1 /* 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 /* 12 * entropy_coding.h 13 * 14 * This header file declares all of the functions used to arithmetically 15 * encode the iSAC bistream 16 * 17 */ 18 19 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_ 20 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_ 21 22 #include "settings.h" 23 #include "structs.h" 24 25 /****************************************************************************** 26 * WebRtcIsac_DecodeSpec() 27 * Decode real and imaginary part of the DFT coefficients, given a bit-stream. 28 * The decoded DFT coefficient can be transformed to time domain by 29 * WebRtcIsac_Time2Spec(). 30 * 31 * Input: 32 * - streamdata : pointer to a stucture containg the encoded 33 * data and theparameters needed for entropy 34 * coding. 35 * - AvgPitchGain_Q12 : average pitch-gain of the frame. This is only 36 * relevant for 0-4 kHz band, and the input value is 37 * not used in other bands. 38 * - band : specifies which band's DFT should be decoded. 39 * 40 * Output: 41 * - *fr : pointer to a buffer where the real part of DFT 42 * coefficients are written to. 43 * - *fi : pointer to a buffer where the imaginary part 44 * of DFT coefficients are written to. 45 * 46 * Return value : < 0 if an error occures 47 * 0 if succeeded. 48 */ 49 int WebRtcIsac_DecodeSpec(Bitstr* streamdata, WebRtc_Word16 AvgPitchGain_Q12, 50 enum ISACBand band, double* fr, double* fi); 51 52 /****************************************************************************** 53 * WebRtcIsac_EncodeSpec() 54 * Encode real and imaginary part of the DFT coefficients into the given 55 * bit-stream. 56 * 57 * Input: 58 * - *fr : pointer to a buffer where the real part of DFT 59 * coefficients are written to. 60 * - *fi : pointer to a buffer where the imaginary part 61 * of DFT coefficients are written to. 62 * - AvgPitchGain_Q12 : average pitch-gain of the frame. This is only 63 * relevant for 0-4 kHz band, and the input value is 64 * not used in other bands. 65 * - band : specifies which band's DFT should be decoded. 66 * 67 * Output: 68 * - streamdata : pointer to a stucture containg the encoded 69 * data and theparameters needed for entropy 70 * coding. 71 * 72 * Return value : < 0 if an error occures 73 * 0 if succeeded. 74 */ 75 int WebRtcIsac_EncodeSpec(const WebRtc_Word16* fr, const WebRtc_Word16* fi, 76 WebRtc_Word16 AvgPitchGain_Q12, enum ISACBand band, 77 Bitstr* streamdata); 78 79 /* decode & dequantize LPC Coef */ 80 int WebRtcIsac_DecodeLpcCoef(Bitstr* streamdata, double* LPCCoef); 81 int WebRtcIsac_DecodeLpcCoefUB(Bitstr* streamdata, double* lpcVecs, 82 double* percepFilterGains, 83 WebRtc_Word16 bandwidth); 84 85 int WebRtcIsac_DecodeLpc(Bitstr* streamdata, double* LPCCoef_lo, 86 double* LPCCoef_hi); 87 88 /* quantize & code LPC Coef */ 89 void WebRtcIsac_EncodeLpcLb(double* LPCCoef_lo, double* LPCCoef_hi, 90 Bitstr* streamdata, ISAC_SaveEncData_t* encData); 91 92 void WebRtcIsac_EncodeLpcGainLb(double* LPCCoef_lo, double* LPCCoef_hi, 93 Bitstr* streamdata, 94 ISAC_SaveEncData_t* encData); 95 96 /****************************************************************************** 97 * WebRtcIsac_EncodeLpcUB() 98 * Encode LPC parameters, given as A-polynomial, of upper-band. The encoding 99 * is performed in LAR domain. 100 * For the upper-band, we compute and encode LPC of some sub-frames, LPC of 101 * other sub-frames are computed by linear interpolation, in LAR domain. This 102 * function performs the interpolation and returns the LPC of all sub-frames. 103 * 104 * Inputs: 105 * - lpcCoef : a buffer containing A-polynomials of sub-frames 106 * (excluding first coefficient that is 1). 107 * - bandwidth : specifies if the codec is operating at 0-12 kHz 108 * or 0-16 kHz mode. 109 * 110 * Input/output: 111 * - streamdata : pointer to a structure containing the encoded 112 * data and the parameters needed for entropy 113 * coding. 114 * 115 * Output: 116 * - interpolLPCCoeff : Decoded and interpolated LPC (A-polynomial) 117 * of all sub-frames. 118 * If LP analysis is of order K, and there are N 119 * sub-frames then this is a buffer of size 120 * (k + 1) * N, each vector starts with the LPC gain 121 * of the corresponding sub-frame. The LPC gains 122 * are encoded and inserted after this function is 123 * called. The first A-coefficient which is 1 is not 124 * included. 125 * 126 * Return value : 0 if encoding is successful, 127 * <0 if failed to encode. 128 */ 129 WebRtc_Word16 WebRtcIsac_EncodeLpcUB(double* lpcCoeff, Bitstr* streamdata, 130 double* interpolLPCCoeff, 131 WebRtc_Word16 bandwidth, 132 ISACUBSaveEncDataStruct* encData); 133 134 /****************************************************************************** 135 * WebRtcIsac_DecodeInterpolLpcUb() 136 * Decode LPC coefficients and interpolate to get the coefficients fo all 137 * sub-frmaes. 138 * 139 * Inputs: 140 * - bandwidth : spepecifies if the codec is in 0-12 kHz or 141 * 0-16 kHz mode. 142 * 143 * Input/output: 144 * - streamdata : pointer to a stucture containg the encoded 145 * data and theparameters needed for entropy 146 * coding. 147 * 148 * Output: 149 * - percepFilterParam : Decoded and interpolated LPC (A-polynomial) of 150 * all sub-frames. 151 * If LP analysis is of order K, and there are N 152 * sub-frames then this is a buffer of size 153 * (k + 1) * N, each vector starts with the LPC gain 154 * of the corresponding sub-frame. The LPC gains 155 * are encoded and inserted after this function is 156 * called. The first A-coefficient which is 1 is not 157 * included. 158 * 159 * Return value : 0 if encoding is successful, 160 * <0 if failed to encode. 161 */ 162 WebRtc_Word16 WebRtcIsac_DecodeInterpolLpcUb(Bitstr* streamdata, 163 double* percepFilterParam, 164 WebRtc_Word16 bandwidth); 165 166 /* Decode & dequantize RC */ 167 int WebRtcIsac_DecodeRc(Bitstr* streamdata, WebRtc_Word16* RCQ15); 168 169 /* Quantize & code RC */ 170 void WebRtcIsac_EncodeRc(WebRtc_Word16* RCQ15, Bitstr* streamdata); 171 172 /* Decode & dequantize squared Gain */ 173 int WebRtcIsac_DecodeGain2(Bitstr* streamdata, WebRtc_Word32* Gain2); 174 175 /* Quantize & code squared Gain (input is squared gain) */ 176 int WebRtcIsac_EncodeGain2(WebRtc_Word32* gain2, Bitstr* streamdata); 177 178 void WebRtcIsac_EncodePitchGain(WebRtc_Word16* PitchGains_Q12, 179 Bitstr* streamdata, 180 ISAC_SaveEncData_t* encData); 181 182 void WebRtcIsac_EncodePitchLag(double* PitchLags, WebRtc_Word16* PitchGain_Q12, 183 Bitstr* streamdata, ISAC_SaveEncData_t* encData); 184 185 int WebRtcIsac_DecodePitchGain(Bitstr* streamdata, 186 WebRtc_Word16* PitchGain_Q12); 187 int WebRtcIsac_DecodePitchLag(Bitstr* streamdata, WebRtc_Word16* PitchGain_Q12, 188 double* PitchLag); 189 190 int WebRtcIsac_DecodeFrameLen(Bitstr* streamdata, WebRtc_Word16* framelength); 191 int WebRtcIsac_EncodeFrameLen(WebRtc_Word16 framelength, Bitstr* streamdata); 192 int WebRtcIsac_DecodeSendBW(Bitstr* streamdata, WebRtc_Word16* BWno); 193 void WebRtcIsac_EncodeReceiveBw(int* BWno, Bitstr* streamdata); 194 195 /* Step-down */ 196 void WebRtcIsac_Poly2Rc(double* a, int N, double* RC); 197 198 /* Step-up */ 199 void WebRtcIsac_Rc2Poly(double* RC, int N, double* a); 200 201 void WebRtcIsac_TranscodeLPCCoef(double* LPCCoef_lo, double* LPCCoef_hi, 202 int* index_g); 203 204 205 /****************************************************************************** 206 * WebRtcIsac_EncodeLpcGainUb() 207 * Encode LPC gains of sub-Frames. 208 * 209 * Input/outputs: 210 * - lpGains : a buffer which contains 'SUBFRAME' number of 211 * LP gains to be encoded. The input values are 212 * overwritten by the quantized values. 213 * - streamdata : pointer to a stucture containg the encoded 214 * data and theparameters needed for entropy 215 * coding. 216 * 217 * Output: 218 * - lpcGainIndex : quantization indices for lpc gains, these will 219 * be stored to be used for FEC. 220 */ 221 void WebRtcIsac_EncodeLpcGainUb(double* lpGains, Bitstr* streamdata, 222 int* lpcGainIndex); 223 224 225 /****************************************************************************** 226 * WebRtcIsac_EncodeLpcGainUb() 227 * Store LPC gains of sub-Frames in 'streamdata'. 228 * 229 * Input: 230 * - lpGains : a buffer which contains 'SUBFRAME' number of 231 * LP gains to be encoded. 232 * Input/outputs: 233 * - streamdata : pointer to a stucture containg the encoded 234 * data and theparameters needed for entropy 235 * coding. 236 * 237 */ 238 void WebRtcIsac_StoreLpcGainUb(double* lpGains, Bitstr* streamdata); 239 240 241 /****************************************************************************** 242 * WebRtcIsac_DecodeLpcGainUb() 243 * Decode the LPC gain of sub-frames. 244 * 245 * Input/output: 246 * - streamdata : pointer to a stucture containg the encoded 247 * data and theparameters needed for entropy 248 * coding. 249 * 250 * Output: 251 * - lpGains : a buffer where decoded LPC gians will be stored. 252 * 253 * Return value : 0 if succeeded. 254 * <0 if failed. 255 */ 256 WebRtc_Word16 WebRtcIsac_DecodeLpcGainUb(double* lpGains, Bitstr* streamdata); 257 258 259 /****************************************************************************** 260 * WebRtcIsac_EncodeBandwidth() 261 * Encode if the bandwidth of encoded audio is 0-12 kHz or 0-16 kHz. 262 * 263 * Input: 264 * - bandwidth : an enumerator specifying if the codec in is 265 * 0-12 kHz or 0-16 kHz mode. 266 * 267 * Input/output: 268 * - streamdata : pointer to a stucture containg the encoded 269 * data and theparameters needed for entropy 270 * coding. 271 * 272 * Return value : 0 if succeeded. 273 * <0 if failed. 274 */ 275 WebRtc_Word16 WebRtcIsac_EncodeBandwidth(enum ISACBandwidth bandwidth, 276 Bitstr* streamData); 277 278 279 /****************************************************************************** 280 * WebRtcIsac_DecodeBandwidth() 281 * Decode the bandwidth of the encoded audio, i.e. if the bandwidth is 0-12 kHz 282 * or 0-16 kHz. 283 * 284 * Input/output: 285 * - streamdata : pointer to a stucture containg the encoded 286 * data and theparameters needed for entropy 287 * coding. 288 * 289 * Output: 290 * - bandwidth : an enumerator specifying if the codec is in 291 * 0-12 kHz or 0-16 kHz mode. 292 * 293 * Return value : 0 if succeeded. 294 * <0 if failed. 295 */ 296 WebRtc_Word16 WebRtcIsac_DecodeBandwidth(Bitstr* streamData, 297 enum ISACBandwidth* bandwidth); 298 299 300 /****************************************************************************** 301 * WebRtcIsac_EncodeJitterInfo() 302 * Decode the jitter information. 303 * 304 * Input/output: 305 * - streamdata : pointer to a stucture containg the encoded 306 * data and theparameters needed for entropy 307 * coding. 308 * 309 * Input: 310 * - jitterInfo : one bit of info specifying if the channel is 311 * in high/low jitter. Zero indicates low jitter 312 * and one indicates high jitter. 313 * 314 * Return value : 0 if succeeded. 315 * <0 if failed. 316 */ 317 WebRtc_Word16 WebRtcIsac_EncodeJitterInfo(WebRtc_Word32 jitterIndex, 318 Bitstr* streamData); 319 320 321 /****************************************************************************** 322 * WebRtcIsac_DecodeJitterInfo() 323 * Decode the jitter information. 324 * 325 * Input/output: 326 * - streamdata : pointer to a stucture containg the encoded 327 * data and theparameters needed for entropy 328 * coding. 329 * 330 * Output: 331 * - jitterInfo : one bit of info specifying if the channel is 332 * in high/low jitter. Zero indicates low jitter 333 * and one indicates high jitter. 334 * 335 * Return value : 0 if succeeded. 336 * <0 if failed. 337 */ 338 WebRtc_Word16 WebRtcIsac_DecodeJitterInfo(Bitstr* streamData, 339 WebRtc_Word32* jitterInfo); 340 341 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_ */ 342
