Sachin Mohan Gadag | 265d94d | 2018-01-04 11:04:00 +0530 | [diff] [blame] | 1 | /* Copyright (c) 2012-2018, The Linux Foundation. All rights reserved. |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 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 | |
| 14 | #include <linux/init.h> |
| 15 | #include <linux/err.h> |
| 16 | #include <linux/module.h> |
| 17 | #include <linux/moduleparam.h> |
| 18 | #include <linux/time.h> |
| 19 | #include <linux/math64.h> |
| 20 | #include <linux/wait.h> |
| 21 | #include <linux/platform_device.h> |
| 22 | #include <linux/slab.h> |
| 23 | #include <sound/core.h> |
| 24 | #include <sound/soc.h> |
| 25 | #include <sound/soc-dapm.h> |
| 26 | #include <sound/pcm.h> |
| 27 | #include <sound/initval.h> |
| 28 | #include <sound/control.h> |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 29 | #include <sound/pcm_params.h> |
| 30 | #include <sound/audio_effects.h> |
| 31 | #include <asm/dma.h> |
| 32 | #include <linux/dma-mapping.h> |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 33 | #include <linux/msm_audio.h> |
| 34 | |
| 35 | #include <sound/timer.h> |
| 36 | #include <sound/tlv.h> |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 37 | #include <sound/compress_params.h> |
| 38 | #include <sound/compress_offload.h> |
| 39 | #include <sound/compress_driver.h> |
Laxminath Kasam | 605b42f | 2017-08-01 22:02:15 +0530 | [diff] [blame] | 40 | |
| 41 | #include <dsp/msm_audio_ion.h> |
| 42 | #include <dsp/apr_audio-v2.h> |
| 43 | #include <dsp/q6asm-v2.h> |
Dieter Luecking | ba7644d | 2018-09-28 15:09:32 +0200 | [diff] [blame] | 44 | #include <dsp/q6core.h> |
Laxminath Kasam | 605b42f | 2017-08-01 22:02:15 +0530 | [diff] [blame] | 45 | #include <dsp/msm-audio-effects-q6-v2.h> |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 46 | #include "msm-pcm-routing-v2.h" |
| 47 | #include "msm-qti-pp-config.h" |
| 48 | |
| 49 | #define DSP_PP_BUFFERING_IN_MSEC 25 |
| 50 | #define PARTIAL_DRAIN_ACK_EARLY_BY_MSEC 150 |
| 51 | #define MP3_OUTPUT_FRAME_SZ 1152 |
| 52 | #define AAC_OUTPUT_FRAME_SZ 1024 |
| 53 | #define AC3_OUTPUT_FRAME_SZ 1536 |
| 54 | #define EAC3_OUTPUT_FRAME_SZ 1536 |
| 55 | #define DSP_NUM_OUTPUT_FRAME_BUFFERED 2 |
| 56 | #define FLAC_BLK_SIZE_LIMIT 65535 |
| 57 | |
| 58 | /* Timestamp mode payload offsets */ |
| 59 | #define CAPTURE_META_DATA_TS_OFFSET_LSW 6 |
| 60 | #define CAPTURE_META_DATA_TS_OFFSET_MSW 7 |
| 61 | |
| 62 | /* decoder parameter length */ |
| 63 | #define DDP_DEC_MAX_NUM_PARAM 18 |
| 64 | |
| 65 | /* Default values used if user space does not set */ |
| 66 | #define COMPR_PLAYBACK_MIN_FRAGMENT_SIZE (8 * 1024) |
| 67 | #define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024) |
| 68 | #define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4) |
| 69 | #define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4) |
| 70 | |
| 71 | #define COMPRESSED_LR_VOL_MAX_STEPS 0x2000 |
| 72 | const DECLARE_TLV_DB_LINEAR(msm_compr_vol_gain, 0, |
| 73 | COMPRESSED_LR_VOL_MAX_STEPS); |
| 74 | |
| 75 | /* Stream id switches between 1 and 2 */ |
| 76 | #define NEXT_STREAM_ID(stream_id) ((stream_id & 1) + 1) |
| 77 | |
| 78 | #define STREAM_ARRAY_INDEX(stream_id) (stream_id - 1) |
| 79 | |
| 80 | #define MAX_NUMBER_OF_STREAMS 2 |
| 81 | |
| 82 | struct msm_compr_gapless_state { |
| 83 | bool set_next_stream_id; |
| 84 | int32_t stream_opened[MAX_NUMBER_OF_STREAMS]; |
| 85 | uint32_t initial_samples_drop; |
| 86 | uint32_t trailing_samples_drop; |
| 87 | uint32_t gapless_transition; |
| 88 | bool use_dsp_gapless_mode; |
| 89 | union snd_codec_options codec_options; |
| 90 | }; |
| 91 | |
| 92 | static unsigned int supported_sample_rates[] = { |
| 93 | 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, |
| 94 | 88200, 96000, 128000, 144000, 176400, 192000, 352800, 384000, 2822400, |
| 95 | 5644800 |
| 96 | }; |
| 97 | |
| 98 | struct msm_compr_pdata { |
| 99 | struct snd_compr_stream *cstream[MSM_FRONTEND_DAI_MAX]; |
| 100 | uint32_t volume[MSM_FRONTEND_DAI_MAX][2]; /* For both L & R */ |
| 101 | struct msm_compr_audio_effects *audio_effects[MSM_FRONTEND_DAI_MAX]; |
| 102 | bool use_dsp_gapless_mode; |
| 103 | bool use_legacy_api; /* indicates use older asm apis*/ |
| 104 | struct msm_compr_dec_params *dec_params[MSM_FRONTEND_DAI_MAX]; |
| 105 | struct msm_compr_ch_map *ch_map[MSM_FRONTEND_DAI_MAX]; |
Aditya Bavanari | 9deef91 | 2017-11-20 13:31:31 +0530 | [diff] [blame] | 106 | bool is_in_use[MSM_FRONTEND_DAI_MAX]; |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 107 | }; |
| 108 | |
| 109 | struct msm_compr_audio { |
| 110 | struct snd_compr_stream *cstream; |
| 111 | struct snd_compr_caps compr_cap; |
| 112 | struct snd_compr_codec_caps codec_caps; |
| 113 | struct snd_compr_params codec_param; |
| 114 | struct audio_client *audio_client; |
| 115 | |
| 116 | uint32_t codec; |
| 117 | uint32_t compr_passthr; |
| 118 | void *buffer; /* virtual address */ |
| 119 | phys_addr_t buffer_paddr; /* physical address */ |
| 120 | uint32_t app_pointer; |
| 121 | uint32_t buffer_size; |
| 122 | uint32_t byte_offset; |
| 123 | uint64_t copied_total; /* bytes consumed by DSP */ |
| 124 | uint64_t bytes_received; /* from userspace */ |
| 125 | uint64_t bytes_sent; /* to DSP */ |
| 126 | |
| 127 | uint64_t received_total; /* bytes received from DSP */ |
| 128 | uint64_t bytes_copied; /* to userspace */ |
| 129 | uint64_t bytes_read; /* from DSP */ |
| 130 | uint32_t bytes_read_offset; /* bytes read offset */ |
| 131 | |
| 132 | uint32_t ts_header_offset; /* holds the timestamp header offset */ |
| 133 | |
| 134 | int32_t first_buffer; |
| 135 | int32_t last_buffer; |
| 136 | int32_t partial_drain_delay; |
| 137 | |
| 138 | uint16_t session_id; |
| 139 | |
| 140 | uint32_t sample_rate; |
| 141 | uint32_t num_channels; |
| 142 | |
| 143 | /* |
| 144 | * convention - commands coming from the same thread |
| 145 | * can use the common cmd_ack var. Others (e.g drain/EOS) |
| 146 | * must use separate vars to track command status. |
| 147 | */ |
| 148 | uint32_t cmd_ack; |
| 149 | uint32_t cmd_interrupt; |
| 150 | uint32_t drain_ready; |
| 151 | uint32_t eos_ack; |
| 152 | |
| 153 | uint32_t stream_available; |
| 154 | uint32_t next_stream; |
| 155 | |
| 156 | uint32_t run_mode; |
| 157 | uint32_t start_delay_lsw; |
| 158 | uint32_t start_delay_msw; |
| 159 | |
| 160 | uint64_t marker_timestamp; |
| 161 | |
| 162 | struct msm_compr_gapless_state gapless_state; |
| 163 | |
| 164 | atomic_t start; |
| 165 | atomic_t eos; |
| 166 | atomic_t drain; |
| 167 | atomic_t xrun; |
| 168 | atomic_t close; |
| 169 | atomic_t wait_on_close; |
| 170 | atomic_t error; |
| 171 | |
| 172 | wait_queue_head_t eos_wait; |
| 173 | wait_queue_head_t drain_wait; |
| 174 | wait_queue_head_t close_wait; |
| 175 | wait_queue_head_t wait_for_stream_avail; |
| 176 | |
| 177 | spinlock_t lock; |
| 178 | }; |
| 179 | |
| 180 | const u32 compr_codecs[] = { |
| 181 | SND_AUDIOCODEC_AC3, SND_AUDIOCODEC_EAC3, SND_AUDIOCODEC_DTS, |
| 182 | SND_AUDIOCODEC_DSD, SND_AUDIOCODEC_TRUEHD, SND_AUDIOCODEC_IEC61937}; |
| 183 | |
| 184 | struct query_audio_effect { |
| 185 | uint32_t mod_id; |
| 186 | uint32_t parm_id; |
| 187 | uint32_t size; |
| 188 | uint32_t offset; |
| 189 | uint32_t device; |
| 190 | }; |
| 191 | |
| 192 | struct msm_compr_audio_effects { |
| 193 | struct bass_boost_params bass_boost; |
| 194 | struct pbe_params pbe; |
| 195 | struct virtualizer_params virtualizer; |
| 196 | struct reverb_params reverb; |
| 197 | struct eq_params equalizer; |
| 198 | struct soft_volume_params volume; |
| 199 | struct query_audio_effect query; |
| 200 | }; |
| 201 | |
| 202 | struct msm_compr_dec_params { |
| 203 | struct snd_dec_ddp ddp_params; |
| 204 | }; |
| 205 | |
| 206 | struct msm_compr_ch_map { |
| 207 | bool set_ch_map; |
Dieter Luecking | ba7644d | 2018-09-28 15:09:32 +0200 | [diff] [blame] | 208 | char channel_map[PCM_FORMAT_MAX_NUM_CHANNEL_V8]; |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 209 | }; |
| 210 | |
| 211 | static int msm_compr_send_dec_params(struct snd_compr_stream *cstream, |
| 212 | struct msm_compr_dec_params *dec_params, |
| 213 | int stream_id); |
| 214 | |
| 215 | static int msm_compr_set_render_mode(struct msm_compr_audio *prtd, |
| 216 | uint32_t render_mode) { |
| 217 | int ret = -EINVAL; |
| 218 | struct audio_client *ac = prtd->audio_client; |
| 219 | |
| 220 | pr_debug("%s, got render mode %u\n", __func__, render_mode); |
| 221 | |
| 222 | if (render_mode == SNDRV_COMPRESS_RENDER_MODE_AUDIO_MASTER) { |
| 223 | render_mode = ASM_SESSION_MTMX_STRTR_PARAM_RENDER_DEFAULT; |
| 224 | } else if (render_mode == SNDRV_COMPRESS_RENDER_MODE_STC_MASTER) { |
| 225 | render_mode = ASM_SESSION_MTMX_STRTR_PARAM_RENDER_LOCAL_STC; |
| 226 | prtd->run_mode = ASM_SESSION_CMD_RUN_STARTIME_RUN_WITH_DELAY; |
| 227 | } else { |
| 228 | pr_err("%s, Invalid render mode %u\n", __func__, |
| 229 | render_mode); |
| 230 | ret = -EINVAL; |
| 231 | goto exit; |
| 232 | } |
| 233 | |
| 234 | ret = q6asm_send_mtmx_strtr_render_mode(ac, render_mode); |
| 235 | if (ret) { |
| 236 | pr_err("%s, Render mode can't be set error %d\n", __func__, |
| 237 | ret); |
| 238 | } |
| 239 | exit: |
| 240 | return ret; |
| 241 | } |
| 242 | |
| 243 | static int msm_compr_set_clk_rec_mode(struct audio_client *ac, |
| 244 | uint32_t clk_rec_mode) { |
| 245 | int ret = -EINVAL; |
| 246 | |
| 247 | pr_debug("%s, got clk rec mode %u\n", __func__, clk_rec_mode); |
| 248 | |
| 249 | if (clk_rec_mode == SNDRV_COMPRESS_CLK_REC_MODE_NONE) { |
| 250 | clk_rec_mode = ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_NONE; |
| 251 | } else if (clk_rec_mode == SNDRV_COMPRESS_CLK_REC_MODE_AUTO) { |
| 252 | clk_rec_mode = ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_AUTO; |
| 253 | } else { |
| 254 | pr_err("%s, Invalid clk rec_mode mode %u\n", __func__, |
| 255 | clk_rec_mode); |
| 256 | ret = -EINVAL; |
| 257 | goto exit; |
| 258 | } |
| 259 | |
| 260 | ret = q6asm_send_mtmx_strtr_clk_rec_mode(ac, clk_rec_mode); |
| 261 | if (ret) { |
| 262 | pr_err("%s, clk rec mode can't be set, error %d\n", __func__, |
| 263 | ret); |
| 264 | } |
| 265 | |
| 266 | exit: |
| 267 | return ret; |
| 268 | } |
| 269 | |
| 270 | static int msm_compr_set_render_window(struct audio_client *ac, |
| 271 | uint32_t ws_lsw, uint32_t ws_msw, |
| 272 | uint32_t we_lsw, uint32_t we_msw) |
| 273 | { |
| 274 | int ret = -EINVAL; |
| 275 | struct asm_session_mtmx_strtr_param_window_v2_t asm_mtmx_strtr_window; |
| 276 | uint32_t param_id; |
| 277 | |
| 278 | pr_debug("%s, ws_lsw 0x%x ws_msw 0x%x we_lsw 0x%x we_ms 0x%x\n", |
| 279 | __func__, ws_lsw, ws_msw, we_lsw, we_msw); |
| 280 | |
| 281 | memset(&asm_mtmx_strtr_window, 0, |
| 282 | sizeof(struct asm_session_mtmx_strtr_param_window_v2_t)); |
| 283 | asm_mtmx_strtr_window.window_lsw = ws_lsw; |
| 284 | asm_mtmx_strtr_window.window_msw = ws_msw; |
| 285 | param_id = ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_START_V2; |
| 286 | ret = q6asm_send_mtmx_strtr_window(ac, &asm_mtmx_strtr_window, |
| 287 | param_id); |
| 288 | if (ret) { |
| 289 | pr_err("%s, start window can't be set error %d\n", __func__, |
| 290 | ret); |
| 291 | goto exit; |
| 292 | } |
| 293 | |
| 294 | asm_mtmx_strtr_window.window_lsw = we_lsw; |
| 295 | asm_mtmx_strtr_window.window_msw = we_msw; |
| 296 | param_id = ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_END_V2; |
| 297 | ret = q6asm_send_mtmx_strtr_window(ac, &asm_mtmx_strtr_window, |
| 298 | param_id); |
| 299 | if (ret) { |
| 300 | pr_err("%s, end window can't be set error %d\n", __func__, |
| 301 | ret); |
| 302 | } |
| 303 | |
| 304 | exit: |
| 305 | return ret; |
| 306 | } |
| 307 | |
| 308 | static int msm_compr_enable_adjust_session_clock(struct audio_client *ac, |
| 309 | bool enable) |
| 310 | { |
| 311 | int ret; |
| 312 | |
| 313 | pr_debug("%s, enable adjust_session %d\n", __func__, enable); |
| 314 | |
| 315 | ret = q6asm_send_mtmx_strtr_enable_adjust_session_clock(ac, enable); |
| 316 | if (ret) |
| 317 | pr_err("%s, adjust session clock can't be set error %d\n", |
| 318 | __func__, ret); |
| 319 | |
| 320 | return ret; |
| 321 | } |
| 322 | |
| 323 | static int msm_compr_adjust_session_clock(struct audio_client *ac, |
| 324 | uint32_t adjust_session_lsw, uint32_t adjust_session_msw) |
| 325 | { |
| 326 | int ret; |
| 327 | |
| 328 | pr_debug("%s, adjust_session_time_msw 0x%x adjust_session_time_lsw 0x%x\n", |
| 329 | __func__, adjust_session_msw, adjust_session_lsw); |
| 330 | |
| 331 | ret = q6asm_adjust_session_clock(ac, |
| 332 | adjust_session_lsw, |
| 333 | adjust_session_msw); |
| 334 | if (ret) |
| 335 | pr_err("%s, adjust session clock can't be set error %d\n", |
| 336 | __func__, ret); |
| 337 | |
| 338 | return ret; |
| 339 | } |
| 340 | |
| 341 | static int msm_compr_set_volume(struct snd_compr_stream *cstream, |
| 342 | uint32_t volume_l, uint32_t volume_r) |
| 343 | { |
| 344 | struct msm_compr_audio *prtd; |
| 345 | int rc = 0; |
| 346 | uint32_t avg_vol, gain_list[VOLUME_CONTROL_MAX_CHANNELS]; |
| 347 | uint32_t num_channels; |
| 348 | struct snd_soc_pcm_runtime *rtd; |
| 349 | struct msm_compr_pdata *pdata; |
| 350 | bool use_default = true; |
| 351 | u8 *chmap = NULL; |
| 352 | |
| 353 | pr_debug("%s: volume_l %d volume_r %d\n", |
| 354 | __func__, volume_l, volume_r); |
| 355 | if (!cstream || !cstream->runtime) { |
| 356 | pr_err("%s: session not active\n", __func__); |
| 357 | return -EPERM; |
| 358 | } |
| 359 | rtd = cstream->private_data; |
| 360 | prtd = cstream->runtime->private_data; |
| 361 | |
| 362 | if (!rtd || !rtd->platform || !prtd || !prtd->audio_client) { |
| 363 | pr_err("%s: invalid rtd, prtd or audio client", __func__); |
| 364 | return rc; |
| 365 | } |
| 366 | pdata = snd_soc_platform_get_drvdata(rtd->platform); |
| 367 | |
| 368 | if (prtd->compr_passthr != LEGACY_PCM) { |
| 369 | pr_debug("%s: No volume config for passthrough %d\n", |
| 370 | __func__, prtd->compr_passthr); |
| 371 | return rc; |
| 372 | } |
| 373 | |
| 374 | use_default = !(pdata->ch_map[rtd->dai_link->id]->set_ch_map); |
| 375 | chmap = pdata->ch_map[rtd->dai_link->id]->channel_map; |
| 376 | num_channels = prtd->num_channels; |
| 377 | |
| 378 | if (prtd->num_channels > 2) { |
| 379 | /* |
| 380 | * Currently the left and right gains are averaged an applied |
| 381 | * to all channels. This might not be desirable. But currently, |
| 382 | * there exists no API in userspace to send a list of gains for |
| 383 | * each channel either. If such an API does become available, |
| 384 | * the mixer control must be updated to accept more than 2 |
| 385 | * channel gains. |
| 386 | * |
| 387 | */ |
| 388 | avg_vol = (volume_l + volume_r) / 2; |
| 389 | rc = q6asm_set_volume(prtd->audio_client, avg_vol); |
| 390 | } else { |
| 391 | gain_list[0] = volume_l; |
| 392 | gain_list[1] = volume_r; |
yidongh | 98526ef | 2017-09-05 17:57:55 +0800 | [diff] [blame] | 393 | gain_list[2] = volume_l; |
| 394 | num_channels = 3; |
| 395 | use_default = true; |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 396 | rc = q6asm_set_multich_gain(prtd->audio_client, num_channels, |
| 397 | gain_list, chmap, use_default); |
| 398 | } |
| 399 | |
| 400 | if (rc < 0) |
| 401 | pr_err("%s: Send vol gain command failed rc=%d\n", |
| 402 | __func__, rc); |
| 403 | |
| 404 | return rc; |
| 405 | } |
| 406 | |
| 407 | static int msm_compr_send_ddp_cfg(struct audio_client *ac, |
| 408 | struct snd_dec_ddp *ddp, |
| 409 | int stream_id) |
| 410 | { |
| 411 | int i, rc; |
| 412 | |
| 413 | pr_debug("%s\n", __func__); |
| 414 | for (i = 0; i < ddp->params_length; i++) { |
| 415 | rc = q6asm_ds1_set_stream_endp_params(ac, ddp->params_id[i], |
| 416 | ddp->params_value[i], |
| 417 | stream_id); |
| 418 | if (rc) { |
| 419 | pr_err("sending params_id: %d failed\n", |
| 420 | ddp->params_id[i]); |
| 421 | return rc; |
| 422 | } |
| 423 | } |
| 424 | return 0; |
| 425 | } |
| 426 | |
| 427 | static int msm_compr_send_buffer(struct msm_compr_audio *prtd) |
| 428 | { |
| 429 | int buffer_length; |
| 430 | uint64_t bytes_available; |
| 431 | struct audio_aio_write_param param; |
| 432 | struct snd_codec_metadata *buff_addr; |
| 433 | |
| 434 | if (!atomic_read(&prtd->start)) { |
| 435 | pr_err("%s: stream is not in started state\n", __func__); |
| 436 | return -EINVAL; |
| 437 | } |
| 438 | |
| 439 | |
| 440 | if (atomic_read(&prtd->xrun)) { |
| 441 | WARN(1, "%s called while xrun is true", __func__); |
| 442 | return -EPERM; |
| 443 | } |
| 444 | |
| 445 | pr_debug("%s: bytes_received = %llu copied_total = %llu\n", |
| 446 | __func__, prtd->bytes_received, prtd->copied_total); |
| 447 | if (prtd->first_buffer && prtd->gapless_state.use_dsp_gapless_mode && |
| 448 | prtd->compr_passthr == LEGACY_PCM) |
| 449 | q6asm_stream_send_meta_data(prtd->audio_client, |
| 450 | prtd->audio_client->stream_id, |
| 451 | prtd->gapless_state.initial_samples_drop, |
| 452 | prtd->gapless_state.trailing_samples_drop); |
| 453 | |
| 454 | buffer_length = prtd->codec_param.buffer.fragment_size; |
| 455 | bytes_available = prtd->bytes_received - prtd->copied_total; |
| 456 | if (bytes_available < prtd->codec_param.buffer.fragment_size) |
| 457 | buffer_length = bytes_available; |
| 458 | |
| 459 | if (prtd->byte_offset + buffer_length > prtd->buffer_size) { |
| 460 | buffer_length = (prtd->buffer_size - prtd->byte_offset); |
| 461 | pr_debug("%s: wrap around situation, send partial data %d now", |
| 462 | __func__, buffer_length); |
| 463 | } |
| 464 | |
| 465 | if (buffer_length) { |
| 466 | param.paddr = prtd->buffer_paddr + prtd->byte_offset; |
| 467 | WARN(prtd->byte_offset % 32 != 0, "offset %x not multiple of 32\n", |
| 468 | prtd->byte_offset); |
| 469 | } else { |
| 470 | param.paddr = prtd->buffer_paddr; |
| 471 | } |
| 472 | param.len = buffer_length; |
| 473 | if (prtd->ts_header_offset) { |
| 474 | buff_addr = (struct snd_codec_metadata *) |
| 475 | (prtd->buffer + prtd->byte_offset); |
| 476 | param.len = buff_addr->length; |
| 477 | param.msw_ts = (uint32_t) |
| 478 | ((buff_addr->timestamp & 0xFFFFFFFF00000000LL) >> 32); |
| 479 | param.lsw_ts = (uint32_t) (buff_addr->timestamp & 0xFFFFFFFFLL); |
| 480 | param.paddr += prtd->ts_header_offset; |
| 481 | param.flags = SET_TIMESTAMP; |
| 482 | param.metadata_len = prtd->ts_header_offset; |
| 483 | } else { |
| 484 | param.msw_ts = 0; |
| 485 | param.lsw_ts = 0; |
| 486 | param.flags = NO_TIMESTAMP; |
| 487 | param.metadata_len = 0; |
| 488 | } |
| 489 | param.uid = buffer_length; |
| 490 | param.last_buffer = prtd->last_buffer; |
| 491 | |
| 492 | pr_debug("%s: sending %d bytes to DSP byte_offset = %d\n", |
| 493 | __func__, param.len, prtd->byte_offset); |
| 494 | if (q6asm_async_write(prtd->audio_client, ¶m) < 0) { |
| 495 | pr_err("%s:q6asm_async_write failed\n", __func__); |
| 496 | } else { |
| 497 | prtd->bytes_sent += buffer_length; |
| 498 | if (prtd->first_buffer) |
| 499 | prtd->first_buffer = 0; |
| 500 | } |
| 501 | |
| 502 | return 0; |
| 503 | } |
| 504 | |
| 505 | static int msm_compr_read_buffer(struct msm_compr_audio *prtd) |
| 506 | { |
| 507 | int buffer_length; |
| 508 | uint64_t bytes_available; |
| 509 | uint64_t buffer_sent; |
| 510 | struct audio_aio_read_param param; |
| 511 | int ret; |
| 512 | |
| 513 | if (!atomic_read(&prtd->start)) { |
| 514 | pr_err("%s: stream is not in started state\n", __func__); |
| 515 | return -EINVAL; |
| 516 | } |
| 517 | |
| 518 | buffer_length = prtd->codec_param.buffer.fragment_size - |
| 519 | prtd->ts_header_offset; |
| 520 | bytes_available = prtd->received_total - prtd->bytes_copied; |
| 521 | buffer_sent = prtd->bytes_read - prtd->bytes_copied; |
| 522 | if (buffer_sent + buffer_length + prtd->ts_header_offset |
| 523 | > prtd->buffer_size) { |
| 524 | pr_debug(" %s : Buffer is Full bytes_available: %llu\n", |
| 525 | __func__, bytes_available); |
| 526 | return 0; |
| 527 | } |
| 528 | |
| 529 | memset(¶m, 0x0, sizeof(struct audio_aio_read_param)); |
| 530 | param.paddr = prtd->buffer_paddr + prtd->bytes_read_offset + |
| 531 | prtd->ts_header_offset; |
| 532 | param.len = buffer_length; |
| 533 | param.uid = buffer_length; |
| 534 | param.flags = prtd->codec_param.codec.flags; |
| 535 | |
| 536 | pr_debug("%s: reading %d bytes from DSP byte_offset = %llu\n", |
| 537 | __func__, buffer_length, prtd->bytes_read); |
| 538 | ret = q6asm_async_read(prtd->audio_client, ¶m); |
| 539 | if (ret < 0) { |
| 540 | pr_err("%s: q6asm_async_read failed - %d\n", |
| 541 | __func__, ret); |
| 542 | return ret; |
| 543 | } |
Sachin Mohan Gadag | 3a52de7 | 2018-07-05 15:48:12 +0530 | [diff] [blame] | 544 | prtd->bytes_read += buffer_length + prtd->ts_header_offset; |
Vikram Panduranga | 40b0ec6 | 2018-01-10 18:04:54 -0800 | [diff] [blame] | 545 | prtd->bytes_read_offset += buffer_length + prtd->ts_header_offset; |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 546 | if (prtd->bytes_read_offset >= prtd->buffer_size) |
| 547 | prtd->bytes_read_offset -= prtd->buffer_size; |
| 548 | |
| 549 | return 0; |
| 550 | } |
| 551 | |
| 552 | static void compr_event_handler(uint32_t opcode, |
| 553 | uint32_t token, uint32_t *payload, void *priv) |
| 554 | { |
| 555 | struct msm_compr_audio *prtd = priv; |
| 556 | struct snd_compr_stream *cstream; |
| 557 | struct audio_client *ac; |
| 558 | uint32_t chan_mode = 0; |
| 559 | uint32_t sample_rate = 0; |
| 560 | uint64_t bytes_available; |
| 561 | int stream_id; |
| 562 | uint32_t stream_index; |
| 563 | unsigned long flags; |
| 564 | uint64_t read_size; |
| 565 | uint32_t *buff_addr; |
| 566 | struct snd_soc_pcm_runtime *rtd; |
| 567 | int ret = 0; |
| 568 | |
| 569 | if (!prtd) { |
| 570 | pr_err("%s: prtd is NULL\n", __func__); |
| 571 | return; |
| 572 | } |
| 573 | cstream = prtd->cstream; |
| 574 | if (!cstream) { |
| 575 | pr_err("%s: cstream is NULL\n", __func__); |
| 576 | return; |
| 577 | } |
| 578 | |
| 579 | ac = prtd->audio_client; |
| 580 | |
| 581 | /* |
| 582 | * Token for rest of the compressed commands use to set |
| 583 | * session id, stream id, dir etc. |
| 584 | */ |
| 585 | stream_id = q6asm_get_stream_id_from_token(token); |
| 586 | |
| 587 | pr_debug("%s opcode =%08x\n", __func__, opcode); |
| 588 | switch (opcode) { |
| 589 | case ASM_DATA_EVENT_WRITE_DONE_V2: |
| 590 | spin_lock_irqsave(&prtd->lock, flags); |
| 591 | |
| 592 | if (payload[3]) { |
| 593 | pr_err("%s: WRITE FAILED w/ err 0x%x !, paddr 0x%x, byte_offset=%d,copied_total=%llu,token=%d\n", |
| 594 | __func__, |
| 595 | payload[3], |
| 596 | payload[0], |
| 597 | prtd->byte_offset, |
| 598 | prtd->copied_total, token); |
| 599 | |
| 600 | if (atomic_cmpxchg(&prtd->drain, 1, 0) && |
| 601 | prtd->last_buffer) { |
| 602 | pr_debug("%s: wake up on drain\n", __func__); |
| 603 | prtd->drain_ready = 1; |
| 604 | wake_up(&prtd->drain_wait); |
| 605 | prtd->last_buffer = 0; |
| 606 | } else { |
| 607 | atomic_set(&prtd->start, 0); |
| 608 | } |
| 609 | } else { |
| 610 | pr_debug("ASM_DATA_EVENT_WRITE_DONE_V2 offset %d, length %d\n", |
| 611 | prtd->byte_offset, token); |
| 612 | } |
| 613 | |
| 614 | /* |
| 615 | * Token for WRITE command represents the amount of data |
| 616 | * written to ADSP in the last write, update offset and |
| 617 | * total copied data accordingly. |
| 618 | */ |
| 619 | if (prtd->ts_header_offset) { |
| 620 | /* Always assume that the data will be sent to DSP on |
| 621 | * frame boundary. |
| 622 | * i.e, one frame of userspace write will result in |
| 623 | * one kernel write to DSP. This is needed as |
| 624 | * timestamp will be sent per frame. |
| 625 | */ |
| 626 | prtd->byte_offset += |
| 627 | prtd->codec_param.buffer.fragment_size; |
| 628 | prtd->copied_total += |
| 629 | prtd->codec_param.buffer.fragment_size; |
| 630 | } else { |
| 631 | prtd->byte_offset += token; |
| 632 | prtd->copied_total += token; |
| 633 | } |
| 634 | if (prtd->byte_offset >= prtd->buffer_size) |
| 635 | prtd->byte_offset -= prtd->buffer_size; |
| 636 | |
| 637 | snd_compr_fragment_elapsed(cstream); |
| 638 | |
| 639 | if (!atomic_read(&prtd->start)) { |
| 640 | /* Writes must be restarted from _copy() */ |
| 641 | pr_debug("write_done received while not started, treat as xrun"); |
| 642 | atomic_set(&prtd->xrun, 1); |
| 643 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 644 | break; |
| 645 | } |
| 646 | |
| 647 | bytes_available = prtd->bytes_received - prtd->copied_total; |
| 648 | if (bytes_available < cstream->runtime->fragment_size) { |
| 649 | pr_debug("WRITE_DONE Insufficient data to send. break out\n"); |
| 650 | atomic_set(&prtd->xrun, 1); |
| 651 | |
| 652 | if (prtd->last_buffer) |
| 653 | prtd->last_buffer = 0; |
| 654 | if (atomic_read(&prtd->drain)) { |
| 655 | pr_debug("wake up on drain\n"); |
| 656 | prtd->drain_ready = 1; |
| 657 | wake_up(&prtd->drain_wait); |
| 658 | atomic_set(&prtd->drain, 0); |
| 659 | } |
| 660 | } else if ((bytes_available == cstream->runtime->fragment_size) |
| 661 | && atomic_read(&prtd->drain)) { |
| 662 | prtd->last_buffer = 1; |
| 663 | msm_compr_send_buffer(prtd); |
| 664 | prtd->last_buffer = 0; |
| 665 | } else |
| 666 | msm_compr_send_buffer(prtd); |
| 667 | |
| 668 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 669 | break; |
| 670 | |
| 671 | case ASM_DATA_EVENT_READ_DONE_V2: |
| 672 | spin_lock_irqsave(&prtd->lock, flags); |
| 673 | |
| 674 | pr_debug("ASM_DATA_EVENT_READ_DONE_V2 offset %d, length %d\n", |
| 675 | prtd->byte_offset, payload[4]); |
| 676 | |
| 677 | if (prtd->ts_header_offset) { |
| 678 | /* Update the header for received buffer */ |
| 679 | buff_addr = prtd->buffer + prtd->byte_offset; |
| 680 | /* Write the length of the buffer */ |
| 681 | *buff_addr = prtd->codec_param.buffer.fragment_size |
| 682 | - prtd->ts_header_offset; |
| 683 | buff_addr++; |
| 684 | /* Write the offset */ |
| 685 | *buff_addr = prtd->ts_header_offset; |
| 686 | buff_addr++; |
| 687 | /* Write the TS LSW */ |
| 688 | *buff_addr = payload[CAPTURE_META_DATA_TS_OFFSET_LSW]; |
| 689 | buff_addr++; |
| 690 | /* Write the TS MSW */ |
| 691 | *buff_addr = payload[CAPTURE_META_DATA_TS_OFFSET_MSW]; |
| 692 | } |
| 693 | /* Always assume read_size is same as fragment_size */ |
| 694 | read_size = prtd->codec_param.buffer.fragment_size; |
| 695 | prtd->byte_offset += read_size; |
| 696 | prtd->received_total += read_size; |
| 697 | if (prtd->byte_offset >= prtd->buffer_size) |
| 698 | prtd->byte_offset -= prtd->buffer_size; |
| 699 | |
| 700 | snd_compr_fragment_elapsed(cstream); |
| 701 | |
| 702 | if (!atomic_read(&prtd->start)) { |
| 703 | pr_debug("read_done received while not started, treat as xrun"); |
| 704 | atomic_set(&prtd->xrun, 1); |
| 705 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 706 | break; |
| 707 | } |
| 708 | msm_compr_read_buffer(prtd); |
| 709 | |
| 710 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 711 | break; |
| 712 | |
| 713 | case ASM_DATA_EVENT_RENDERED_EOS: |
| 714 | spin_lock_irqsave(&prtd->lock, flags); |
| 715 | pr_debug("%s: ASM_DATA_CMDRSP_EOS token 0x%x,stream id %d\n", |
| 716 | __func__, token, stream_id); |
| 717 | if (atomic_read(&prtd->eos) && |
| 718 | !prtd->gapless_state.set_next_stream_id) { |
| 719 | pr_debug("ASM_DATA_CMDRSP_EOS wake up\n"); |
| 720 | prtd->eos_ack = 1; |
| 721 | wake_up(&prtd->eos_wait); |
| 722 | } |
| 723 | atomic_set(&prtd->eos, 0); |
| 724 | stream_index = STREAM_ARRAY_INDEX(stream_id); |
| 725 | if (stream_index >= MAX_NUMBER_OF_STREAMS || |
| 726 | stream_index < 0) { |
| 727 | pr_err("%s: Invalid stream index %d", __func__, |
| 728 | stream_index); |
| 729 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 730 | break; |
| 731 | } |
| 732 | |
| 733 | if (prtd->gapless_state.set_next_stream_id && |
| 734 | prtd->gapless_state.stream_opened[stream_index]) { |
| 735 | pr_debug("%s: CMD_CLOSE stream_id %d\n", |
| 736 | __func__, stream_id); |
| 737 | q6asm_stream_cmd_nowait(ac, CMD_CLOSE, stream_id); |
| 738 | atomic_set(&prtd->close, 1); |
| 739 | prtd->gapless_state.stream_opened[stream_index] = 0; |
| 740 | prtd->gapless_state.set_next_stream_id = false; |
| 741 | } |
| 742 | if (prtd->gapless_state.gapless_transition) |
| 743 | prtd->gapless_state.gapless_transition = 0; |
| 744 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 745 | break; |
| 746 | case ASM_STREAM_PP_EVENT: |
| 747 | case ASM_STREAM_CMD_ENCDEC_EVENTS: |
| 748 | pr_debug("%s: ASM_STREAM_EVENT(0x%x)\n", __func__, opcode); |
| 749 | rtd = cstream->private_data; |
| 750 | if (!rtd) { |
| 751 | pr_err("%s: rtd is NULL\n", __func__); |
| 752 | return; |
| 753 | } |
| 754 | |
| 755 | ret = msm_adsp_inform_mixer_ctl(rtd, payload); |
| 756 | if (ret) { |
| 757 | pr_err("%s: failed to inform mixer ctrl. err = %d\n", |
| 758 | __func__, ret); |
| 759 | return; |
| 760 | } |
| 761 | break; |
| 762 | case ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY: |
| 763 | case ASM_DATA_EVENT_ENC_SR_CM_CHANGE_NOTIFY: { |
| 764 | pr_debug("ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY\n"); |
| 765 | chan_mode = payload[1] >> 16; |
| 766 | sample_rate = payload[2] >> 16; |
| 767 | if (prtd && (chan_mode != prtd->num_channels || |
| 768 | sample_rate != prtd->sample_rate)) { |
| 769 | prtd->num_channels = chan_mode; |
| 770 | prtd->sample_rate = sample_rate; |
| 771 | } |
| 772 | } |
| 773 | /* Fallthrough here */ |
| 774 | case APR_BASIC_RSP_RESULT: { |
| 775 | switch (payload[0]) { |
| 776 | case ASM_SESSION_CMD_RUN_V2: |
| 777 | /* check if the first buffer need to be sent to DSP */ |
| 778 | pr_debug("ASM_SESSION_CMD_RUN_V2\n"); |
| 779 | |
| 780 | /* FIXME: A state is a better way, dealing with this */ |
| 781 | spin_lock_irqsave(&prtd->lock, flags); |
| 782 | |
| 783 | if (cstream->direction == SND_COMPRESS_CAPTURE) { |
| 784 | atomic_set(&prtd->start, 1); |
| 785 | msm_compr_read_buffer(prtd); |
| 786 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 787 | break; |
| 788 | } |
| 789 | |
| 790 | if (!prtd->bytes_sent) { |
| 791 | bytes_available = prtd->bytes_received - |
| 792 | prtd->copied_total; |
| 793 | if (bytes_available < |
| 794 | cstream->runtime->fragment_size) { |
| 795 | pr_debug("CMD_RUN_V2 Insufficient data to send. break out\n"); |
| 796 | atomic_set(&prtd->xrun, 1); |
| 797 | } else { |
| 798 | msm_compr_send_buffer(prtd); |
| 799 | } |
| 800 | } |
| 801 | |
| 802 | /* |
| 803 | * The condition below ensures playback finishes in the |
| 804 | * follow cornercase |
| 805 | * WRITE(last buffer) |
| 806 | * WAIT_FOR_DRAIN |
| 807 | * PAUSE |
| 808 | * WRITE_DONE(X) |
| 809 | * RESUME |
| 810 | */ |
| 811 | if ((prtd->copied_total == prtd->bytes_sent) && |
| 812 | atomic_read(&prtd->drain)) { |
| 813 | pr_debug("RUN ack, wake up & continue pending drain\n"); |
| 814 | |
| 815 | if (prtd->last_buffer) |
| 816 | prtd->last_buffer = 0; |
| 817 | |
| 818 | prtd->drain_ready = 1; |
| 819 | wake_up(&prtd->drain_wait); |
| 820 | atomic_set(&prtd->drain, 0); |
| 821 | } |
| 822 | |
| 823 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 824 | break; |
| 825 | case ASM_STREAM_CMD_FLUSH: |
| 826 | pr_debug("%s: ASM_STREAM_CMD_FLUSH:", __func__); |
| 827 | pr_debug("token 0x%x, stream id %d\n", token, |
| 828 | stream_id); |
| 829 | prtd->cmd_ack = 1; |
| 830 | break; |
| 831 | case ASM_DATA_CMD_REMOVE_INITIAL_SILENCE: |
| 832 | pr_debug("%s: ASM_DATA_CMD_REMOVE_INITIAL_SILENCE:", |
| 833 | __func__); |
| 834 | pr_debug("token 0x%x, stream id = %d\n", token, |
| 835 | stream_id); |
| 836 | break; |
| 837 | case ASM_DATA_CMD_REMOVE_TRAILING_SILENCE: |
| 838 | pr_debug("%s: ASM_DATA_CMD_REMOVE_TRAILING_SILENCE:", |
| 839 | __func__); |
| 840 | pr_debug("token = 0x%x, stream id = %d\n", token, |
| 841 | stream_id); |
| 842 | break; |
| 843 | case ASM_STREAM_CMD_CLOSE: |
| 844 | pr_debug("%s: ASM_DATA_CMD_CLOSE:", __func__); |
| 845 | pr_debug("token 0x%x, stream id %d\n", token, |
| 846 | stream_id); |
| 847 | /* |
| 848 | * wakeup wait for stream avail on stream 3 |
| 849 | * after stream 1 ends. |
| 850 | */ |
| 851 | if (prtd->next_stream) { |
| 852 | pr_debug("%s:CLOSE:wakeup wait for stream\n", |
| 853 | __func__); |
| 854 | prtd->stream_available = 1; |
| 855 | wake_up(&prtd->wait_for_stream_avail); |
| 856 | prtd->next_stream = 0; |
| 857 | } |
| 858 | if (atomic_read(&prtd->close) && |
| 859 | atomic_read(&prtd->wait_on_close)) { |
| 860 | prtd->cmd_ack = 1; |
| 861 | wake_up(&prtd->close_wait); |
| 862 | } |
| 863 | atomic_set(&prtd->close, 0); |
| 864 | break; |
| 865 | case ASM_STREAM_CMD_REGISTER_PP_EVENTS: |
| 866 | pr_debug("%s: ASM_STREAM_CMD_REGISTER_PP_EVENTS:", |
| 867 | __func__); |
| 868 | break; |
| 869 | default: |
| 870 | break; |
| 871 | } |
| 872 | break; |
| 873 | } |
| 874 | case ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3: |
| 875 | pr_debug("%s: ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3\n", |
| 876 | __func__); |
| 877 | break; |
| 878 | case RESET_EVENTS: |
| 879 | pr_err("%s: Received reset events CB, move to error state", |
| 880 | __func__); |
| 881 | spin_lock_irqsave(&prtd->lock, flags); |
| 882 | /* |
| 883 | * Since ADSP is down, let this driver pretend that it copied |
| 884 | * all the bytes received, so that next write will be triggered |
| 885 | */ |
| 886 | prtd->copied_total = prtd->bytes_received; |
| 887 | snd_compr_fragment_elapsed(cstream); |
| 888 | atomic_set(&prtd->error, 1); |
| 889 | wake_up(&prtd->drain_wait); |
| 890 | if (atomic_cmpxchg(&prtd->eos, 1, 0)) { |
| 891 | pr_debug("%s:unblock eos wait queues", __func__); |
| 892 | wake_up(&prtd->eos_wait); |
| 893 | } |
| 894 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 895 | break; |
| 896 | default: |
| 897 | pr_debug("%s: Not Supported Event opcode[0x%x]\n", |
| 898 | __func__, opcode); |
| 899 | break; |
| 900 | } |
| 901 | } |
| 902 | |
| 903 | static int msm_compr_get_partial_drain_delay(int frame_sz, int sample_rate) |
| 904 | { |
| 905 | int delay_time_ms = 0; |
| 906 | |
| 907 | delay_time_ms = ((DSP_NUM_OUTPUT_FRAME_BUFFERED * frame_sz * 1000) / |
| 908 | sample_rate) + DSP_PP_BUFFERING_IN_MSEC; |
| 909 | delay_time_ms = delay_time_ms > PARTIAL_DRAIN_ACK_EARLY_BY_MSEC ? |
| 910 | delay_time_ms - PARTIAL_DRAIN_ACK_EARLY_BY_MSEC : 0; |
| 911 | |
| 912 | pr_debug("%s: frame_sz %d, sample_rate %d, partial drain delay %d\n", |
| 913 | __func__, frame_sz, sample_rate, delay_time_ms); |
| 914 | return delay_time_ms; |
| 915 | } |
| 916 | |
| 917 | static void populate_codec_list(struct msm_compr_audio *prtd) |
| 918 | { |
| 919 | pr_debug("%s\n", __func__); |
| 920 | prtd->compr_cap.direction = SND_COMPRESS_PLAYBACK; |
| 921 | prtd->compr_cap.min_fragment_size = |
| 922 | COMPR_PLAYBACK_MIN_FRAGMENT_SIZE; |
| 923 | prtd->compr_cap.max_fragment_size = |
| 924 | COMPR_PLAYBACK_MAX_FRAGMENT_SIZE; |
| 925 | prtd->compr_cap.min_fragments = |
| 926 | COMPR_PLAYBACK_MIN_NUM_FRAGMENTS; |
| 927 | prtd->compr_cap.max_fragments = |
| 928 | COMPR_PLAYBACK_MAX_NUM_FRAGMENTS; |
| 929 | prtd->compr_cap.num_codecs = 17; |
| 930 | prtd->compr_cap.codecs[0] = SND_AUDIOCODEC_MP3; |
| 931 | prtd->compr_cap.codecs[1] = SND_AUDIOCODEC_AAC; |
| 932 | prtd->compr_cap.codecs[2] = SND_AUDIOCODEC_AC3; |
| 933 | prtd->compr_cap.codecs[3] = SND_AUDIOCODEC_EAC3; |
| 934 | prtd->compr_cap.codecs[4] = SND_AUDIOCODEC_MP2; |
| 935 | prtd->compr_cap.codecs[5] = SND_AUDIOCODEC_PCM; |
| 936 | prtd->compr_cap.codecs[6] = SND_AUDIOCODEC_WMA; |
| 937 | prtd->compr_cap.codecs[7] = SND_AUDIOCODEC_WMA_PRO; |
| 938 | prtd->compr_cap.codecs[8] = SND_AUDIOCODEC_FLAC; |
| 939 | prtd->compr_cap.codecs[9] = SND_AUDIOCODEC_VORBIS; |
| 940 | prtd->compr_cap.codecs[10] = SND_AUDIOCODEC_ALAC; |
| 941 | prtd->compr_cap.codecs[11] = SND_AUDIOCODEC_APE; |
| 942 | prtd->compr_cap.codecs[12] = SND_AUDIOCODEC_DTS; |
| 943 | prtd->compr_cap.codecs[13] = SND_AUDIOCODEC_DSD; |
| 944 | prtd->compr_cap.codecs[14] = SND_AUDIOCODEC_APTX; |
| 945 | prtd->compr_cap.codecs[15] = SND_AUDIOCODEC_TRUEHD; |
| 946 | prtd->compr_cap.codecs[16] = SND_AUDIOCODEC_IEC61937; |
| 947 | } |
| 948 | |
| 949 | static int msm_compr_send_media_format_block(struct snd_compr_stream *cstream, |
| 950 | int stream_id, |
| 951 | bool use_gapless_codec_options) |
| 952 | { |
| 953 | struct snd_compr_runtime *runtime = cstream->runtime; |
| 954 | struct msm_compr_audio *prtd = runtime->private_data; |
| 955 | struct snd_soc_pcm_runtime *rtd = cstream->private_data; |
| 956 | struct msm_compr_pdata *pdata = |
| 957 | snd_soc_platform_get_drvdata(rtd->platform); |
| 958 | struct asm_aac_cfg aac_cfg; |
| 959 | struct asm_wma_cfg wma_cfg; |
| 960 | struct asm_wmapro_cfg wma_pro_cfg; |
| 961 | struct asm_flac_cfg flac_cfg; |
| 962 | struct asm_vorbis_cfg vorbis_cfg; |
| 963 | struct asm_alac_cfg alac_cfg; |
| 964 | struct asm_ape_cfg ape_cfg; |
| 965 | struct asm_dsd_cfg dsd_cfg; |
| 966 | struct aptx_dec_bt_addr_cfg aptx_cfg; |
| 967 | union snd_codec_options *codec_options; |
| 968 | |
| 969 | int ret = 0; |
| 970 | uint16_t bit_width; |
| 971 | bool use_default_chmap = true; |
| 972 | char *chmap = NULL; |
| 973 | uint16_t sample_word_size; |
| 974 | |
| 975 | pr_debug("%s: use_gapless_codec_options %d\n", |
| 976 | __func__, use_gapless_codec_options); |
| 977 | |
| 978 | if (use_gapless_codec_options) |
| 979 | codec_options = &(prtd->gapless_state.codec_options); |
| 980 | else |
| 981 | codec_options = &(prtd->codec_param.codec.options); |
| 982 | |
| 983 | if (!codec_options) { |
| 984 | pr_err("%s: codec_options is NULL\n", __func__); |
| 985 | return -EINVAL; |
| 986 | } |
| 987 | |
| 988 | switch (prtd->codec) { |
| 989 | case FORMAT_LINEAR_PCM: |
| 990 | pr_debug("SND_AUDIOCODEC_PCM\n"); |
| 991 | if (pdata->ch_map[rtd->dai_link->id]) { |
| 992 | use_default_chmap = |
| 993 | !(pdata->ch_map[rtd->dai_link->id]->set_ch_map); |
| 994 | chmap = |
| 995 | pdata->ch_map[rtd->dai_link->id]->channel_map; |
| 996 | } |
| 997 | |
| 998 | switch (prtd->codec_param.codec.format) { |
| 999 | case SNDRV_PCM_FORMAT_S32_LE: |
| 1000 | bit_width = 32; |
| 1001 | sample_word_size = 32; |
| 1002 | break; |
| 1003 | case SNDRV_PCM_FORMAT_S24_LE: |
| 1004 | bit_width = 24; |
| 1005 | sample_word_size = 32; |
| 1006 | break; |
| 1007 | case SNDRV_PCM_FORMAT_S24_3LE: |
| 1008 | bit_width = 24; |
| 1009 | sample_word_size = 24; |
| 1010 | break; |
| 1011 | case SNDRV_PCM_FORMAT_S16_LE: |
| 1012 | default: |
| 1013 | bit_width = 16; |
| 1014 | sample_word_size = 16; |
| 1015 | break; |
| 1016 | } |
Dieter Luecking | ba7644d | 2018-09-28 15:09:32 +0200 | [diff] [blame] | 1017 | |
| 1018 | if (q6core_get_avcs_api_version_per_service( |
| 1019 | APRV2_IDS_SERVICE_ID_ADSP_ASM_V) >= |
| 1020 | ADSP_ASM_API_VERSION_V2) { |
| 1021 | ret = q6asm_media_format_block_pcm_format_support_v5( |
| 1022 | prtd->audio_client, |
| 1023 | prtd->sample_rate, |
| 1024 | prtd->num_channels, |
| 1025 | bit_width, stream_id, |
| 1026 | use_default_chmap, |
| 1027 | chmap, |
| 1028 | sample_word_size, |
| 1029 | ASM_LITTLE_ENDIAN, |
| 1030 | DEFAULT_QF); |
| 1031 | } else { |
| 1032 | ret = q6asm_media_format_block_pcm_format_support_v4( |
| 1033 | prtd->audio_client, |
| 1034 | prtd->sample_rate, |
| 1035 | prtd->num_channels, |
| 1036 | bit_width, stream_id, |
| 1037 | use_default_chmap, |
| 1038 | chmap, |
| 1039 | sample_word_size, |
| 1040 | ASM_LITTLE_ENDIAN, |
| 1041 | DEFAULT_QF); |
| 1042 | } |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 1043 | if (ret < 0) |
| 1044 | pr_err("%s: CMD Format block failed\n", __func__); |
| 1045 | |
| 1046 | break; |
| 1047 | case FORMAT_MP3: |
| 1048 | pr_debug("SND_AUDIOCODEC_MP3\n"); |
| 1049 | /* no media format block needed */ |
| 1050 | break; |
| 1051 | case FORMAT_MPEG4_AAC: |
| 1052 | pr_debug("SND_AUDIOCODEC_AAC\n"); |
| 1053 | memset(&aac_cfg, 0x0, sizeof(struct asm_aac_cfg)); |
| 1054 | aac_cfg.aot = AAC_ENC_MODE_EAAC_P; |
| 1055 | if (prtd->codec_param.codec.format == |
| 1056 | SND_AUDIOSTREAMFORMAT_MP4ADTS) |
| 1057 | aac_cfg.format = 0x0; |
| 1058 | else if (prtd->codec_param.codec.format == |
| 1059 | SND_AUDIOSTREAMFORMAT_MP4LATM) |
| 1060 | aac_cfg.format = 0x04; |
| 1061 | else |
| 1062 | aac_cfg.format = 0x03; |
| 1063 | aac_cfg.ch_cfg = prtd->num_channels; |
| 1064 | aac_cfg.sample_rate = prtd->sample_rate; |
| 1065 | ret = q6asm_stream_media_format_block_aac(prtd->audio_client, |
| 1066 | &aac_cfg, stream_id); |
| 1067 | if (ret < 0) |
| 1068 | pr_err("%s: CMD Format block failed\n", __func__); |
| 1069 | break; |
| 1070 | case FORMAT_AC3: |
| 1071 | pr_debug("SND_AUDIOCODEC_AC3\n"); |
| 1072 | break; |
| 1073 | case FORMAT_EAC3: |
| 1074 | pr_debug("SND_AUDIOCODEC_EAC3\n"); |
| 1075 | break; |
| 1076 | case FORMAT_WMA_V9: |
| 1077 | pr_debug("SND_AUDIOCODEC_WMA\n"); |
| 1078 | memset(&wma_cfg, 0x0, sizeof(struct asm_wma_cfg)); |
| 1079 | wma_cfg.format_tag = prtd->codec_param.codec.format; |
| 1080 | wma_cfg.ch_cfg = prtd->codec_param.codec.ch_in; |
| 1081 | wma_cfg.sample_rate = prtd->sample_rate; |
| 1082 | wma_cfg.avg_bytes_per_sec = codec_options->wma.avg_bit_rate/8; |
| 1083 | wma_cfg.block_align = codec_options->wma.super_block_align; |
| 1084 | wma_cfg.valid_bits_per_sample = |
| 1085 | codec_options->wma.bits_per_sample; |
| 1086 | wma_cfg.ch_mask = codec_options->wma.channelmask; |
| 1087 | wma_cfg.encode_opt = codec_options->wma.encodeopt; |
| 1088 | ret = q6asm_media_format_block_wma(prtd->audio_client, |
| 1089 | &wma_cfg, stream_id); |
| 1090 | if (ret < 0) |
| 1091 | pr_err("%s: CMD Format block failed\n", __func__); |
| 1092 | break; |
| 1093 | case FORMAT_WMA_V10PRO: |
| 1094 | pr_debug("SND_AUDIOCODEC_WMA_PRO\n"); |
| 1095 | memset(&wma_pro_cfg, 0x0, sizeof(struct asm_wmapro_cfg)); |
| 1096 | wma_pro_cfg.format_tag = prtd->codec_param.codec.format; |
| 1097 | wma_pro_cfg.ch_cfg = prtd->codec_param.codec.ch_in; |
| 1098 | wma_pro_cfg.sample_rate = prtd->sample_rate; |
| 1099 | wma_cfg.avg_bytes_per_sec = codec_options->wma.avg_bit_rate/8; |
| 1100 | wma_pro_cfg.block_align = codec_options->wma.super_block_align; |
| 1101 | wma_pro_cfg.valid_bits_per_sample = |
| 1102 | codec_options->wma.bits_per_sample; |
| 1103 | wma_pro_cfg.ch_mask = codec_options->wma.channelmask; |
| 1104 | wma_pro_cfg.encode_opt = codec_options->wma.encodeopt; |
| 1105 | wma_pro_cfg.adv_encode_opt = codec_options->wma.encodeopt1; |
| 1106 | wma_pro_cfg.adv_encode_opt2 = codec_options->wma.encodeopt2; |
| 1107 | ret = q6asm_media_format_block_wmapro(prtd->audio_client, |
| 1108 | &wma_pro_cfg, stream_id); |
| 1109 | if (ret < 0) |
| 1110 | pr_err("%s: CMD Format block failed\n", __func__); |
| 1111 | break; |
| 1112 | case FORMAT_MP2: |
| 1113 | pr_debug("%s: SND_AUDIOCODEC_MP2\n", __func__); |
| 1114 | break; |
| 1115 | case FORMAT_FLAC: |
| 1116 | pr_debug("%s: SND_AUDIOCODEC_FLAC\n", __func__); |
| 1117 | memset(&flac_cfg, 0x0, sizeof(struct asm_flac_cfg)); |
| 1118 | flac_cfg.ch_cfg = prtd->num_channels; |
| 1119 | flac_cfg.sample_rate = prtd->sample_rate; |
| 1120 | flac_cfg.stream_info_present = 1; |
| 1121 | flac_cfg.sample_size = codec_options->flac_dec.sample_size; |
| 1122 | flac_cfg.min_blk_size = codec_options->flac_dec.min_blk_size; |
| 1123 | flac_cfg.max_blk_size = codec_options->flac_dec.max_blk_size; |
| 1124 | flac_cfg.max_frame_size = |
| 1125 | codec_options->flac_dec.max_frame_size; |
| 1126 | flac_cfg.min_frame_size = |
| 1127 | codec_options->flac_dec.min_frame_size; |
| 1128 | |
| 1129 | ret = q6asm_stream_media_format_block_flac(prtd->audio_client, |
| 1130 | &flac_cfg, stream_id); |
| 1131 | if (ret < 0) |
| 1132 | pr_err("%s: CMD Format block failed ret %d\n", |
| 1133 | __func__, ret); |
| 1134 | |
| 1135 | break; |
| 1136 | case FORMAT_VORBIS: |
| 1137 | pr_debug("%s: SND_AUDIOCODEC_VORBIS\n", __func__); |
| 1138 | memset(&vorbis_cfg, 0x0, sizeof(struct asm_vorbis_cfg)); |
| 1139 | vorbis_cfg.bit_stream_fmt = |
| 1140 | codec_options->vorbis_dec.bit_stream_fmt; |
| 1141 | |
| 1142 | ret = q6asm_stream_media_format_block_vorbis( |
| 1143 | prtd->audio_client, &vorbis_cfg, |
| 1144 | stream_id); |
| 1145 | if (ret < 0) |
| 1146 | pr_err("%s: CMD Format block failed ret %d\n", |
| 1147 | __func__, ret); |
| 1148 | |
| 1149 | break; |
| 1150 | case FORMAT_ALAC: |
| 1151 | pr_debug("%s: SND_AUDIOCODEC_ALAC\n", __func__); |
| 1152 | memset(&alac_cfg, 0x0, sizeof(struct asm_alac_cfg)); |
| 1153 | alac_cfg.num_channels = prtd->num_channels; |
| 1154 | alac_cfg.sample_rate = prtd->sample_rate; |
| 1155 | alac_cfg.frame_length = codec_options->alac.frame_length; |
| 1156 | alac_cfg.compatible_version = |
| 1157 | codec_options->alac.compatible_version; |
| 1158 | alac_cfg.bit_depth = codec_options->alac.bit_depth; |
| 1159 | alac_cfg.pb = codec_options->alac.pb; |
| 1160 | alac_cfg.mb = codec_options->alac.mb; |
| 1161 | alac_cfg.kb = codec_options->alac.kb; |
| 1162 | alac_cfg.max_run = codec_options->alac.max_run; |
| 1163 | alac_cfg.max_frame_bytes = codec_options->alac.max_frame_bytes; |
| 1164 | alac_cfg.avg_bit_rate = codec_options->alac.avg_bit_rate; |
| 1165 | alac_cfg.channel_layout_tag = |
| 1166 | codec_options->alac.channel_layout_tag; |
| 1167 | |
| 1168 | ret = q6asm_media_format_block_alac(prtd->audio_client, |
| 1169 | &alac_cfg, stream_id); |
| 1170 | if (ret < 0) |
| 1171 | pr_err("%s: CMD Format block failed ret %d\n", |
| 1172 | __func__, ret); |
| 1173 | break; |
| 1174 | case FORMAT_APE: |
| 1175 | pr_debug("%s: SND_AUDIOCODEC_APE\n", __func__); |
| 1176 | memset(&ape_cfg, 0x0, sizeof(struct asm_ape_cfg)); |
| 1177 | ape_cfg.num_channels = prtd->num_channels; |
| 1178 | ape_cfg.sample_rate = prtd->sample_rate; |
| 1179 | ape_cfg.compatible_version = |
| 1180 | codec_options->ape.compatible_version; |
| 1181 | ape_cfg.compression_level = |
| 1182 | codec_options->ape.compression_level; |
| 1183 | ape_cfg.format_flags = codec_options->ape.format_flags; |
| 1184 | ape_cfg.blocks_per_frame = codec_options->ape.blocks_per_frame; |
| 1185 | ape_cfg.final_frame_blocks = |
| 1186 | codec_options->ape.final_frame_blocks; |
| 1187 | ape_cfg.total_frames = codec_options->ape.total_frames; |
| 1188 | ape_cfg.bits_per_sample = codec_options->ape.bits_per_sample; |
| 1189 | ape_cfg.seek_table_present = |
| 1190 | codec_options->ape.seek_table_present; |
| 1191 | |
| 1192 | ret = q6asm_media_format_block_ape(prtd->audio_client, |
| 1193 | &ape_cfg, stream_id); |
| 1194 | |
| 1195 | if (ret < 0) |
| 1196 | pr_err("%s: CMD Format block failed ret %d\n", |
| 1197 | __func__, ret); |
| 1198 | break; |
| 1199 | case FORMAT_DTS: |
| 1200 | pr_debug("SND_AUDIOCODEC_DTS\n"); |
| 1201 | /* no media format block needed */ |
| 1202 | break; |
| 1203 | case FORMAT_DSD: |
| 1204 | pr_debug("%s: SND_AUDIOCODEC_DSD\n", __func__); |
| 1205 | memset(&dsd_cfg, 0x0, sizeof(struct asm_dsd_cfg)); |
| 1206 | dsd_cfg.num_channels = prtd->num_channels; |
| 1207 | dsd_cfg.dsd_data_rate = prtd->sample_rate; |
| 1208 | dsd_cfg.num_version = 0; |
| 1209 | dsd_cfg.is_bitwise_big_endian = 1; |
| 1210 | dsd_cfg.dsd_channel_block_size = 1; |
| 1211 | ret = q6asm_media_format_block_dsd(prtd->audio_client, |
| 1212 | &dsd_cfg, stream_id); |
| 1213 | if (ret < 0) |
| 1214 | pr_err("%s: CMD DSD Format block failed ret %d\n", |
| 1215 | __func__, ret); |
| 1216 | break; |
| 1217 | case FORMAT_TRUEHD: |
| 1218 | pr_debug("SND_AUDIOCODEC_TRUEHD\n"); |
| 1219 | /* no media format block needed */ |
| 1220 | break; |
| 1221 | case FORMAT_IEC61937: |
| 1222 | pr_debug("SND_AUDIOCODEC_IEC61937\n"); |
| 1223 | ret = q6asm_media_format_block_iec(prtd->audio_client, |
| 1224 | prtd->sample_rate, |
| 1225 | prtd->num_channels); |
| 1226 | if (ret < 0) |
| 1227 | pr_err("%s: CMD IEC61937 Format block failed ret %d\n", |
| 1228 | __func__, ret); |
| 1229 | break; |
| 1230 | case FORMAT_APTX: |
| 1231 | pr_debug("SND_AUDIOCODEC_APTX\n"); |
| 1232 | memset(&aptx_cfg, 0x0, sizeof(struct aptx_dec_bt_addr_cfg)); |
| 1233 | ret = q6asm_stream_media_format_block_aptx_dec( |
| 1234 | prtd->audio_client, |
| 1235 | prtd->sample_rate, |
| 1236 | stream_id); |
| 1237 | if (ret >= 0) { |
| 1238 | aptx_cfg.nap = codec_options->aptx_dec.nap; |
| 1239 | aptx_cfg.uap = codec_options->aptx_dec.uap; |
| 1240 | aptx_cfg.lap = codec_options->aptx_dec.lap; |
| 1241 | q6asm_set_aptx_dec_bt_addr(prtd->audio_client, |
| 1242 | &aptx_cfg); |
| 1243 | } else { |
| 1244 | pr_err("%s: CMD Format block failed ret %d\n", |
| 1245 | __func__, ret); |
| 1246 | } |
| 1247 | break; |
| 1248 | default: |
| 1249 | pr_debug("%s, unsupported format, skip", __func__); |
| 1250 | break; |
| 1251 | } |
| 1252 | return ret; |
| 1253 | } |
| 1254 | |
| 1255 | static int msm_compr_init_pp_params(struct snd_compr_stream *cstream, |
| 1256 | struct audio_client *ac) |
| 1257 | { |
| 1258 | int ret = 0; |
| 1259 | struct asm_softvolume_params softvol = { |
| 1260 | .period = SOFT_VOLUME_PERIOD, |
| 1261 | .step = SOFT_VOLUME_STEP, |
| 1262 | .rampingcurve = SOFT_VOLUME_CURVE_LINEAR, |
| 1263 | }; |
| 1264 | |
| 1265 | switch (ac->topology) { |
| 1266 | default: |
| 1267 | ret = q6asm_set_softvolume_v2(ac, &softvol, |
| 1268 | SOFT_VOLUME_INSTANCE_1); |
| 1269 | if (ret < 0) |
| 1270 | pr_err("%s: Send SoftVolume Param failed ret=%d\n", |
| 1271 | __func__, ret); |
| 1272 | |
| 1273 | break; |
| 1274 | } |
| 1275 | return ret; |
| 1276 | } |
| 1277 | |
| 1278 | static int msm_compr_configure_dsp_for_playback |
| 1279 | (struct snd_compr_stream *cstream) |
| 1280 | { |
| 1281 | struct snd_compr_runtime *runtime = cstream->runtime; |
| 1282 | struct msm_compr_audio *prtd = runtime->private_data; |
| 1283 | struct snd_soc_pcm_runtime *soc_prtd = cstream->private_data; |
| 1284 | uint16_t bits_per_sample = 16; |
| 1285 | int dir = IN, ret = 0; |
| 1286 | struct audio_client *ac = prtd->audio_client; |
| 1287 | uint32_t stream_index; |
| 1288 | struct asm_softpause_params softpause = { |
| 1289 | .enable = SOFT_PAUSE_ENABLE, |
| 1290 | .period = SOFT_PAUSE_PERIOD, |
| 1291 | .step = SOFT_PAUSE_STEP, |
| 1292 | .rampingcurve = SOFT_PAUSE_CURVE_LINEAR, |
| 1293 | }; |
| 1294 | struct asm_softvolume_params softvol = { |
| 1295 | .period = SOFT_VOLUME_PERIOD, |
| 1296 | .step = SOFT_VOLUME_STEP, |
| 1297 | .rampingcurve = SOFT_VOLUME_CURVE_LINEAR, |
| 1298 | }; |
Laxminath Kasam | 8f7ccc2 | 2017-08-28 17:35:04 +0530 | [diff] [blame] | 1299 | struct snd_kcontrol *kctl; |
| 1300 | struct snd_ctl_elem_value kctl_elem_value; |
| 1301 | uint16_t target_asm_bit_width = 0; |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 1302 | |
| 1303 | pr_debug("%s: stream_id %d\n", __func__, ac->stream_id); |
| 1304 | stream_index = STREAM_ARRAY_INDEX(ac->stream_id); |
| 1305 | if (stream_index >= MAX_NUMBER_OF_STREAMS || stream_index < 0) { |
| 1306 | pr_err("%s: Invalid stream index:%d", __func__, stream_index); |
| 1307 | return -EINVAL; |
| 1308 | } |
| 1309 | |
Laxminath Kasam | 8f7ccc2 | 2017-08-28 17:35:04 +0530 | [diff] [blame] | 1310 | kctl = snd_soc_card_get_kcontrol(soc_prtd->card, |
| 1311 | DSP_BIT_WIDTH_MIXER_CTL); |
| 1312 | if (kctl) { |
| 1313 | kctl->get(kctl, &kctl_elem_value); |
| 1314 | target_asm_bit_width = kctl_elem_value.value.integer.value[0]; |
| 1315 | if (target_asm_bit_width > 0) { |
| 1316 | pr_debug("%s enforce ASM bitwidth to %d from %d\n", |
| 1317 | __func__, |
| 1318 | target_asm_bit_width, |
| 1319 | bits_per_sample); |
| 1320 | bits_per_sample = target_asm_bit_width; |
| 1321 | } |
| 1322 | } else { |
| 1323 | pr_info("%s: failed to get mixer ctl for %s.\n", |
| 1324 | __func__, DSP_BIT_WIDTH_MIXER_CTL); |
| 1325 | } |
| 1326 | |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 1327 | if ((prtd->codec_param.codec.format == SNDRV_PCM_FORMAT_S24_LE) || |
| 1328 | (prtd->codec_param.codec.format == SNDRV_PCM_FORMAT_S24_3LE)) |
| 1329 | bits_per_sample = 24; |
| 1330 | else if (prtd->codec_param.codec.format == SNDRV_PCM_FORMAT_S32_LE) |
| 1331 | bits_per_sample = 32; |
| 1332 | |
| 1333 | if (prtd->compr_passthr != LEGACY_PCM) { |
| 1334 | ret = q6asm_open_write_compressed(ac, prtd->codec, |
| 1335 | prtd->compr_passthr); |
| 1336 | if (ret < 0) { |
| 1337 | pr_err("%s:ASM open write err[%d] for compr_type[%d]\n", |
| 1338 | __func__, ret, prtd->compr_passthr); |
| 1339 | return ret; |
| 1340 | } |
| 1341 | prtd->gapless_state.stream_opened[stream_index] = 1; |
| 1342 | |
| 1343 | ret = msm_pcm_routing_reg_phy_compr_stream( |
| 1344 | soc_prtd->dai_link->id, |
| 1345 | ac->perf_mode, |
| 1346 | prtd->session_id, |
| 1347 | SNDRV_PCM_STREAM_PLAYBACK, |
| 1348 | prtd->compr_passthr); |
| 1349 | if (ret) { |
| 1350 | pr_err("%s: compr stream reg failed:%d\n", __func__, |
| 1351 | ret); |
| 1352 | return ret; |
| 1353 | } |
| 1354 | } else { |
| 1355 | pr_debug("%s: stream_id %d bits_per_sample %d\n", |
| 1356 | __func__, ac->stream_id, bits_per_sample); |
Dieter Luecking | ba7644d | 2018-09-28 15:09:32 +0200 | [diff] [blame] | 1357 | |
| 1358 | if (q6core_get_avcs_api_version_per_service( |
| 1359 | APRV2_IDS_SERVICE_ID_ADSP_ASM_V) >= |
| 1360 | ADSP_ASM_API_VERSION_V2) |
| 1361 | ret = q6asm_stream_open_write_v5(ac, |
| 1362 | prtd->codec, bits_per_sample, |
| 1363 | ac->stream_id, |
| 1364 | prtd->gapless_state.use_dsp_gapless_mode); |
| 1365 | else |
| 1366 | ret = q6asm_stream_open_write_v4(ac, |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 1367 | prtd->codec, bits_per_sample, |
| 1368 | ac->stream_id, |
| 1369 | prtd->gapless_state.use_dsp_gapless_mode); |
| 1370 | if (ret < 0) { |
| 1371 | pr_err("%s:ASM open write err[%d] for compr type[%d]\n", |
| 1372 | __func__, ret, prtd->compr_passthr); |
| 1373 | return -ENOMEM; |
| 1374 | } |
| 1375 | prtd->gapless_state.stream_opened[stream_index] = 1; |
| 1376 | |
| 1377 | pr_debug("%s: BE id %d\n", __func__, soc_prtd->dai_link->id); |
| 1378 | ret = msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->id, |
| 1379 | ac->perf_mode, |
| 1380 | prtd->session_id, |
| 1381 | SNDRV_PCM_STREAM_PLAYBACK); |
| 1382 | if (ret) { |
| 1383 | pr_err("%s: stream reg failed:%d\n", __func__, ret); |
| 1384 | return ret; |
| 1385 | } |
| 1386 | } |
| 1387 | |
| 1388 | ret = msm_compr_set_volume(cstream, 0, 0); |
| 1389 | if (ret < 0) |
| 1390 | pr_err("%s : Set Volume failed : %d", __func__, ret); |
| 1391 | |
| 1392 | if (prtd->compr_passthr != LEGACY_PCM) { |
| 1393 | pr_debug("%s : Don't send cal and PP params for compress path", |
| 1394 | __func__); |
| 1395 | } else { |
| 1396 | ret = q6asm_send_cal(ac); |
| 1397 | if (ret < 0) |
| 1398 | pr_debug("%s : Send cal failed : %d", __func__, ret); |
| 1399 | |
| 1400 | ret = q6asm_set_softpause(ac, &softpause); |
| 1401 | if (ret < 0) |
| 1402 | pr_err("%s: Send SoftPause Param failed ret=%d\n", |
| 1403 | __func__, ret); |
| 1404 | |
| 1405 | ret = q6asm_set_softvolume(ac, &softvol); |
| 1406 | if (ret < 0) |
| 1407 | pr_err("%s: Send SoftVolume Param failed ret=%d\n", |
| 1408 | __func__, ret); |
| 1409 | } |
| 1410 | ret = q6asm_set_io_mode(ac, (COMPRESSED_STREAM_IO | ASYNC_IO_MODE)); |
| 1411 | if (ret < 0) { |
| 1412 | pr_err("%s: Set IO mode failed\n", __func__); |
| 1413 | return -EINVAL; |
| 1414 | } |
| 1415 | |
| 1416 | runtime->fragments = prtd->codec_param.buffer.fragments; |
| 1417 | runtime->fragment_size = prtd->codec_param.buffer.fragment_size; |
| 1418 | pr_debug("allocate %d buffers each of size %d\n", |
| 1419 | runtime->fragments, |
| 1420 | runtime->fragment_size); |
| 1421 | ret = q6asm_audio_client_buf_alloc_contiguous(dir, ac, |
| 1422 | runtime->fragment_size, |
| 1423 | runtime->fragments); |
| 1424 | if (ret < 0) { |
| 1425 | pr_err("Audio Start: Buffer Allocation failed rc = %d\n", ret); |
| 1426 | return -ENOMEM; |
| 1427 | } |
| 1428 | |
| 1429 | prtd->byte_offset = 0; |
| 1430 | prtd->copied_total = 0; |
| 1431 | prtd->app_pointer = 0; |
| 1432 | prtd->bytes_received = 0; |
| 1433 | prtd->bytes_sent = 0; |
| 1434 | prtd->buffer = ac->port[dir].buf[0].data; |
| 1435 | prtd->buffer_paddr = ac->port[dir].buf[0].phys; |
| 1436 | prtd->buffer_size = runtime->fragments * runtime->fragment_size; |
| 1437 | |
| 1438 | /* Bit-0 of flags represent timestamp mode */ |
| 1439 | if (prtd->codec_param.codec.flags & COMPRESSED_TIMESTAMP_FLAG) |
| 1440 | prtd->ts_header_offset = sizeof(struct snd_codec_metadata); |
| 1441 | else |
| 1442 | prtd->ts_header_offset = 0; |
| 1443 | |
| 1444 | ret = msm_compr_send_media_format_block(cstream, ac->stream_id, false); |
| 1445 | if (ret < 0) |
| 1446 | pr_err("%s, failed to send media format block\n", __func__); |
| 1447 | |
| 1448 | return ret; |
| 1449 | } |
| 1450 | |
| 1451 | static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream) |
| 1452 | { |
| 1453 | struct snd_compr_runtime *runtime = cstream->runtime; |
| 1454 | struct msm_compr_audio *prtd = runtime->private_data; |
| 1455 | struct snd_soc_pcm_runtime *soc_prtd = cstream->private_data; |
| 1456 | uint16_t bits_per_sample; |
| 1457 | uint16_t sample_word_size; |
| 1458 | int dir = OUT, ret = 0; |
| 1459 | struct audio_client *ac = prtd->audio_client; |
| 1460 | uint32_t stream_index; |
| 1461 | |
| 1462 | switch (prtd->codec_param.codec.format) { |
| 1463 | case SNDRV_PCM_FORMAT_S24_LE: |
| 1464 | bits_per_sample = 24; |
| 1465 | sample_word_size = 32; |
| 1466 | break; |
| 1467 | case SNDRV_PCM_FORMAT_S24_3LE: |
| 1468 | bits_per_sample = 24; |
| 1469 | sample_word_size = 24; |
| 1470 | break; |
| 1471 | case SNDRV_PCM_FORMAT_S32_LE: |
| 1472 | bits_per_sample = 32; |
| 1473 | sample_word_size = 32; |
| 1474 | break; |
| 1475 | case SNDRV_PCM_FORMAT_S16_LE: |
| 1476 | default: |
| 1477 | bits_per_sample = 16; |
| 1478 | sample_word_size = 16; |
| 1479 | break; |
| 1480 | } |
| 1481 | |
| 1482 | pr_debug("%s: stream_id %d bits_per_sample %d\n", |
| 1483 | __func__, ac->stream_id, bits_per_sample); |
| 1484 | |
| 1485 | if (prtd->codec_param.codec.flags & COMPRESSED_TIMESTAMP_FLAG) { |
| 1486 | ret = q6asm_open_read_v4(prtd->audio_client, FORMAT_LINEAR_PCM, |
| 1487 | bits_per_sample, true); |
| 1488 | } else { |
| 1489 | ret = q6asm_open_read_v4(prtd->audio_client, FORMAT_LINEAR_PCM, |
| 1490 | bits_per_sample, false); |
| 1491 | } |
| 1492 | if (ret < 0) { |
| 1493 | pr_err("%s: q6asm_open_read failed:%d\n", __func__, ret); |
| 1494 | return ret; |
| 1495 | } |
| 1496 | |
| 1497 | ret = msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->id, |
| 1498 | ac->perf_mode, |
| 1499 | prtd->session_id, |
| 1500 | SNDRV_PCM_STREAM_CAPTURE); |
| 1501 | if (ret) { |
| 1502 | pr_err("%s: stream reg failed:%d\n", __func__, ret); |
| 1503 | return ret; |
| 1504 | } |
| 1505 | |
| 1506 | ret = q6asm_set_io_mode(ac, (COMPRESSED_STREAM_IO | ASYNC_IO_MODE)); |
| 1507 | if (ret < 0) { |
| 1508 | pr_err("%s: Set IO mode failed\n", __func__); |
| 1509 | return -EINVAL; |
| 1510 | } |
| 1511 | |
| 1512 | stream_index = STREAM_ARRAY_INDEX(ac->stream_id); |
| 1513 | if (stream_index >= MAX_NUMBER_OF_STREAMS || stream_index < 0) { |
| 1514 | pr_err("%s: Invalid stream index:%d", __func__, stream_index); |
| 1515 | return -EINVAL; |
| 1516 | } |
| 1517 | |
| 1518 | runtime->fragments = prtd->codec_param.buffer.fragments; |
| 1519 | runtime->fragment_size = prtd->codec_param.buffer.fragment_size; |
| 1520 | pr_debug("%s: allocate %d buffers each of size %d\n", |
| 1521 | __func__, runtime->fragments, |
| 1522 | runtime->fragment_size); |
| 1523 | ret = q6asm_audio_client_buf_alloc_contiguous(dir, ac, |
| 1524 | runtime->fragment_size, |
| 1525 | runtime->fragments); |
| 1526 | if (ret < 0) { |
| 1527 | pr_err("Audio Start: Buffer Allocation failed rc = %d\n", ret); |
| 1528 | return -ENOMEM; |
| 1529 | } |
| 1530 | |
| 1531 | prtd->byte_offset = 0; |
| 1532 | prtd->received_total = 0; |
| 1533 | prtd->app_pointer = 0; |
| 1534 | prtd->bytes_copied = 0; |
| 1535 | prtd->bytes_read = 0; |
| 1536 | prtd->bytes_read_offset = 0; |
| 1537 | prtd->buffer = ac->port[dir].buf[0].data; |
| 1538 | prtd->buffer_paddr = ac->port[dir].buf[0].phys; |
| 1539 | prtd->buffer_size = runtime->fragments * runtime->fragment_size; |
| 1540 | |
| 1541 | /* Bit-0 of flags represent timestamp mode */ |
| 1542 | if (prtd->codec_param.codec.flags & COMPRESSED_TIMESTAMP_FLAG) |
| 1543 | prtd->ts_header_offset = sizeof(struct snd_codec_metadata); |
| 1544 | else |
| 1545 | prtd->ts_header_offset = 0; |
| 1546 | |
| 1547 | pr_debug("%s: sample_rate = %d channels = %d bps = %d sample_word_size = %d\n", |
| 1548 | __func__, prtd->sample_rate, prtd->num_channels, |
| 1549 | bits_per_sample, sample_word_size); |
Sachin Mohan Gadag | 265d94d | 2018-01-04 11:04:00 +0530 | [diff] [blame] | 1550 | ret = q6asm_enc_cfg_blk_pcm_format_support_v4(prtd->audio_client, |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 1551 | prtd->sample_rate, prtd->num_channels, |
Sachin Mohan Gadag | 265d94d | 2018-01-04 11:04:00 +0530 | [diff] [blame] | 1552 | bits_per_sample, sample_word_size, |
| 1553 | ASM_LITTLE_ENDIAN, DEFAULT_QF); |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 1554 | |
| 1555 | return ret; |
| 1556 | } |
| 1557 | |
| 1558 | static int msm_compr_playback_open(struct snd_compr_stream *cstream) |
| 1559 | { |
| 1560 | struct snd_compr_runtime *runtime = cstream->runtime; |
| 1561 | struct snd_soc_pcm_runtime *rtd = cstream->private_data; |
Aditya Bavanari | 9deef91 | 2017-11-20 13:31:31 +0530 | [diff] [blame] | 1562 | struct msm_compr_audio *prtd = NULL; |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 1563 | struct msm_compr_pdata *pdata = |
| 1564 | snd_soc_platform_get_drvdata(rtd->platform); |
| 1565 | |
| 1566 | pr_debug("%s\n", __func__); |
Aditya Bavanari | 9deef91 | 2017-11-20 13:31:31 +0530 | [diff] [blame] | 1567 | if (pdata->is_in_use[rtd->dai_link->id] == true) { |
| 1568 | pr_err("%s: %s is already in use, err: %d\n", |
| 1569 | __func__, rtd->dai_link->cpu_dai_name, -EBUSY); |
| 1570 | return -EBUSY; |
| 1571 | } |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 1572 | prtd = kzalloc(sizeof(struct msm_compr_audio), GFP_KERNEL); |
| 1573 | if (prtd == NULL) { |
| 1574 | pr_err("Failed to allocate memory for msm_compr_audio\n"); |
| 1575 | return -ENOMEM; |
| 1576 | } |
| 1577 | |
| 1578 | runtime->private_data = NULL; |
| 1579 | prtd->cstream = cstream; |
| 1580 | pdata->cstream[rtd->dai_link->id] = cstream; |
| 1581 | pdata->audio_effects[rtd->dai_link->id] = |
| 1582 | kzalloc(sizeof(struct msm_compr_audio_effects), GFP_KERNEL); |
Aditya Bavanari | 9deef91 | 2017-11-20 13:31:31 +0530 | [diff] [blame] | 1583 | if (pdata->audio_effects[rtd->dai_link->id] == NULL) { |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 1584 | pr_err("%s: Could not allocate memory for effects\n", __func__); |
| 1585 | pdata->cstream[rtd->dai_link->id] = NULL; |
| 1586 | kfree(prtd); |
| 1587 | return -ENOMEM; |
| 1588 | } |
| 1589 | pdata->dec_params[rtd->dai_link->id] = |
| 1590 | kzalloc(sizeof(struct msm_compr_dec_params), GFP_KERNEL); |
Aditya Bavanari | 9deef91 | 2017-11-20 13:31:31 +0530 | [diff] [blame] | 1591 | if (pdata->dec_params[rtd->dai_link->id] == NULL) { |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 1592 | pr_err("%s: Could not allocate memory for dec params\n", |
| 1593 | __func__); |
| 1594 | kfree(pdata->audio_effects[rtd->dai_link->id]); |
Aditya Bavanari | 9deef91 | 2017-11-20 13:31:31 +0530 | [diff] [blame] | 1595 | pdata->audio_effects[rtd->dai_link->id] = NULL; |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 1596 | pdata->cstream[rtd->dai_link->id] = NULL; |
| 1597 | kfree(prtd); |
| 1598 | return -ENOMEM; |
| 1599 | } |
| 1600 | prtd->codec = FORMAT_MP3; |
| 1601 | prtd->bytes_received = 0; |
| 1602 | prtd->bytes_sent = 0; |
| 1603 | prtd->copied_total = 0; |
| 1604 | prtd->byte_offset = 0; |
| 1605 | prtd->sample_rate = 44100; |
| 1606 | prtd->num_channels = 2; |
| 1607 | prtd->drain_ready = 0; |
| 1608 | prtd->last_buffer = 0; |
| 1609 | prtd->first_buffer = 1; |
| 1610 | prtd->partial_drain_delay = 0; |
| 1611 | prtd->next_stream = 0; |
| 1612 | memset(&prtd->gapless_state, 0, sizeof(struct msm_compr_gapless_state)); |
| 1613 | /* |
| 1614 | * Update the use_dsp_gapless_mode from gapless struture with the value |
| 1615 | * part of platform data. |
| 1616 | */ |
| 1617 | prtd->gapless_state.use_dsp_gapless_mode = pdata->use_dsp_gapless_mode; |
| 1618 | |
| 1619 | pr_debug("%s: gapless mode %d", __func__, pdata->use_dsp_gapless_mode); |
| 1620 | |
| 1621 | spin_lock_init(&prtd->lock); |
| 1622 | |
| 1623 | atomic_set(&prtd->eos, 0); |
| 1624 | atomic_set(&prtd->start, 0); |
| 1625 | atomic_set(&prtd->drain, 0); |
| 1626 | atomic_set(&prtd->xrun, 0); |
| 1627 | atomic_set(&prtd->close, 0); |
| 1628 | atomic_set(&prtd->wait_on_close, 0); |
| 1629 | atomic_set(&prtd->error, 0); |
| 1630 | |
| 1631 | init_waitqueue_head(&prtd->eos_wait); |
| 1632 | init_waitqueue_head(&prtd->drain_wait); |
| 1633 | init_waitqueue_head(&prtd->close_wait); |
| 1634 | init_waitqueue_head(&prtd->wait_for_stream_avail); |
| 1635 | |
| 1636 | runtime->private_data = prtd; |
| 1637 | populate_codec_list(prtd); |
| 1638 | prtd->audio_client = q6asm_audio_client_alloc( |
| 1639 | (app_cb)compr_event_handler, prtd); |
Aditya Bavanari | 9deef91 | 2017-11-20 13:31:31 +0530 | [diff] [blame] | 1640 | if (prtd->audio_client == NULL) { |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 1641 | pr_err("%s: Could not allocate memory for client\n", __func__); |
| 1642 | kfree(pdata->audio_effects[rtd->dai_link->id]); |
Aditya Bavanari | 9deef91 | 2017-11-20 13:31:31 +0530 | [diff] [blame] | 1643 | pdata->audio_effects[rtd->dai_link->id] = NULL; |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 1644 | kfree(pdata->dec_params[rtd->dai_link->id]); |
Aditya Bavanari | 9deef91 | 2017-11-20 13:31:31 +0530 | [diff] [blame] | 1645 | pdata->dec_params[rtd->dai_link->id] = NULL; |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 1646 | pdata->cstream[rtd->dai_link->id] = NULL; |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 1647 | kfree(prtd); |
Aditya Bavanari | 9deef91 | 2017-11-20 13:31:31 +0530 | [diff] [blame] | 1648 | runtime->private_data = NULL; |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 1649 | return -ENOMEM; |
| 1650 | } |
| 1651 | pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session); |
| 1652 | prtd->audio_client->perf_mode = false; |
| 1653 | prtd->session_id = prtd->audio_client->session; |
| 1654 | msm_adsp_init_mixer_ctl_pp_event_queue(rtd); |
Aditya Bavanari | 9deef91 | 2017-11-20 13:31:31 +0530 | [diff] [blame] | 1655 | pdata->is_in_use[rtd->dai_link->id] = true; |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 1656 | return 0; |
| 1657 | } |
| 1658 | |
| 1659 | static int msm_compr_capture_open(struct snd_compr_stream *cstream) |
| 1660 | { |
| 1661 | struct snd_compr_runtime *runtime = cstream->runtime; |
| 1662 | struct snd_soc_pcm_runtime *rtd = cstream->private_data; |
| 1663 | struct msm_compr_audio *prtd; |
| 1664 | struct msm_compr_pdata *pdata = |
| 1665 | snd_soc_platform_get_drvdata(rtd->platform); |
| 1666 | |
| 1667 | pr_debug("%s\n", __func__); |
| 1668 | prtd = kzalloc(sizeof(struct msm_compr_audio), GFP_KERNEL); |
| 1669 | if (prtd == NULL) { |
| 1670 | pr_err("Failed to allocate memory for msm_compr_audio\n"); |
| 1671 | return -ENOMEM; |
| 1672 | } |
| 1673 | |
| 1674 | runtime->private_data = NULL; |
| 1675 | prtd->cstream = cstream; |
| 1676 | pdata->cstream[rtd->dai_link->id] = cstream; |
| 1677 | |
| 1678 | prtd->audio_client = q6asm_audio_client_alloc( |
| 1679 | (app_cb)compr_event_handler, prtd); |
| 1680 | if (!prtd->audio_client) { |
| 1681 | pr_err("%s: Could not allocate memory for client\n", __func__); |
| 1682 | pdata->cstream[rtd->dai_link->id] = NULL; |
| 1683 | kfree(prtd); |
| 1684 | return -ENOMEM; |
| 1685 | } |
| 1686 | pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session); |
| 1687 | prtd->audio_client->perf_mode = false; |
| 1688 | prtd->session_id = prtd->audio_client->session; |
| 1689 | prtd->codec = FORMAT_LINEAR_PCM; |
| 1690 | prtd->bytes_copied = 0; |
| 1691 | prtd->bytes_read = 0; |
| 1692 | prtd->bytes_read_offset = 0; |
| 1693 | prtd->received_total = 0; |
| 1694 | prtd->byte_offset = 0; |
| 1695 | prtd->sample_rate = 48000; |
| 1696 | prtd->num_channels = 2; |
| 1697 | prtd->first_buffer = 0; |
| 1698 | |
| 1699 | spin_lock_init(&prtd->lock); |
| 1700 | |
| 1701 | atomic_set(&prtd->eos, 0); |
| 1702 | atomic_set(&prtd->start, 0); |
| 1703 | atomic_set(&prtd->drain, 0); |
| 1704 | atomic_set(&prtd->xrun, 0); |
| 1705 | atomic_set(&prtd->close, 0); |
| 1706 | atomic_set(&prtd->wait_on_close, 0); |
| 1707 | atomic_set(&prtd->error, 0); |
| 1708 | |
| 1709 | runtime->private_data = prtd; |
| 1710 | |
| 1711 | return 0; |
| 1712 | } |
| 1713 | |
| 1714 | static int msm_compr_open(struct snd_compr_stream *cstream) |
| 1715 | { |
| 1716 | int ret = 0; |
| 1717 | |
| 1718 | if (cstream->direction == SND_COMPRESS_PLAYBACK) |
| 1719 | ret = msm_compr_playback_open(cstream); |
| 1720 | else if (cstream->direction == SND_COMPRESS_CAPTURE) |
| 1721 | ret = msm_compr_capture_open(cstream); |
| 1722 | return ret; |
| 1723 | } |
| 1724 | |
| 1725 | static int msm_compr_playback_free(struct snd_compr_stream *cstream) |
| 1726 | { |
| 1727 | struct snd_compr_runtime *runtime; |
| 1728 | struct msm_compr_audio *prtd; |
| 1729 | struct snd_soc_pcm_runtime *soc_prtd; |
| 1730 | struct msm_compr_pdata *pdata; |
| 1731 | struct audio_client *ac; |
| 1732 | int dir = IN, ret = 0, stream_id; |
| 1733 | unsigned long flags; |
| 1734 | uint32_t stream_index; |
| 1735 | |
| 1736 | pr_debug("%s\n", __func__); |
| 1737 | |
| 1738 | if (!cstream) { |
| 1739 | pr_err("%s cstream is null\n", __func__); |
| 1740 | return 0; |
| 1741 | } |
| 1742 | runtime = cstream->runtime; |
| 1743 | soc_prtd = cstream->private_data; |
| 1744 | if (!runtime || !soc_prtd || !(soc_prtd->platform)) { |
| 1745 | pr_err("%s runtime or soc_prtd or platform is null\n", |
| 1746 | __func__); |
| 1747 | return 0; |
| 1748 | } |
| 1749 | prtd = runtime->private_data; |
| 1750 | if (!prtd) { |
| 1751 | pr_err("%s prtd is null\n", __func__); |
| 1752 | return 0; |
| 1753 | } |
| 1754 | prtd->cmd_interrupt = 1; |
| 1755 | wake_up(&prtd->drain_wait); |
| 1756 | pdata = snd_soc_platform_get_drvdata(soc_prtd->platform); |
| 1757 | ac = prtd->audio_client; |
| 1758 | if (!pdata || !ac) { |
| 1759 | pr_err("%s pdata or ac is null\n", __func__); |
| 1760 | return 0; |
| 1761 | } |
| 1762 | if (atomic_read(&prtd->eos)) { |
| 1763 | ret = wait_event_timeout(prtd->eos_wait, |
| 1764 | prtd->eos_ack, 5 * HZ); |
| 1765 | if (!ret) |
| 1766 | pr_err("%s: CMD_EOS failed\n", __func__); |
| 1767 | } |
| 1768 | if (atomic_read(&prtd->close)) { |
| 1769 | prtd->cmd_ack = 0; |
| 1770 | atomic_set(&prtd->wait_on_close, 1); |
| 1771 | ret = wait_event_timeout(prtd->close_wait, |
| 1772 | prtd->cmd_ack, 5 * HZ); |
| 1773 | if (!ret) |
| 1774 | pr_err("%s: CMD_CLOSE failed\n", __func__); |
| 1775 | } |
| 1776 | |
| 1777 | spin_lock_irqsave(&prtd->lock, flags); |
| 1778 | stream_id = ac->stream_id; |
| 1779 | stream_index = STREAM_ARRAY_INDEX(NEXT_STREAM_ID(stream_id)); |
| 1780 | |
| 1781 | if ((stream_index < MAX_NUMBER_OF_STREAMS && stream_index >= 0) && |
| 1782 | (prtd->gapless_state.stream_opened[stream_index])) { |
| 1783 | prtd->gapless_state.stream_opened[stream_index] = 0; |
| 1784 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 1785 | pr_debug(" close stream %d", NEXT_STREAM_ID(stream_id)); |
| 1786 | q6asm_stream_cmd(ac, CMD_CLOSE, NEXT_STREAM_ID(stream_id)); |
| 1787 | spin_lock_irqsave(&prtd->lock, flags); |
| 1788 | } |
| 1789 | |
| 1790 | stream_index = STREAM_ARRAY_INDEX(stream_id); |
| 1791 | if ((stream_index < MAX_NUMBER_OF_STREAMS && stream_index >= 0) && |
| 1792 | (prtd->gapless_state.stream_opened[stream_index])) { |
| 1793 | prtd->gapless_state.stream_opened[stream_index] = 0; |
| 1794 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 1795 | pr_debug("close stream %d", stream_id); |
| 1796 | q6asm_stream_cmd(ac, CMD_CLOSE, stream_id); |
| 1797 | spin_lock_irqsave(&prtd->lock, flags); |
| 1798 | } |
| 1799 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 1800 | |
| 1801 | pdata->cstream[soc_prtd->dai_link->id] = NULL; |
| 1802 | if (cstream->direction == SND_COMPRESS_PLAYBACK) { |
| 1803 | msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->id, |
| 1804 | SNDRV_PCM_STREAM_PLAYBACK); |
| 1805 | } |
| 1806 | |
| 1807 | q6asm_audio_client_buf_free_contiguous(dir, ac); |
| 1808 | |
| 1809 | q6asm_audio_client_free(ac); |
| 1810 | msm_adsp_clean_mixer_ctl_pp_event_queue(soc_prtd); |
Aditya Bavanari | 9deef91 | 2017-11-20 13:31:31 +0530 | [diff] [blame] | 1811 | if (pdata->audio_effects[soc_prtd->dai_link->id] != NULL) { |
| 1812 | kfree(pdata->audio_effects[soc_prtd->dai_link->id]); |
| 1813 | pdata->audio_effects[soc_prtd->dai_link->id] = NULL; |
| 1814 | } |
| 1815 | if (pdata->dec_params[soc_prtd->dai_link->id] != NULL) { |
| 1816 | kfree(pdata->dec_params[soc_prtd->dai_link->id]); |
| 1817 | pdata->dec_params[soc_prtd->dai_link->id] = NULL; |
| 1818 | } |
| 1819 | pdata->is_in_use[soc_prtd->dai_link->id] = false; |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 1820 | kfree(prtd); |
| 1821 | runtime->private_data = NULL; |
| 1822 | |
| 1823 | return 0; |
| 1824 | } |
| 1825 | |
| 1826 | static int msm_compr_capture_free(struct snd_compr_stream *cstream) |
| 1827 | { |
| 1828 | struct snd_compr_runtime *runtime; |
| 1829 | struct msm_compr_audio *prtd; |
| 1830 | struct snd_soc_pcm_runtime *soc_prtd; |
| 1831 | struct msm_compr_pdata *pdata; |
| 1832 | struct audio_client *ac; |
| 1833 | int dir = OUT, stream_id; |
| 1834 | unsigned long flags; |
| 1835 | uint32_t stream_index; |
| 1836 | |
| 1837 | if (!cstream) { |
| 1838 | pr_err("%s cstream is null\n", __func__); |
| 1839 | return 0; |
| 1840 | } |
| 1841 | runtime = cstream->runtime; |
| 1842 | soc_prtd = cstream->private_data; |
| 1843 | if (!runtime || !soc_prtd || !(soc_prtd->platform)) { |
| 1844 | pr_err("%s runtime or soc_prtd or platform is null\n", |
| 1845 | __func__); |
| 1846 | return 0; |
| 1847 | } |
| 1848 | prtd = runtime->private_data; |
| 1849 | if (!prtd) { |
| 1850 | pr_err("%s prtd is null\n", __func__); |
| 1851 | return 0; |
| 1852 | } |
| 1853 | pdata = snd_soc_platform_get_drvdata(soc_prtd->platform); |
| 1854 | ac = prtd->audio_client; |
| 1855 | if (!pdata || !ac) { |
| 1856 | pr_err("%s pdata or ac is null\n", __func__); |
| 1857 | return 0; |
| 1858 | } |
| 1859 | |
| 1860 | spin_lock_irqsave(&prtd->lock, flags); |
| 1861 | stream_id = ac->stream_id; |
| 1862 | |
| 1863 | stream_index = STREAM_ARRAY_INDEX(stream_id); |
| 1864 | if ((stream_index < MAX_NUMBER_OF_STREAMS && stream_index >= 0)) { |
| 1865 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 1866 | pr_debug("close stream %d", stream_id); |
| 1867 | q6asm_stream_cmd(ac, CMD_CLOSE, stream_id); |
| 1868 | spin_lock_irqsave(&prtd->lock, flags); |
| 1869 | } |
| 1870 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 1871 | |
| 1872 | pdata->cstream[soc_prtd->dai_link->id] = NULL; |
| 1873 | msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->id, |
| 1874 | SNDRV_PCM_STREAM_CAPTURE); |
| 1875 | |
| 1876 | q6asm_audio_client_buf_free_contiguous(dir, ac); |
| 1877 | |
| 1878 | q6asm_audio_client_free(ac); |
| 1879 | |
| 1880 | kfree(prtd); |
| 1881 | runtime->private_data = NULL; |
| 1882 | |
| 1883 | return 0; |
| 1884 | } |
| 1885 | |
| 1886 | static int msm_compr_free(struct snd_compr_stream *cstream) |
| 1887 | { |
| 1888 | int ret = 0; |
| 1889 | |
| 1890 | if (cstream->direction == SND_COMPRESS_PLAYBACK) |
| 1891 | ret = msm_compr_playback_free(cstream); |
| 1892 | else if (cstream->direction == SND_COMPRESS_CAPTURE) |
| 1893 | ret = msm_compr_capture_free(cstream); |
| 1894 | return ret; |
| 1895 | } |
| 1896 | |
| 1897 | static bool msm_compr_validate_codec_compr(__u32 codec_id) |
| 1898 | { |
| 1899 | int32_t i; |
| 1900 | |
| 1901 | for (i = 0; i < ARRAY_SIZE(compr_codecs); i++) { |
| 1902 | if (compr_codecs[i] == codec_id) |
| 1903 | return true; |
| 1904 | } |
| 1905 | return false; |
| 1906 | } |
| 1907 | |
| 1908 | /* compress stream operations */ |
| 1909 | static int msm_compr_set_params(struct snd_compr_stream *cstream, |
| 1910 | struct snd_compr_params *params) |
| 1911 | { |
| 1912 | struct snd_compr_runtime *runtime = cstream->runtime; |
| 1913 | struct msm_compr_audio *prtd = runtime->private_data; |
| 1914 | int ret = 0, frame_sz = 0; |
| 1915 | int i, num_rates; |
| 1916 | bool is_format_gapless = false; |
| 1917 | |
| 1918 | pr_debug("%s\n", __func__); |
| 1919 | |
| 1920 | num_rates = sizeof(supported_sample_rates)/sizeof(unsigned int); |
| 1921 | for (i = 0; i < num_rates; i++) |
| 1922 | if (params->codec.sample_rate == supported_sample_rates[i]) |
| 1923 | break; |
| 1924 | if (i == num_rates) |
| 1925 | return -EINVAL; |
| 1926 | |
| 1927 | memcpy(&prtd->codec_param, params, sizeof(struct snd_compr_params)); |
| 1928 | /* ToDo: remove duplicates */ |
| 1929 | prtd->num_channels = prtd->codec_param.codec.ch_in; |
| 1930 | prtd->sample_rate = prtd->codec_param.codec.sample_rate; |
| 1931 | pr_debug("%s: sample_rate %d\n", __func__, prtd->sample_rate); |
| 1932 | |
| 1933 | if ((prtd->codec_param.codec.compr_passthr >= LEGACY_PCM && |
| 1934 | prtd->codec_param. |
| 1935 | codec.compr_passthr <= COMPRESSED_PASSTHROUGH_DSD) || |
| 1936 | (prtd->codec_param. |
| 1937 | codec.compr_passthr == COMPRESSED_PASSTHROUGH_IEC61937)) |
| 1938 | prtd->compr_passthr = prtd->codec_param.codec.compr_passthr; |
| 1939 | else |
| 1940 | prtd->compr_passthr = LEGACY_PCM; |
| 1941 | pr_debug("%s: compr_passthr = %d", __func__, prtd->compr_passthr); |
| 1942 | if (prtd->compr_passthr != LEGACY_PCM) { |
| 1943 | pr_debug("%s: Reset gapless mode playback for compr_type[%d]\n", |
| 1944 | __func__, prtd->compr_passthr); |
| 1945 | prtd->gapless_state.use_dsp_gapless_mode = 0; |
| 1946 | if (!msm_compr_validate_codec_compr(params->codec.id)) { |
| 1947 | pr_err("%s codec not supported in passthrough,id =%d\n", |
| 1948 | __func__, params->codec.id); |
| 1949 | return -EINVAL; |
| 1950 | } |
| 1951 | } |
| 1952 | |
| 1953 | switch (params->codec.id) { |
| 1954 | case SND_AUDIOCODEC_PCM: { |
| 1955 | pr_debug("SND_AUDIOCODEC_PCM\n"); |
| 1956 | prtd->codec = FORMAT_LINEAR_PCM; |
| 1957 | is_format_gapless = true; |
| 1958 | break; |
| 1959 | } |
| 1960 | |
| 1961 | case SND_AUDIOCODEC_MP3: { |
| 1962 | pr_debug("SND_AUDIOCODEC_MP3\n"); |
| 1963 | prtd->codec = FORMAT_MP3; |
| 1964 | frame_sz = MP3_OUTPUT_FRAME_SZ; |
| 1965 | is_format_gapless = true; |
| 1966 | break; |
| 1967 | } |
| 1968 | |
| 1969 | case SND_AUDIOCODEC_AAC: { |
| 1970 | pr_debug("SND_AUDIOCODEC_AAC\n"); |
| 1971 | prtd->codec = FORMAT_MPEG4_AAC; |
| 1972 | frame_sz = AAC_OUTPUT_FRAME_SZ; |
| 1973 | is_format_gapless = true; |
| 1974 | break; |
| 1975 | } |
| 1976 | |
| 1977 | case SND_AUDIOCODEC_AC3: { |
| 1978 | pr_debug("SND_AUDIOCODEC_AC3\n"); |
| 1979 | prtd->codec = FORMAT_AC3; |
| 1980 | frame_sz = AC3_OUTPUT_FRAME_SZ; |
| 1981 | is_format_gapless = true; |
| 1982 | break; |
| 1983 | } |
| 1984 | |
| 1985 | case SND_AUDIOCODEC_EAC3: { |
| 1986 | pr_debug("SND_AUDIOCODEC_EAC3\n"); |
| 1987 | prtd->codec = FORMAT_EAC3; |
| 1988 | frame_sz = EAC3_OUTPUT_FRAME_SZ; |
| 1989 | is_format_gapless = true; |
| 1990 | break; |
| 1991 | } |
| 1992 | |
| 1993 | case SND_AUDIOCODEC_MP2: { |
| 1994 | pr_debug("SND_AUDIOCODEC_MP2\n"); |
| 1995 | prtd->codec = FORMAT_MP2; |
| 1996 | break; |
| 1997 | } |
| 1998 | |
| 1999 | case SND_AUDIOCODEC_WMA: { |
| 2000 | pr_debug("SND_AUDIOCODEC_WMA\n"); |
| 2001 | prtd->codec = FORMAT_WMA_V9; |
| 2002 | break; |
| 2003 | } |
| 2004 | |
| 2005 | case SND_AUDIOCODEC_WMA_PRO: { |
| 2006 | pr_debug("SND_AUDIOCODEC_WMA_PRO\n"); |
| 2007 | prtd->codec = FORMAT_WMA_V10PRO; |
| 2008 | break; |
| 2009 | } |
| 2010 | |
| 2011 | case SND_AUDIOCODEC_FLAC: { |
| 2012 | pr_debug("%s: SND_AUDIOCODEC_FLAC\n", __func__); |
| 2013 | prtd->codec = FORMAT_FLAC; |
| 2014 | /* |
| 2015 | * DSP bufferring is based on blk size, |
| 2016 | * consider mininum buffering to rule out any false wait |
| 2017 | */ |
| 2018 | frame_sz = |
| 2019 | prtd->codec_param.codec.options.flac_dec.min_blk_size; |
| 2020 | is_format_gapless = true; |
| 2021 | break; |
| 2022 | } |
| 2023 | |
| 2024 | case SND_AUDIOCODEC_VORBIS: { |
| 2025 | pr_debug("%s: SND_AUDIOCODEC_VORBIS\n", __func__); |
| 2026 | prtd->codec = FORMAT_VORBIS; |
| 2027 | break; |
| 2028 | } |
| 2029 | |
| 2030 | case SND_AUDIOCODEC_ALAC: { |
| 2031 | pr_debug("%s: SND_AUDIOCODEC_ALAC\n", __func__); |
| 2032 | prtd->codec = FORMAT_ALAC; |
| 2033 | break; |
| 2034 | } |
| 2035 | |
| 2036 | case SND_AUDIOCODEC_APE: { |
| 2037 | pr_debug("%s: SND_AUDIOCODEC_APE\n", __func__); |
| 2038 | prtd->codec = FORMAT_APE; |
| 2039 | break; |
| 2040 | } |
| 2041 | |
| 2042 | case SND_AUDIOCODEC_DTS: { |
| 2043 | pr_debug("%s: SND_AUDIOCODEC_DTS\n", __func__); |
| 2044 | prtd->codec = FORMAT_DTS; |
| 2045 | break; |
| 2046 | } |
| 2047 | |
| 2048 | case SND_AUDIOCODEC_DSD: { |
| 2049 | pr_debug("%s: SND_AUDIOCODEC_DSD\n", __func__); |
| 2050 | prtd->codec = FORMAT_DSD; |
| 2051 | break; |
| 2052 | } |
| 2053 | |
| 2054 | case SND_AUDIOCODEC_TRUEHD: { |
| 2055 | pr_debug("%s: SND_AUDIOCODEC_TRUEHD\n", __func__); |
| 2056 | prtd->codec = FORMAT_TRUEHD; |
| 2057 | break; |
| 2058 | } |
| 2059 | |
| 2060 | case SND_AUDIOCODEC_IEC61937: { |
| 2061 | pr_debug("%s: SND_AUDIOCODEC_IEC61937\n", __func__); |
| 2062 | prtd->codec = FORMAT_IEC61937; |
| 2063 | break; |
| 2064 | } |
| 2065 | |
| 2066 | case SND_AUDIOCODEC_APTX: { |
| 2067 | pr_debug("%s: SND_AUDIOCODEC_APTX\n", __func__); |
| 2068 | prtd->codec = FORMAT_APTX; |
| 2069 | break; |
| 2070 | } |
| 2071 | |
| 2072 | default: |
| 2073 | pr_err("codec not supported, id =%d\n", params->codec.id); |
| 2074 | return -EINVAL; |
| 2075 | } |
| 2076 | |
| 2077 | if (!is_format_gapless) |
| 2078 | prtd->gapless_state.use_dsp_gapless_mode = false; |
| 2079 | |
| 2080 | prtd->partial_drain_delay = |
| 2081 | msm_compr_get_partial_drain_delay(frame_sz, prtd->sample_rate); |
| 2082 | |
| 2083 | if (cstream->direction == SND_COMPRESS_PLAYBACK) |
| 2084 | ret = msm_compr_configure_dsp_for_playback(cstream); |
| 2085 | else if (cstream->direction == SND_COMPRESS_CAPTURE) |
| 2086 | ret = msm_compr_configure_dsp_for_capture(cstream); |
| 2087 | |
| 2088 | return ret; |
| 2089 | } |
| 2090 | |
| 2091 | static int msm_compr_drain_buffer(struct msm_compr_audio *prtd, |
| 2092 | unsigned long *flags) |
| 2093 | { |
| 2094 | int rc = 0; |
| 2095 | |
| 2096 | atomic_set(&prtd->drain, 1); |
| 2097 | prtd->drain_ready = 0; |
| 2098 | spin_unlock_irqrestore(&prtd->lock, *flags); |
| 2099 | pr_debug("%s: wait for buffer to be drained\n", __func__); |
| 2100 | rc = wait_event_interruptible(prtd->drain_wait, |
| 2101 | prtd->drain_ready || |
| 2102 | prtd->cmd_interrupt || |
| 2103 | atomic_read(&prtd->xrun) || |
| 2104 | atomic_read(&prtd->error)); |
| 2105 | pr_debug("%s: out of buffer drain wait with ret %d\n", __func__, rc); |
| 2106 | spin_lock_irqsave(&prtd->lock, *flags); |
| 2107 | if (prtd->cmd_interrupt) { |
| 2108 | pr_debug("%s: buffer drain interrupted by flush)\n", __func__); |
| 2109 | rc = -EINTR; |
| 2110 | prtd->cmd_interrupt = 0; |
| 2111 | } |
| 2112 | if (atomic_read(&prtd->error)) { |
| 2113 | pr_err("%s: Got RESET EVENTS notification, return\n", |
| 2114 | __func__); |
| 2115 | rc = -ENETRESET; |
| 2116 | } |
| 2117 | return rc; |
| 2118 | } |
| 2119 | |
| 2120 | static int msm_compr_wait_for_stream_avail(struct msm_compr_audio *prtd, |
| 2121 | unsigned long *flags) |
| 2122 | { |
| 2123 | int rc = 0; |
| 2124 | |
| 2125 | pr_debug("next session is already in opened state\n"); |
| 2126 | prtd->next_stream = 1; |
| 2127 | prtd->cmd_interrupt = 0; |
| 2128 | spin_unlock_irqrestore(&prtd->lock, *flags); |
| 2129 | /* |
| 2130 | * Wait for stream to be available, or the wait to be interrupted by |
| 2131 | * commands like flush or till a timeout of one second. |
| 2132 | */ |
| 2133 | rc = wait_event_timeout(prtd->wait_for_stream_avail, |
| 2134 | prtd->stream_available || prtd->cmd_interrupt, 1 * HZ); |
| 2135 | pr_err("%s:prtd->stream_available %d, prtd->cmd_interrupt %d rc %d\n", |
| 2136 | __func__, prtd->stream_available, prtd->cmd_interrupt, rc); |
| 2137 | |
| 2138 | spin_lock_irqsave(&prtd->lock, *flags); |
| 2139 | if (rc == 0) { |
| 2140 | pr_err("%s: wait_for_stream_avail timed out\n", |
| 2141 | __func__); |
| 2142 | rc = -ETIMEDOUT; |
| 2143 | } else if (prtd->cmd_interrupt == 1) { |
| 2144 | /* |
| 2145 | * This scenario might not happen as we do not allow |
| 2146 | * flush in transition state. |
| 2147 | */ |
| 2148 | pr_debug("%s: wait_for_stream_avail interrupted\n", __func__); |
| 2149 | prtd->cmd_interrupt = 0; |
| 2150 | prtd->stream_available = 0; |
| 2151 | rc = -EINTR; |
| 2152 | } else { |
| 2153 | prtd->stream_available = 0; |
| 2154 | rc = 0; |
| 2155 | } |
| 2156 | pr_debug("%s : rc = %d", __func__, rc); |
| 2157 | return rc; |
| 2158 | } |
| 2159 | |
| 2160 | static int msm_compr_trigger(struct snd_compr_stream *cstream, int cmd) |
| 2161 | { |
| 2162 | struct snd_compr_runtime *runtime = cstream->runtime; |
| 2163 | struct msm_compr_audio *prtd = runtime->private_data; |
| 2164 | struct snd_soc_pcm_runtime *rtd = cstream->private_data; |
| 2165 | struct msm_compr_pdata *pdata = |
| 2166 | snd_soc_platform_get_drvdata(rtd->platform); |
| 2167 | uint32_t *volume = pdata->volume[rtd->dai_link->id]; |
| 2168 | struct audio_client *ac = prtd->audio_client; |
| 2169 | unsigned long fe_id = rtd->dai_link->id; |
| 2170 | int rc = 0; |
| 2171 | int bytes_to_write; |
| 2172 | unsigned long flags; |
| 2173 | int stream_id; |
| 2174 | uint32_t stream_index; |
| 2175 | uint16_t bits_per_sample = 16; |
| 2176 | |
| 2177 | spin_lock_irqsave(&prtd->lock, flags); |
| 2178 | if (atomic_read(&prtd->error)) { |
| 2179 | pr_err("%s Got RESET EVENTS notification, return immediately", |
| 2180 | __func__); |
| 2181 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2182 | return 0; |
| 2183 | } |
| 2184 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2185 | |
| 2186 | switch (cmd) { |
| 2187 | case SNDRV_PCM_TRIGGER_START: |
| 2188 | pr_debug("%s: SNDRV_PCM_TRIGGER_START\n", __func__); |
| 2189 | atomic_set(&prtd->start, 1); |
| 2190 | |
| 2191 | /* |
| 2192 | * compr_set_volume and compr_init_pp_params |
| 2193 | * are used to configure ASM volume hence not |
| 2194 | * needed for compress passthrough playback. |
| 2195 | * |
| 2196 | * compress passthrough volume is controlled in |
| 2197 | * ADM by adm_send_compressed_device_mute() |
| 2198 | */ |
| 2199 | if (prtd->compr_passthr == LEGACY_PCM && |
| 2200 | cstream->direction == SND_COMPRESS_PLAYBACK) { |
| 2201 | /* set volume for the stream before RUN */ |
| 2202 | rc = msm_compr_set_volume(cstream, |
| 2203 | volume[0], volume[1]); |
| 2204 | if (rc) |
| 2205 | pr_err("%s : Set Volume failed : %d\n", |
| 2206 | __func__, rc); |
| 2207 | |
| 2208 | rc = msm_compr_init_pp_params(cstream, ac); |
| 2209 | if (rc) |
| 2210 | pr_err("%s : init PP params failed : %d\n", |
| 2211 | __func__, rc); |
| 2212 | } else { |
| 2213 | msm_compr_read_buffer(prtd); |
| 2214 | } |
| 2215 | /* issue RUN command for the stream */ |
| 2216 | q6asm_run_nowait(prtd->audio_client, prtd->run_mode, |
| 2217 | prtd->start_delay_msw, prtd->start_delay_lsw); |
| 2218 | break; |
| 2219 | case SNDRV_PCM_TRIGGER_STOP: |
| 2220 | spin_lock_irqsave(&prtd->lock, flags); |
| 2221 | pr_debug("%s: SNDRV_PCM_TRIGGER_STOP transition %d\n", __func__, |
| 2222 | prtd->gapless_state.gapless_transition); |
| 2223 | stream_id = ac->stream_id; |
| 2224 | atomic_set(&prtd->start, 0); |
| 2225 | if (cstream->direction == SND_COMPRESS_CAPTURE) { |
| 2226 | q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE); |
| 2227 | atomic_set(&prtd->xrun, 0); |
| 2228 | prtd->received_total = 0; |
| 2229 | prtd->bytes_copied = 0; |
| 2230 | prtd->bytes_read = 0; |
| 2231 | prtd->bytes_read_offset = 0; |
| 2232 | prtd->byte_offset = 0; |
| 2233 | prtd->app_pointer = 0; |
| 2234 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2235 | break; |
| 2236 | } |
| 2237 | if (prtd->next_stream) { |
| 2238 | pr_debug("%s: interrupt next track wait queues\n", |
| 2239 | __func__); |
| 2240 | prtd->cmd_interrupt = 1; |
| 2241 | wake_up(&prtd->wait_for_stream_avail); |
| 2242 | prtd->next_stream = 0; |
| 2243 | } |
| 2244 | if (atomic_read(&prtd->eos)) { |
| 2245 | pr_debug("%s: interrupt eos wait queues", __func__); |
| 2246 | /* |
| 2247 | * Gapless playback does not wait for eos, do not set |
| 2248 | * cmd_int and do not wake up eos_wait during gapless |
| 2249 | * transition |
| 2250 | */ |
| 2251 | if (!prtd->gapless_state.gapless_transition) { |
| 2252 | prtd->cmd_interrupt = 1; |
| 2253 | wake_up(&prtd->eos_wait); |
| 2254 | } |
| 2255 | atomic_set(&prtd->eos, 0); |
| 2256 | } |
| 2257 | if (atomic_read(&prtd->drain)) { |
| 2258 | pr_debug("%s: interrupt drain wait queues", __func__); |
| 2259 | prtd->cmd_interrupt = 1; |
| 2260 | prtd->drain_ready = 1; |
| 2261 | wake_up(&prtd->drain_wait); |
| 2262 | atomic_set(&prtd->drain, 0); |
| 2263 | } |
| 2264 | prtd->last_buffer = 0; |
| 2265 | prtd->cmd_ack = 0; |
| 2266 | if (!prtd->gapless_state.gapless_transition) { |
| 2267 | pr_debug("issue CMD_FLUSH stream_id %d\n", stream_id); |
| 2268 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2269 | q6asm_stream_cmd( |
| 2270 | prtd->audio_client, CMD_FLUSH, stream_id); |
| 2271 | spin_lock_irqsave(&prtd->lock, flags); |
| 2272 | } else { |
| 2273 | prtd->first_buffer = 0; |
| 2274 | } |
| 2275 | /* FIXME. only reset if flush was successful */ |
| 2276 | prtd->byte_offset = 0; |
| 2277 | prtd->copied_total = 0; |
| 2278 | prtd->app_pointer = 0; |
| 2279 | prtd->bytes_received = 0; |
| 2280 | prtd->bytes_sent = 0; |
| 2281 | prtd->marker_timestamp = 0; |
| 2282 | |
| 2283 | atomic_set(&prtd->xrun, 0); |
| 2284 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2285 | break; |
| 2286 | case SNDRV_PCM_TRIGGER_PAUSE_PUSH: |
| 2287 | pr_debug("SNDRV_PCM_TRIGGER_PAUSE_PUSH transition %d\n", |
| 2288 | prtd->gapless_state.gapless_transition); |
| 2289 | if (!prtd->gapless_state.gapless_transition) { |
| 2290 | pr_debug("issue CMD_PAUSE stream_id %d\n", |
| 2291 | ac->stream_id); |
| 2292 | q6asm_stream_cmd_nowait(ac, CMD_PAUSE, ac->stream_id); |
| 2293 | atomic_set(&prtd->start, 0); |
| 2294 | } |
| 2295 | break; |
| 2296 | case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: |
| 2297 | pr_debug("SNDRV_PCM_TRIGGER_PAUSE_RELEASE transition %d\n", |
| 2298 | prtd->gapless_state.gapless_transition); |
| 2299 | if (!prtd->gapless_state.gapless_transition) { |
| 2300 | atomic_set(&prtd->start, 1); |
| 2301 | q6asm_run_nowait(prtd->audio_client, prtd->run_mode, |
| 2302 | 0, 0); |
| 2303 | } |
| 2304 | break; |
| 2305 | case SND_COMPR_TRIGGER_PARTIAL_DRAIN: |
| 2306 | pr_debug("%s: SND_COMPR_TRIGGER_PARTIAL_DRAIN\n", __func__); |
| 2307 | if (!prtd->gapless_state.use_dsp_gapless_mode) { |
| 2308 | pr_debug("%s: set partial drain as drain\n", __func__); |
| 2309 | cmd = SND_COMPR_TRIGGER_DRAIN; |
| 2310 | } |
| 2311 | case SND_COMPR_TRIGGER_DRAIN: |
| 2312 | pr_debug("%s: SNDRV_COMPRESS_DRAIN\n", __func__); |
| 2313 | /* Make sure all the data is sent to DSP before sending EOS */ |
| 2314 | spin_lock_irqsave(&prtd->lock, flags); |
| 2315 | |
| 2316 | if (!atomic_read(&prtd->start)) { |
| 2317 | pr_err("%s: stream is not in started state\n", |
| 2318 | __func__); |
| 2319 | rc = -EPERM; |
| 2320 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2321 | break; |
| 2322 | } |
| 2323 | if (prtd->bytes_received > prtd->copied_total) { |
| 2324 | pr_debug("%s: wait till all the data is sent to dsp\n", |
| 2325 | __func__); |
| 2326 | rc = msm_compr_drain_buffer(prtd, &flags); |
| 2327 | if (rc || !atomic_read(&prtd->start)) { |
| 2328 | if (rc != -ENETRESET) |
| 2329 | rc = -EINTR; |
| 2330 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2331 | break; |
| 2332 | } |
| 2333 | /* |
| 2334 | * FIXME: Bug. |
| 2335 | * Write(32767) |
| 2336 | * Start |
| 2337 | * Drain <- Indefinite wait |
| 2338 | * sol1 : if (prtd->copied_total) then wait? |
| 2339 | * sol2 : (prtd->cmd_interrupt || prtd->drain_ready || |
| 2340 | * atomic_read(xrun) |
| 2341 | */ |
| 2342 | bytes_to_write = prtd->bytes_received |
| 2343 | - prtd->copied_total; |
| 2344 | WARN(bytes_to_write > runtime->fragment_size, |
| 2345 | "last write %d cannot be > than fragment_size", |
| 2346 | bytes_to_write); |
| 2347 | |
| 2348 | if (bytes_to_write > 0) { |
| 2349 | pr_debug("%s: send %d partial bytes at the end", |
| 2350 | __func__, bytes_to_write); |
| 2351 | atomic_set(&prtd->xrun, 0); |
| 2352 | prtd->last_buffer = 1; |
| 2353 | msm_compr_send_buffer(prtd); |
| 2354 | } |
| 2355 | } |
| 2356 | |
| 2357 | if ((cmd == SND_COMPR_TRIGGER_PARTIAL_DRAIN) && |
| 2358 | (prtd->gapless_state.set_next_stream_id)) { |
| 2359 | /* wait for the last buffer to be returned */ |
| 2360 | |
| 2361 | if (prtd->last_buffer) { |
| 2362 | pr_debug("%s: last buffer drain\n", __func__); |
| 2363 | rc = msm_compr_drain_buffer(prtd, &flags); |
| 2364 | if (rc || !atomic_read(&prtd->start)) { |
| 2365 | spin_unlock_irqrestore(&prtd->lock, |
| 2366 | flags); |
| 2367 | break; |
| 2368 | } |
| 2369 | } |
| 2370 | /* send EOS */ |
| 2371 | prtd->eos_ack = 0; |
| 2372 | atomic_set(&prtd->eos, 1); |
| 2373 | pr_debug("issue CMD_EOS stream_id %d\n", ac->stream_id); |
| 2374 | q6asm_stream_cmd_nowait(ac, CMD_EOS, ac->stream_id); |
| 2375 | pr_info("PARTIAL DRAIN, do not wait for EOS ack\n"); |
| 2376 | |
| 2377 | /* send a zero length buffer */ |
| 2378 | atomic_set(&prtd->xrun, 0); |
| 2379 | msm_compr_send_buffer(prtd); |
| 2380 | |
| 2381 | /* wait for the zero length buffer to be returned */ |
| 2382 | pr_debug("%s: zero length buffer drain\n", __func__); |
| 2383 | rc = msm_compr_drain_buffer(prtd, &flags); |
| 2384 | if (rc || !atomic_read(&prtd->start)) { |
| 2385 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2386 | break; |
| 2387 | } |
| 2388 | |
| 2389 | /* sleep for additional duration partial drain */ |
| 2390 | atomic_set(&prtd->drain, 1); |
| 2391 | prtd->drain_ready = 0; |
| 2392 | pr_debug("%s, additional sleep: %d\n", __func__, |
| 2393 | prtd->partial_drain_delay); |
| 2394 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2395 | rc = wait_event_timeout(prtd->drain_wait, |
| 2396 | prtd->drain_ready || prtd->cmd_interrupt, |
| 2397 | msecs_to_jiffies(prtd->partial_drain_delay)); |
| 2398 | pr_debug("%s: out of additional wait for low sample rate\n", |
| 2399 | __func__); |
| 2400 | spin_lock_irqsave(&prtd->lock, flags); |
| 2401 | if (prtd->cmd_interrupt) { |
| 2402 | pr_debug("%s: additional wait interrupted by flush)\n", |
| 2403 | __func__); |
| 2404 | rc = -EINTR; |
| 2405 | prtd->cmd_interrupt = 0; |
| 2406 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2407 | break; |
| 2408 | } |
| 2409 | |
| 2410 | /* move to next stream and reset vars */ |
| 2411 | pr_debug("%s: Moving to next stream in gapless\n", |
| 2412 | __func__); |
| 2413 | ac->stream_id = NEXT_STREAM_ID(ac->stream_id); |
| 2414 | prtd->byte_offset = 0; |
| 2415 | prtd->app_pointer = 0; |
| 2416 | prtd->first_buffer = 1; |
| 2417 | prtd->last_buffer = 0; |
| 2418 | /* |
| 2419 | * Set gapless transition flag only if EOS hasn't been |
| 2420 | * acknowledged already. |
| 2421 | */ |
| 2422 | if (atomic_read(&prtd->eos)) |
| 2423 | prtd->gapless_state.gapless_transition = 1; |
| 2424 | prtd->marker_timestamp = 0; |
| 2425 | |
| 2426 | /* |
| 2427 | * Don't reset these as these vars map to |
| 2428 | * total_bytes_transferred and total_bytes_available |
| 2429 | * directly, only total_bytes_transferred will be |
| 2430 | * updated in the next avail() ioctl |
| 2431 | * prtd->copied_total = 0; |
| 2432 | * prtd->bytes_received = 0; |
| 2433 | */ |
| 2434 | atomic_set(&prtd->drain, 0); |
| 2435 | atomic_set(&prtd->xrun, 1); |
| 2436 | pr_debug("%s: issue CMD_RUN", __func__); |
| 2437 | q6asm_run_nowait(prtd->audio_client, 0, 0, 0); |
| 2438 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2439 | break; |
| 2440 | } |
| 2441 | /* |
| 2442 | * moving to next stream failed, so reset the gapless state |
| 2443 | * set next stream id for the same session so that the same |
| 2444 | * stream can be used for gapless playback |
| 2445 | */ |
| 2446 | prtd->gapless_state.set_next_stream_id = false; |
| 2447 | prtd->gapless_state.gapless_transition = 0; |
| 2448 | pr_debug("%s:CMD_EOS stream_id %d\n", __func__, ac->stream_id); |
| 2449 | |
| 2450 | prtd->eos_ack = 0; |
| 2451 | atomic_set(&prtd->eos, 1); |
| 2452 | q6asm_stream_cmd_nowait(ac, CMD_EOS, ac->stream_id); |
| 2453 | |
| 2454 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2455 | |
| 2456 | |
| 2457 | /* Wait indefinitely for DRAIN. Flush can also signal this*/ |
| 2458 | rc = wait_event_interruptible(prtd->eos_wait, |
| 2459 | (prtd->eos_ack || |
| 2460 | prtd->cmd_interrupt || |
| 2461 | atomic_read(&prtd->error))); |
| 2462 | |
| 2463 | if (rc < 0) |
| 2464 | pr_err("%s: EOS wait failed\n", __func__); |
| 2465 | |
| 2466 | pr_debug("%s: SNDRV_COMPRESS_DRAIN out of wait for EOS\n", |
| 2467 | __func__); |
| 2468 | |
| 2469 | if (prtd->cmd_interrupt) |
| 2470 | rc = -EINTR; |
| 2471 | |
| 2472 | if (atomic_read(&prtd->error)) { |
| 2473 | pr_err("%s: Got RESET EVENTS notification, return\n", |
| 2474 | __func__); |
| 2475 | rc = -ENETRESET; |
| 2476 | } |
| 2477 | |
| 2478 | /*FIXME : what if a flush comes while PC is here */ |
| 2479 | if (rc == 0) { |
| 2480 | /* |
| 2481 | * Failed to open second stream in DSP for gapless |
| 2482 | * so prepare the current stream in session |
| 2483 | * for gapless playback |
| 2484 | */ |
| 2485 | spin_lock_irqsave(&prtd->lock, flags); |
| 2486 | pr_debug("%s:issue CMD_PAUSE stream_id %d", |
| 2487 | __func__, ac->stream_id); |
| 2488 | q6asm_stream_cmd_nowait(ac, CMD_PAUSE, ac->stream_id); |
| 2489 | prtd->cmd_ack = 0; |
| 2490 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2491 | |
| 2492 | /* |
| 2493 | * Cache this time as last known time |
| 2494 | */ |
| 2495 | if (pdata->use_legacy_api) |
| 2496 | q6asm_get_session_time_legacy( |
| 2497 | prtd->audio_client, |
| 2498 | &prtd->marker_timestamp); |
| 2499 | else |
| 2500 | q6asm_get_session_time(prtd->audio_client, |
| 2501 | &prtd->marker_timestamp); |
| 2502 | |
| 2503 | spin_lock_irqsave(&prtd->lock, flags); |
| 2504 | /* |
| 2505 | * Don't reset these as these vars map to |
| 2506 | * total_bytes_transferred and total_bytes_available. |
| 2507 | * Just total_bytes_transferred will be updated |
| 2508 | * in the next avail() ioctl. |
| 2509 | * prtd->copied_total = 0; |
| 2510 | * prtd->bytes_received = 0; |
| 2511 | * do not reset prtd->bytes_sent as well as the same |
| 2512 | * session is used for gapless playback |
| 2513 | */ |
| 2514 | prtd->byte_offset = 0; |
| 2515 | |
| 2516 | prtd->app_pointer = 0; |
| 2517 | prtd->first_buffer = 1; |
| 2518 | prtd->last_buffer = 0; |
| 2519 | atomic_set(&prtd->drain, 0); |
| 2520 | atomic_set(&prtd->xrun, 1); |
| 2521 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2522 | |
| 2523 | pr_debug("%s:issue CMD_FLUSH ac->stream_id %d", |
| 2524 | __func__, ac->stream_id); |
| 2525 | q6asm_stream_cmd(ac, CMD_FLUSH, ac->stream_id); |
| 2526 | |
| 2527 | q6asm_run_nowait(prtd->audio_client, 0, 0, 0); |
| 2528 | } |
| 2529 | prtd->cmd_interrupt = 0; |
| 2530 | break; |
| 2531 | case SND_COMPR_TRIGGER_NEXT_TRACK: |
| 2532 | if (!prtd->gapless_state.use_dsp_gapless_mode) { |
| 2533 | pr_debug("%s: ignore trigger next track\n", __func__); |
| 2534 | rc = 0; |
| 2535 | break; |
| 2536 | } |
| 2537 | pr_debug("%s: SND_COMPR_TRIGGER_NEXT_TRACK\n", __func__); |
| 2538 | spin_lock_irqsave(&prtd->lock, flags); |
| 2539 | rc = 0; |
| 2540 | /* next stream in gapless */ |
| 2541 | stream_id = NEXT_STREAM_ID(ac->stream_id); |
| 2542 | /* |
| 2543 | * Wait if stream 1 has not completed before honoring next |
| 2544 | * track for stream 3. Scenario happens if second clip is |
| 2545 | * small and fills in one buffer so next track will be |
| 2546 | * called immediately. |
| 2547 | */ |
| 2548 | stream_index = STREAM_ARRAY_INDEX(stream_id); |
| 2549 | if (stream_index >= MAX_NUMBER_OF_STREAMS || |
| 2550 | stream_index < 0) { |
| 2551 | pr_err("%s: Invalid stream index: %d", __func__, |
| 2552 | stream_index); |
| 2553 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2554 | rc = -EINVAL; |
| 2555 | break; |
| 2556 | } |
| 2557 | |
| 2558 | if (prtd->gapless_state.stream_opened[stream_index]) { |
| 2559 | if (prtd->gapless_state.gapless_transition) { |
| 2560 | rc = msm_compr_wait_for_stream_avail(prtd, |
| 2561 | &flags); |
| 2562 | } else { |
| 2563 | /* |
| 2564 | * If session is already opened break out if |
| 2565 | * the state is not gapless transition. This |
| 2566 | * is when seek happens after the last buffer |
| 2567 | * is sent to the driver. Next track would be |
| 2568 | * called again after last buffer is sent. |
| 2569 | */ |
| 2570 | pr_debug("next session is in opened state\n"); |
| 2571 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2572 | break; |
| 2573 | } |
| 2574 | } |
| 2575 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2576 | if (rc < 0) { |
| 2577 | /* |
| 2578 | * if return type EINTR then reset to zero. Tiny |
| 2579 | * compress treats EINTR as error and prevents PARTIAL |
| 2580 | * DRAIN. EINTR is not an error. wait for stream avail |
| 2581 | * is interrupted by some other command like FLUSH. |
| 2582 | */ |
| 2583 | if (rc == -EINTR) { |
| 2584 | pr_debug("%s: EINTR reset rc to 0\n", __func__); |
| 2585 | rc = 0; |
| 2586 | } |
| 2587 | break; |
| 2588 | } |
| 2589 | |
| 2590 | if (prtd->codec_param.codec.format == SNDRV_PCM_FORMAT_S24_LE) |
| 2591 | bits_per_sample = 24; |
| 2592 | else if (prtd->codec_param.codec.format == |
| 2593 | SNDRV_PCM_FORMAT_S32_LE) |
| 2594 | bits_per_sample = 32; |
| 2595 | |
| 2596 | pr_debug("%s: open_write stream_id %d bits_per_sample %d", |
| 2597 | __func__, stream_id, bits_per_sample); |
| 2598 | rc = q6asm_stream_open_write_v4(prtd->audio_client, |
| 2599 | prtd->codec, bits_per_sample, |
| 2600 | stream_id, |
| 2601 | prtd->gapless_state.use_dsp_gapless_mode); |
| 2602 | if (rc < 0) { |
| 2603 | pr_err("%s: Session out open failed for gapless\n", |
| 2604 | __func__); |
| 2605 | break; |
| 2606 | } |
| 2607 | |
| 2608 | spin_lock_irqsave(&prtd->lock, flags); |
| 2609 | prtd->gapless_state.stream_opened[stream_index] = 1; |
| 2610 | prtd->gapless_state.set_next_stream_id = true; |
| 2611 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2612 | |
| 2613 | rc = msm_compr_send_media_format_block(cstream, |
| 2614 | stream_id, false); |
| 2615 | if (rc < 0) { |
| 2616 | pr_err("%s, failed to send media format block\n", |
| 2617 | __func__); |
| 2618 | break; |
| 2619 | } |
| 2620 | msm_compr_send_dec_params(cstream, pdata->dec_params[fe_id], |
| 2621 | stream_id); |
| 2622 | break; |
| 2623 | } |
| 2624 | |
| 2625 | return rc; |
| 2626 | } |
| 2627 | |
| 2628 | static int msm_compr_pointer(struct snd_compr_stream *cstream, |
| 2629 | struct snd_compr_tstamp *arg) |
| 2630 | { |
| 2631 | struct snd_compr_runtime *runtime = cstream->runtime; |
| 2632 | struct snd_soc_pcm_runtime *rtd = cstream->private_data; |
| 2633 | struct msm_compr_audio *prtd = runtime->private_data; |
| 2634 | struct msm_compr_pdata *pdata = NULL; |
| 2635 | struct snd_compr_tstamp tstamp; |
| 2636 | uint64_t timestamp = 0; |
| 2637 | int rc = 0, first_buffer; |
| 2638 | unsigned long flags; |
| 2639 | uint32_t gapless_transition; |
| 2640 | |
| 2641 | pdata = snd_soc_platform_get_drvdata(rtd->platform); |
| 2642 | pr_debug("%s\n", __func__); |
| 2643 | memset(&tstamp, 0x0, sizeof(struct snd_compr_tstamp)); |
| 2644 | |
| 2645 | spin_lock_irqsave(&prtd->lock, flags); |
| 2646 | tstamp.sampling_rate = prtd->sample_rate; |
| 2647 | tstamp.byte_offset = prtd->byte_offset; |
| 2648 | if (cstream->direction == SND_COMPRESS_PLAYBACK) |
| 2649 | tstamp.copied_total = prtd->copied_total; |
| 2650 | else if (cstream->direction == SND_COMPRESS_CAPTURE) |
| 2651 | tstamp.copied_total = prtd->received_total; |
| 2652 | first_buffer = prtd->first_buffer; |
| 2653 | if (atomic_read(&prtd->error)) { |
Vatsal Bucha | 0527c56 | 2017-10-04 20:38:49 +0530 | [diff] [blame] | 2654 | pr_err_ratelimited("%s Got RESET EVENTS notification, return error\n", |
| 2655 | __func__); |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 2656 | if (cstream->direction == SND_COMPRESS_PLAYBACK) |
| 2657 | runtime->total_bytes_transferred = tstamp.copied_total; |
| 2658 | else |
| 2659 | runtime->total_bytes_available = tstamp.copied_total; |
| 2660 | tstamp.pcm_io_frames = 0; |
| 2661 | memcpy(arg, &tstamp, sizeof(struct snd_compr_tstamp)); |
| 2662 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2663 | return -ENETRESET; |
| 2664 | } |
| 2665 | if (cstream->direction == SND_COMPRESS_PLAYBACK) { |
| 2666 | |
| 2667 | gapless_transition = prtd->gapless_state.gapless_transition; |
| 2668 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2669 | if (gapless_transition) |
| 2670 | pr_debug("%s session time in gapless transition", |
| 2671 | __func__); |
| 2672 | /* |
| 2673 | *- Do not query if no buffer has been given. |
| 2674 | *- Do not query on a gapless transition. |
| 2675 | * Playback for the 2nd stream can start (thus returning time |
| 2676 | * starting from 0) before the driver knows about EOS of first |
| 2677 | * stream. |
| 2678 | */ |
| 2679 | if (!first_buffer || gapless_transition) { |
| 2680 | |
| 2681 | if (pdata->use_legacy_api) |
| 2682 | rc = q6asm_get_session_time_legacy( |
| 2683 | prtd->audio_client, &prtd->marker_timestamp); |
| 2684 | else |
| 2685 | rc = q6asm_get_session_time( |
| 2686 | prtd->audio_client, &prtd->marker_timestamp); |
| 2687 | if (rc < 0) { |
| 2688 | pr_err("%s: Get Session Time return =%lld\n", |
| 2689 | __func__, timestamp); |
| 2690 | if (atomic_read(&prtd->error)) |
| 2691 | return -ENETRESET; |
| 2692 | else |
| 2693 | return -EAGAIN; |
| 2694 | } |
| 2695 | } |
| 2696 | } else { |
| 2697 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2698 | } |
| 2699 | timestamp = prtd->marker_timestamp; |
| 2700 | |
| 2701 | /* DSP returns timestamp in usec */ |
| 2702 | pr_debug("%s: timestamp = %lld usec\n", __func__, timestamp); |
| 2703 | timestamp *= prtd->sample_rate; |
| 2704 | tstamp.pcm_io_frames = (snd_pcm_uframes_t)div64_u64(timestamp, 1000000); |
| 2705 | memcpy(arg, &tstamp, sizeof(struct snd_compr_tstamp)); |
| 2706 | |
| 2707 | return 0; |
| 2708 | } |
| 2709 | |
| 2710 | static int msm_compr_ack(struct snd_compr_stream *cstream, |
| 2711 | size_t count) |
| 2712 | { |
| 2713 | struct snd_compr_runtime *runtime = cstream->runtime; |
| 2714 | struct msm_compr_audio *prtd = runtime->private_data; |
| 2715 | void *src, *dstn; |
| 2716 | size_t copy; |
| 2717 | unsigned long flags; |
| 2718 | |
| 2719 | WARN(1, "This path is untested"); |
| 2720 | return -EINVAL; |
| 2721 | |
| 2722 | pr_debug("%s: count = %zd\n", __func__, count); |
| 2723 | if (!prtd->buffer) { |
| 2724 | pr_err("%s: Buffer is not allocated yet ??\n", __func__); |
| 2725 | return -EINVAL; |
| 2726 | } |
| 2727 | src = runtime->buffer + prtd->app_pointer; |
| 2728 | dstn = prtd->buffer + prtd->app_pointer; |
| 2729 | if (count < prtd->buffer_size - prtd->app_pointer) { |
| 2730 | memcpy(dstn, src, count); |
| 2731 | prtd->app_pointer += count; |
| 2732 | } else { |
| 2733 | copy = prtd->buffer_size - prtd->app_pointer; |
| 2734 | memcpy(dstn, src, copy); |
| 2735 | memcpy(prtd->buffer, runtime->buffer, count - copy); |
| 2736 | prtd->app_pointer = count - copy; |
| 2737 | } |
| 2738 | |
| 2739 | /* |
| 2740 | * If the stream is started and all the bytes received were |
| 2741 | * copied to DSP, the newly received bytes should be |
| 2742 | * sent right away |
| 2743 | */ |
| 2744 | spin_lock_irqsave(&prtd->lock, flags); |
| 2745 | |
| 2746 | if (atomic_read(&prtd->start) && |
| 2747 | prtd->bytes_received == prtd->copied_total) { |
| 2748 | prtd->bytes_received += count; |
| 2749 | msm_compr_send_buffer(prtd); |
| 2750 | } else |
| 2751 | prtd->bytes_received += count; |
| 2752 | |
| 2753 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2754 | |
| 2755 | return 0; |
| 2756 | } |
| 2757 | |
| 2758 | static int msm_compr_playback_copy(struct snd_compr_stream *cstream, |
| 2759 | char __user *buf, size_t count) |
| 2760 | { |
| 2761 | struct snd_compr_runtime *runtime = cstream->runtime; |
| 2762 | struct msm_compr_audio *prtd = runtime->private_data; |
| 2763 | void *dstn; |
| 2764 | size_t copy; |
| 2765 | uint64_t bytes_available = 0; |
| 2766 | unsigned long flags; |
| 2767 | |
| 2768 | pr_debug("%s: count = %zd\n", __func__, count); |
| 2769 | if (!prtd->buffer) { |
| 2770 | pr_err("%s: Buffer is not allocated yet ??", __func__); |
| 2771 | return 0; |
| 2772 | } |
| 2773 | |
| 2774 | spin_lock_irqsave(&prtd->lock, flags); |
| 2775 | if (atomic_read(&prtd->error)) { |
| 2776 | pr_err("%s Got RESET EVENTS notification", __func__); |
| 2777 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2778 | return -ENETRESET; |
| 2779 | } |
| 2780 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2781 | |
| 2782 | dstn = prtd->buffer + prtd->app_pointer; |
| 2783 | if (count < prtd->buffer_size - prtd->app_pointer) { |
| 2784 | if (copy_from_user(dstn, buf, count)) |
| 2785 | return -EFAULT; |
| 2786 | prtd->app_pointer += count; |
| 2787 | } else { |
| 2788 | copy = prtd->buffer_size - prtd->app_pointer; |
| 2789 | if (copy_from_user(dstn, buf, copy)) |
| 2790 | return -EFAULT; |
| 2791 | if (copy_from_user(prtd->buffer, buf + copy, count - copy)) |
| 2792 | return -EFAULT; |
| 2793 | prtd->app_pointer = count - copy; |
| 2794 | } |
| 2795 | |
| 2796 | /* |
| 2797 | * If stream is started and there has been an xrun, |
| 2798 | * since the available bytes fits fragment_size, copy the data |
| 2799 | * right away. |
| 2800 | */ |
| 2801 | spin_lock_irqsave(&prtd->lock, flags); |
| 2802 | prtd->bytes_received += count; |
| 2803 | if (atomic_read(&prtd->start)) { |
| 2804 | if (atomic_read(&prtd->xrun)) { |
| 2805 | pr_debug("%s: in xrun, count = %zd\n", __func__, count); |
| 2806 | bytes_available = prtd->bytes_received - |
| 2807 | prtd->copied_total; |
| 2808 | if (bytes_available >= runtime->fragment_size) { |
| 2809 | pr_debug("%s: handle xrun, bytes_to_write = %llu\n", |
| 2810 | __func__, bytes_available); |
| 2811 | atomic_set(&prtd->xrun, 0); |
| 2812 | msm_compr_send_buffer(prtd); |
| 2813 | } /* else not sufficient data */ |
| 2814 | } /* writes will continue on the next write_done */ |
| 2815 | } |
| 2816 | |
| 2817 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2818 | |
| 2819 | return count; |
| 2820 | } |
| 2821 | |
| 2822 | static int msm_compr_capture_copy(struct snd_compr_stream *cstream, |
| 2823 | char __user *buf, size_t count) |
| 2824 | { |
| 2825 | struct snd_compr_runtime *runtime = cstream->runtime; |
| 2826 | struct msm_compr_audio *prtd = runtime->private_data; |
| 2827 | void *source; |
| 2828 | unsigned long flags; |
| 2829 | |
| 2830 | pr_debug("%s: count = %zd\n", __func__, count); |
| 2831 | if (!prtd->buffer) { |
| 2832 | pr_err("%s: Buffer is not allocated yet ??", __func__); |
| 2833 | return 0; |
| 2834 | } |
| 2835 | |
| 2836 | spin_lock_irqsave(&prtd->lock, flags); |
| 2837 | if (atomic_read(&prtd->error)) { |
| 2838 | pr_err("%s Got RESET EVENTS notification", __func__); |
| 2839 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2840 | return -ENETRESET; |
| 2841 | } |
| 2842 | |
| 2843 | source = prtd->buffer + prtd->app_pointer; |
| 2844 | /* check if we have requested amount of data to copy to user*/ |
| 2845 | if (count <= prtd->received_total - prtd->bytes_copied) { |
| 2846 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2847 | if (copy_to_user(buf, source, count)) { |
| 2848 | pr_err("copy_to_user failed"); |
| 2849 | return -EFAULT; |
| 2850 | } |
| 2851 | spin_lock_irqsave(&prtd->lock, flags); |
| 2852 | prtd->app_pointer += count; |
| 2853 | if (prtd->app_pointer >= prtd->buffer_size) |
| 2854 | prtd->app_pointer -= prtd->buffer_size; |
| 2855 | prtd->bytes_copied += count; |
| 2856 | } |
| 2857 | msm_compr_read_buffer(prtd); |
| 2858 | |
| 2859 | spin_unlock_irqrestore(&prtd->lock, flags); |
| 2860 | return count; |
| 2861 | } |
| 2862 | |
| 2863 | static int msm_compr_copy(struct snd_compr_stream *cstream, |
| 2864 | char __user *buf, size_t count) |
| 2865 | { |
| 2866 | int ret = 0; |
| 2867 | |
| 2868 | pr_debug(" In %s\n", __func__); |
| 2869 | if (cstream->direction == SND_COMPRESS_PLAYBACK) |
| 2870 | ret = msm_compr_playback_copy(cstream, buf, count); |
| 2871 | else if (cstream->direction == SND_COMPRESS_CAPTURE) |
| 2872 | ret = msm_compr_capture_copy(cstream, buf, count); |
| 2873 | return ret; |
| 2874 | } |
| 2875 | |
| 2876 | static int msm_compr_get_caps(struct snd_compr_stream *cstream, |
| 2877 | struct snd_compr_caps *arg) |
| 2878 | { |
| 2879 | struct snd_compr_runtime *runtime = cstream->runtime; |
| 2880 | struct msm_compr_audio *prtd = runtime->private_data; |
| 2881 | int ret = 0; |
| 2882 | |
| 2883 | pr_debug("%s\n", __func__); |
| 2884 | if ((arg != NULL) && (prtd != NULL)) { |
| 2885 | memcpy(arg, &prtd->compr_cap, sizeof(struct snd_compr_caps)); |
| 2886 | } else { |
| 2887 | ret = -EINVAL; |
| 2888 | pr_err("%s: arg (0x%pK), prtd (0x%pK)\n", __func__, arg, prtd); |
| 2889 | } |
| 2890 | |
| 2891 | return ret; |
| 2892 | } |
| 2893 | |
| 2894 | static int msm_compr_get_codec_caps(struct snd_compr_stream *cstream, |
| 2895 | struct snd_compr_codec_caps *codec) |
| 2896 | { |
| 2897 | pr_debug("%s\n", __func__); |
| 2898 | |
| 2899 | switch (codec->codec) { |
| 2900 | case SND_AUDIOCODEC_MP3: |
| 2901 | codec->num_descriptors = 2; |
| 2902 | codec->descriptor[0].max_ch = 2; |
| 2903 | memcpy(codec->descriptor[0].sample_rates, |
| 2904 | supported_sample_rates, |
| 2905 | sizeof(supported_sample_rates)); |
| 2906 | codec->descriptor[0].num_sample_rates = |
| 2907 | sizeof(supported_sample_rates)/sizeof(unsigned int); |
| 2908 | codec->descriptor[0].bit_rate[0] = 320; /* 320kbps */ |
| 2909 | codec->descriptor[0].bit_rate[1] = 128; |
| 2910 | codec->descriptor[0].num_bitrates = 2; |
| 2911 | codec->descriptor[0].profiles = 0; |
| 2912 | codec->descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO; |
| 2913 | codec->descriptor[0].formats = 0; |
| 2914 | break; |
| 2915 | case SND_AUDIOCODEC_AAC: |
| 2916 | codec->num_descriptors = 2; |
| 2917 | codec->descriptor[1].max_ch = 2; |
| 2918 | memcpy(codec->descriptor[1].sample_rates, |
| 2919 | supported_sample_rates, |
| 2920 | sizeof(supported_sample_rates)); |
| 2921 | codec->descriptor[1].num_sample_rates = |
| 2922 | sizeof(supported_sample_rates)/sizeof(unsigned int); |
| 2923 | codec->descriptor[1].bit_rate[0] = 320; /* 320kbps */ |
| 2924 | codec->descriptor[1].bit_rate[1] = 128; |
| 2925 | codec->descriptor[1].num_bitrates = 2; |
| 2926 | codec->descriptor[1].profiles = 0; |
| 2927 | codec->descriptor[1].modes = 0; |
| 2928 | codec->descriptor[1].formats = |
| 2929 | (SND_AUDIOSTREAMFORMAT_MP4ADTS | |
| 2930 | SND_AUDIOSTREAMFORMAT_RAW); |
| 2931 | break; |
| 2932 | case SND_AUDIOCODEC_AC3: |
| 2933 | case SND_AUDIOCODEC_EAC3: |
| 2934 | case SND_AUDIOCODEC_FLAC: |
| 2935 | case SND_AUDIOCODEC_VORBIS: |
| 2936 | case SND_AUDIOCODEC_ALAC: |
| 2937 | case SND_AUDIOCODEC_APE: |
| 2938 | case SND_AUDIOCODEC_DTS: |
| 2939 | case SND_AUDIOCODEC_DSD: |
| 2940 | case SND_AUDIOCODEC_TRUEHD: |
| 2941 | case SND_AUDIOCODEC_IEC61937: |
| 2942 | case SND_AUDIOCODEC_APTX: |
| 2943 | break; |
| 2944 | default: |
| 2945 | pr_err("%s: Unsupported audio codec %d\n", |
| 2946 | __func__, codec->codec); |
| 2947 | return -EINVAL; |
| 2948 | } |
| 2949 | |
| 2950 | return 0; |
| 2951 | } |
| 2952 | |
| 2953 | static int msm_compr_set_metadata(struct snd_compr_stream *cstream, |
| 2954 | struct snd_compr_metadata *metadata) |
| 2955 | { |
| 2956 | struct msm_compr_audio *prtd; |
| 2957 | struct audio_client *ac; |
| 2958 | pr_debug("%s\n", __func__); |
| 2959 | |
| 2960 | if (!metadata || !cstream) |
| 2961 | return -EINVAL; |
| 2962 | |
| 2963 | prtd = cstream->runtime->private_data; |
| 2964 | if (!prtd || !prtd->audio_client) { |
| 2965 | pr_err("%s: prtd or audio client is NULL\n", __func__); |
| 2966 | return -EINVAL; |
| 2967 | } |
| 2968 | |
| 2969 | if (((metadata->key == SNDRV_COMPRESS_ENCODER_PADDING) || |
| 2970 | (metadata->key == SNDRV_COMPRESS_ENCODER_DELAY)) && |
| 2971 | (prtd->compr_passthr != LEGACY_PCM)) { |
| 2972 | pr_debug("%s: No trailing silence for compress_type[%d]\n", |
| 2973 | __func__, prtd->compr_passthr); |
| 2974 | return 0; |
| 2975 | } |
| 2976 | |
| 2977 | ac = prtd->audio_client; |
| 2978 | if (metadata->key == SNDRV_COMPRESS_ENCODER_PADDING) { |
| 2979 | pr_debug("%s, got encoder padding %u", |
| 2980 | __func__, metadata->value[0]); |
| 2981 | prtd->gapless_state.trailing_samples_drop = metadata->value[0]; |
| 2982 | } else if (metadata->key == SNDRV_COMPRESS_ENCODER_DELAY) { |
| 2983 | pr_debug("%s, got encoder delay %u", |
| 2984 | __func__, metadata->value[0]); |
| 2985 | prtd->gapless_state.initial_samples_drop = metadata->value[0]; |
| 2986 | } else if (metadata->key == SNDRV_COMPRESS_RENDER_MODE) { |
| 2987 | return msm_compr_set_render_mode(prtd, metadata->value[0]); |
| 2988 | } else if (metadata->key == SNDRV_COMPRESS_CLK_REC_MODE) { |
| 2989 | return msm_compr_set_clk_rec_mode(ac, metadata->value[0]); |
| 2990 | } else if (metadata->key == SNDRV_COMPRESS_RENDER_WINDOW) { |
| 2991 | return msm_compr_set_render_window( |
| 2992 | ac, |
| 2993 | metadata->value[0], |
| 2994 | metadata->value[1], |
| 2995 | metadata->value[2], |
| 2996 | metadata->value[3]); |
| 2997 | } else if (metadata->key == SNDRV_COMPRESS_START_DELAY) { |
| 2998 | prtd->start_delay_lsw = metadata->value[0]; |
| 2999 | prtd->start_delay_msw = metadata->value[1]; |
| 3000 | } else if (metadata->key == |
| 3001 | SNDRV_COMPRESS_ENABLE_ADJUST_SESSION_CLOCK) { |
| 3002 | return msm_compr_enable_adjust_session_clock(ac, |
| 3003 | metadata->value[0]); |
| 3004 | } else if (metadata->key == SNDRV_COMPRESS_ADJUST_SESSION_CLOCK) { |
| 3005 | return msm_compr_adjust_session_clock(ac, |
| 3006 | metadata->value[0], |
| 3007 | metadata->value[1]); |
| 3008 | } |
| 3009 | |
| 3010 | return 0; |
| 3011 | } |
| 3012 | |
| 3013 | static int msm_compr_get_metadata(struct snd_compr_stream *cstream, |
| 3014 | struct snd_compr_metadata *metadata) |
| 3015 | { |
| 3016 | struct msm_compr_audio *prtd; |
| 3017 | struct audio_client *ac; |
| 3018 | int ret = -EINVAL; |
| 3019 | |
| 3020 | pr_debug("%s\n", __func__); |
| 3021 | |
| 3022 | if (!metadata || !cstream || !cstream->runtime) |
| 3023 | return ret; |
| 3024 | |
| 3025 | if (metadata->key != SNDRV_COMPRESS_PATH_DELAY) { |
| 3026 | pr_err("%s, unsupported key %d\n", __func__, metadata->key); |
| 3027 | return ret; |
| 3028 | } |
| 3029 | |
| 3030 | prtd = cstream->runtime->private_data; |
| 3031 | if (!prtd || !prtd->audio_client) { |
| 3032 | pr_err("%s: prtd or audio client is NULL\n", __func__); |
| 3033 | return ret; |
| 3034 | } |
| 3035 | |
| 3036 | ac = prtd->audio_client; |
| 3037 | ret = q6asm_get_path_delay(prtd->audio_client); |
| 3038 | if (ret) { |
| 3039 | pr_err("%s: get_path_delay failed, ret=%d\n", __func__, ret); |
| 3040 | return ret; |
| 3041 | } |
| 3042 | |
| 3043 | pr_debug("%s, path delay(in us) %u\n", __func__, ac->path_delay); |
| 3044 | |
| 3045 | metadata->value[0] = ac->path_delay; |
| 3046 | |
| 3047 | return ret; |
| 3048 | } |
| 3049 | |
| 3050 | |
| 3051 | static int msm_compr_set_next_track_param(struct snd_compr_stream *cstream, |
| 3052 | union snd_codec_options *codec_options) |
| 3053 | { |
| 3054 | struct msm_compr_audio *prtd; |
| 3055 | struct audio_client *ac; |
| 3056 | int ret = 0; |
| 3057 | |
| 3058 | if (!codec_options || !cstream) |
| 3059 | return -EINVAL; |
| 3060 | |
| 3061 | prtd = cstream->runtime->private_data; |
| 3062 | if (!prtd || !prtd->audio_client) { |
| 3063 | pr_err("%s: prtd or audio client is NULL\n", __func__); |
| 3064 | return -EINVAL; |
| 3065 | } |
| 3066 | |
| 3067 | ac = prtd->audio_client; |
| 3068 | |
| 3069 | pr_debug("%s: got codec options for codec type %u", |
| 3070 | __func__, prtd->codec); |
| 3071 | switch (prtd->codec) { |
| 3072 | case FORMAT_WMA_V9: |
| 3073 | case FORMAT_WMA_V10PRO: |
| 3074 | case FORMAT_FLAC: |
| 3075 | case FORMAT_VORBIS: |
| 3076 | case FORMAT_ALAC: |
| 3077 | case FORMAT_APE: |
| 3078 | memcpy(&(prtd->gapless_state.codec_options), |
| 3079 | codec_options, |
| 3080 | sizeof(union snd_codec_options)); |
| 3081 | ret = msm_compr_send_media_format_block(cstream, |
| 3082 | ac->stream_id, true); |
| 3083 | if (ret < 0) { |
| 3084 | pr_err("%s: failed to send media format block\n", |
| 3085 | __func__); |
| 3086 | } |
| 3087 | break; |
| 3088 | |
| 3089 | default: |
| 3090 | pr_debug("%s: Ignore sending CMD Format block\n", |
| 3091 | __func__); |
| 3092 | break; |
| 3093 | } |
| 3094 | |
| 3095 | return ret; |
| 3096 | } |
| 3097 | |
| 3098 | static int msm_compr_volume_put(struct snd_kcontrol *kcontrol, |
| 3099 | struct snd_ctl_elem_value *ucontrol) |
| 3100 | { |
| 3101 | struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| 3102 | unsigned long fe_id = kcontrol->private_value; |
| 3103 | struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| 3104 | snd_soc_component_get_drvdata(comp); |
| 3105 | struct snd_compr_stream *cstream = NULL; |
| 3106 | uint32_t *volume = NULL; |
| 3107 | |
| 3108 | if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| 3109 | pr_err("%s Received out of bounds fe_id %lu\n", |
| 3110 | __func__, fe_id); |
| 3111 | return -EINVAL; |
| 3112 | } |
| 3113 | |
| 3114 | cstream = pdata->cstream[fe_id]; |
| 3115 | volume = pdata->volume[fe_id]; |
| 3116 | |
| 3117 | volume[0] = ucontrol->value.integer.value[0]; |
| 3118 | volume[1] = ucontrol->value.integer.value[1]; |
| 3119 | pr_debug("%s: fe_id %lu left_vol %d right_vol %d\n", |
| 3120 | __func__, fe_id, volume[0], volume[1]); |
| 3121 | if (cstream) |
| 3122 | msm_compr_set_volume(cstream, volume[0], volume[1]); |
| 3123 | return 0; |
| 3124 | } |
| 3125 | |
| 3126 | static int msm_compr_volume_get(struct snd_kcontrol *kcontrol, |
| 3127 | struct snd_ctl_elem_value *ucontrol) |
| 3128 | { |
| 3129 | struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| 3130 | unsigned long fe_id = kcontrol->private_value; |
| 3131 | |
| 3132 | struct msm_compr_pdata *pdata = |
| 3133 | snd_soc_component_get_drvdata(comp); |
| 3134 | uint32_t *volume = NULL; |
| 3135 | |
| 3136 | if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| 3137 | pr_err("%s Received out of bound fe_id %lu\n", __func__, fe_id); |
| 3138 | return -EINVAL; |
| 3139 | } |
| 3140 | |
| 3141 | volume = pdata->volume[fe_id]; |
| 3142 | pr_debug("%s: fe_id %lu\n", __func__, fe_id); |
| 3143 | ucontrol->value.integer.value[0] = volume[0]; |
| 3144 | ucontrol->value.integer.value[1] = volume[1]; |
| 3145 | |
| 3146 | return 0; |
| 3147 | } |
| 3148 | |
| 3149 | static int msm_compr_audio_effects_config_put(struct snd_kcontrol *kcontrol, |
| 3150 | struct snd_ctl_elem_value *ucontrol) |
| 3151 | { |
| 3152 | struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| 3153 | unsigned long fe_id = kcontrol->private_value; |
| 3154 | struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| 3155 | snd_soc_component_get_drvdata(comp); |
| 3156 | struct msm_compr_audio_effects *audio_effects = NULL; |
| 3157 | struct snd_compr_stream *cstream = NULL; |
| 3158 | struct msm_compr_audio *prtd = NULL; |
| 3159 | long *values = &(ucontrol->value.integer.value[0]); |
| 3160 | int effects_module; |
| 3161 | |
| 3162 | pr_debug("%s\n", __func__); |
| 3163 | if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| 3164 | pr_err("%s Received out of bounds fe_id %lu\n", |
| 3165 | __func__, fe_id); |
| 3166 | return -EINVAL; |
| 3167 | } |
| 3168 | cstream = pdata->cstream[fe_id]; |
| 3169 | audio_effects = pdata->audio_effects[fe_id]; |
| 3170 | if (!cstream || !audio_effects) { |
| 3171 | pr_err("%s: stream or effects inactive\n", __func__); |
| 3172 | return -EINVAL; |
| 3173 | } |
| 3174 | prtd = cstream->runtime->private_data; |
| 3175 | if (!prtd) { |
| 3176 | pr_err("%s: cannot set audio effects\n", __func__); |
| 3177 | return -EINVAL; |
| 3178 | } |
| 3179 | if (prtd->compr_passthr != LEGACY_PCM) { |
| 3180 | pr_debug("%s: No effects for compr_type[%d]\n", |
| 3181 | __func__, prtd->compr_passthr); |
| 3182 | return 0; |
| 3183 | } |
| 3184 | pr_debug("%s: Effects supported for compr_type[%d]\n", |
| 3185 | __func__, prtd->compr_passthr); |
| 3186 | |
| 3187 | effects_module = *values++; |
| 3188 | switch (effects_module) { |
| 3189 | case VIRTUALIZER_MODULE: |
| 3190 | pr_debug("%s: VIRTUALIZER_MODULE\n", __func__); |
| 3191 | if (msm_audio_effects_is_effmodule_supp_in_top(effects_module, |
| 3192 | prtd->audio_client->topology)) |
| 3193 | msm_audio_effects_virtualizer_handler( |
| 3194 | prtd->audio_client, |
| 3195 | &(audio_effects->virtualizer), |
| 3196 | values); |
| 3197 | break; |
| 3198 | case REVERB_MODULE: |
| 3199 | pr_debug("%s: REVERB_MODULE\n", __func__); |
| 3200 | if (msm_audio_effects_is_effmodule_supp_in_top(effects_module, |
| 3201 | prtd->audio_client->topology)) |
| 3202 | msm_audio_effects_reverb_handler(prtd->audio_client, |
| 3203 | &(audio_effects->reverb), |
| 3204 | values); |
| 3205 | break; |
| 3206 | case BASS_BOOST_MODULE: |
| 3207 | pr_debug("%s: BASS_BOOST_MODULE\n", __func__); |
| 3208 | if (msm_audio_effects_is_effmodule_supp_in_top(effects_module, |
| 3209 | prtd->audio_client->topology)) |
| 3210 | msm_audio_effects_bass_boost_handler(prtd->audio_client, |
| 3211 | &(audio_effects->bass_boost), |
| 3212 | values); |
| 3213 | break; |
| 3214 | case PBE_MODULE: |
| 3215 | pr_debug("%s: PBE_MODULE\n", __func__); |
| 3216 | if (msm_audio_effects_is_effmodule_supp_in_top(effects_module, |
| 3217 | prtd->audio_client->topology)) |
| 3218 | msm_audio_effects_pbe_handler(prtd->audio_client, |
| 3219 | &(audio_effects->pbe), |
| 3220 | values); |
| 3221 | break; |
| 3222 | case EQ_MODULE: |
| 3223 | pr_debug("%s: EQ_MODULE\n", __func__); |
| 3224 | if (msm_audio_effects_is_effmodule_supp_in_top(effects_module, |
| 3225 | prtd->audio_client->topology)) |
| 3226 | msm_audio_effects_popless_eq_handler(prtd->audio_client, |
| 3227 | &(audio_effects->equalizer), |
| 3228 | values); |
| 3229 | break; |
| 3230 | case SOFT_VOLUME_MODULE: |
| 3231 | pr_debug("%s: SOFT_VOLUME_MODULE\n", __func__); |
| 3232 | break; |
| 3233 | case SOFT_VOLUME2_MODULE: |
| 3234 | pr_debug("%s: SOFT_VOLUME2_MODULE\n", __func__); |
| 3235 | if (msm_audio_effects_is_effmodule_supp_in_top(effects_module, |
| 3236 | prtd->audio_client->topology)) |
| 3237 | msm_audio_effects_volume_handler_v2(prtd->audio_client, |
| 3238 | &(audio_effects->volume), |
| 3239 | values, SOFT_VOLUME_INSTANCE_2); |
| 3240 | break; |
| 3241 | default: |
| 3242 | pr_err("%s Invalid effects config module\n", __func__); |
| 3243 | return -EINVAL; |
| 3244 | } |
| 3245 | return 0; |
| 3246 | } |
| 3247 | |
| 3248 | static int msm_compr_audio_effects_config_get(struct snd_kcontrol *kcontrol, |
| 3249 | struct snd_ctl_elem_value *ucontrol) |
| 3250 | { |
| 3251 | struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| 3252 | unsigned long fe_id = kcontrol->private_value; |
| 3253 | struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| 3254 | snd_soc_component_get_drvdata(comp); |
| 3255 | struct msm_compr_audio_effects *audio_effects = NULL; |
| 3256 | struct snd_compr_stream *cstream = NULL; |
| 3257 | struct msm_compr_audio *prtd = NULL; |
| 3258 | |
| 3259 | pr_debug("%s\n", __func__); |
| 3260 | if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| 3261 | pr_err("%s Received out of bounds fe_id %lu\n", |
| 3262 | __func__, fe_id); |
| 3263 | return -EINVAL; |
| 3264 | } |
| 3265 | cstream = pdata->cstream[fe_id]; |
| 3266 | audio_effects = pdata->audio_effects[fe_id]; |
| 3267 | if (!cstream || !audio_effects) { |
| 3268 | pr_err("%s: stream or effects inactive\n", __func__); |
| 3269 | return -EINVAL; |
| 3270 | } |
| 3271 | prtd = cstream->runtime->private_data; |
| 3272 | if (!prtd) { |
| 3273 | pr_err("%s: cannot set audio effects\n", __func__); |
| 3274 | return -EINVAL; |
| 3275 | } |
| 3276 | |
| 3277 | return 0; |
| 3278 | } |
| 3279 | |
| 3280 | static int msm_compr_query_audio_effect_put(struct snd_kcontrol *kcontrol, |
| 3281 | struct snd_ctl_elem_value *ucontrol) |
| 3282 | { |
| 3283 | struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| 3284 | unsigned long fe_id = kcontrol->private_value; |
| 3285 | struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| 3286 | snd_soc_component_get_drvdata(comp); |
| 3287 | struct msm_compr_audio_effects *audio_effects = NULL; |
| 3288 | struct snd_compr_stream *cstream = NULL; |
| 3289 | struct msm_compr_audio *prtd = NULL; |
| 3290 | long *values = &(ucontrol->value.integer.value[0]); |
| 3291 | |
| 3292 | if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| 3293 | pr_err("%s Received out of bounds fe_id %lu\n", |
| 3294 | __func__, fe_id); |
| 3295 | return -EINVAL; |
| 3296 | } |
| 3297 | cstream = pdata->cstream[fe_id]; |
| 3298 | audio_effects = pdata->audio_effects[fe_id]; |
| 3299 | if (!cstream || !audio_effects) { |
| 3300 | pr_err("%s: stream or effects inactive\n", __func__); |
| 3301 | return -EINVAL; |
| 3302 | } |
| 3303 | prtd = cstream->runtime->private_data; |
| 3304 | if (!prtd) { |
| 3305 | pr_err("%s: cannot set audio effects\n", __func__); |
| 3306 | return -EINVAL; |
| 3307 | } |
| 3308 | if (prtd->compr_passthr != LEGACY_PCM) { |
| 3309 | pr_err("%s: No effects for compr_type[%d]\n", |
| 3310 | __func__, prtd->compr_passthr); |
| 3311 | return -EPERM; |
| 3312 | } |
| 3313 | audio_effects->query.mod_id = (u32)*values++; |
| 3314 | audio_effects->query.parm_id = (u32)*values++; |
| 3315 | audio_effects->query.size = (u32)*values++; |
| 3316 | audio_effects->query.offset = (u32)*values++; |
| 3317 | audio_effects->query.device = (u32)*values++; |
| 3318 | return 0; |
| 3319 | } |
| 3320 | |
| 3321 | static int msm_compr_query_audio_effect_get(struct snd_kcontrol *kcontrol, |
| 3322 | struct snd_ctl_elem_value *ucontrol) |
| 3323 | { |
| 3324 | struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| 3325 | unsigned long fe_id = kcontrol->private_value; |
| 3326 | struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| 3327 | snd_soc_component_get_drvdata(comp); |
| 3328 | struct msm_compr_audio_effects *audio_effects = NULL; |
| 3329 | struct snd_compr_stream *cstream = NULL; |
| 3330 | struct msm_compr_audio *prtd = NULL; |
| 3331 | long *values = &(ucontrol->value.integer.value[0]); |
| 3332 | |
| 3333 | if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| 3334 | pr_err("%s Received out of bounds fe_id %lu\n", |
| 3335 | __func__, fe_id); |
| 3336 | return -EINVAL; |
| 3337 | } |
| 3338 | cstream = pdata->cstream[fe_id]; |
| 3339 | audio_effects = pdata->audio_effects[fe_id]; |
| 3340 | if (!cstream || !audio_effects) { |
| 3341 | pr_debug("%s: stream or effects inactive\n", __func__); |
| 3342 | return -EINVAL; |
| 3343 | } |
| 3344 | prtd = cstream->runtime->private_data; |
| 3345 | if (!prtd) { |
| 3346 | pr_err("%s: cannot set audio effects\n", __func__); |
| 3347 | return -EINVAL; |
| 3348 | } |
| 3349 | values[0] = (long)audio_effects->query.mod_id; |
| 3350 | values[1] = (long)audio_effects->query.parm_id; |
| 3351 | values[2] = (long)audio_effects->query.size; |
| 3352 | values[3] = (long)audio_effects->query.offset; |
| 3353 | values[4] = (long)audio_effects->query.device; |
| 3354 | return 0; |
| 3355 | } |
| 3356 | |
| 3357 | static int msm_compr_send_dec_params(struct snd_compr_stream *cstream, |
| 3358 | struct msm_compr_dec_params *dec_params, |
| 3359 | int stream_id) |
| 3360 | { |
| 3361 | |
| 3362 | int rc = 0; |
| 3363 | struct msm_compr_audio *prtd = NULL; |
| 3364 | struct snd_dec_ddp *ddp = &dec_params->ddp_params; |
| 3365 | |
| 3366 | if (!cstream || !dec_params) { |
| 3367 | pr_err("%s: stream or dec_params inactive\n", __func__); |
| 3368 | rc = -EINVAL; |
| 3369 | goto end; |
| 3370 | } |
| 3371 | prtd = cstream->runtime->private_data; |
| 3372 | if (!prtd) { |
| 3373 | pr_err("%s: cannot set dec_params\n", __func__); |
| 3374 | rc = -EINVAL; |
| 3375 | goto end; |
| 3376 | } |
| 3377 | switch (prtd->codec) { |
| 3378 | case FORMAT_MP3: |
| 3379 | case FORMAT_MPEG4_AAC: |
| 3380 | case FORMAT_TRUEHD: |
| 3381 | case FORMAT_IEC61937: |
| 3382 | case FORMAT_APTX: |
| 3383 | pr_debug("%s: no runtime parameters for codec: %d\n", __func__, |
| 3384 | prtd->codec); |
| 3385 | break; |
| 3386 | case FORMAT_AC3: |
| 3387 | case FORMAT_EAC3: |
| 3388 | if (prtd->compr_passthr != LEGACY_PCM) { |
| 3389 | pr_debug("%s: No DDP param for compr_type[%d]\n", |
| 3390 | __func__, prtd->compr_passthr); |
| 3391 | break; |
| 3392 | } |
| 3393 | rc = msm_compr_send_ddp_cfg(prtd->audio_client, ddp, stream_id); |
| 3394 | if (rc < 0) |
| 3395 | pr_err("%s: DDP CMD CFG failed %d\n", __func__, rc); |
| 3396 | break; |
| 3397 | default: |
| 3398 | break; |
| 3399 | } |
| 3400 | end: |
| 3401 | return rc; |
| 3402 | |
| 3403 | } |
| 3404 | static int msm_compr_dec_params_put(struct snd_kcontrol *kcontrol, |
| 3405 | struct snd_ctl_elem_value *ucontrol) |
| 3406 | { |
| 3407 | struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| 3408 | unsigned long fe_id = kcontrol->private_value; |
| 3409 | struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| 3410 | snd_soc_component_get_drvdata(comp); |
| 3411 | struct msm_compr_dec_params *dec_params = NULL; |
| 3412 | struct snd_compr_stream *cstream = NULL; |
| 3413 | struct msm_compr_audio *prtd = NULL; |
| 3414 | long *values = &(ucontrol->value.integer.value[0]); |
| 3415 | int rc = 0; |
| 3416 | |
| 3417 | pr_debug("%s\n", __func__); |
| 3418 | if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| 3419 | pr_err("%s Received out of bounds fe_id %lu\n", |
| 3420 | __func__, fe_id); |
| 3421 | rc = -EINVAL; |
| 3422 | goto end; |
| 3423 | } |
| 3424 | |
| 3425 | cstream = pdata->cstream[fe_id]; |
| 3426 | dec_params = pdata->dec_params[fe_id]; |
| 3427 | |
| 3428 | if (!cstream || !dec_params) { |
| 3429 | pr_err("%s: stream or dec_params inactive\n", __func__); |
| 3430 | rc = -EINVAL; |
| 3431 | goto end; |
| 3432 | } |
| 3433 | prtd = cstream->runtime->private_data; |
| 3434 | if (!prtd) { |
| 3435 | pr_err("%s: cannot set dec_params\n", __func__); |
| 3436 | rc = -EINVAL; |
| 3437 | goto end; |
| 3438 | } |
| 3439 | |
| 3440 | switch (prtd->codec) { |
| 3441 | case FORMAT_MP3: |
| 3442 | case FORMAT_MPEG4_AAC: |
| 3443 | case FORMAT_FLAC: |
| 3444 | case FORMAT_VORBIS: |
| 3445 | case FORMAT_ALAC: |
| 3446 | case FORMAT_APE: |
| 3447 | case FORMAT_DTS: |
| 3448 | case FORMAT_DSD: |
| 3449 | case FORMAT_TRUEHD: |
| 3450 | case FORMAT_IEC61937: |
| 3451 | case FORMAT_APTX: |
| 3452 | pr_debug("%s: no runtime parameters for codec: %d\n", __func__, |
| 3453 | prtd->codec); |
| 3454 | break; |
| 3455 | case FORMAT_AC3: |
| 3456 | case FORMAT_EAC3: { |
| 3457 | struct snd_dec_ddp *ddp = &dec_params->ddp_params; |
| 3458 | int cnt; |
| 3459 | |
| 3460 | if (prtd->compr_passthr != LEGACY_PCM) { |
| 3461 | pr_debug("%s: No DDP param for compr_type[%d]\n", |
| 3462 | __func__, prtd->compr_passthr); |
| 3463 | break; |
| 3464 | } |
| 3465 | |
| 3466 | ddp->params_length = (*values++); |
| 3467 | if (ddp->params_length > DDP_DEC_MAX_NUM_PARAM) { |
| 3468 | pr_err("%s: invalid num of params:: %d\n", __func__, |
| 3469 | ddp->params_length); |
| 3470 | rc = -EINVAL; |
| 3471 | goto end; |
| 3472 | } |
| 3473 | for (cnt = 0; cnt < ddp->params_length; cnt++) { |
| 3474 | ddp->params_id[cnt] = *values++; |
| 3475 | ddp->params_value[cnt] = *values++; |
| 3476 | } |
| 3477 | prtd = cstream->runtime->private_data; |
| 3478 | if (prtd && prtd->audio_client) |
| 3479 | rc = msm_compr_send_dec_params(cstream, dec_params, |
| 3480 | prtd->audio_client->stream_id); |
| 3481 | break; |
| 3482 | } |
| 3483 | default: |
| 3484 | break; |
| 3485 | } |
| 3486 | end: |
| 3487 | pr_debug("%s: ret %d\n", __func__, rc); |
| 3488 | return rc; |
| 3489 | } |
| 3490 | |
| 3491 | static int msm_compr_dec_params_get(struct snd_kcontrol *kcontrol, |
| 3492 | struct snd_ctl_elem_value *ucontrol) |
| 3493 | { |
| 3494 | /* dummy function */ |
| 3495 | return 0; |
| 3496 | } |
| 3497 | |
| 3498 | static int msm_compr_playback_app_type_cfg_put(struct snd_kcontrol *kcontrol, |
| 3499 | struct snd_ctl_elem_value *ucontrol) |
| 3500 | { |
| 3501 | u64 fe_id = kcontrol->private_value; |
| 3502 | int session_type = SESSION_TYPE_RX; |
| 3503 | int be_id = ucontrol->value.integer.value[3]; |
| 3504 | struct msm_pcm_stream_app_type_cfg cfg_data = {0, 0, 48000}; |
| 3505 | int ret = 0; |
| 3506 | |
| 3507 | cfg_data.app_type = ucontrol->value.integer.value[0]; |
| 3508 | cfg_data.acdb_dev_id = ucontrol->value.integer.value[1]; |
| 3509 | if (ucontrol->value.integer.value[2] != 0) |
| 3510 | cfg_data.sample_rate = ucontrol->value.integer.value[2]; |
| 3511 | pr_debug("%s: fe_id- %llu session_type- %d be_id- %d app_type- %d acdb_dev_id- %d sample_rate- %d\n", |
| 3512 | __func__, fe_id, session_type, be_id, |
| 3513 | cfg_data.app_type, cfg_data.acdb_dev_id, cfg_data.sample_rate); |
| 3514 | ret = msm_pcm_routing_reg_stream_app_type_cfg(fe_id, session_type, |
| 3515 | be_id, &cfg_data); |
| 3516 | if (ret < 0) |
| 3517 | pr_err("%s: msm_pcm_routing_reg_stream_app_type_cfg failed returned %d\n", |
| 3518 | __func__, ret); |
| 3519 | |
| 3520 | return ret; |
| 3521 | } |
| 3522 | |
| 3523 | static int msm_compr_playback_app_type_cfg_get(struct snd_kcontrol *kcontrol, |
| 3524 | struct snd_ctl_elem_value *ucontrol) |
| 3525 | { |
| 3526 | u64 fe_id = kcontrol->private_value; |
| 3527 | int session_type = SESSION_TYPE_RX; |
| 3528 | int be_id = 0; |
| 3529 | struct msm_pcm_stream_app_type_cfg cfg_data = {0}; |
| 3530 | int ret = 0; |
| 3531 | |
| 3532 | ret = msm_pcm_routing_get_stream_app_type_cfg(fe_id, session_type, |
| 3533 | &be_id, &cfg_data); |
| 3534 | if (ret < 0) { |
| 3535 | pr_err("%s: msm_pcm_routing_get_stream_app_type_cfg failed returned %d\n", |
| 3536 | __func__, ret); |
| 3537 | goto done; |
| 3538 | } |
| 3539 | |
| 3540 | ucontrol->value.integer.value[0] = cfg_data.app_type; |
| 3541 | ucontrol->value.integer.value[1] = cfg_data.acdb_dev_id; |
| 3542 | ucontrol->value.integer.value[2] = cfg_data.sample_rate; |
| 3543 | ucontrol->value.integer.value[3] = be_id; |
| 3544 | pr_debug("%s: fedai_id %llu, session_type %d, be_id %d, app_type %d, acdb_dev_id %d, sample_rate %d\n", |
| 3545 | __func__, fe_id, session_type, be_id, |
| 3546 | cfg_data.app_type, cfg_data.acdb_dev_id, cfg_data.sample_rate); |
| 3547 | done: |
| 3548 | return ret; |
| 3549 | } |
| 3550 | |
| 3551 | static int msm_compr_capture_app_type_cfg_put(struct snd_kcontrol *kcontrol, |
| 3552 | struct snd_ctl_elem_value *ucontrol) |
| 3553 | { |
| 3554 | u64 fe_id = kcontrol->private_value; |
| 3555 | int session_type = SESSION_TYPE_TX; |
| 3556 | int be_id = ucontrol->value.integer.value[3]; |
| 3557 | struct msm_pcm_stream_app_type_cfg cfg_data = {0, 0, 48000}; |
| 3558 | int ret = 0; |
| 3559 | |
| 3560 | cfg_data.app_type = ucontrol->value.integer.value[0]; |
| 3561 | cfg_data.acdb_dev_id = ucontrol->value.integer.value[1]; |
| 3562 | if (ucontrol->value.integer.value[2] != 0) |
| 3563 | cfg_data.sample_rate = ucontrol->value.integer.value[2]; |
| 3564 | pr_debug("%s: fe_id- %llu session_type- %d be_id- %d app_type- %d acdb_dev_id- %d sample_rate- %d\n", |
| 3565 | __func__, fe_id, session_type, be_id, |
| 3566 | cfg_data.app_type, cfg_data.acdb_dev_id, cfg_data.sample_rate); |
| 3567 | ret = msm_pcm_routing_reg_stream_app_type_cfg(fe_id, session_type, |
| 3568 | be_id, &cfg_data); |
| 3569 | if (ret < 0) |
| 3570 | pr_err("%s: msm_pcm_routing_reg_stream_app_type_cfg failed returned %d\n", |
| 3571 | __func__, ret); |
| 3572 | |
| 3573 | return ret; |
| 3574 | } |
| 3575 | |
| 3576 | static int msm_compr_capture_app_type_cfg_get(struct snd_kcontrol *kcontrol, |
| 3577 | struct snd_ctl_elem_value *ucontrol) |
| 3578 | { |
| 3579 | u64 fe_id = kcontrol->private_value; |
| 3580 | int session_type = SESSION_TYPE_TX; |
| 3581 | int be_id = 0; |
| 3582 | struct msm_pcm_stream_app_type_cfg cfg_data = {0}; |
| 3583 | int ret = 0; |
| 3584 | |
| 3585 | ret = msm_pcm_routing_get_stream_app_type_cfg(fe_id, session_type, |
| 3586 | &be_id, &cfg_data); |
| 3587 | if (ret < 0) { |
| 3588 | pr_err("%s: msm_pcm_routing_get_stream_app_type_cfg failed returned %d\n", |
| 3589 | __func__, ret); |
| 3590 | goto done; |
| 3591 | } |
| 3592 | |
| 3593 | ucontrol->value.integer.value[0] = cfg_data.app_type; |
| 3594 | ucontrol->value.integer.value[1] = cfg_data.acdb_dev_id; |
| 3595 | ucontrol->value.integer.value[2] = cfg_data.sample_rate; |
| 3596 | ucontrol->value.integer.value[3] = be_id; |
| 3597 | pr_debug("%s: fedai_id %llu, session_type %d, be_id %d, app_type %d, acdb_dev_id %d, sample_rate %d\n", |
| 3598 | __func__, fe_id, session_type, be_id, |
| 3599 | cfg_data.app_type, cfg_data.acdb_dev_id, cfg_data.sample_rate); |
| 3600 | done: |
| 3601 | return ret; |
| 3602 | } |
| 3603 | |
| 3604 | static int msm_compr_channel_map_put(struct snd_kcontrol *kcontrol, |
| 3605 | struct snd_ctl_elem_value *ucontrol) |
| 3606 | { |
| 3607 | struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| 3608 | u64 fe_id = kcontrol->private_value; |
| 3609 | struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| 3610 | snd_soc_component_get_drvdata(comp); |
| 3611 | int rc = 0, i; |
| 3612 | |
| 3613 | pr_debug("%s: fe_id- %llu\n", __func__, fe_id); |
| 3614 | |
| 3615 | if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| 3616 | pr_err("%s Received out of bounds fe_id %llu\n", |
| 3617 | __func__, fe_id); |
| 3618 | rc = -EINVAL; |
| 3619 | goto end; |
| 3620 | } |
| 3621 | |
| 3622 | if (pdata->ch_map[fe_id]) { |
| 3623 | pdata->ch_map[fe_id]->set_ch_map = true; |
Dieter Luecking | ba7644d | 2018-09-28 15:09:32 +0200 | [diff] [blame] | 3624 | for (i = 0; i < PCM_FORMAT_MAX_NUM_CHANNEL_V8; i++) |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 3625 | pdata->ch_map[fe_id]->channel_map[i] = |
| 3626 | (char)(ucontrol->value.integer.value[i]); |
| 3627 | } else { |
| 3628 | pr_debug("%s: no memory for ch_map, default will be set\n", |
| 3629 | __func__); |
| 3630 | } |
| 3631 | end: |
| 3632 | pr_debug("%s: ret %d\n", __func__, rc); |
| 3633 | return rc; |
| 3634 | } |
| 3635 | |
| 3636 | static int msm_compr_channel_map_get(struct snd_kcontrol *kcontrol, |
| 3637 | struct snd_ctl_elem_value *ucontrol) |
| 3638 | { |
| 3639 | struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| 3640 | u64 fe_id = kcontrol->private_value; |
| 3641 | struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| 3642 | snd_soc_component_get_drvdata(comp); |
| 3643 | int rc = 0, i; |
| 3644 | |
| 3645 | pr_debug("%s: fe_id- %llu\n", __func__, fe_id); |
| 3646 | if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| 3647 | pr_err("%s: Received out of bounds fe_id %llu\n", |
| 3648 | __func__, fe_id); |
| 3649 | rc = -EINVAL; |
| 3650 | goto end; |
| 3651 | } |
| 3652 | if (pdata->ch_map[fe_id]) { |
Dieter Luecking | ba7644d | 2018-09-28 15:09:32 +0200 | [diff] [blame] | 3653 | for (i = 0; i < PCM_FORMAT_MAX_NUM_CHANNEL_V8; i++) |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 3654 | ucontrol->value.integer.value[i] = |
| 3655 | pdata->ch_map[fe_id]->channel_map[i]; |
| 3656 | } |
| 3657 | end: |
| 3658 | pr_debug("%s: ret %d\n", __func__, rc); |
| 3659 | return rc; |
| 3660 | } |
| 3661 | |
| 3662 | static int msm_compr_adsp_stream_cmd_put(struct snd_kcontrol *kcontrol, |
| 3663 | struct snd_ctl_elem_value *ucontrol) |
| 3664 | { |
| 3665 | struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| 3666 | unsigned long fe_id = kcontrol->private_value; |
| 3667 | struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| 3668 | snd_soc_component_get_drvdata(comp); |
| 3669 | struct snd_compr_stream *cstream = NULL; |
| 3670 | struct msm_compr_audio *prtd; |
| 3671 | int ret = 0; |
| 3672 | struct msm_adsp_event_data *event_data = NULL; |
Aditya Bavanari | 2e3341d | 2018-02-23 12:58:57 +0530 | [diff] [blame] | 3673 | uint64_t actual_payload_len = 0; |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 3674 | |
| 3675 | if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| 3676 | pr_err("%s Received invalid fe_id %lu\n", |
| 3677 | __func__, fe_id); |
| 3678 | ret = -EINVAL; |
| 3679 | goto done; |
| 3680 | } |
| 3681 | |
| 3682 | cstream = pdata->cstream[fe_id]; |
| 3683 | if (cstream == NULL) { |
| 3684 | pr_err("%s cstream is null\n", __func__); |
| 3685 | ret = -EINVAL; |
| 3686 | goto done; |
| 3687 | } |
| 3688 | |
| 3689 | prtd = cstream->runtime->private_data; |
| 3690 | if (!prtd) { |
| 3691 | pr_err("%s: prtd is null\n", __func__); |
| 3692 | ret = -EINVAL; |
| 3693 | goto done; |
| 3694 | } |
| 3695 | |
| 3696 | if (prtd->audio_client == NULL) { |
| 3697 | pr_err("%s: audio_client is null\n", __func__); |
| 3698 | ret = -EINVAL; |
| 3699 | goto done; |
| 3700 | } |
| 3701 | |
| 3702 | event_data = (struct msm_adsp_event_data *)ucontrol->value.bytes.data; |
| 3703 | if ((event_data->event_type < ADSP_STREAM_PP_EVENT) || |
| 3704 | (event_data->event_type >= ADSP_STREAM_EVENT_MAX)) { |
| 3705 | pr_err("%s: invalid event_type=%d", |
| 3706 | __func__, event_data->event_type); |
| 3707 | ret = -EINVAL; |
| 3708 | goto done; |
| 3709 | } |
| 3710 | |
Aditya Bavanari | 2e3341d | 2018-02-23 12:58:57 +0530 | [diff] [blame] | 3711 | actual_payload_len = sizeof(struct msm_adsp_event_data) + |
| 3712 | event_data->payload_len; |
| 3713 | if (actual_payload_len >= U32_MAX) { |
| 3714 | pr_err("%s payload length 0x%X exceeds limit", |
| 3715 | __func__, event_data->payload_len); |
| 3716 | ret = -EINVAL; |
| 3717 | goto done; |
| 3718 | } |
| 3719 | |
Xiaojun Sang | ae3d886 | 2018-03-23 08:57:33 +0800 | [diff] [blame] | 3720 | if (event_data->payload_len > sizeof(ucontrol->value.bytes.data) |
| 3721 | - sizeof(struct msm_adsp_event_data)) { |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 3722 | pr_err("%s param length=%d exceeds limit", |
| 3723 | __func__, event_data->payload_len); |
| 3724 | ret = -EINVAL; |
| 3725 | goto done; |
| 3726 | } |
| 3727 | |
| 3728 | ret = q6asm_send_stream_cmd(prtd->audio_client, event_data); |
| 3729 | if (ret < 0) |
| 3730 | pr_err("%s: failed to send stream event cmd, err = %d\n", |
| 3731 | __func__, ret); |
| 3732 | done: |
| 3733 | return ret; |
| 3734 | } |
| 3735 | |
| 3736 | static int msm_compr_ion_fd_map_put(struct snd_kcontrol *kcontrol, |
| 3737 | struct snd_ctl_elem_value *ucontrol) |
| 3738 | { |
| 3739 | struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| 3740 | unsigned long fe_id = kcontrol->private_value; |
| 3741 | struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| 3742 | snd_soc_component_get_drvdata(comp); |
| 3743 | struct snd_compr_stream *cstream = NULL; |
| 3744 | struct msm_compr_audio *prtd; |
| 3745 | int fd; |
| 3746 | int ret = 0; |
| 3747 | |
| 3748 | if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| 3749 | pr_err("%s Received out of bounds invalid fe_id %lu\n", |
| 3750 | __func__, fe_id); |
| 3751 | ret = -EINVAL; |
| 3752 | goto done; |
| 3753 | } |
| 3754 | |
| 3755 | cstream = pdata->cstream[fe_id]; |
| 3756 | if (cstream == NULL) { |
| 3757 | pr_err("%s cstream is null\n", __func__); |
| 3758 | ret = -EINVAL; |
| 3759 | goto done; |
| 3760 | } |
| 3761 | |
| 3762 | prtd = cstream->runtime->private_data; |
| 3763 | if (!prtd) { |
| 3764 | pr_err("%s: prtd is null\n", __func__); |
| 3765 | ret = -EINVAL; |
| 3766 | goto done; |
| 3767 | } |
| 3768 | |
| 3769 | if (prtd->audio_client == NULL) { |
| 3770 | pr_err("%s: audio_client is null\n", __func__); |
| 3771 | ret = -EINVAL; |
| 3772 | goto done; |
| 3773 | } |
| 3774 | |
| 3775 | memcpy(&fd, ucontrol->value.bytes.data, sizeof(fd)); |
| 3776 | ret = q6asm_send_ion_fd(prtd->audio_client, fd); |
| 3777 | if (ret < 0) |
| 3778 | pr_err("%s: failed to register ion fd\n", __func__); |
| 3779 | done: |
| 3780 | return ret; |
| 3781 | } |
| 3782 | |
| 3783 | static int msm_compr_rtic_event_ack_put(struct snd_kcontrol *kcontrol, |
| 3784 | struct snd_ctl_elem_value *ucontrol) |
| 3785 | { |
| 3786 | struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| 3787 | unsigned long fe_id = kcontrol->private_value; |
| 3788 | struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| 3789 | snd_soc_component_get_drvdata(comp); |
| 3790 | struct snd_compr_stream *cstream = NULL; |
| 3791 | struct msm_compr_audio *prtd; |
| 3792 | int ret = 0; |
| 3793 | int param_length = 0; |
| 3794 | |
| 3795 | if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| 3796 | pr_err("%s Received invalid fe_id %lu\n", |
| 3797 | __func__, fe_id); |
| 3798 | ret = -EINVAL; |
| 3799 | goto done; |
| 3800 | } |
| 3801 | |
| 3802 | cstream = pdata->cstream[fe_id]; |
| 3803 | if (cstream == NULL) { |
| 3804 | pr_err("%s cstream is null\n", __func__); |
| 3805 | ret = -EINVAL; |
| 3806 | goto done; |
| 3807 | } |
| 3808 | |
| 3809 | prtd = cstream->runtime->private_data; |
| 3810 | if (!prtd) { |
| 3811 | pr_err("%s: prtd is null\n", __func__); |
| 3812 | ret = -EINVAL; |
| 3813 | goto done; |
| 3814 | } |
| 3815 | |
| 3816 | if (prtd->audio_client == NULL) { |
| 3817 | pr_err("%s: audio_client is null\n", __func__); |
| 3818 | ret = -EINVAL; |
| 3819 | goto done; |
| 3820 | } |
| 3821 | |
| 3822 | memcpy(¶m_length, ucontrol->value.bytes.data, |
| 3823 | sizeof(param_length)); |
| 3824 | if ((param_length + sizeof(param_length)) |
| 3825 | >= sizeof(ucontrol->value.bytes.data)) { |
| 3826 | pr_err("%s param length=%d exceeds limit", |
| 3827 | __func__, param_length); |
| 3828 | ret = -EINVAL; |
| 3829 | goto done; |
| 3830 | } |
| 3831 | |
| 3832 | ret = q6asm_send_rtic_event_ack(prtd->audio_client, |
| 3833 | ucontrol->value.bytes.data + sizeof(param_length), |
| 3834 | param_length); |
| 3835 | if (ret < 0) |
| 3836 | pr_err("%s: failed to send rtic event ack, err = %d\n", |
| 3837 | __func__, ret); |
| 3838 | done: |
| 3839 | return ret; |
| 3840 | } |
| 3841 | |
| 3842 | static int msm_compr_gapless_put(struct snd_kcontrol *kcontrol, |
| 3843 | struct snd_ctl_elem_value *ucontrol) |
| 3844 | { |
| 3845 | struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| 3846 | struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| 3847 | snd_soc_component_get_drvdata(comp); |
| 3848 | pdata->use_dsp_gapless_mode = ucontrol->value.integer.value[0]; |
| 3849 | pr_debug("%s: value: %ld\n", __func__, |
| 3850 | ucontrol->value.integer.value[0]); |
| 3851 | |
| 3852 | return 0; |
| 3853 | } |
| 3854 | |
| 3855 | static int msm_compr_gapless_get(struct snd_kcontrol *kcontrol, |
| 3856 | struct snd_ctl_elem_value *ucontrol) |
| 3857 | { |
| 3858 | struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| 3859 | struct msm_compr_pdata *pdata = |
| 3860 | snd_soc_component_get_drvdata(comp); |
| 3861 | pr_debug("%s:gapless mode %d\n", __func__, pdata->use_dsp_gapless_mode); |
| 3862 | ucontrol->value.integer.value[0] = pdata->use_dsp_gapless_mode; |
| 3863 | |
| 3864 | return 0; |
| 3865 | } |
| 3866 | |
| 3867 | static const struct snd_kcontrol_new msm_compr_gapless_controls[] = { |
| 3868 | SOC_SINGLE_EXT("Compress Gapless Playback", |
| 3869 | 0, 0, 1, 0, |
| 3870 | msm_compr_gapless_get, |
| 3871 | msm_compr_gapless_put), |
| 3872 | }; |
| 3873 | |
| 3874 | static int msm_compr_probe(struct snd_soc_platform *platform) |
| 3875 | { |
| 3876 | struct msm_compr_pdata *pdata; |
| 3877 | int i; |
| 3878 | int rc; |
| 3879 | const char *qdsp_version; |
| 3880 | |
| 3881 | pr_debug("%s\n", __func__); |
| 3882 | pdata = (struct msm_compr_pdata *) |
| 3883 | kzalloc(sizeof(*pdata), GFP_KERNEL); |
| 3884 | if (!pdata) |
| 3885 | return -ENOMEM; |
| 3886 | |
| 3887 | snd_soc_platform_set_drvdata(platform, pdata); |
| 3888 | |
| 3889 | for (i = 0; i < MSM_FRONTEND_DAI_MAX; i++) { |
| 3890 | pdata->volume[i][0] = COMPRESSED_LR_VOL_MAX_STEPS; |
| 3891 | pdata->volume[i][1] = COMPRESSED_LR_VOL_MAX_STEPS; |
| 3892 | pdata->audio_effects[i] = NULL; |
| 3893 | pdata->dec_params[i] = NULL; |
| 3894 | pdata->cstream[i] = NULL; |
| 3895 | pdata->ch_map[i] = NULL; |
Aditya Bavanari | 9deef91 | 2017-11-20 13:31:31 +0530 | [diff] [blame] | 3896 | pdata->is_in_use[i] = false; |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 3897 | } |
| 3898 | |
| 3899 | snd_soc_add_platform_controls(platform, msm_compr_gapless_controls, |
| 3900 | ARRAY_SIZE(msm_compr_gapless_controls)); |
| 3901 | |
| 3902 | rc = of_property_read_string(platform->dev->of_node, |
| 3903 | "qcom,adsp-version", &qdsp_version); |
| 3904 | if (!rc) { |
| 3905 | if (!strcmp(qdsp_version, "MDSP 1.2")) |
| 3906 | pdata->use_legacy_api = true; |
| 3907 | else |
| 3908 | pdata->use_legacy_api = false; |
| 3909 | } else |
| 3910 | pdata->use_legacy_api = false; |
| 3911 | |
| 3912 | pr_debug("%s: use legacy api %d\n", __func__, pdata->use_legacy_api); |
| 3913 | /* |
| 3914 | * use_dsp_gapless_mode part of platform data(pdata) is updated from HAL |
| 3915 | * through a mixer control before compress driver is opened. The mixer |
| 3916 | * control is used to decide if dsp gapless mode needs to be enabled. |
| 3917 | * Gapless is disabled by default. |
| 3918 | */ |
| 3919 | pdata->use_dsp_gapless_mode = false; |
| 3920 | return 0; |
| 3921 | } |
| 3922 | |
Dhanalakshmi Siddani | 040e026 | 2018-11-26 23:01:26 +0530 | [diff] [blame] | 3923 | static int msm_compr_chmix_cfg_ctl_info(struct snd_kcontrol *kcontrol, |
| 3924 | struct snd_ctl_elem_info *uinfo) |
| 3925 | { |
| 3926 | uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| 3927 | uinfo->count = 128; |
| 3928 | uinfo->value.integer.min = 0; |
| 3929 | uinfo->value.integer.max = 0xFFFFFFFF; |
| 3930 | return 0; |
| 3931 | } |
| 3932 | |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 3933 | static int msm_compr_volume_info(struct snd_kcontrol *kcontrol, |
| 3934 | struct snd_ctl_elem_info *uinfo) |
| 3935 | { |
| 3936 | uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| 3937 | uinfo->count = 2; |
| 3938 | uinfo->value.integer.min = 0; |
| 3939 | uinfo->value.integer.max = COMPRESSED_LR_VOL_MAX_STEPS; |
| 3940 | return 0; |
| 3941 | } |
| 3942 | |
| 3943 | static int msm_compr_audio_effects_config_info(struct snd_kcontrol *kcontrol, |
| 3944 | struct snd_ctl_elem_info *uinfo) |
| 3945 | { |
| 3946 | uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| 3947 | uinfo->count = MAX_PP_PARAMS_SZ; |
| 3948 | uinfo->value.integer.min = 0; |
| 3949 | uinfo->value.integer.max = 0xFFFFFFFF; |
| 3950 | return 0; |
| 3951 | } |
| 3952 | |
| 3953 | static int msm_compr_query_audio_effect_info(struct snd_kcontrol *kcontrol, |
| 3954 | struct snd_ctl_elem_info *uinfo) |
| 3955 | { |
| 3956 | uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| 3957 | uinfo->count = 128; |
| 3958 | uinfo->value.integer.min = 0; |
| 3959 | uinfo->value.integer.max = 0xFFFFFFFF; |
| 3960 | return 0; |
| 3961 | } |
| 3962 | |
| 3963 | static int msm_compr_dec_params_info(struct snd_kcontrol *kcontrol, |
| 3964 | struct snd_ctl_elem_info *uinfo) |
| 3965 | { |
| 3966 | uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| 3967 | uinfo->count = 128; |
| 3968 | uinfo->value.integer.min = 0; |
| 3969 | uinfo->value.integer.max = 0xFFFFFFFF; |
| 3970 | return 0; |
| 3971 | } |
| 3972 | |
| 3973 | static int msm_compr_app_type_cfg_info(struct snd_kcontrol *kcontrol, |
| 3974 | struct snd_ctl_elem_info *uinfo) |
| 3975 | { |
| 3976 | uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| 3977 | uinfo->count = 5; |
| 3978 | uinfo->value.integer.min = 0; |
| 3979 | uinfo->value.integer.max = 0xFFFFFFFF; |
| 3980 | return 0; |
| 3981 | } |
| 3982 | |
| 3983 | static int msm_compr_channel_map_info(struct snd_kcontrol *kcontrol, |
| 3984 | struct snd_ctl_elem_info *uinfo) |
| 3985 | { |
| 3986 | uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
Dieter Luecking | ba7644d | 2018-09-28 15:09:32 +0200 | [diff] [blame] | 3987 | uinfo->count = PCM_FORMAT_MAX_NUM_CHANNEL_V8; |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 3988 | uinfo->value.integer.min = 0; |
Dieter Luecking | ba7644d | 2018-09-28 15:09:32 +0200 | [diff] [blame] | 3989 | /* See PCM_CHANNEL_RSD=34 in apr_audio-v2.h */ |
| 3990 | uinfo->value.integer.max = 34; |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 3991 | return 0; |
| 3992 | } |
| 3993 | |
| 3994 | static int msm_compr_add_volume_control(struct snd_soc_pcm_runtime *rtd) |
| 3995 | { |
| 3996 | const char *mixer_ctl_name = "Compress Playback"; |
| 3997 | const char *deviceNo = "NN"; |
| 3998 | const char *suffix = "Volume"; |
| 3999 | char *mixer_str = NULL; |
| 4000 | int ctl_len; |
| 4001 | struct snd_kcontrol_new fe_volume_control[1] = { |
| 4002 | { |
| 4003 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| 4004 | .name = "?", |
| 4005 | .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | |
| 4006 | SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| 4007 | .info = msm_compr_volume_info, |
| 4008 | .tlv.p = msm_compr_vol_gain, |
| 4009 | .get = msm_compr_volume_get, |
| 4010 | .put = msm_compr_volume_put, |
| 4011 | .private_value = 0, |
| 4012 | } |
| 4013 | }; |
| 4014 | |
| 4015 | if (!rtd) { |
| 4016 | pr_err("%s NULL rtd\n", __func__); |
| 4017 | return 0; |
| 4018 | } |
| 4019 | pr_debug("%s: added new compr FE with name %s, id %d, cpu dai %s, device no %d\n", |
| 4020 | __func__, rtd->dai_link->name, rtd->dai_link->id, |
| 4021 | rtd->dai_link->cpu_dai_name, rtd->pcm->device); |
| 4022 | ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1 + |
| 4023 | strlen(suffix) + 1; |
| 4024 | mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| 4025 | if (!mixer_str) { |
| 4026 | pr_err("failed to allocate mixer ctrl str of len %d", ctl_len); |
| 4027 | return 0; |
| 4028 | } |
| 4029 | snprintf(mixer_str, ctl_len, "%s %d %s", mixer_ctl_name, |
| 4030 | rtd->pcm->device, suffix); |
| 4031 | fe_volume_control[0].name = mixer_str; |
| 4032 | fe_volume_control[0].private_value = rtd->dai_link->id; |
| 4033 | pr_debug("Registering new mixer ctl %s", mixer_str); |
| 4034 | snd_soc_add_platform_controls(rtd->platform, fe_volume_control, |
| 4035 | ARRAY_SIZE(fe_volume_control)); |
| 4036 | kfree(mixer_str); |
| 4037 | return 0; |
| 4038 | } |
| 4039 | |
| 4040 | static int msm_compr_add_audio_effects_control(struct snd_soc_pcm_runtime *rtd) |
| 4041 | { |
| 4042 | const char *mixer_ctl_name = "Audio Effects Config"; |
| 4043 | const char *deviceNo = "NN"; |
| 4044 | char *mixer_str = NULL; |
| 4045 | int ctl_len; |
| 4046 | struct snd_kcontrol_new fe_audio_effects_config_control[1] = { |
| 4047 | { |
| 4048 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| 4049 | .name = "?", |
| 4050 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| 4051 | .info = msm_compr_audio_effects_config_info, |
| 4052 | .get = msm_compr_audio_effects_config_get, |
| 4053 | .put = msm_compr_audio_effects_config_put, |
| 4054 | .private_value = 0, |
| 4055 | } |
| 4056 | }; |
| 4057 | |
| 4058 | |
| 4059 | if (!rtd) { |
| 4060 | pr_err("%s NULL rtd\n", __func__); |
| 4061 | return 0; |
| 4062 | } |
| 4063 | |
| 4064 | pr_debug("%s: added new compr FE with name %s, id %d, cpu dai %s, device no %d\n", |
| 4065 | __func__, rtd->dai_link->name, rtd->dai_link->id, |
| 4066 | rtd->dai_link->cpu_dai_name, rtd->pcm->device); |
| 4067 | |
| 4068 | ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1; |
| 4069 | mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| 4070 | |
| 4071 | if (!mixer_str) |
| 4072 | return 0; |
| 4073 | |
| 4074 | snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device); |
| 4075 | |
| 4076 | fe_audio_effects_config_control[0].name = mixer_str; |
| 4077 | fe_audio_effects_config_control[0].private_value = rtd->dai_link->id; |
| 4078 | pr_debug("Registering new mixer ctl %s\n", mixer_str); |
| 4079 | snd_soc_add_platform_controls(rtd->platform, |
| 4080 | fe_audio_effects_config_control, |
| 4081 | ARRAY_SIZE(fe_audio_effects_config_control)); |
| 4082 | kfree(mixer_str); |
| 4083 | return 0; |
| 4084 | } |
| 4085 | |
| 4086 | static int msm_compr_add_query_audio_effect_control( |
| 4087 | struct snd_soc_pcm_runtime *rtd) |
| 4088 | { |
| 4089 | const char *mixer_ctl_name = "Query Audio Effect Param"; |
| 4090 | const char *deviceNo = "NN"; |
| 4091 | char *mixer_str = NULL; |
| 4092 | int ctl_len; |
| 4093 | struct snd_kcontrol_new fe_query_audio_effect_control[1] = { |
| 4094 | { |
| 4095 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| 4096 | .name = "?", |
| 4097 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| 4098 | .info = msm_compr_query_audio_effect_info, |
| 4099 | .get = msm_compr_query_audio_effect_get, |
| 4100 | .put = msm_compr_query_audio_effect_put, |
| 4101 | .private_value = 0, |
| 4102 | } |
| 4103 | }; |
| 4104 | if (!rtd) { |
| 4105 | pr_err("%s NULL rtd\n", __func__); |
| 4106 | return 0; |
| 4107 | } |
| 4108 | pr_debug("%s: added new compr FE with name %s, id %d, cpu dai %s, device no %d\n", |
| 4109 | __func__, rtd->dai_link->name, rtd->dai_link->id, |
| 4110 | rtd->dai_link->cpu_dai_name, rtd->pcm->device); |
| 4111 | ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1; |
| 4112 | mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| 4113 | if (!mixer_str) { |
| 4114 | pr_err("failed to allocate mixer ctrl str of len %d", ctl_len); |
| 4115 | return 0; |
| 4116 | } |
| 4117 | snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device); |
| 4118 | fe_query_audio_effect_control[0].name = mixer_str; |
| 4119 | fe_query_audio_effect_control[0].private_value = rtd->dai_link->id; |
| 4120 | pr_debug("%s: registering new mixer ctl %s\n", __func__, mixer_str); |
| 4121 | snd_soc_add_platform_controls(rtd->platform, |
| 4122 | fe_query_audio_effect_control, |
| 4123 | ARRAY_SIZE(fe_query_audio_effect_control)); |
| 4124 | kfree(mixer_str); |
| 4125 | return 0; |
| 4126 | } |
| 4127 | |
| 4128 | static int msm_compr_add_audio_adsp_stream_cmd_control( |
| 4129 | struct snd_soc_pcm_runtime *rtd) |
| 4130 | { |
| 4131 | const char *mixer_ctl_name = DSP_STREAM_CMD; |
| 4132 | const char *deviceNo = "NN"; |
| 4133 | char *mixer_str = NULL; |
| 4134 | int ctl_len = 0, ret = 0; |
| 4135 | struct snd_kcontrol_new fe_audio_adsp_stream_cmd_config_control[1] = { |
| 4136 | { |
| 4137 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| 4138 | .name = "?", |
| 4139 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| 4140 | .info = msm_adsp_stream_cmd_info, |
| 4141 | .put = msm_compr_adsp_stream_cmd_put, |
| 4142 | .private_value = 0, |
| 4143 | } |
| 4144 | }; |
| 4145 | |
| 4146 | if (!rtd) { |
| 4147 | pr_err("%s NULL rtd\n", __func__); |
| 4148 | ret = -EINVAL; |
| 4149 | goto done; |
| 4150 | } |
| 4151 | |
| 4152 | ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1; |
| 4153 | mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| 4154 | if (!mixer_str) { |
| 4155 | ret = -ENOMEM; |
| 4156 | goto done; |
| 4157 | } |
| 4158 | |
| 4159 | snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device); |
| 4160 | fe_audio_adsp_stream_cmd_config_control[0].name = mixer_str; |
| 4161 | fe_audio_adsp_stream_cmd_config_control[0].private_value = |
| 4162 | rtd->dai_link->id; |
| 4163 | pr_debug("%s: Registering new mixer ctl %s\n", __func__, mixer_str); |
| 4164 | ret = snd_soc_add_platform_controls(rtd->platform, |
| 4165 | fe_audio_adsp_stream_cmd_config_control, |
| 4166 | ARRAY_SIZE(fe_audio_adsp_stream_cmd_config_control)); |
| 4167 | if (ret < 0) |
| 4168 | pr_err("%s: failed to add ctl %s. err = %d\n", |
| 4169 | __func__, mixer_str, ret); |
| 4170 | |
| 4171 | kfree(mixer_str); |
| 4172 | done: |
| 4173 | return ret; |
| 4174 | } |
| 4175 | |
| 4176 | static int msm_compr_add_audio_adsp_stream_callback_control( |
| 4177 | struct snd_soc_pcm_runtime *rtd) |
| 4178 | { |
| 4179 | const char *mixer_ctl_name = DSP_STREAM_CALLBACK; |
| 4180 | const char *deviceNo = "NN"; |
| 4181 | char *mixer_str = NULL; |
| 4182 | int ctl_len = 0, ret = 0; |
| 4183 | struct snd_kcontrol *kctl; |
| 4184 | |
| 4185 | struct snd_kcontrol_new fe_audio_adsp_callback_config_control[1] = { |
| 4186 | { |
| 4187 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| 4188 | .name = "?", |
| 4189 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| 4190 | .info = msm_adsp_stream_callback_info, |
| 4191 | .get = msm_adsp_stream_callback_get, |
| 4192 | .private_value = 0, |
| 4193 | } |
| 4194 | }; |
| 4195 | |
| 4196 | if (!rtd) { |
| 4197 | pr_err("%s: rtd is NULL\n", __func__); |
| 4198 | ret = -EINVAL; |
| 4199 | goto done; |
| 4200 | } |
| 4201 | |
| 4202 | ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1; |
| 4203 | mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| 4204 | if (!mixer_str) { |
| 4205 | ret = -ENOMEM; |
| 4206 | goto done; |
| 4207 | } |
| 4208 | |
| 4209 | snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device); |
| 4210 | fe_audio_adsp_callback_config_control[0].name = mixer_str; |
| 4211 | fe_audio_adsp_callback_config_control[0].private_value = |
| 4212 | rtd->dai_link->id; |
| 4213 | pr_debug("%s: Registering new mixer ctl %s\n", __func__, mixer_str); |
| 4214 | ret = snd_soc_add_platform_controls(rtd->platform, |
| 4215 | fe_audio_adsp_callback_config_control, |
| 4216 | ARRAY_SIZE(fe_audio_adsp_callback_config_control)); |
| 4217 | if (ret < 0) { |
| 4218 | pr_err("%s: failed to add ctl %s. err = %d\n", |
| 4219 | __func__, mixer_str, ret); |
| 4220 | ret = -EINVAL; |
| 4221 | goto free_mixer_str; |
| 4222 | } |
| 4223 | |
| 4224 | kctl = snd_soc_card_get_kcontrol(rtd->card, mixer_str); |
| 4225 | if (!kctl) { |
| 4226 | pr_err("%s: failed to get kctl %s.\n", __func__, mixer_str); |
| 4227 | ret = -EINVAL; |
| 4228 | goto free_mixer_str; |
| 4229 | } |
| 4230 | |
| 4231 | kctl->private_data = NULL; |
| 4232 | |
| 4233 | free_mixer_str: |
| 4234 | kfree(mixer_str); |
| 4235 | done: |
| 4236 | return ret; |
| 4237 | } |
| 4238 | |
| 4239 | static int msm_compr_add_dec_runtime_params_control( |
| 4240 | struct snd_soc_pcm_runtime *rtd) |
| 4241 | { |
| 4242 | const char *mixer_ctl_name = "Audio Stream"; |
| 4243 | const char *deviceNo = "NN"; |
| 4244 | const char *suffix = "Dec Params"; |
| 4245 | char *mixer_str = NULL; |
| 4246 | int ctl_len; |
| 4247 | struct snd_kcontrol_new fe_dec_params_control[1] = { |
| 4248 | { |
| 4249 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| 4250 | .name = "?", |
| 4251 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| 4252 | .info = msm_compr_dec_params_info, |
| 4253 | .get = msm_compr_dec_params_get, |
| 4254 | .put = msm_compr_dec_params_put, |
| 4255 | .private_value = 0, |
| 4256 | } |
| 4257 | }; |
| 4258 | |
| 4259 | if (!rtd) { |
| 4260 | pr_err("%s NULL rtd\n", __func__); |
| 4261 | return 0; |
| 4262 | } |
| 4263 | |
| 4264 | pr_debug("%s: added new compr FE with name %s, id %d, cpu dai %s, device no %d\n", |
| 4265 | __func__, rtd->dai_link->name, rtd->dai_link->id, |
| 4266 | rtd->dai_link->cpu_dai_name, rtd->pcm->device); |
| 4267 | |
| 4268 | ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1 + |
| 4269 | strlen(suffix) + 1; |
| 4270 | mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| 4271 | |
| 4272 | if (!mixer_str) |
| 4273 | return 0; |
| 4274 | |
| 4275 | snprintf(mixer_str, ctl_len, "%s %d %s", mixer_ctl_name, |
| 4276 | rtd->pcm->device, suffix); |
| 4277 | |
| 4278 | fe_dec_params_control[0].name = mixer_str; |
| 4279 | fe_dec_params_control[0].private_value = rtd->dai_link->id; |
| 4280 | pr_debug("Registering new mixer ctl %s", mixer_str); |
| 4281 | snd_soc_add_platform_controls(rtd->platform, |
| 4282 | fe_dec_params_control, |
| 4283 | ARRAY_SIZE(fe_dec_params_control)); |
| 4284 | kfree(mixer_str); |
| 4285 | return 0; |
| 4286 | } |
| 4287 | |
| 4288 | static int msm_compr_add_app_type_cfg_control(struct snd_soc_pcm_runtime *rtd) |
| 4289 | { |
| 4290 | const char *playback_mixer_ctl_name = "Audio Stream"; |
| 4291 | const char *capture_mixer_ctl_name = "Audio Stream Capture"; |
| 4292 | const char *deviceNo = "NN"; |
| 4293 | const char *suffix = "App Type Cfg"; |
| 4294 | char *mixer_str = NULL; |
| 4295 | int ctl_len; |
| 4296 | struct snd_kcontrol_new fe_app_type_cfg_control[1] = { |
| 4297 | { |
| 4298 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| 4299 | .name = "?", |
| 4300 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| 4301 | .info = msm_compr_app_type_cfg_info, |
| 4302 | .put = msm_compr_playback_app_type_cfg_put, |
| 4303 | .get = msm_compr_playback_app_type_cfg_get, |
| 4304 | .private_value = 0, |
| 4305 | } |
| 4306 | }; |
| 4307 | |
| 4308 | if (!rtd) { |
| 4309 | pr_err("%s NULL rtd\n", __func__); |
| 4310 | return 0; |
| 4311 | } |
| 4312 | |
| 4313 | pr_debug("%s: added new compr FE ctl with name %s, id %d, cpu dai %s, device no %d\n", |
| 4314 | __func__, rtd->dai_link->name, rtd->dai_link->id, |
| 4315 | rtd->dai_link->cpu_dai_name, rtd->pcm->device); |
| 4316 | if (rtd->compr->direction == SND_COMPRESS_PLAYBACK) |
| 4317 | ctl_len = strlen(playback_mixer_ctl_name) + 1 + strlen(deviceNo) |
| 4318 | + 1 + strlen(suffix) + 1; |
| 4319 | else |
| 4320 | ctl_len = strlen(capture_mixer_ctl_name) + 1 + strlen(deviceNo) |
| 4321 | + 1 + strlen(suffix) + 1; |
| 4322 | |
| 4323 | mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| 4324 | |
| 4325 | if (!mixer_str) |
| 4326 | return 0; |
| 4327 | |
| 4328 | if (rtd->compr->direction == SND_COMPRESS_PLAYBACK) |
| 4329 | snprintf(mixer_str, ctl_len, "%s %d %s", |
| 4330 | playback_mixer_ctl_name, rtd->pcm->device, suffix); |
| 4331 | else |
| 4332 | snprintf(mixer_str, ctl_len, "%s %d %s", |
| 4333 | capture_mixer_ctl_name, rtd->pcm->device, suffix); |
| 4334 | |
| 4335 | fe_app_type_cfg_control[0].name = mixer_str; |
| 4336 | fe_app_type_cfg_control[0].private_value = rtd->dai_link->id; |
| 4337 | |
| 4338 | if (rtd->compr->direction == SND_COMPRESS_PLAYBACK) { |
| 4339 | fe_app_type_cfg_control[0].put = |
| 4340 | msm_compr_playback_app_type_cfg_put; |
| 4341 | fe_app_type_cfg_control[0].get = |
| 4342 | msm_compr_playback_app_type_cfg_get; |
| 4343 | } else { |
| 4344 | fe_app_type_cfg_control[0].put = |
| 4345 | msm_compr_capture_app_type_cfg_put; |
| 4346 | fe_app_type_cfg_control[0].get = |
| 4347 | msm_compr_capture_app_type_cfg_get; |
| 4348 | } |
| 4349 | pr_debug("Registering new mixer ctl %s", mixer_str); |
| 4350 | snd_soc_add_platform_controls(rtd->platform, |
| 4351 | fe_app_type_cfg_control, |
| 4352 | ARRAY_SIZE(fe_app_type_cfg_control)); |
| 4353 | kfree(mixer_str); |
| 4354 | return 0; |
| 4355 | } |
| 4356 | |
Dhanalakshmi Siddani | 040e026 | 2018-11-26 23:01:26 +0530 | [diff] [blame] | 4357 | static int msm_compr_chmix_cfg_ctl_put(struct snd_kcontrol *kcontrol, |
| 4358 | struct snd_ctl_elem_value *ucontrol) |
| 4359 | { |
| 4360 | struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); |
| 4361 | struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) |
| 4362 | snd_soc_component_get_drvdata(comp); |
| 4363 | struct snd_compr_stream *cstream = NULL; |
| 4364 | struct snd_soc_pcm_runtime *rtd = NULL; |
| 4365 | u64 fe_id = kcontrol->private_value; |
| 4366 | int ip_channel_cnt, op_channel_cnt; |
| 4367 | int i, index = 0; |
| 4368 | int ch_coeff[PCM_FORMAT_MAX_NUM_CHANNEL * PCM_FORMAT_MAX_NUM_CHANNEL]; |
| 4369 | bool use_default_chmap = true; |
| 4370 | char *chmap = NULL; |
| 4371 | |
| 4372 | pr_debug("%s: fe_id- %llu\n", __func__, fe_id); |
| 4373 | if (fe_id >= MSM_FRONTEND_DAI_MAX) { |
| 4374 | pr_err("%s: Received out of bounds fe_id %llu\n", |
| 4375 | __func__, fe_id); |
| 4376 | return -EINVAL; |
| 4377 | } |
| 4378 | cstream = pdata->cstream[fe_id]; |
| 4379 | if (!cstream) { |
| 4380 | pr_err("%s: stream inactive\n", __func__); |
| 4381 | return -EINVAL; |
| 4382 | } |
| 4383 | rtd = cstream->private_data; |
| 4384 | if (!rtd) { |
| 4385 | pr_err("%s: stream inactive\n", __func__); |
| 4386 | return -EINVAL; |
| 4387 | } |
| 4388 | |
| 4389 | use_default_chmap = !(pdata->ch_map[rtd->dai_link->id]->set_ch_map); |
| 4390 | chmap = pdata->ch_map[rtd->dai_link->id]->channel_map; |
| 4391 | |
| 4392 | ip_channel_cnt = ucontrol->value.integer.value[index++]; |
| 4393 | op_channel_cnt = ucontrol->value.integer.value[index++]; |
| 4394 | /* |
| 4395 | * wght coeff of first out channel corresponding to each in channel |
| 4396 | * are sent followed by second out channel for each in channel etc. |
| 4397 | */ |
| 4398 | memset(ch_coeff, 0, sizeof(ch_coeff)); |
| 4399 | for (i = 0; i < op_channel_cnt * ip_channel_cnt; i++) { |
| 4400 | ch_coeff[i] = |
| 4401 | ucontrol->value.integer.value[index++]; |
| 4402 | } |
| 4403 | |
| 4404 | msm_pcm_routing_send_chmix_cfg(fe_id, ip_channel_cnt, op_channel_cnt, |
| 4405 | ch_coeff, SESSION_TYPE_RX, use_default_chmap, chmap); |
| 4406 | |
| 4407 | return 0; |
| 4408 | } |
| 4409 | |
| 4410 | static int msm_compr_chmix_cfg_ctl_get(struct snd_kcontrol *kcontrol, |
| 4411 | struct snd_ctl_elem_value *ucontrol) |
| 4412 | { |
| 4413 | return 0; |
| 4414 | } |
| 4415 | |
| 4416 | static int msm_compr_add_chmix_cfg_controls(struct snd_soc_pcm_runtime *rtd) |
| 4417 | { |
| 4418 | const char *mixer_ctl_name = "Audio Stream"; |
| 4419 | const char *deviceNo = "NN"; |
| 4420 | const char *suffix = "Channel Mix Cfg"; |
| 4421 | int ctl_len; |
| 4422 | char *mixer_str = NULL; |
| 4423 | struct snd_kcontrol_new chmix_cfg_controls[1] = { |
| 4424 | { |
| 4425 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| 4426 | .name = "?", |
| 4427 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| 4428 | .info = msm_compr_chmix_cfg_ctl_info, |
| 4429 | .get = msm_compr_chmix_cfg_ctl_get, |
| 4430 | .put = msm_compr_chmix_cfg_ctl_put, |
| 4431 | .private_value = 0, |
| 4432 | } |
| 4433 | }; |
| 4434 | |
| 4435 | if (!rtd) { |
| 4436 | pr_err("%s NULL rtd\n", __func__); |
| 4437 | return -EINVAL; |
| 4438 | } |
| 4439 | |
| 4440 | pr_debug("%s: added new compr FE with name %s, id %d, cpu dai %s, device no %d\n", |
| 4441 | __func__, rtd->dai_link->name, rtd->dai_link->id, |
| 4442 | rtd->dai_link->cpu_dai_name, rtd->pcm->device); |
| 4443 | |
| 4444 | ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1 + |
| 4445 | strlen(suffix) + 1; |
| 4446 | mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| 4447 | if (!mixer_str) |
| 4448 | return -ENOMEM; |
| 4449 | |
| 4450 | snprintf(mixer_str, ctl_len, "%s %d %s", mixer_ctl_name, |
| 4451 | rtd->pcm->device, suffix); |
| 4452 | |
| 4453 | chmix_cfg_controls[0].name = mixer_str; |
| 4454 | chmix_cfg_controls[0].private_value = rtd->dai_link->id; |
| 4455 | pr_debug("%s: Registering new mixer ctl %s", __func__, mixer_str); |
| 4456 | snd_soc_add_platform_controls(rtd->platform, |
| 4457 | chmix_cfg_controls, |
| 4458 | ARRAY_SIZE(chmix_cfg_controls)); |
| 4459 | kfree(mixer_str); |
| 4460 | return 0; |
| 4461 | } |
| 4462 | |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 4463 | static int msm_compr_add_channel_map_control(struct snd_soc_pcm_runtime *rtd) |
| 4464 | { |
| 4465 | const char *mixer_ctl_name = "Playback Channel Map"; |
| 4466 | const char *deviceNo = "NN"; |
| 4467 | char *mixer_str = NULL; |
| 4468 | struct msm_compr_pdata *pdata = NULL; |
| 4469 | int ctl_len; |
| 4470 | struct snd_kcontrol_new fe_channel_map_control[1] = { |
| 4471 | { |
| 4472 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| 4473 | .name = "?", |
| 4474 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| 4475 | .info = msm_compr_channel_map_info, |
| 4476 | .get = msm_compr_channel_map_get, |
| 4477 | .put = msm_compr_channel_map_put, |
| 4478 | .private_value = 0, |
| 4479 | } |
| 4480 | }; |
| 4481 | |
| 4482 | if (!rtd) { |
| 4483 | pr_err("%s: NULL rtd\n", __func__); |
| 4484 | return -EINVAL; |
| 4485 | } |
| 4486 | |
| 4487 | pr_debug("%s: added new compr FE with name %s, id %d, cpu dai %s, device no %d\n", |
| 4488 | __func__, rtd->dai_link->name, rtd->dai_link->id, |
| 4489 | rtd->dai_link->cpu_dai_name, rtd->pcm->device); |
| 4490 | |
| 4491 | ctl_len = strlen(mixer_ctl_name) + strlen(deviceNo) + 1; |
| 4492 | mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| 4493 | |
| 4494 | if (!mixer_str) |
| 4495 | return -ENOMEM; |
| 4496 | |
| 4497 | snprintf(mixer_str, ctl_len, "%s%d", mixer_ctl_name, rtd->pcm->device); |
| 4498 | |
| 4499 | fe_channel_map_control[0].name = mixer_str; |
| 4500 | fe_channel_map_control[0].private_value = rtd->dai_link->id; |
| 4501 | pr_debug("%s: Registering new mixer ctl %s\n", __func__, mixer_str); |
| 4502 | snd_soc_add_platform_controls(rtd->platform, |
| 4503 | fe_channel_map_control, |
| 4504 | ARRAY_SIZE(fe_channel_map_control)); |
| 4505 | |
| 4506 | pdata = snd_soc_platform_get_drvdata(rtd->platform); |
| 4507 | pdata->ch_map[rtd->dai_link->id] = |
| 4508 | kzalloc(sizeof(struct msm_compr_ch_map), GFP_KERNEL); |
| 4509 | if (!pdata->ch_map[rtd->dai_link->id]) { |
| 4510 | pr_err("%s: Could not allocate memory for channel map\n", |
| 4511 | __func__); |
| 4512 | kfree(mixer_str); |
| 4513 | return -ENOMEM; |
| 4514 | } |
| 4515 | kfree(mixer_str); |
| 4516 | return 0; |
| 4517 | } |
| 4518 | |
| 4519 | static int msm_compr_add_io_fd_cmd_control(struct snd_soc_pcm_runtime *rtd) |
| 4520 | { |
| 4521 | const char *mixer_ctl_name = "Playback ION FD"; |
| 4522 | const char *deviceNo = "NN"; |
| 4523 | char *mixer_str = NULL; |
| 4524 | int ctl_len = 0, ret = 0; |
| 4525 | struct snd_kcontrol_new fe_ion_fd_config_control[1] = { |
| 4526 | { |
| 4527 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| 4528 | .name = "?", |
| 4529 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| 4530 | .info = msm_adsp_stream_cmd_info, |
| 4531 | .put = msm_compr_ion_fd_map_put, |
| 4532 | .private_value = 0, |
| 4533 | } |
| 4534 | }; |
| 4535 | |
| 4536 | if (!rtd) { |
| 4537 | pr_err("%s NULL rtd\n", __func__); |
| 4538 | ret = -EINVAL; |
| 4539 | goto done; |
| 4540 | } |
| 4541 | |
| 4542 | ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1; |
| 4543 | mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| 4544 | if (!mixer_str) { |
| 4545 | ret = -ENOMEM; |
| 4546 | goto done; |
| 4547 | } |
| 4548 | |
| 4549 | snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device); |
| 4550 | fe_ion_fd_config_control[0].name = mixer_str; |
| 4551 | fe_ion_fd_config_control[0].private_value = rtd->dai_link->id; |
| 4552 | pr_debug("%s: Registering new mixer ctl %s\n", __func__, mixer_str); |
| 4553 | ret = snd_soc_add_platform_controls(rtd->platform, |
| 4554 | fe_ion_fd_config_control, |
| 4555 | ARRAY_SIZE(fe_ion_fd_config_control)); |
| 4556 | if (ret < 0) |
| 4557 | pr_err("%s: failed to add ctl %s\n", __func__, mixer_str); |
| 4558 | |
| 4559 | kfree(mixer_str); |
| 4560 | done: |
| 4561 | return ret; |
| 4562 | } |
| 4563 | |
| 4564 | static int msm_compr_add_event_ack_cmd_control(struct snd_soc_pcm_runtime *rtd) |
| 4565 | { |
| 4566 | const char *mixer_ctl_name = "Playback Event Ack"; |
| 4567 | const char *deviceNo = "NN"; |
| 4568 | char *mixer_str = NULL; |
| 4569 | int ctl_len = 0, ret = 0; |
| 4570 | struct snd_kcontrol_new fe_event_ack_config_control[1] = { |
| 4571 | { |
| 4572 | .iface = SNDRV_CTL_ELEM_IFACE_MIXER, |
| 4573 | .name = "?", |
| 4574 | .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, |
| 4575 | .info = msm_adsp_stream_cmd_info, |
| 4576 | .put = msm_compr_rtic_event_ack_put, |
| 4577 | .private_value = 0, |
| 4578 | } |
| 4579 | }; |
| 4580 | |
| 4581 | if (!rtd) { |
| 4582 | pr_err("%s NULL rtd\n", __func__); |
| 4583 | ret = -EINVAL; |
| 4584 | goto done; |
| 4585 | } |
| 4586 | |
| 4587 | ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1; |
| 4588 | mixer_str = kzalloc(ctl_len, GFP_KERNEL); |
| 4589 | if (!mixer_str) { |
| 4590 | ret = -ENOMEM; |
| 4591 | goto done; |
| 4592 | } |
| 4593 | |
| 4594 | snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device); |
| 4595 | fe_event_ack_config_control[0].name = mixer_str; |
| 4596 | fe_event_ack_config_control[0].private_value = rtd->dai_link->id; |
| 4597 | pr_debug("%s: Registering new mixer ctl %s\n", __func__, mixer_str); |
| 4598 | ret = snd_soc_add_platform_controls(rtd->platform, |
| 4599 | fe_event_ack_config_control, |
| 4600 | ARRAY_SIZE(fe_event_ack_config_control)); |
| 4601 | if (ret < 0) |
| 4602 | pr_err("%s: failed to add ctl %s\n", __func__, mixer_str); |
| 4603 | |
| 4604 | kfree(mixer_str); |
| 4605 | done: |
| 4606 | return ret; |
| 4607 | } |
| 4608 | |
| 4609 | static int msm_compr_new(struct snd_soc_pcm_runtime *rtd) |
| 4610 | { |
| 4611 | int rc; |
| 4612 | |
| 4613 | rc = msm_compr_add_volume_control(rtd); |
| 4614 | if (rc) |
| 4615 | pr_err("%s: Could not add Compr Volume Control\n", __func__); |
| 4616 | |
| 4617 | rc = msm_compr_add_audio_effects_control(rtd); |
| 4618 | if (rc) |
| 4619 | pr_err("%s: Could not add Compr Audio Effects Control\n", |
| 4620 | __func__); |
| 4621 | |
| 4622 | rc = msm_compr_add_audio_adsp_stream_cmd_control(rtd); |
| 4623 | if (rc) |
| 4624 | pr_err("%s: Could not add Compr ADSP Stream Cmd Control\n", |
| 4625 | __func__); |
| 4626 | |
| 4627 | rc = msm_compr_add_audio_adsp_stream_callback_control(rtd); |
| 4628 | if (rc) |
| 4629 | pr_err("%s: Could not add Compr ADSP Stream Callback Control\n", |
| 4630 | __func__); |
| 4631 | |
| 4632 | rc = msm_compr_add_io_fd_cmd_control(rtd); |
| 4633 | if (rc) |
| 4634 | pr_err("%s: Could not add Compr ion fd Control\n", |
| 4635 | __func__); |
| 4636 | |
| 4637 | rc = msm_compr_add_event_ack_cmd_control(rtd); |
| 4638 | if (rc) |
| 4639 | pr_err("%s: Could not add Compr event ack Control\n", |
| 4640 | __func__); |
| 4641 | |
| 4642 | rc = msm_compr_add_query_audio_effect_control(rtd); |
| 4643 | if (rc) |
| 4644 | pr_err("%s: Could not add Compr Query Audio Effect Control\n", |
| 4645 | __func__); |
| 4646 | |
| 4647 | rc = msm_compr_add_dec_runtime_params_control(rtd); |
| 4648 | if (rc) |
| 4649 | pr_err("%s: Could not add Compr Dec runtime params Control\n", |
| 4650 | __func__); |
| 4651 | rc = msm_compr_add_app_type_cfg_control(rtd); |
| 4652 | if (rc) |
| 4653 | pr_err("%s: Could not add Compr App Type Cfg Control\n", |
| 4654 | __func__); |
| 4655 | rc = msm_compr_add_channel_map_control(rtd); |
| 4656 | if (rc) |
| 4657 | pr_err("%s: Could not add Compr Channel Map Control\n", |
| 4658 | __func__); |
Dhanalakshmi Siddani | 040e026 | 2018-11-26 23:01:26 +0530 | [diff] [blame] | 4659 | rc = msm_compr_add_chmix_cfg_controls(rtd); |
| 4660 | if (rc) |
| 4661 | pr_err("%s: add chmix cfg controls failed:%d\n", __func__, rc); |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 4662 | return 0; |
| 4663 | } |
| 4664 | |
| 4665 | static struct snd_compr_ops msm_compr_ops = { |
| 4666 | .open = msm_compr_open, |
| 4667 | .free = msm_compr_free, |
| 4668 | .trigger = msm_compr_trigger, |
| 4669 | .pointer = msm_compr_pointer, |
| 4670 | .set_params = msm_compr_set_params, |
| 4671 | .set_metadata = msm_compr_set_metadata, |
| 4672 | .get_metadata = msm_compr_get_metadata, |
| 4673 | .set_next_track_param = msm_compr_set_next_track_param, |
| 4674 | .ack = msm_compr_ack, |
| 4675 | .copy = msm_compr_copy, |
| 4676 | .get_caps = msm_compr_get_caps, |
| 4677 | .get_codec_caps = msm_compr_get_codec_caps, |
| 4678 | }; |
| 4679 | |
| 4680 | static struct snd_soc_platform_driver msm_soc_platform = { |
| 4681 | .probe = msm_compr_probe, |
| 4682 | .compr_ops = &msm_compr_ops, |
| 4683 | .pcm_new = msm_compr_new, |
| 4684 | }; |
| 4685 | |
| 4686 | static int msm_compr_dev_probe(struct platform_device *pdev) |
| 4687 | { |
| 4688 | |
| 4689 | pr_debug("%s: dev name %s\n", __func__, dev_name(&pdev->dev)); |
| 4690 | return snd_soc_register_platform(&pdev->dev, |
| 4691 | &msm_soc_platform); |
| 4692 | } |
| 4693 | |
| 4694 | static int msm_compr_remove(struct platform_device *pdev) |
| 4695 | { |
| 4696 | snd_soc_unregister_platform(&pdev->dev); |
| 4697 | return 0; |
| 4698 | } |
| 4699 | |
| 4700 | static const struct of_device_id msm_compr_dt_match[] = { |
| 4701 | {.compatible = "qcom,msm-compress-dsp"}, |
| 4702 | {} |
| 4703 | }; |
| 4704 | MODULE_DEVICE_TABLE(of, msm_compr_dt_match); |
| 4705 | |
| 4706 | static struct platform_driver msm_compr_driver = { |
| 4707 | .driver = { |
| 4708 | .name = "msm-compress-dsp", |
| 4709 | .owner = THIS_MODULE, |
| 4710 | .of_match_table = msm_compr_dt_match, |
| 4711 | }, |
| 4712 | .probe = msm_compr_dev_probe, |
| 4713 | .remove = msm_compr_remove, |
| 4714 | }; |
| 4715 | |
Laxminath Kasam | 8b1366a | 2017-10-05 01:44:16 +0530 | [diff] [blame] | 4716 | int __init msm_compress_dsp_init(void) |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 4717 | { |
| 4718 | return platform_driver_register(&msm_compr_driver); |
| 4719 | } |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 4720 | |
Asish Bhattacharya | 5faacb3 | 2017-12-04 17:23:15 +0530 | [diff] [blame] | 4721 | void msm_compress_dsp_exit(void) |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 4722 | { |
| 4723 | platform_driver_unregister(&msm_compr_driver); |
| 4724 | } |
Asish Bhattacharya | 8e2277f | 2017-07-20 18:31:55 +0530 | [diff] [blame] | 4725 | |
| 4726 | MODULE_DESCRIPTION("Compress Offload platform driver"); |
| 4727 | MODULE_LICENSE("GPL v2"); |