
993 {
994         struct snd_compr_runtime *runtime = stream->runtime;
995         struct q6asm_dai_rtd *prtd = runtime->private_data;
996         int ret = 0;
997 
998         switch (cmd) {
999         case SNDRV_PCM_TRIGGER_START:
1000         case SNDRV_PCM_TRIGGER_RESUME:
1001         case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
1002                 ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id,
1003                                        0, 0, 0);
1004                 break;
1005         case SNDRV_PCM_TRIGGER_STOP:
1006                 prtd->state = Q6ASM_STREAM_STOPPED;
1007                 ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
1008                                        CMD_EOS);
1009                 break;
1010         case SNDRV_PCM_TRIGGER_SUSPEND:
1011         case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
1012                 ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
1013                                        CMD_PAUSE);
1014                 break;
1015         case SND_COMPR_TRIGGER_NEXT_TRACK:
1016                 prtd->next_track = true;
1017                 prtd->next_track_stream_id = (prtd->stream_id == 1 ? 2 : 1);
1018                 break;
1019         case SND_COMPR_TRIGGER_DRAIN:
1020         case SND_COMPR_TRIGGER_PARTIAL_DRAIN:
1021                 prtd->notify_on_drain = true;
1022                 break;
1023         default:
1024                 ret = -EINVAL;
1025                 break;
1026         }
1027 
1028         return ret;
1029 }
1030 
1031 static int q6asm_dai_compr_pointer(struct snd_soc_component *component,
1032                                    struct snd_compr_stream *stream,
1033                                    struct snd_compr_tstamp *tstamp)
1034 {
1035         struct snd_compr_runtime *runtime = stream->runtime;
1036         struct q6asm_dai_rtd *prtd = runtime->private_data;
1037         unsigned long flags;
1038 
1039         spin_lock_irqsave(&prtd->lock, flags);
1040 
1041         tstamp->copied_total = prtd->copied_total;
1042         tstamp->byte_offset = prtd->copied_total % prtd->pcm_size;
1043 
1044         spin_unlock_irqrestore(&prtd->lock, flags);
1045 
1046         return 0;
1047 }
1048 
1049 static int q6asm_compr_copy(struct snd_soc_component *component,
1050                             struct snd_compr_stream *stream, char __user *buf,
1051                             size_t count)
1052 {
1053         struct snd_compr_runtime *runtime = stream->runtime;
1054         struct q6asm_dai_rtd *prtd = runtime->private_data;
1055         unsigned long flags;
1056         u32 wflags = 0;
1057         int avail, bytes_in_flight = 0;
1058         void *dstn;
1059         size_t copy;
1060         u32 app_pointer;
1061         u32 bytes_received;
1062 
1063         bytes_received = prtd->bytes_received;
1064 
1065         /**
1066          * Make sure that next track data pointer is aligned at 32 bit boundary
1067          * This is a Mandatory requirement from DSP data buffers alignment
1068          */
1069         if (prtd->next_track)
1070                 bytes_received = ALIGN(prtd->bytes_received, prtd->pcm_count);
1071 
1072         app_pointer = bytes_received/prtd->pcm_size;
1073         app_pointer = bytes_received -  (app_pointer * prtd->pcm_size);
1074         dstn = prtd->dma_buffer.area + app_pointer;
1075 
1076         if (count < prtd->pcm_size - app_pointer) {
1077                 if (copy_from_user(dstn, buf, count))
1078                         return -EFAULT;
1079         } else {
1080                 copy = prtd->pcm_size - app_pointer;
1081                 if (copy_from_user(dstn, buf, copy))
1082                         return -EFAULT;
1083                 if (copy_from_user(prtd->dma_buffer.area, buf + copy,
1084                                    count - copy))
1085                         return -EFAULT;
1086         }
1087 
1088         spin_lock_irqsave(&prtd->lock, flags);
1089 
1090         bytes_in_flight = prtd->bytes_received - prtd->copied_total;
1091 
1092         if (prtd->next_track) {
1093                 prtd->next_track = false;
1094                 prtd->copied_total = ALIGN(prtd->copied_total, prtd->pcm_count);
1095                 prtd->bytes_sent = ALIGN(prtd->bytes_sent, prtd->pcm_count);
1096         }
1097 
1098         prtd->bytes_received = bytes_received + count;
1099 
1100         /* Kick off the data to dsp if its starving!! */
1101         if (prtd->state == Q6ASM_STREAM_RUNNING && (bytes_in_flight == 0)) {
1102                 uint32_t bytes_to_write = prtd->pcm_count;
1103 
1104                 avail = prtd->bytes_received - prtd->bytes_sent;
1105 
1106                 if (avail < prtd->pcm_count)
1107                         bytes_to_write = avail;
1108 
1109                 q6asm_write_async(prtd->audio_client, prtd->stream_id,
1110                                   bytes_to_write, 0, 0, wflags);
1111                 prtd->bytes_sent += bytes_to_write;
1112         }
1113 
1114         spin_unlock_irqrestore(&prtd->lock, flags);
1115 
1116         return count;
1117 }
1118 
1119 static int q6asm_dai_compr_mmap(struct snd_soc_component *component,
1120                                 struct snd_compr_stream *stream,
1121                                 struct vm_area_struct *vma)
1122 {
1123         struct snd_compr_runtime *runtime = stream->runtime;
1124         struct q6asm_dai_rtd *prtd = runtime->private_data;
1125         struct device *dev = component->dev;
1126 
1127         return dma_mmap_coherent(dev, vma,
1128                         prtd->dma_buffer.area, prtd->dma_buffer.addr,
1129                         prtd->dma_buffer.bytes);
1130 }
1131 
1132 static int q6asm_dai_compr_get_caps(struct snd_soc_component *component,
1133                                     struct snd_compr_stream *stream,
1134                                     struct snd_compr_caps *caps)
1135 {
1136         caps->direction = SND_COMPRESS_PLAYBACK;
1137         caps->min_fragment_size = COMPR_PLAYBACK_MIN_FRAGMENT_SIZE;
1138         caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE;
1139         caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS;
1140         caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
1141         caps->num_codecs = 5;
1142         caps->codecs[0] = SND_AUDIOCODEC_MP3;
1143         caps->codecs[1] = SND_AUDIOCODEC_FLAC;
1144         caps->codecs[2] = SND_AUDIOCODEC_WMA;
1145         caps->codecs[3] = SND_AUDIOCODEC_ALAC;
1146         caps->codecs[4] = SND_AUDIOCODEC_APE;
1147 
1148         return 0;
1149 }
1150 
1151 static int q6asm_dai_compr_get_codec_caps(struct snd_soc_component *component,
1152                                           struct snd_compr_stream *stream,
1153                                           struct snd_compr_codec_caps *codec)
1154 {
1155         switch (codec->codec) {
1156         case SND_AUDIOCODEC_MP3:
1157                 *codec = q6asm_compr_caps;
1158                 break;
1159         default:
1160                 break;
1161         }
1162 
1163         return 0;
1164 }
1165 
1166 static const struct snd_compress_ops q6asm_dai_compress_ops = {
1167         .open           = q6asm_dai_compr_open,
1168         .free           = q6asm_dai_compr_free,
1169         .set_params     = q6asm_dai_compr_set_params,
1170         .set_metadata   = q6asm_dai_compr_set_metadata,
1171         .pointer        = q6asm_dai_compr_pointer,
1172         .trigger        = q6asm_dai_compr_trigger,
1173         .get_caps       = q6asm_dai_compr_get_caps,
1174         .get_codec_caps = q6asm_dai_compr_get_codec_caps,
1175         .mmap           = q6asm_dai_compr_mmap,
1176         .copy           = q6asm_compr_copy,
1177 };
1178 
1179 static int q6asm_dai_pcm_new(struct snd_soc_component *component,
1180                              struct snd_soc_pcm_runtime *rtd)
1181 {
1182         struct snd_pcm *pcm = rtd->pcm;
1183         size_t size = q6asm_dai_hardware_playback.buffer_bytes_max;
1184 
1185         return snd_pcm_set_fixed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV,
1186                                             component->dev, size);
1187 }
1188 
1189 static const struct snd_soc_dapm_widget q6asm_dapm_widgets[] = {
1190         SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, SND_SOC_NOPM, 0, 0),
1191         SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, SND_SOC_NOPM, 0, 0),
1192         SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, SND_SOC_NOPM, 0, 0),
1193         SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, SND_SOC_NOPM, 0, 0),
1194         SND_SOC_DAPM_AIF_IN("MM_DL5", "MultiMedia5 Playback", 0, SND_SOC_NOPM, 0, 0),
1195         SND_SOC_DAPM_AIF_IN("MM_DL6", "MultiMedia6 Playback", 0, SND_SOC_NOPM, 0, 0),
1196         SND_SOC_DAPM_AIF_IN("MM_DL7", "MultiMedia7 Playback", 0, SND_SOC_NOPM, 0, 0),
1197         SND_SOC_DAPM_AIF_IN("MM_DL8", "MultiMedia8 Playback", 0, SND_SOC_NOPM, 0, 0),
1198         SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, SND_SOC_NOPM, 0, 0),
1199         SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, SND_SOC_NOPM, 0, 0),
1200         SND_SOC_DAPM_AIF_OUT("MM_UL3", "MultiMedia3 Capture", 0, SND_SOC_NOPM, 0, 0),
1201         SND_SOC_DAPM_AIF_OUT("MM_UL4", "MultiMedia4 Capture", 0, SND_SOC_NOPM, 0, 0),
1202         SND_SOC_DAPM_AIF_OUT("MM_UL5", "MultiMedia5 Capture", 0, SND_SOC_NOPM, 0, 0),
1203         SND_SOC_DAPM_AIF_OUT("MM_UL6", "MultiMedia6 Capture", 0, SND_SOC_NOPM, 0, 0),
1204         SND_SOC_DAPM_AIF_OUT("MM_UL7", "MultiMedia7 Capture", 0, SND_SOC_NOPM, 0, 0),
1205         SND_SOC_DAPM_AIF_OUT("MM_UL8", "MultiMedia8 Capture", 0, SND_SOC_NOPM, 0, 0),
1206 };
1207 
1208 static const struct snd_soc_component_driver q6asm_fe_dai_component = {
1209         .name                   = DRV_NAME,
1210         .open                   = q6asm_dai_open,
1211         .hw_params              = q6asm_dai_hw_params,
1212         .close                  = q6asm_dai_close,
1213         .prepare                = q6asm_dai_prepare,
1214         .trigger                = q6asm_dai_trigger,
1215         .pointer                = q6asm_dai_pointer,
1216         .pcm_construct          = q6asm_dai_pcm_new,
1217         .compress_ops           = &q6asm_dai_compress_ops,
1218         .dapm_widgets           = q6asm_dapm_widgets,
1219         .num_dapm_widgets       = ARRAY_SIZE(q6asm_dapm_widgets),
1220         .legacy_dai_naming      = 1,
1221 };
1222 
1223 static struct snd_soc_dai_driver q6asm_fe_dais_template[] = {
1224         Q6ASM_FEDAI_DRIVER(1),
1225         Q6ASM_FEDAI_DRIVER(2),
1226         Q6ASM_FEDAI_DRIVER(3),
1227         Q6ASM_FEDAI_DRIVER(4),
1228         Q6ASM_FEDAI_DRIVER(5),
1229         Q6ASM_FEDAI_DRIVER(6),
1230         Q6ASM_FEDAI_DRIVER(7),
1231         Q6ASM_FEDAI_DRIVER(8),
1232 };
1233 
1234 static const struct snd_soc_dai_ops q6asm_dai_ops = {
1235         .compress_new = snd_soc_new_compress,
1236 };
1237 
1238 static int of_q6asm_parse_dai_data(struct device *dev,
1239                                     struct q6asm_dai_data *pdata)
1240 {
1241         struct snd_soc_dai_driver *dai_drv;
1242         struct snd_soc_pcm_stream empty_stream;
1243         struct device_node *node;
1244         int ret, id, dir, idx = 0;
1245 
1246 
1247         pdata->num_dais = of_get_child_count(dev->of_node);
1248         if (!pdata->num_dais) {
1249                 dev_err(dev, "No dais found in DT\n");
1250                 return -EINVAL;
1251         }
1252 
1253         pdata->dais = devm_kcalloc(dev, pdata->num_dais, sizeof(*dai_drv),
1254                                    GFP_KERNEL);
1255         if (!pdata->dais)
1256                 return -ENOMEM;
1257 
1258         memset(&empty_stream, 0, sizeof(empty_stream));
1259 
1260         for_each_child_of_node(dev->of_node, node) {
1261                 ret = of_property_read_u32(node, "reg", &id);
1262                 if (ret || id >= MAX_SESSIONS || id < 0) {
1263                         dev_err(dev, "valid dai id not found:%d\n", ret);
1264                         continue;
1265                 }
1266 
1267                 dai_drv = &pdata->dais[idx++];
1268                 *dai_drv = q6asm_fe_dais_template[id];
1269 
1270                 ret = of_property_read_u32(node, "direction", &dir);
1271                 if (ret)
1272                         continue;
1273 
1274                 if (dir == Q6ASM_DAI_RX)
1275                         dai_drv->capture = empty_stream;
1276                 else if (dir == Q6ASM_DAI_TX)
1277                         dai_drv->playback = empty_stream;
1278 
1279                 if (of_property_read_bool(node, "is-compress-dai"))
1280                         dai_drv->ops = &q6asm_dai_ops;
1281         }
1282 
1283         return 0;
1284 }
1285 
1286 static int q6asm_dai_probe(struct platform_device *pdev)
1287 {
1288         struct device *dev = &pdev->dev;
1289         struct device_node *node = dev->of_node;
1290         struct of_phandle_args args;
1291         struct q6asm_dai_data *pdata;
1292         int rc;
1293 
1294         pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL);
1295         if (!pdata)
1296                 return -ENOMEM;
1297 
1298         rc = of_parse_phandle_with_fixed_args(node, "iommus", 1, 0, &args);
1299         if (rc < 0)
1300                 pdata->sid = -1;
1301         else
1302                 pdata->sid = args.args[0] & SID_MASK_DEFAULT;
1303 
1304         dev_set_drvdata(dev, pdata);
1305 
1306         rc = of_q6asm_parse_dai_data(dev, pdata);
1307         if (rc)
1308                 return rc;
1309 
1310         return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component,
1311                                                pdata->dais, pdata->num_dais);
1312 }
1313 
1314 #ifdef CONFIG_OF
1315 static const struct of_device_id q6asm_dai_device_id[] = {
1316         { .compatible = "qcom,q6asm-dais" },
1317         {},
1318 };
1319 MODULE_DEVICE_TABLE(of, q6asm_dai_device_id);
1320 #endif
1321 
1322 static struct platform_driver q6asm_dai_platform_driver = {
1323         .driver = {
1324                 .name = "q6asm-dai",
1325                 .of_match_table = of_match_ptr(q6asm_dai_device_id),
1326         },
1327         .probe = q6asm_dai_probe,
1328 };
1329 module_platform_driver(q6asm_dai_platform_driver);
1330 
1331 MODULE_DESCRIPTION("Q6ASM dai driver");
1332 MODULE_LICENSE("GPL v2");
1333 

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