Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 1 | /* Copyright (c) 2017, The Linux Foundation. All rights reserved. |
| 2 | * |
| 3 | * This program is free software; you can redistribute it and/or modify |
| 4 | * it under the terms of the GNU General Public License version 2 and |
| 5 | * only version 2 as published by the Free Software Foundation. |
| 6 | * |
| 7 | * This program is distributed in the hope that it will be useful, |
| 8 | * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| 9 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
| 10 | * GNU General Public License for more details. |
| 11 | */ |
| 12 | |
| 13 | #include <linux/init.h> |
| 14 | #include <linux/err.h> |
| 15 | #include <linux/module.h> |
| 16 | #include <linux/moduleparam.h> |
| 17 | #include <linux/time.h> |
| 18 | #include <linux/math64.h> |
| 19 | #include <linux/wait.h> |
| 20 | #include <linux/platform_device.h> |
| 21 | #include <linux/slab.h> |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 22 | #include <sound/core.h> |
| 23 | #include <sound/soc.h> |
| 24 | #include <sound/soc-dapm.h> |
| 25 | #include <sound/pcm.h> |
| 26 | #include <sound/initval.h> |
| 27 | #include <sound/control.h> |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 28 | #include <sound/pcm_params.h> |
| 29 | #include <sound/timer.h> |
| 30 | #include <sound/tlv.h> |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 31 | #include <sound/compress_params.h> |
| 32 | #include <sound/compress_offload.h> |
| 33 | #include <sound/compress_driver.h> |
Laxminath Kasam | 605b42f | 2017-08-01 22:02:15 +0530 | [diff] [blame] | 34 | #include <dsp/msm_audio_ion.h> |
| 35 | #include <dsp/apr_audio-v2.h> |
| 36 | #include <dsp/q6asm-v2.h> |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 37 | |
| 38 | #include "msm-pcm-routing-v2.h" |
| 39 | #include "msm-qti-pp-config.h" |
| 40 | |
| 41 | #define LOOPBACK_SESSION_MAX_NUM_STREAMS 2 |
| 42 | |
| 43 | static DEFINE_MUTEX(transcode_loopback_session_lock); |
| 44 | |
| 45 | struct trans_loopback_pdata { |
| 46 | struct snd_compr_stream *cstream[MSM_FRONTEND_DAI_MAX]; |
| 47 | }; |
| 48 | |
| 49 | struct loopback_stream { |
| 50 | struct snd_compr_stream *cstream; |
| 51 | uint32_t codec_format; |
| 52 | bool start; |
| 53 | }; |
| 54 | |
| 55 | enum loopback_session_state { |
| 56 | /* One or both streams not opened */ |
| 57 | LOOPBACK_SESSION_CLOSE = 0, |
| 58 | /* Loopback streams opened */ |
| 59 | LOOPBACK_SESSION_READY, |
| 60 | /* Loopback streams opened and formats configured */ |
| 61 | LOOPBACK_SESSION_START, |
| 62 | /* Trigger issued on either of streams when in START state */ |
| 63 | LOOPBACK_SESSION_RUN |
| 64 | }; |
| 65 | |
| 66 | struct msm_transcode_loopback { |
| 67 | struct loopback_stream source; |
| 68 | struct loopback_stream sink; |
| 69 | |
| 70 | struct snd_compr_caps source_compr_cap; |
| 71 | struct snd_compr_caps sink_compr_cap; |
| 72 | |
| 73 | uint32_t instance; |
| 74 | uint32_t num_streams; |
| 75 | int session_state; |
| 76 | |
| 77 | struct mutex lock; |
| 78 | |
| 79 | int session_id; |
| 80 | struct audio_client *audio_client; |
| 81 | }; |
| 82 | |
| 83 | /* Transcode loopback global info struct */ |
| 84 | static struct msm_transcode_loopback transcode_info; |
| 85 | |
| 86 | static void loopback_event_handler(uint32_t opcode, |
| 87 | uint32_t token, uint32_t *payload, void *priv) |
| 88 | { |
| 89 | struct msm_transcode_loopback *trans = |
| 90 | (struct msm_transcode_loopback *)priv; |
| 91 | struct snd_soc_pcm_runtime *rtd; |
| 92 | struct snd_compr_stream *cstream; |
| 93 | struct audio_client *ac; |
| 94 | int stream_id; |
| 95 | int ret; |
| 96 | |
| 97 | if (!trans || !payload) { |
| 98 | pr_err("%s: rtd or payload is NULL\n", __func__); |
| 99 | return; |
| 100 | } |
| 101 | |
| 102 | cstream = trans->source.cstream; |
| 103 | ac = trans->audio_client; |
| 104 | |
| 105 | /* |
| 106 | * Token for rest of the compressed commands use to set |
| 107 | * session id, stream id, dir etc. |
| 108 | */ |
| 109 | stream_id = q6asm_get_stream_id_from_token(token); |
| 110 | |
| 111 | switch (opcode) { |
| 112 | case ASM_STREAM_CMD_ENCDEC_EVENTS: |
| 113 | case ASM_IEC_61937_MEDIA_FMT_EVENT: |
| 114 | pr_debug("%s: ASM_IEC_61937_MEDIA_FMT_EVENT\n", __func__); |
| 115 | rtd = cstream->private_data; |
| 116 | if (!rtd) { |
| 117 | pr_err("%s: rtd is NULL\n", __func__); |
| 118 | return; |
| 119 | } |
| 120 | |
| 121 | ret = msm_adsp_inform_mixer_ctl(rtd, payload); |
| 122 | if (ret) { |
| 123 | pr_err("%s: failed to inform mixer ctrl. err = %d\n", |
| 124 | __func__, ret); |
| 125 | return; |
| 126 | } |
| 127 | break; |
| 128 | case APR_BASIC_RSP_RESULT: { |
| 129 | switch (payload[0]) { |
| 130 | case ASM_SESSION_CMD_RUN_V2: |
| 131 | pr_debug("%s: ASM_SESSION_CMD_RUN_V2:", __func__); |
| 132 | pr_debug("token 0x%x, stream id %d\n", token, |
| 133 | stream_id); |
| 134 | break; |
| 135 | case ASM_STREAM_CMD_CLOSE: |
| 136 | pr_debug("%s: ASM_DATA_CMD_CLOSE:", __func__); |
| 137 | pr_debug("token 0x%x, stream id %d\n", token, |
| 138 | stream_id); |
| 139 | break; |
| 140 | default: |
| 141 | break; |
| 142 | } |
| 143 | break; |
| 144 | } |
| 145 | default: |
| 146 | pr_debug("%s: Not Supported Event opcode[0x%x]\n", |
| 147 | __func__, opcode); |
| 148 | break; |
| 149 | } |
| 150 | } |
| 151 | |
| 152 | static void populate_codec_list(struct msm_transcode_loopback *trans, |
| 153 | struct snd_compr_stream *cstream) |
| 154 | { |
| 155 | struct snd_compr_caps compr_cap; |
| 156 | |
| 157 | pr_debug("%s\n", __func__); |
| 158 | |
| 159 | memset(&compr_cap, 0, sizeof(struct snd_compr_caps)); |
| 160 | |
| 161 | if (cstream->direction == SND_COMPRESS_CAPTURE) { |
| 162 | compr_cap.direction = SND_COMPRESS_CAPTURE; |
| 163 | compr_cap.num_codecs = 3; |
| 164 | compr_cap.codecs[0] = SND_AUDIOCODEC_PCM; |
| 165 | compr_cap.codecs[1] = SND_AUDIOCODEC_AC3; |
| 166 | compr_cap.codecs[2] = SND_AUDIOCODEC_EAC3; |
| 167 | memcpy(&trans->source_compr_cap, &compr_cap, |
| 168 | sizeof(struct snd_compr_caps)); |
| 169 | } |
| 170 | |
| 171 | if (cstream->direction == SND_COMPRESS_PLAYBACK) { |
| 172 | compr_cap.direction = SND_COMPRESS_PLAYBACK; |
| 173 | compr_cap.num_codecs = 1; |
| 174 | compr_cap.codecs[0] = SND_AUDIOCODEC_PCM; |
| 175 | memcpy(&trans->sink_compr_cap, &compr_cap, |
| 176 | sizeof(struct snd_compr_caps)); |
| 177 | } |
| 178 | } |
| 179 | |
| 180 | static int msm_transcode_loopback_open(struct snd_compr_stream *cstream) |
| 181 | { |
| 182 | int ret = 0; |
| 183 | struct snd_compr_runtime *runtime; |
| 184 | struct snd_soc_pcm_runtime *rtd; |
| 185 | struct msm_transcode_loopback *trans = &transcode_info; |
| 186 | struct trans_loopback_pdata *pdata; |
| 187 | |
| 188 | if (cstream == NULL) { |
| 189 | pr_err("%s: Invalid substream\n", __func__); |
| 190 | return -EINVAL; |
| 191 | } |
| 192 | runtime = cstream->runtime; |
| 193 | rtd = snd_pcm_substream_chip(cstream); |
| 194 | pdata = snd_soc_platform_get_drvdata(rtd->platform); |
| 195 | pdata->cstream[rtd->dai_link->id] = cstream; |
| 196 | |
| 197 | mutex_lock(&trans->lock); |
| 198 | if (trans->num_streams > LOOPBACK_SESSION_MAX_NUM_STREAMS) { |
| 199 | pr_err("msm_transcode_open failed..invalid stream\n"); |
| 200 | ret = -EINVAL; |
| 201 | goto exit; |
| 202 | } |
| 203 | |
| 204 | if (cstream->direction == SND_COMPRESS_CAPTURE) { |
| 205 | if (trans->source.cstream == NULL) { |
| 206 | trans->source.cstream = cstream; |
| 207 | trans->num_streams++; |
| 208 | } else { |
| 209 | pr_err("%s: capture stream already opened\n", |
| 210 | __func__); |
| 211 | ret = -EINVAL; |
| 212 | goto exit; |
| 213 | } |
| 214 | } else if (cstream->direction == SND_COMPRESS_PLAYBACK) { |
| 215 | if (trans->sink.cstream == NULL) { |
| 216 | trans->sink.cstream = cstream; |
| 217 | trans->num_streams++; |
| 218 | } else { |
| 219 | pr_debug("%s: playback stream already opened\n", |
| 220 | __func__); |
| 221 | ret = -EINVAL; |
| 222 | goto exit; |
| 223 | } |
| 224 | } |
| 225 | |
| 226 | pr_debug("%s: num stream%d, stream name %s\n", __func__, |
| 227 | trans->num_streams, cstream->name); |
| 228 | |
| 229 | populate_codec_list(trans, cstream); |
| 230 | |
| 231 | if (trans->num_streams == LOOPBACK_SESSION_MAX_NUM_STREAMS) { |
| 232 | pr_debug("%s: Moving loopback session to READY state %d\n", |
| 233 | __func__, trans->session_state); |
| 234 | trans->session_state = LOOPBACK_SESSION_READY; |
| 235 | } |
| 236 | |
| 237 | runtime->private_data = trans; |
| 238 | if (trans->num_streams == 1) |
| 239 | msm_adsp_init_mixer_ctl_pp_event_queue(rtd); |
| 240 | exit: |
| 241 | mutex_unlock(&trans->lock); |
| 242 | return ret; |
| 243 | } |
| 244 | |
| 245 | static void stop_transcoding(struct msm_transcode_loopback *trans) |
| 246 | { |
| 247 | struct snd_soc_pcm_runtime *soc_pcm_rx; |
| 248 | struct snd_soc_pcm_runtime *soc_pcm_tx; |
| 249 | |
| 250 | if (trans->audio_client != NULL) { |
| 251 | q6asm_cmd(trans->audio_client, CMD_CLOSE); |
| 252 | |
| 253 | if (trans->sink.cstream != NULL) { |
| 254 | soc_pcm_rx = trans->sink.cstream->private_data; |
| 255 | msm_pcm_routing_dereg_phy_stream( |
| 256 | soc_pcm_rx->dai_link->id, |
| 257 | SND_COMPRESS_PLAYBACK); |
| 258 | } |
| 259 | if (trans->source.cstream != NULL) { |
| 260 | soc_pcm_tx = trans->source.cstream->private_data; |
| 261 | msm_pcm_routing_dereg_phy_stream( |
| 262 | soc_pcm_tx->dai_link->id, |
| 263 | SND_COMPRESS_CAPTURE); |
| 264 | } |
| 265 | q6asm_audio_client_free(trans->audio_client); |
| 266 | trans->audio_client = NULL; |
| 267 | } |
| 268 | } |
| 269 | |
| 270 | static int msm_transcode_loopback_free(struct snd_compr_stream *cstream) |
| 271 | { |
| 272 | struct snd_compr_runtime *runtime = cstream->runtime; |
| 273 | struct msm_transcode_loopback *trans = runtime->private_data; |
| 274 | struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(cstream); |
| 275 | int ret = 0; |
| 276 | |
| 277 | mutex_lock(&trans->lock); |
| 278 | |
| 279 | pr_debug("%s: Transcode loopback end:%d, streams %d\n", __func__, |
| 280 | cstream->direction, trans->num_streams); |
| 281 | trans->num_streams--; |
| 282 | stop_transcoding(trans); |
| 283 | |
| 284 | if (cstream->direction == SND_COMPRESS_PLAYBACK) |
| 285 | memset(&trans->sink, 0, sizeof(struct loopback_stream)); |
| 286 | else if (cstream->direction == SND_COMPRESS_CAPTURE) |
| 287 | memset(&trans->source, 0, sizeof(struct loopback_stream)); |
| 288 | |
| 289 | trans->session_state = LOOPBACK_SESSION_CLOSE; |
| 290 | if (trans->num_streams == 1) |
| 291 | msm_adsp_clean_mixer_ctl_pp_event_queue(rtd); |
| 292 | mutex_unlock(&trans->lock); |
| 293 | return ret; |
| 294 | } |
| 295 | |
| 296 | static int msm_transcode_loopback_trigger(struct snd_compr_stream *cstream, |
| 297 | int cmd) |
| 298 | { |
| 299 | struct snd_compr_runtime *runtime = cstream->runtime; |
| 300 | struct msm_transcode_loopback *trans = runtime->private_data; |
| 301 | |
| 302 | switch (cmd) { |
| 303 | case SNDRV_PCM_TRIGGER_START: |
| 304 | case SNDRV_PCM_TRIGGER_RESUME: |
| 305 | case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: |
| 306 | |
| 307 | if (trans->session_state == LOOPBACK_SESSION_START) { |
| 308 | pr_debug("%s: Issue Loopback session %d RUN\n", |
| 309 | __func__, trans->instance); |
| 310 | q6asm_run_nowait(trans->audio_client, 0, 0, 0); |
| 311 | trans->session_state = LOOPBACK_SESSION_RUN; |
| 312 | } |
| 313 | break; |
| 314 | case SNDRV_PCM_TRIGGER_SUSPEND: |
| 315 | case SNDRV_PCM_TRIGGER_PAUSE_PUSH: |
| 316 | case SNDRV_PCM_TRIGGER_STOP: |
| 317 | pr_debug("%s: Issue Loopback session %d STOP\n", __func__, |
| 318 | trans->instance); |
| 319 | if (trans->session_state == LOOPBACK_SESSION_RUN) |
| 320 | q6asm_cmd_nowait(trans->audio_client, CMD_PAUSE); |
| 321 | trans->session_state = LOOPBACK_SESSION_START; |
| 322 | break; |
| 323 | |
| 324 | default: |
| 325 | break; |
| 326 | } |
| 327 | return 0; |
| 328 | } |
| 329 | |
| 330 | static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream, |
| 331 | struct snd_compr_params *codec_param) |
| 332 | { |
| 333 | |
| 334 | struct snd_compr_runtime *runtime = cstream->runtime; |
| 335 | struct msm_transcode_loopback *trans = runtime->private_data; |
| 336 | struct snd_soc_pcm_runtime *soc_pcm_rx; |
| 337 | struct snd_soc_pcm_runtime *soc_pcm_tx; |
| 338 | uint32_t bit_width = 16; |
| 339 | int ret = 0; |
| 340 | |
| 341 | if (trans == NULL) { |
| 342 | pr_err("%s: Invalid param\n", __func__); |
| 343 | return -EINVAL; |
| 344 | } |
| 345 | |
| 346 | mutex_lock(&trans->lock); |
| 347 | |
| 348 | if (cstream->direction == SND_COMPRESS_PLAYBACK) { |
| 349 | if (codec_param->codec.id == SND_AUDIOCODEC_PCM) { |
| 350 | trans->sink.codec_format = |
| 351 | FORMAT_LINEAR_PCM; |
| 352 | switch (codec_param->codec.format) { |
| 353 | case SNDRV_PCM_FORMAT_S32_LE: |
| 354 | bit_width = 32; |
| 355 | break; |
| 356 | case SNDRV_PCM_FORMAT_S24_LE: |
| 357 | bit_width = 24; |
| 358 | break; |
| 359 | case SNDRV_PCM_FORMAT_S24_3LE: |
| 360 | bit_width = 24; |
| 361 | break; |
| 362 | case SNDRV_PCM_FORMAT_S16_LE: |
| 363 | default: |
| 364 | bit_width = 16; |
| 365 | break; |
| 366 | } |
| 367 | } else { |
| 368 | pr_debug("%s: unknown sink codec\n", __func__); |
| 369 | ret = -EINVAL; |
| 370 | goto exit; |
| 371 | } |
| 372 | trans->sink.start = true; |
| 373 | } |
| 374 | |
| 375 | if (cstream->direction == SND_COMPRESS_CAPTURE) { |
| 376 | switch (codec_param->codec.id) { |
| 377 | case SND_AUDIOCODEC_PCM: |
| 378 | pr_debug("Source SND_AUDIOCODEC_PCM\n"); |
| 379 | trans->source.codec_format = |
| 380 | FORMAT_LINEAR_PCM; |
| 381 | break; |
| 382 | case SND_AUDIOCODEC_AC3: |
| 383 | pr_debug("Source SND_AUDIOCODEC_AC3\n"); |
| 384 | trans->source.codec_format = |
| 385 | FORMAT_AC3; |
| 386 | break; |
| 387 | case SND_AUDIOCODEC_EAC3: |
| 388 | pr_debug("Source SND_AUDIOCODEC_EAC3\n"); |
| 389 | trans->source.codec_format = |
| 390 | FORMAT_EAC3; |
| 391 | break; |
| 392 | default: |
| 393 | pr_debug("%s: unknown source codec\n", __func__); |
| 394 | ret = -EINVAL; |
| 395 | goto exit; |
| 396 | } |
| 397 | trans->source.start = true; |
| 398 | } |
| 399 | |
| 400 | pr_debug("%s: trans->source.start %d trans->sink.start %d trans->source.cstream %pK trans->sink.cstream %pK trans->session_state %d\n", |
| 401 | __func__, trans->source.start, trans->sink.start, |
| 402 | trans->source.cstream, trans->sink.cstream, |
| 403 | trans->session_state); |
| 404 | |
| 405 | if ((trans->session_state == LOOPBACK_SESSION_READY) && |
| 406 | trans->source.start && trans->sink.start) { |
| 407 | pr_debug("%s: Moving loopback session to start state\n", |
| 408 | __func__); |
| 409 | trans->session_state = LOOPBACK_SESSION_START; |
| 410 | } |
| 411 | |
| 412 | if (trans->session_state == LOOPBACK_SESSION_START) { |
| 413 | if (trans->audio_client != NULL) { |
| 414 | pr_debug("%s: ASM client already opened, closing\n", |
| 415 | __func__); |
| 416 | stop_transcoding(trans); |
| 417 | } |
| 418 | |
| 419 | trans->audio_client = q6asm_audio_client_alloc( |
| 420 | (app_cb)loopback_event_handler, trans); |
| 421 | if (!trans->audio_client) { |
| 422 | pr_err("%s: Could not allocate memory\n", __func__); |
| 423 | ret = -EINVAL; |
| 424 | goto exit; |
| 425 | } |
| 426 | pr_debug("%s: ASM client allocated, callback %pK\n", __func__, |
| 427 | loopback_event_handler); |
| 428 | trans->session_id = trans->audio_client->session; |
| 429 | trans->audio_client->perf_mode = false; |
| 430 | ret = q6asm_open_transcode_loopback(trans->audio_client, |
| 431 | bit_width, |
| 432 | trans->source.codec_format, |
| 433 | trans->sink.codec_format); |
| 434 | if (ret < 0) { |
| 435 | pr_err("%s: Session transcode loopback open failed\n", |
| 436 | __func__); |
| 437 | q6asm_audio_client_free(trans->audio_client); |
| 438 | trans->audio_client = NULL; |
| 439 | goto exit; |
| 440 | } |
| 441 | |
| 442 | pr_debug("%s: Starting ADM open for loopback\n", __func__); |
| 443 | soc_pcm_rx = trans->sink.cstream->private_data; |
| 444 | soc_pcm_tx = trans->source.cstream->private_data; |
| 445 | if (trans->source.codec_format != FORMAT_LINEAR_PCM) |
| 446 | msm_pcm_routing_reg_phy_compr_stream( |
| 447 | soc_pcm_tx->dai_link->id, |
| 448 | trans->audio_client->perf_mode, |
| 449 | trans->session_id, |
| 450 | SNDRV_PCM_STREAM_CAPTURE, |
| 451 | true); |
| 452 | else |
| 453 | msm_pcm_routing_reg_phy_stream( |
| 454 | soc_pcm_tx->dai_link->id, |
| 455 | trans->audio_client->perf_mode, |
| 456 | trans->session_id, |
| 457 | SNDRV_PCM_STREAM_CAPTURE); |
| 458 | |
| 459 | msm_pcm_routing_reg_phy_stream( |
| 460 | soc_pcm_rx->dai_link->id, |
| 461 | trans->audio_client->perf_mode, |
| 462 | trans->session_id, |
| 463 | SNDRV_PCM_STREAM_PLAYBACK); |
| 464 | pr_debug("%s: Successfully opened ADM sessions\n", __func__); |
| 465 | } |
| 466 | exit: |
| 467 | mutex_unlock(&trans->lock); |
| 468 | return ret; |
| 469 | } |
| 470 | |
| 471 | static int msm_transcode_loopback_get_caps(struct snd_compr_stream *cstream, |
| 472 | struct snd_compr_caps *arg) |
| 473 | { |
| 474 | struct snd_compr_runtime *runtime; |
| 475 | struct msm_transcode_loopback *trans; |
| 476 | |
| 477 | if (!arg || !cstream) { |
| 478 | pr_err("%s: Invalid arguments\n", __func__); |
| 479 | return -EINVAL; |
| 480 | } |
| 481 | |
| 482 | runtime = cstream->runtime; |
| 483 | trans = runtime->private_data; |
| 484 | pr_debug("%s\n", __func__); |
| 485 | if (cstream->direction == SND_COMPRESS_CAPTURE) |
| 486 | memcpy(arg, &trans->source_compr_cap, |
| 487 | sizeof(struct snd_compr_caps)); |
| 488 | else |
| 489 | memcpy(arg, &trans->sink_compr_cap, |
| 490 | sizeof(struct snd_compr_caps)); |
| 491 | return 0; |
| 492 | } |
| 493 | |
| 494 | static int msm_transcode_stream_cmd_put(struct snd_kcontrol *kcontrol, |
| 495 | struct snd_ctl_elem_value *ucontrol) |
| 496 | { |
| 497 | struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| 498 | unsigned long fe_id = kcontrol->private_value; |
| 499 | struct trans_loopback_pdata *pdata = (struct trans_loopback_pdata *) |
| 500 | snd_soc_component_get_drvdata(comp); |
| 501 | struct snd_compr_stream *cstream = NULL; |
| 502 | struct msm_transcode_loopback *prtd; |
| 503 | int ret = 0; |
| 504 | struct msm_adsp_event_data *event_data = NULL; |
| 505 | |
| 506 | if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| 507 | pr_err("%s Received invalid fe_id %lu\n", |
| 508 | __func__, fe_id); |
| 509 | ret = -EINVAL; |
| 510 | goto done; |
| 511 | } |
| 512 | |
| 513 | cstream = pdata->cstream[fe_id]; |
| 514 | if (cstream == NULL) { |
| 515 | pr_err("%s cstream is null.\n", __func__); |
| 516 | ret = -EINVAL; |
| 517 | goto done; |
| 518 | } |
| 519 | |
| 520 | prtd = cstream->runtime->private_data; |
| 521 | if (!prtd) { |
| 522 | pr_err("%s: prtd is null.\n", __func__); |
| 523 | ret = -EINVAL; |
| 524 | goto done; |
| 525 | } |
| 526 | |
| 527 | if (prtd->audio_client == NULL) { |
| 528 | pr_err("%s: audio_client is null.\n", __func__); |
| 529 | ret = -EINVAL; |
| 530 | goto done; |
| 531 | } |
| 532 | |
| 533 | event_data = (struct msm_adsp_event_data *)ucontrol->value.bytes.data; |
| 534 | if ((event_data->event_type < ADSP_STREAM_PP_EVENT) || |
| 535 | (event_data->event_type >= ADSP_STREAM_EVENT_MAX)) { |
| 536 | pr_err("%s: invalid event_type=%d", |
| 537 | __func__, event_data->event_type); |
| 538 | ret = -EINVAL; |
| 539 | goto done; |
| 540 | } |
| 541 | |
| 542 | if ((sizeof(struct msm_adsp_event_data) + event_data->payload_len) >= |
| 543 | sizeof(ucontrol->value.bytes.data)) { |
| 544 | pr_err("%s param length=%d exceeds limit", |
| 545 | __func__, event_data->payload_len); |
| 546 | ret = -EINVAL; |
| 547 | goto done; |
| 548 | } |
| 549 | |
| 550 | ret = q6asm_send_stream_cmd(prtd->audio_client, event_data); |
| 551 | if (ret < 0) |
| 552 | pr_err("%s: failed to send stream event cmd, err = %d\n", |
| 553 | __func__, ret); |
| 554 | done: |
| 555 | return ret; |
| 556 | } |
| 557 | |
| 558 | static int msm_transcode_ion_fd_map_put(struct snd_kcontrol *kcontrol, |
| 559 | struct snd_ctl_elem_value *ucontrol) |
| 560 | { |
| 561 | struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| 562 | unsigned long fe_id = kcontrol->private_value; |
| 563 | struct trans_loopback_pdata *pdata = (struct trans_loopback_pdata *) |
| 564 | snd_soc_component_get_drvdata(comp); |
| 565 | struct snd_compr_stream *cstream = NULL; |
| 566 | struct msm_transcode_loopback *prtd; |
| 567 | int fd; |
| 568 | int ret = 0; |
| 569 | |
| 570 | if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| 571 | pr_err("%s Received out of bounds invalid fe_id %lu\n", |
| 572 | __func__, fe_id); |
| 573 | ret = -EINVAL; |
| 574 | goto done; |
| 575 | } |
| 576 | |
| 577 | cstream = pdata->cstream[fe_id]; |
| 578 | if (cstream == NULL) { |
| 579 | pr_err("%s cstream is null\n", __func__); |
| 580 | ret = -EINVAL; |
| 581 | goto done; |
| 582 | } |
| 583 | |
| 584 | prtd = cstream->runtime->private_data; |
| 585 | if (!prtd) { |
| 586 | pr_err("%s: prtd is null\n", __func__); |
| 587 | ret = -EINVAL; |
| 588 | goto done; |
| 589 | } |
| 590 | |
| 591 | if (prtd->audio_client == NULL) { |
| 592 | pr_err("%s: audio_client is null\n", __func__); |
| 593 | ret = -EINVAL; |
| 594 | goto done; |
| 595 | } |
| 596 | |
| 597 | memcpy(&fd, ucontrol->value.bytes.data, sizeof(fd)); |
| 598 | ret = q6asm_send_ion_fd(prtd->audio_client, fd); |
| 599 | if (ret < 0) |
| 600 | pr_err("%s: failed to register ion fd\n", __func__); |
| 601 | done: |
| 602 | return ret; |
| 603 | } |
| 604 | |
| 605 | static int msm_transcode_rtic_event_ack_put(struct snd_kcontrol *kcontrol, |
| 606 | struct snd_ctl_elem_value *ucontrol) |
| 607 | { |
| 608 | struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| 609 | unsigned long fe_id = kcontrol->private_value; |
| 610 | struct trans_loopback_pdata *pdata = (struct trans_loopback_pdata *) |
| 611 | snd_soc_component_get_drvdata(comp); |
| 612 | struct snd_compr_stream *cstream = NULL; |
| 613 | struct msm_transcode_loopback *prtd; |
| 614 | int ret = 0; |
| 615 | int param_length = 0; |
| 616 | |
| 617 | if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| 618 | pr_err("%s Received invalid fe_id %lu\n", |
| 619 | __func__, fe_id); |
| 620 | ret = -EINVAL; |
| 621 | goto done; |
| 622 | } |
| 623 | |
| 624 | cstream = pdata->cstream[fe_id]; |
| 625 | if (cstream == NULL) { |
| 626 | pr_err("%s cstream is null\n", __func__); |
| 627 | ret = -EINVAL; |
| 628 | goto done; |
| 629 | } |
| 630 | |
| 631 | prtd = cstream->runtime->private_data; |
| 632 | if (!prtd) { |
| 633 | pr_err("%s: prtd is null\n", __func__); |
| 634 | ret = -EINVAL; |
| 635 | goto done; |
| 636 | } |
| 637 | |
| 638 | if (prtd->audio_client == NULL) { |
| 639 | pr_err("%s: audio_client is null\n", __func__); |
| 640 | ret = -EINVAL; |
| 641 | goto done; |
| 642 | } |
| 643 | |
| 644 | memcpy(¶m_length, ucontrol->value.bytes.data, |
| 645 | sizeof(param_length)); |
| 646 | if ((param_length + sizeof(param_length)) |
| 647 | >= sizeof(ucontrol->value.bytes.data)) { |
| 648 | pr_err("%s param length=%d exceeds limit", |
| 649 | __func__, param_length); |
| 650 | ret = -EINVAL; |
| 651 | goto done; |
| 652 | } |
| 653 | |
| 654 | ret = q6asm_send_rtic_event_ack(prtd->audio_client, |
| 655 | ucontrol->value.bytes.data + sizeof(param_length), |
| 656 | param_length); |
| 657 | if (ret < 0) |
| 658 | pr_err("%s: failed to send rtic event ack, err = %d\n", |
| 659 | __func__, ret); |
| 660 | done: |
| 661 | return ret; |
| 662 | } |
| 663 | |
| 664 | static int msm_transcode_stream_cmd_control( |
| 665 | struct snd_soc_pcm_runtime *rtd) |
| 666 | { |
| 667 | const char *mixer_ctl_name = DSP_STREAM_CMD; |
| 668 | const char *deviceNo = "NN"; |
| 669 | char *mixer_str = NULL; |
| 670 | int ctl_len = 0, ret = 0; |
| 671 | struct snd_kcontrol_new fe_loopback_stream_cmd_config_control[1] = { |
| 672 | { |
| 673 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| 674 | .name = "?", |
| 675 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| 676 | .info = msm_adsp_stream_cmd_info, |
| 677 | .put = msm_transcode_stream_cmd_put, |
| 678 | .private_value = 0, |
| 679 | } |
| 680 | }; |
| 681 | |
| 682 | if (!rtd) { |
| 683 | pr_err("%s NULL rtd\n", __func__); |
| 684 | ret = -EINVAL; |
| 685 | goto done; |
| 686 | } |
| 687 | |
| 688 | ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1; |
| 689 | mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| 690 | if (!mixer_str) { |
| 691 | ret = -ENOMEM; |
| 692 | goto done; |
| 693 | } |
| 694 | |
| 695 | snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device); |
| 696 | fe_loopback_stream_cmd_config_control[0].name = mixer_str; |
| 697 | fe_loopback_stream_cmd_config_control[0].private_value = |
| 698 | rtd->dai_link->id; |
| 699 | pr_debug("%s: Registering new mixer ctl %s\n", __func__, mixer_str); |
| 700 | ret = snd_soc_add_platform_controls(rtd->platform, |
| 701 | fe_loopback_stream_cmd_config_control, |
| 702 | ARRAY_SIZE(fe_loopback_stream_cmd_config_control)); |
| 703 | if (ret < 0) |
| 704 | pr_err("%s: failed to add ctl %s. err = %d\n", |
| 705 | __func__, mixer_str, ret); |
| 706 | |
| 707 | kfree(mixer_str); |
| 708 | done: |
| 709 | return ret; |
| 710 | } |
| 711 | |
| 712 | static int msm_transcode_stream_callback_control( |
| 713 | struct snd_soc_pcm_runtime *rtd) |
| 714 | { |
| 715 | const char *mixer_ctl_name = DSP_STREAM_CALLBACK; |
| 716 | const char *deviceNo = "NN"; |
| 717 | char *mixer_str = NULL; |
| 718 | int ctl_len = 0, ret = 0; |
| 719 | struct snd_kcontrol *kctl; |
| 720 | |
| 721 | struct snd_kcontrol_new fe_loopback_callback_config_control[1] = { |
| 722 | { |
| 723 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| 724 | .name = "?", |
| 725 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| 726 | .info = msm_adsp_stream_callback_info, |
| 727 | .get = msm_adsp_stream_callback_get, |
| 728 | .private_value = 0, |
| 729 | } |
| 730 | }; |
| 731 | |
| 732 | if (!rtd) { |
| 733 | pr_err("%s: rtd is NULL\n", __func__); |
| 734 | ret = -EINVAL; |
| 735 | goto done; |
| 736 | } |
| 737 | |
| 738 | ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1; |
| 739 | mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| 740 | if (!mixer_str) { |
| 741 | ret = -ENOMEM; |
| 742 | goto done; |
| 743 | } |
| 744 | |
| 745 | snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device); |
| 746 | fe_loopback_callback_config_control[0].name = mixer_str; |
| 747 | fe_loopback_callback_config_control[0].private_value = |
| 748 | rtd->dai_link->id; |
| 749 | pr_debug("%s: Registering new mixer ctl %s\n", __func__, mixer_str); |
| 750 | ret = snd_soc_add_platform_controls(rtd->platform, |
| 751 | fe_loopback_callback_config_control, |
| 752 | ARRAY_SIZE(fe_loopback_callback_config_control)); |
| 753 | if (ret < 0) { |
| 754 | pr_err("%s: failed to add ctl %s. err = %d\n", |
| 755 | __func__, mixer_str, ret); |
| 756 | ret = -EINVAL; |
| 757 | goto free_mixer_str; |
| 758 | } |
| 759 | |
| 760 | kctl = snd_soc_card_get_kcontrol(rtd->card, mixer_str); |
| 761 | if (!kctl) { |
| 762 | pr_err("%s: failed to get kctl %s.\n", __func__, mixer_str); |
| 763 | ret = -EINVAL; |
| 764 | goto free_mixer_str; |
| 765 | } |
| 766 | |
| 767 | kctl->private_data = NULL; |
| 768 | free_mixer_str: |
| 769 | kfree(mixer_str); |
| 770 | done: |
| 771 | return ret; |
| 772 | } |
| 773 | |
| 774 | static int msm_transcode_add_ion_fd_cmd_control(struct snd_soc_pcm_runtime *rtd) |
| 775 | { |
| 776 | const char *mixer_ctl_name = "Playback ION FD"; |
| 777 | const char *deviceNo = "NN"; |
| 778 | char *mixer_str = NULL; |
| 779 | int ctl_len = 0, ret = 0; |
| 780 | struct snd_kcontrol_new fe_ion_fd_config_control[1] = { |
| 781 | { |
| 782 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| 783 | .name = "?", |
| 784 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| 785 | .info = msm_adsp_stream_cmd_info, |
| 786 | .put = msm_transcode_ion_fd_map_put, |
| 787 | .private_value = 0, |
| 788 | } |
| 789 | }; |
| 790 | |
| 791 | if (!rtd) { |
| 792 | pr_err("%s NULL rtd\n", __func__); |
| 793 | ret = -EINVAL; |
| 794 | goto done; |
| 795 | } |
| 796 | |
| 797 | ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1; |
| 798 | mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| 799 | if (!mixer_str) { |
| 800 | ret = -ENOMEM; |
| 801 | goto done; |
| 802 | } |
| 803 | |
| 804 | snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device); |
| 805 | fe_ion_fd_config_control[0].name = mixer_str; |
| 806 | fe_ion_fd_config_control[0].private_value = rtd->dai_link->id; |
| 807 | pr_debug("%s: Registering new mixer ctl %s\n", __func__, mixer_str); |
| 808 | ret = snd_soc_add_platform_controls(rtd->platform, |
| 809 | fe_ion_fd_config_control, |
| 810 | ARRAY_SIZE(fe_ion_fd_config_control)); |
| 811 | if (ret < 0) |
| 812 | pr_err("%s: failed to add ctl %s\n", __func__, mixer_str); |
| 813 | |
| 814 | kfree(mixer_str); |
| 815 | done: |
| 816 | return ret; |
| 817 | } |
| 818 | |
| 819 | static int msm_transcode_add_event_ack_cmd_control( |
| 820 | struct snd_soc_pcm_runtime *rtd) |
| 821 | { |
| 822 | const char *mixer_ctl_name = "Playback Event Ack"; |
| 823 | const char *deviceNo = "NN"; |
| 824 | char *mixer_str = NULL; |
| 825 | int ctl_len = 0, ret = 0; |
| 826 | struct snd_kcontrol_new fe_event_ack_config_control[1] = { |
| 827 | { |
| 828 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| 829 | .name = "?", |
| 830 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| 831 | .info = msm_adsp_stream_cmd_info, |
| 832 | .put = msm_transcode_rtic_event_ack_put, |
| 833 | .private_value = 0, |
| 834 | } |
| 835 | }; |
| 836 | |
| 837 | if (!rtd) { |
| 838 | pr_err("%s NULL rtd\n", __func__); |
| 839 | ret = -EINVAL; |
| 840 | goto done; |
| 841 | } |
| 842 | |
| 843 | ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1; |
| 844 | mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| 845 | if (!mixer_str) { |
| 846 | ret = -ENOMEM; |
| 847 | goto done; |
| 848 | } |
| 849 | |
| 850 | snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device); |
| 851 | fe_event_ack_config_control[0].name = mixer_str; |
| 852 | fe_event_ack_config_control[0].private_value = rtd->dai_link->id; |
| 853 | pr_debug("%s: Registering new mixer ctl %s\n", __func__, mixer_str); |
| 854 | ret = snd_soc_add_platform_controls(rtd->platform, |
| 855 | fe_event_ack_config_control, |
| 856 | ARRAY_SIZE(fe_event_ack_config_control)); |
| 857 | if (ret < 0) |
| 858 | pr_err("%s: failed to add ctl %s\n", __func__, mixer_str); |
| 859 | |
| 860 | kfree(mixer_str); |
| 861 | done: |
| 862 | return ret; |
| 863 | } |
| 864 | |
| 865 | static int msm_transcode_loopback_new(struct snd_soc_pcm_runtime *rtd) |
| 866 | { |
| 867 | int rc; |
| 868 | |
| 869 | rc = msm_transcode_stream_cmd_control(rtd); |
| 870 | if (rc) |
| 871 | pr_err("%s: ADSP Stream Cmd Control open failed\n", __func__); |
| 872 | |
| 873 | rc = msm_transcode_stream_callback_control(rtd); |
| 874 | if (rc) |
| 875 | pr_err("%s: ADSP Stream callback Control open failed\n", |
| 876 | __func__); |
| 877 | |
| 878 | rc = msm_transcode_add_ion_fd_cmd_control(rtd); |
| 879 | if (rc) |
| 880 | pr_err("%s: Could not add transcode ion fd Control\n", |
| 881 | __func__); |
| 882 | |
| 883 | rc = msm_transcode_add_event_ack_cmd_control(rtd); |
| 884 | if (rc) |
| 885 | pr_err("%s: Could not add transcode event ack Control\n", |
| 886 | __func__); |
| 887 | |
| 888 | return 0; |
| 889 | } |
| 890 | |
| 891 | static struct snd_compr_ops msm_transcode_loopback_ops = { |
| 892 | .open = msm_transcode_loopback_open, |
| 893 | .free = msm_transcode_loopback_free, |
| 894 | .trigger = msm_transcode_loopback_trigger, |
| 895 | .set_params = msm_transcode_loopback_set_params, |
| 896 | .get_caps = msm_transcode_loopback_get_caps, |
| 897 | }; |
| 898 | |
| 899 | |
| 900 | static int msm_transcode_loopback_probe(struct snd_soc_platform *platform) |
| 901 | { |
| 902 | struct trans_loopback_pdata *pdata = NULL; |
| 903 | |
| 904 | pr_debug("%s\n", __func__); |
| 905 | pdata = (struct trans_loopback_pdata *) |
| 906 | kzalloc(sizeof(struct trans_loopback_pdata), |
| 907 | GFP_KERNEL); |
| 908 | if (!pdata) |
| 909 | return -ENOMEM; |
| 910 | |
| 911 | snd_soc_platform_set_drvdata(platform, pdata); |
| 912 | return 0; |
| 913 | } |
| 914 | |
| 915 | static struct snd_soc_platform_driver msm_soc_platform = { |
| 916 | .probe = msm_transcode_loopback_probe, |
| 917 | .compr_ops = &msm_transcode_loopback_ops, |
| 918 | .pcm_new = msm_transcode_loopback_new, |
| 919 | }; |
| 920 | |
| 921 | static int msm_transcode_dev_probe(struct platform_device *pdev) |
| 922 | { |
| 923 | |
| 924 | pr_debug("%s: dev name %s\n", __func__, dev_name(&pdev->dev)); |
| 925 | if (pdev->dev.of_node) |
| 926 | dev_set_name(&pdev->dev, "%s", "msm-transcode-loopback"); |
| 927 | |
| 928 | return snd_soc_register_platform(&pdev->dev, |
| 929 | &msm_soc_platform); |
| 930 | } |
| 931 | |
| 932 | static int msm_transcode_remove(struct platform_device *pdev) |
| 933 | { |
| 934 | snd_soc_unregister_platform(&pdev->dev); |
| 935 | return 0; |
| 936 | } |
| 937 | |
| 938 | static const struct of_device_id msm_transcode_loopback_dt_match[] = { |
| 939 | {.compatible = "qcom,msm-transcode-loopback"}, |
| 940 | {} |
| 941 | }; |
| 942 | MODULE_DEVICE_TABLE(of, msm_transcode_loopback_dt_match); |
| 943 | |
| 944 | static struct platform_driver msm_transcode_loopback_driver = { |
| 945 | .driver = { |
| 946 | .name = "msm-transcode-loopback", |
| 947 | .owner = THIS_MODULE, |
| 948 | .of_match_table = msm_transcode_loopback_dt_match, |
| 949 | }, |
| 950 | .probe = msm_transcode_dev_probe, |
| 951 | .remove = msm_transcode_remove, |
| 952 | }; |
| 953 | |
Laxminath Kasam | 8b1366a | 2017-10-05 01:44:16 +0530 | [diff] [blame] | 954 | int __init msm_transcode_loopback_init(void) |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 955 | { |
| 956 | memset(&transcode_info, 0, sizeof(struct msm_transcode_loopback)); |
| 957 | mutex_init(&transcode_info.lock); |
| 958 | return platform_driver_register(&msm_transcode_loopback_driver); |
| 959 | } |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 960 | |
Laxminath Kasam | 8b1366a | 2017-10-05 01:44:16 +0530 | [diff] [blame] | 961 | void __exit msm_transcode_loopback_exit(void) |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 962 | { |
| 963 | mutex_destroy(&transcode_info.lock); |
| 964 | platform_driver_unregister(&msm_transcode_loopback_driver); |
| 965 | } |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 966 | |
| 967 | MODULE_DESCRIPTION("Transcode loopback platform driver"); |
| 968 | MODULE_LICENSE("GPL v2"); |