1# SPDX-License-Identifier: BSD-3-Clause 2 3menu "Audio components" 4 5config COMP_COPIER 6 bool "COPIER component" 7 default y 8 depends on IPC_MAJOR_4 9 help 10 Select for COPIER component 11config COMP_DAI 12 bool "DAI component" 13 default y 14 help 15 Select for DAI component 16 17config COMP_VOLUME 18 bool "Volume component" 19 default y 20 help 21 Select for Volume component 22 23config COMP_SRC 24 bool "SRC component" 25 default y 26 help 27 Select for SRC component 28 29if COMP_SRC 30 31choice 32 prompt "SRC coefficient set" 33 default COMP_SRC_STD 34 35config COMP_SRC_STD 36 bool "Full conversions support and high quality" 37 help 38 This coefficients set supports a large number of conversions 39 with 8 - 192 kHz input and 8 - 48 kHz output. The coefficients 40 are 32 bits for high conversion quality. The coefficients 41 storage consume 79 kB. The runtime needs 8 kB. Use this if there 42 is critical quality usage for SRC like music via headphones and 43 need high sample rates. 44 45config COMP_SRC_SMALL 46 bool "Smaller conversions support and high quality" 47 help 48 This coefficients set supports a small number of conversions 49 with 8 - 48 kHz input and output. The coefficients are 32 bits 50 for high conversion quality. The coefficients storage consumes 51 30 kB RAM. Runtime needs 4 kB. Use this if there is critical 52 quality usage for SRC like music via headphones. 53 54config COMP_SRC_TINY 55 bool "Reduced conversions support and reduced quality" 56 help 57 This coefficients set supports a minimum number of conversions 58 with 8 - 48 kHz input and output. The coefficients are 16 bits 59 that restricts the possible stop-attenuation and could cause 60 some audible aliasing. The bandwidth of the filters is restricted 61 so the full 20 kHz band is not met even if sample rate would 62 allow. The coefficients storage consumes 5 kB RAM. The runtime 63 needs 6 kB. Use this to save RAM for other applications when SRC 64 has no critical usage or when only need with lower quality 65 endpoint like miniature speakers. 66 67endchoice 68 69endif # SRC 70 71config MATH_FIR 72 bool "FIR filter library" 73 default n 74 help 75 This option builds FIR (Finite Impulse Response) filter library. It 76 is selected by components for their digital signal processing. A FIR 77 filter calculates a convolution of input PCM sample and a configurable 78 impulse response. 79 80config MATH_FFT 81 bool "FFT library" 82 default n 83 help 84 Enable Fast Fourier Transform library, this should not be selected 85 directly, please select it from other audio components where need it. 86 87config COMP_FIR 88 bool "FIR component" 89 select MATH_FIR 90 default y 91 help 92 Select for FIR component. FIR performance can differ between DSP 93 ISA and compiler used. FIR will perform best when MAC instructions 94 are available in ISA and are used by the compiler. i.e. xt-xcc on 95 xtensa will generate MAC instructions but GCC on xtensa won't. 96 Filter tap count can be severely restricted to reduce FIR cycles 97 and FIR performance for DSP/compilers with no MAC support 98 99config COMP_IIR 100 bool "IIR component" 101 default y 102 help 103 Select for IIR component 104 105config CORDIC_FIXED 106 bool "Sine Cosine library function" 107 default n 108 help 109 This option builds the 32 and 16 bit sin_fixed() library function. 110 The input is Q4.28 format and the output is Q1.31. The cordic sine 111 cos algorithm converges, when the angle is in the range [-pi/2, pi/2). 112 If an angle is outside of this range, then a multiple of pi/2 is added 113 or subtracted from the angle until it is within the range [-pi/2,pi/2). 114 Start with the angle in the range [-2*pi, 2*pi) and output has range in 115 [-1.0 to 1.0] 116 117config COMP_TONE 118 bool "Tone component" 119 default y 120 select CORDIC_FIXED 121 help 122 Select for Tone component 123 124config COMP_MIXER 125 bool "Mixer component" 126 default y 127 help 128 Select for Mixer component 129 130config COMP_MUX 131 bool "MUX component" 132 default y 133 help 134 Select for MUX component 135 136config COMP_SWITCH 137 bool "Switch component" 138 default y 139 help 140 Select for Switch component 141 142config COMP_KPB 143 bool "KPB component" 144 default y 145 help 146 Select for KPB component 147if COMP_KPB 148 149config KPB_FORCE_COPY_TYPE_NORMAL 150 bool "KPB force copy type normal" 151 default y 152 help 153 Select this to force the kpb draining copy type to normal. 154 Unselecting this will keep the kpb sink copy type unchanged. 155 156endif # COMP_KPB 157 158config COMP_GOOGLE_HOTWORD_DETECT 159 bool "Google hotword detector component" 160 default n 161 help 162 Select for Google hotword detector component. It uses the Google 163 hotword library to do keyword detection. A language model needs to 164 be set using the byte control 'Hotword Model' before running the 165 detector. 166 167config COMP_SEL 168 bool "Channel selector component" 169 default y 170 help 171 Select for SEL component 172 173config COMP_CROSSOVER 174 bool "Crossover Filter component" 175 default n 176 help 177 Select for Crossover Filter component. A crossover can be used to 178 split a signal into two or more frequency ranges, so that the outputs 179 can be sent to drivers that are designed for those ranges. 180 181config COMP_DRC 182 bool "Dynamic Range Compressor component" 183 select CORDIC_FIXED 184 default n 185 help 186 Select for Dynamic Range Compressor (DRC) component. A DRC can be used 187 to reduce the volume of loud sounds and amplify silent sounds thus 188 compressing an audio signal's dynamic range. 189 190config DRC_MAX_PRE_DELAY_FRAMES 191 depends on COMP_DRC 192 int "DRC max number of pre-delay frames" 193 default 512 194 help 195 Determines the length of pre-delay frame buffers which are allocated 196 runtime on DRC setup. It requires to be a 2^N number. 512 is 197 suggested by inference to avoid memory waste and provide reasonable 198 length for pre-delay frames. 199 200config COMP_MULTIBAND_DRC 201 depends on COMP_IIR && COMP_CROSSOVER && COMP_DRC 202 bool "Multiband Dynamic Range Compressor component" 203 select CORDIC_FIXED 204 default n 205 help 206 Select for Multiband Dynamic Range Compressor (DRC) component. It 207 consists of Emphasis Equalizer, n-way Crossover Filter, per-band DRC, 208 and Deemphasis Equalizer. 209 210config COMP_DCBLOCK 211 bool "DC Blocking Filter component" 212 default y 213 help 214 Select for DC Blocking Filter component. This component filters out 215 the DC offset which often originates from a microphone's output. 216 217config MAXIM_DSM 218 depends on CAVS && !CAVS_VERSION_1_5 219 bool "Maxim DSM component" 220 default n 221 help 222 Select for Maxim DSM(Dynamic Speaker Management) component. 223 This component protect the speaker from overheating and 224 excursion violation. This consists of two parts which are 225 feedforward processing block running on playback thread and 226 feedback processing block running on capture thread with 227 amp current and voltage feedback. 228 229config COMP_ASRC 230 bool "ASRC component" 231 default y 232 help 233 Select for Asynchronous sample rate conversion (ASRC) 234 component. The ASRC has the capabilities of (synchronous) 235 SRC but can in addition track a slave DAI that is not in 236 sync with firmware timer. The ASRC can be setup for 237 synchronous operation without DAI tracking via topology 238 though it has a bit higher computational load than SRC. The 239 RAM footprint of ASRC is smaller than SRC because it does 240 not have pre-computed filter coefficients for every 241 conversion fraction that SRC does. 242 243if COMP_ASRC 244 245choice 246 prompt "ASRC down sampling conversions set" 247 default COMP_ASRC_DOWNSAMPLING_FULL 248 249config COMP_ASRC_DOWNSAMPLING_FULL 250 bool "Full downsampling conversions support" 251 select ASRC_SUPPORT_CONVERSION_24000_TO_08000 252 select ASRC_SUPPORT_CONVERSION_24000_TO_16000 253 select ASRC_SUPPORT_CONVERSION_48000_TO_08000 254 select ASRC_SUPPORT_CONVERSION_48000_TO_11025 255 select ASRC_SUPPORT_CONVERSION_48000_TO_12000 256 select ASRC_SUPPORT_CONVERSION_48000_TO_16000 257 select ASRC_SUPPORT_CONVERSION_48000_TO_22050 258 select ASRC_SUPPORT_CONVERSION_48000_TO_24000 259 select ASRC_SUPPORT_CONVERSION_48000_TO_32000 260 select ASRC_SUPPORT_CONVERSION_48000_TO_44100 261 help 262 This option enables all supported downsampling conversions 263 for the asynchronous sample rate conversion. All the 264 upsampling conversions use the same filter always enabled 265 coefficients set so there is no configuration options for 266 them. The enabled conversions consume about 18 kB of memory. 267 268config COMP_ASRC_DOWNSAMPLING_CUSTOM 269 bool "Custom downsampling conversions support" 270 help 271 This option used to manually select each downsampling ratio. 272 In order to optimize the text code size of the sample rate 273 converter, non-required conversion ratios can be 274 deactivated. Disregarding these settings, the sample rate 275 converter always supports all conversion ratios with fs_in 276 less or equal to fs_out. Therefore, it is possible to 277 deactivate all conversion ratios listed below, if we only 278 need an (asynchronous) 1:1 sample rate conversion, e.g, from 279 16 kHz to 16 kHz. This option allows to trim down the coefficient 280 memory for conversion down from default 18 kB. Each conversion 281 needs about 2 kB. 282 283endchoice 284 285menu "Supported downsampling conversions" 286 visible if COMP_ASRC_DOWNSAMPLING_CUSTOM 287 288config ASRC_SUPPORT_CONVERSION_24000_TO_08000 289 bool "Downsample 24 kHz to 8 kHz" 290 default n 291 help 292 This option enables downsampling from 24 kHz to 8 kHz into 293 the build. The consumption of memory is 2.1 kB. 294 295config ASRC_SUPPORT_CONVERSION_24000_TO_16000 296 bool "Downsample 24 kHz to 16 kHz" 297 default n 298 help 299 This option enables downsampling from 24 kHz to 16 kHz into 300 the build. The consumption of memory is 1.9 kB. 301 302config ASRC_SUPPORT_CONVERSION_48000_TO_08000 303 bool "Downsample 48 kHz to 8 kHz" 304 default n 305 help 306 This option enables downsampling from 48 kHz to 8 kHz into 307 the build. The consumption of memory is 2.1 kB. 308 309config ASRC_SUPPORT_CONVERSION_48000_TO_11025 310 bool "Downsample 48 kHz to 11.025 kHz" 311 default n 312 help 313 This option enables downsampling from 48 kHz to 11.025 kHz 314 into the build. The consumption of memory is 1.5 kB. 315 316config ASRC_SUPPORT_CONVERSION_48000_TO_12000 317 bool "Downsample 48 kHz to 12 kHz" 318 default n 319 help 320 This option enables downsampling from 48 kHz to 12 kHz into 321 the build. The consumption of memory is 1.5 kB. 322 323config ASRC_SUPPORT_CONVERSION_48000_TO_16000 324 bool "Downsample 48 kHz to 16 kHz" 325 default n 326 help 327 This option enables downsampling from 48 kHz to 16 kHz into 328 the build. The consumption of memory is 1.9 kB. 329 330config ASRC_SUPPORT_CONVERSION_48000_TO_22050 331 bool "Downsample 48 kHz to 22.05 kHz" 332 default n 333 help 334 This option enables downsampling from 48 kHz to 22.05 kHz into 335 the build. The consumption of memory is 1.6 kB. 336 337config ASRC_SUPPORT_CONVERSION_48000_TO_24000 338 bool "Downsample 48 kHz to 24 kHz" 339 default n 340 help 341 This option enables downsampling from 48 kHz to 24 kHz into 342 the build. The consumption of memory is 1.6 kB. 343 344config ASRC_SUPPORT_CONVERSION_48000_TO_32000 345 bool "Downsample 48 kHz to 32 kHz" 346 default n 347 help 348 This option enables downsampling from 48 kHz to 32 kHz into 349 the build. The consumption of memory is 1.9 kB. 350 351config ASRC_SUPPORT_CONVERSION_48000_TO_44100 352 bool "Downsample 48 kHz to 44.1 kHz" 353 default n 354 help 355 This option enables downsampling from 48 kHz to 44.1 kHz 356 into the build. The consumption of memory is 1.8 kB. 357 358endmenu # "Downsampling ratios" 359 360endif # COMP_ASRC 361 362config COMP_TDFB 363 bool "TDFB component" 364 select MATH_FIR 365 default y 366 help 367 Select for time domain fixed beamformer (TDFB) component. The 368 beamformer component enhances microphone capture by spatial 369 suppression of noise and improvement of signal-to-noise ratio of 370 capture from beam direction. The component is essentially a generic 371 single rate FIR filter bank that performs microphone array 372 directivity enhancement when programmed with suitable configuration 373 for channels selection, channel filter coefficients, and output 374 streams mixing. 375 376config COMP_CODEC_ADAPTER 377 bool "Codec adapter" 378 default n 379 help 380 This component is an adapter between SoF components and any external third 381 party codecs/libraries. In order to make use of it the library itself should 382 be statically linked with the SoF FW binary image and the codec details, such as its 383 ID or specific methods provided in generic interface object located under 384 "src\include\sof\audio\codec_adapter\interfaces.h". It is possible to link several 385 different codecs and use them in parallel. 386 387rsource "codec_adapter/Kconfig" 388 389config COMP_IGO_NR 390 bool "IGO NR component" 391 default n 392 help 393 This option enables Intelligo non-speech noise reduction. The feature links to a proprietary 394 binary libigonr.a that currently is supported on different Xtensa DSP platforms. Please email 395 info@intelli-go.com for any questions about the binary. 396 397config COMP_RTNR 398 bool "RTNR component" 399 default n 400 help 401 Select for Realtek noise reduction/suppression(NR/NS) component. 402 Noise Suppression technology reduces stationary and transient noises in 403 single-channel speech signals, which increases the signal-to-noise ratio, 404 improves speech intelligibility and reduces listening fatigue. It estimates 405 ambient noise and signal levels and then passes or attenuates the signals 406 according to improve signal-to-noise ratios. The feature links to a 407 proprietary binary libSOF_RTK_MA_API.a, libSuite_rename.a, libNet.a and libPreset.a. 408 Please contact antz0525@realtek.com for any question about the binary. 409 410endmenu # "Audio components" 411 412menu "Data formats" 413 414config FORMAT_S16LE 415 bool "Support S16LE" 416 default y 417 help 418 Support 16 bit processing data format with sign and in little endian format 419 420config FORMAT_S24LE 421 bool "Support S24LE" 422 default y 423 help 424 Support 24 bit processing data format with sign and in little endian format 425 426config FORMAT_S32LE 427 bool "Support S32LE" 428 default y 429 help 430 Support 32 bit processing data format with sign and in little endian format 431 432config FORMAT_FLOAT 433 bool "Support float" 434 default y 435 help 436 Support floating point processing data format 437 438config FORMAT_CONVERT_HIFI3 439 bool "HIFI3 optimized conversion" 440 default y 441 help 442 Use HIFI3 extensions for optimized format conversion (experimental). 443 444config PCM_CONVERTER_FORMAT_S16LE 445 bool "Support S16LE" 446 default y 447 help 448 Support 16 bit processing data format with sign and in little endian format 449 450config PCM_CONVERTER_FORMAT_S24LE 451 bool "Support S24LE" 452 default y 453 help 454 Support 24 bit processing data format with sign and in little endian format 455 456config PCM_CONVERTER_FORMAT_S32LE 457 bool "Support S32LE" 458 default y 459 help 460 Support 32 bit processing data format with sign and in little endian format 461 462config PCM_CONVERTER_FORMAT_FLOAT 463 bool "Support float" 464 default y 465 help 466 Support floating point processing data format 467 468config PCM_CONVERTER_FORMAT_S16_C16_AND_S16_C32 469 bool "Support S16C16 <-> S16C32" 470 default n 471 help 472 Support conversion between 16 bit valid sample size in 16 bit container 473 and 16 bit valid sample size in 32 bit container 474 475config PCM_CONVERTER_FORMAT_S16_C32_AND_S32_C32 476 bool "Support S16C32 <-> S32C32" 477 default n 478 help 479 Support conversion between 16 bit valid sample size in 32 bit container 480 and 32 bit valid sample size in 32 bit container 481 482config PCM_CONVERTER_FORMAT_S16_C32_AND_S24_C32 483 bool "Support S16C32 <-> S24C32" 484 default n 485 help 486 Support conversion between 16 bit valid sample size in 32 bit container 487 and 24 bit valid sample size in 32 bit container 488 489config PCM_CONVERTER_FORMAT_CONVERT_HIFI3 490 bool "HIFI3 optimized conversion" 491 default y 492 help 493 Use HIFI3 extensions for optimized format conversion (experimental). 494 495config TRACE_CHANNEL 496 int "TRACE DMA Channel configuration" 497 default 0 498 help 499 This option is used for custom configuration of trace dma channel other 500 than zero dma channel.Define macro DMA_TRACE_CHANNEL in platform to 501 configure the required channel number. 502 503config WRAP_ACTUAL_POSITION 504 bool "Buffer wrapping" 505 default n 506 help 507 This option is to update the actual position information on wrap of buffer. 508 It is not necessary that on wrap, the buffer position would be zero.At wrap, 509 in some cases based on the period size, the frame may not exactly be at the 510 end of the buffer and roll over for some bytes from the beginning of the buffer. 511endmenu 512
