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libavcodec/alacenc.c

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00001 
00022 #include "avcodec.h"
00023 #include "put_bits.h"
00024 #include "dsputil.h"
00025 #include "lpc.h"
00026 #include "mathops.h"
00027 
00028 #define DEFAULT_FRAME_SIZE        4096
00029 #define DEFAULT_SAMPLE_SIZE       16
00030 #define MAX_CHANNELS              8
00031 #define ALAC_EXTRADATA_SIZE       36
00032 #define ALAC_FRAME_HEADER_SIZE    55
00033 #define ALAC_FRAME_FOOTER_SIZE    3
00034 
00035 #define ALAC_ESCAPE_CODE          0x1FF
00036 #define ALAC_MAX_LPC_ORDER        30
00037 #define DEFAULT_MAX_PRED_ORDER    6
00038 #define DEFAULT_MIN_PRED_ORDER    4
00039 #define ALAC_MAX_LPC_PRECISION    9
00040 #define ALAC_MAX_LPC_SHIFT        9
00041 
00042 #define ALAC_CHMODE_LEFT_RIGHT    0
00043 #define ALAC_CHMODE_LEFT_SIDE     1
00044 #define ALAC_CHMODE_RIGHT_SIDE    2
00045 #define ALAC_CHMODE_MID_SIDE      3
00046 
00047 typedef struct RiceContext {
00048     int history_mult;
00049     int initial_history;
00050     int k_modifier;
00051     int rice_modifier;
00052 } RiceContext;
00053 
00054 typedef struct LPCContext {
00055     int lpc_order;
00056     int lpc_coeff[ALAC_MAX_LPC_ORDER+1];
00057     int lpc_quant;
00058 } LPCContext;
00059 
00060 typedef struct AlacEncodeContext {
00061     int compression_level;
00062     int min_prediction_order;
00063     int max_prediction_order;
00064     int max_coded_frame_size;
00065     int write_sample_size;
00066     int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE];
00067     int32_t predictor_buf[DEFAULT_FRAME_SIZE];
00068     int interlacing_shift;
00069     int interlacing_leftweight;
00070     PutBitContext pbctx;
00071     RiceContext rc;
00072     LPCContext lpc[MAX_CHANNELS];
00073     DSPContext dspctx;
00074     AVCodecContext *avctx;
00075 } AlacEncodeContext;
00076 
00077 
00078 static void init_sample_buffers(AlacEncodeContext *s, int16_t *input_samples)
00079 {
00080     int ch, i;
00081 
00082     for(ch=0;ch<s->avctx->channels;ch++) {
00083         int16_t *sptr = input_samples + ch;
00084         for(i=0;i<s->avctx->frame_size;i++) {
00085             s->sample_buf[ch][i] = *sptr;
00086             sptr += s->avctx->channels;
00087         }
00088     }
00089 }
00090 
00091 static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size)
00092 {
00093     int divisor, q, r;
00094 
00095     k = FFMIN(k, s->rc.k_modifier);
00096     divisor = (1<<k) - 1;
00097     q = x / divisor;
00098     r = x % divisor;
00099 
00100     if(q > 8) {
00101         // write escape code and sample value directly
00102         put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE);
00103         put_bits(&s->pbctx, write_sample_size, x);
00104     } else {
00105         if(q)
00106             put_bits(&s->pbctx, q, (1<<q) - 1);
00107         put_bits(&s->pbctx, 1, 0);
00108 
00109         if(k != 1) {
00110             if(r > 0)
00111                 put_bits(&s->pbctx, k, r+1);
00112             else
00113                 put_bits(&s->pbctx, k-1, 0);
00114         }
00115     }
00116 }
00117 
00118 static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
00119 {
00120     put_bits(&s->pbctx, 3,  s->avctx->channels-1);          // No. of channels -1
00121     put_bits(&s->pbctx, 16, 0);                             // Seems to be zero
00122     put_bits(&s->pbctx, 1,  1);                             // Sample count is in the header
00123     put_bits(&s->pbctx, 2,  0);                             // FIXME: Wasted bytes field
00124     put_bits(&s->pbctx, 1,  is_verbatim);                   // Audio block is verbatim
00125     put_bits32(&s->pbctx, s->avctx->frame_size);            // No. of samples in the frame
00126 }
00127 
00128 static void calc_predictor_params(AlacEncodeContext *s, int ch)
00129 {
00130     int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER];
00131     int shift[MAX_LPC_ORDER];
00132     int opt_order;
00133 
00134     if (s->compression_level == 1) {
00135         s->lpc[ch].lpc_order = 6;
00136         s->lpc[ch].lpc_quant = 6;
00137         s->lpc[ch].lpc_coeff[0] =  160;
00138         s->lpc[ch].lpc_coeff[1] = -190;
00139         s->lpc[ch].lpc_coeff[2] =  170;
00140         s->lpc[ch].lpc_coeff[3] = -130;
00141         s->lpc[ch].lpc_coeff[4] =   80;
00142         s->lpc[ch].lpc_coeff[5] =  -25;
00143     } else {
00144         opt_order = ff_lpc_calc_coefs(&s->dspctx, s->sample_buf[ch],
00145                                       s->avctx->frame_size,
00146                                       s->min_prediction_order,
00147                                       s->max_prediction_order,
00148                                       ALAC_MAX_LPC_PRECISION, coefs, shift, 1,
00149                                       ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1);
00150 
00151         s->lpc[ch].lpc_order = opt_order;
00152         s->lpc[ch].lpc_quant = shift[opt_order-1];
00153         memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int));
00154     }
00155 }
00156 
00157 static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
00158 {
00159     int i, best;
00160     int32_t lt, rt;
00161     uint64_t sum[4];
00162     uint64_t score[4];
00163 
00164     /* calculate sum of 2nd order residual for each channel */
00165     sum[0] = sum[1] = sum[2] = sum[3] = 0;
00166     for(i=2; i<n; i++) {
00167         lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2];
00168         rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2];
00169         sum[2] += FFABS((lt + rt) >> 1);
00170         sum[3] += FFABS(lt - rt);
00171         sum[0] += FFABS(lt);
00172         sum[1] += FFABS(rt);
00173     }
00174 
00175     /* calculate score for each mode */
00176     score[0] = sum[0] + sum[1];
00177     score[1] = sum[0] + sum[3];
00178     score[2] = sum[1] + sum[3];
00179     score[3] = sum[2] + sum[3];
00180 
00181     /* return mode with lowest score */
00182     best = 0;
00183     for(i=1; i<4; i++) {
00184         if(score[i] < score[best]) {
00185             best = i;
00186         }
00187     }
00188     return best;
00189 }
00190 
00191 static void alac_stereo_decorrelation(AlacEncodeContext *s)
00192 {
00193     int32_t *left = s->sample_buf[0], *right = s->sample_buf[1];
00194     int i, mode, n = s->avctx->frame_size;
00195     int32_t tmp;
00196 
00197     mode = estimate_stereo_mode(left, right, n);
00198 
00199     switch(mode)
00200     {
00201         case ALAC_CHMODE_LEFT_RIGHT:
00202             s->interlacing_leftweight = 0;
00203             s->interlacing_shift = 0;
00204             break;
00205 
00206         case ALAC_CHMODE_LEFT_SIDE:
00207             for(i=0; i<n; i++) {
00208                 right[i] = left[i] - right[i];
00209             }
00210             s->interlacing_leftweight = 1;
00211             s->interlacing_shift = 0;
00212             break;
00213 
00214         case ALAC_CHMODE_RIGHT_SIDE:
00215             for(i=0; i<n; i++) {
00216                 tmp = right[i];
00217                 right[i] = left[i] - right[i];
00218                 left[i] = tmp + (right[i] >> 31);
00219             }
00220             s->interlacing_leftweight = 1;
00221             s->interlacing_shift = 31;
00222             break;
00223 
00224         default:
00225             for(i=0; i<n; i++) {
00226                 tmp = left[i];
00227                 left[i] = (tmp + right[i]) >> 1;
00228                 right[i] = tmp - right[i];
00229             }
00230             s->interlacing_leftweight = 1;
00231             s->interlacing_shift = 1;
00232             break;
00233     }
00234 }
00235 
00236 static void alac_linear_predictor(AlacEncodeContext *s, int ch)
00237 {
00238     int i;
00239     LPCContext lpc = s->lpc[ch];
00240 
00241     if(lpc.lpc_order == 31) {
00242         s->predictor_buf[0] = s->sample_buf[ch][0];
00243 
00244         for(i=1; i<s->avctx->frame_size; i++)
00245             s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1];
00246 
00247         return;
00248     }
00249 
00250     // generalised linear predictor
00251 
00252     if(lpc.lpc_order > 0) {
00253         int32_t *samples  = s->sample_buf[ch];
00254         int32_t *residual = s->predictor_buf;
00255 
00256         // generate warm-up samples
00257         residual[0] = samples[0];
00258         for(i=1;i<=lpc.lpc_order;i++)
00259             residual[i] = samples[i] - samples[i-1];
00260 
00261         // perform lpc on remaining samples
00262         for(i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) {
00263             int sum = 1 << (lpc.lpc_quant - 1), res_val, j;
00264 
00265             for (j = 0; j < lpc.lpc_order; j++) {
00266                 sum += (samples[lpc.lpc_order-j] - samples[0]) *
00267                         lpc.lpc_coeff[j];
00268             }
00269 
00270             sum >>= lpc.lpc_quant;
00271             sum += samples[0];
00272             residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum,
00273                                       s->write_sample_size);
00274             res_val = residual[i];
00275 
00276             if(res_val) {
00277                 int index = lpc.lpc_order - 1;
00278                 int neg = (res_val < 0);
00279 
00280                 while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) {
00281                     int val = samples[0] - samples[lpc.lpc_order - index];
00282                     int sign = (val ? FFSIGN(val) : 0);
00283 
00284                     if(neg)
00285                         sign*=-1;
00286 
00287                     lpc.lpc_coeff[index] -= sign;
00288                     val *= sign;
00289                     res_val -= ((val >> lpc.lpc_quant) *
00290                             (lpc.lpc_order - index));
00291                     index--;
00292                 }
00293             }
00294             samples++;
00295         }
00296     }
00297 }
00298 
00299 static void alac_entropy_coder(AlacEncodeContext *s)
00300 {
00301     unsigned int history = s->rc.initial_history;
00302     int sign_modifier = 0, i, k;
00303     int32_t *samples = s->predictor_buf;
00304 
00305     for(i=0;i < s->avctx->frame_size;) {
00306         int x;
00307 
00308         k = av_log2((history >> 9) + 3);
00309 
00310         x = -2*(*samples)-1;
00311         x ^= (x>>31);
00312 
00313         samples++;
00314         i++;
00315 
00316         encode_scalar(s, x - sign_modifier, k, s->write_sample_size);
00317 
00318         history += x * s->rc.history_mult
00319                    - ((history * s->rc.history_mult) >> 9);
00320 
00321         sign_modifier = 0;
00322         if(x > 0xFFFF)
00323             history = 0xFFFF;
00324 
00325         if((history < 128) && (i < s->avctx->frame_size)) {
00326             unsigned int block_size = 0;
00327 
00328             k = 7 - av_log2(history) + ((history + 16) >> 6);
00329 
00330             while((*samples == 0) && (i < s->avctx->frame_size)) {
00331                 samples++;
00332                 i++;
00333                 block_size++;
00334             }
00335             encode_scalar(s, block_size, k, 16);
00336 
00337             sign_modifier = (block_size <= 0xFFFF);
00338 
00339             history = 0;
00340         }
00341 
00342     }
00343 }
00344 
00345 static void write_compressed_frame(AlacEncodeContext *s)
00346 {
00347     int i, j;
00348 
00349     if(s->avctx->channels == 2)
00350         alac_stereo_decorrelation(s);
00351     put_bits(&s->pbctx, 8, s->interlacing_shift);
00352     put_bits(&s->pbctx, 8, s->interlacing_leftweight);
00353 
00354     for(i=0;i<s->avctx->channels;i++) {
00355 
00356         calc_predictor_params(s, i);
00357 
00358         put_bits(&s->pbctx, 4, 0);  // prediction type : currently only type 0 has been RE'd
00359         put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant);
00360 
00361         put_bits(&s->pbctx, 3, s->rc.rice_modifier);
00362         put_bits(&s->pbctx, 5, s->lpc[i].lpc_order);
00363         // predictor coeff. table
00364         for(j=0;j<s->lpc[i].lpc_order;j++) {
00365             put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]);
00366         }
00367     }
00368 
00369     // apply lpc and entropy coding to audio samples
00370 
00371     for(i=0;i<s->avctx->channels;i++) {
00372         alac_linear_predictor(s, i);
00373         alac_entropy_coder(s);
00374     }
00375 }
00376 
00377 static av_cold int alac_encode_init(AVCodecContext *avctx)
00378 {
00379     AlacEncodeContext *s    = avctx->priv_data;
00380     uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1);
00381 
00382     avctx->frame_size      = DEFAULT_FRAME_SIZE;
00383     avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE;
00384 
00385     if(avctx->sample_fmt != SAMPLE_FMT_S16) {
00386         av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
00387         return -1;
00388     }
00389 
00390     // Set default compression level
00391     if(avctx->compression_level == FF_COMPRESSION_DEFAULT)
00392         s->compression_level = 2;
00393     else
00394         s->compression_level = av_clip(avctx->compression_level, 0, 2);
00395 
00396     // Initialize default Rice parameters
00397     s->rc.history_mult    = 40;
00398     s->rc.initial_history = 10;
00399     s->rc.k_modifier      = 14;
00400     s->rc.rice_modifier   = 4;
00401 
00402     s->max_coded_frame_size = 8 + (avctx->frame_size*avctx->channels*avctx->bits_per_coded_sample>>3);
00403 
00404     s->write_sample_size  = avctx->bits_per_coded_sample + avctx->channels - 1; // FIXME: consider wasted_bytes
00405 
00406     AV_WB32(alac_extradata,    ALAC_EXTRADATA_SIZE);
00407     AV_WB32(alac_extradata+4,  MKBETAG('a','l','a','c'));
00408     AV_WB32(alac_extradata+12, avctx->frame_size);
00409     AV_WB8 (alac_extradata+17, avctx->bits_per_coded_sample);
00410     AV_WB8 (alac_extradata+21, avctx->channels);
00411     AV_WB32(alac_extradata+24, s->max_coded_frame_size);
00412     AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_coded_sample); // average bitrate
00413     AV_WB32(alac_extradata+32, avctx->sample_rate);
00414 
00415     // Set relevant extradata fields
00416     if(s->compression_level > 0) {
00417         AV_WB8(alac_extradata+18, s->rc.history_mult);
00418         AV_WB8(alac_extradata+19, s->rc.initial_history);
00419         AV_WB8(alac_extradata+20, s->rc.k_modifier);
00420     }
00421 
00422     s->min_prediction_order = DEFAULT_MIN_PRED_ORDER;
00423     if(avctx->min_prediction_order >= 0) {
00424         if(avctx->min_prediction_order < MIN_LPC_ORDER ||
00425            avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) {
00426             av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order);
00427                 return -1;
00428         }
00429 
00430         s->min_prediction_order = avctx->min_prediction_order;
00431     }
00432 
00433     s->max_prediction_order = DEFAULT_MAX_PRED_ORDER;
00434     if(avctx->max_prediction_order >= 0) {
00435         if(avctx->max_prediction_order < MIN_LPC_ORDER ||
00436            avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) {
00437             av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", avctx->max_prediction_order);
00438                 return -1;
00439         }
00440 
00441         s->max_prediction_order = avctx->max_prediction_order;
00442     }
00443 
00444     if(s->max_prediction_order < s->min_prediction_order) {
00445         av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n",
00446                s->min_prediction_order, s->max_prediction_order);
00447         return -1;
00448     }
00449 
00450     avctx->extradata = alac_extradata;
00451     avctx->extradata_size = ALAC_EXTRADATA_SIZE;
00452 
00453     avctx->coded_frame = avcodec_alloc_frame();
00454     avctx->coded_frame->key_frame = 1;
00455 
00456     s->avctx = avctx;
00457     dsputil_init(&s->dspctx, avctx);
00458 
00459     return 0;
00460 }
00461 
00462 static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
00463                              int buf_size, void *data)
00464 {
00465     AlacEncodeContext *s = avctx->priv_data;
00466     PutBitContext *pb = &s->pbctx;
00467     int i, out_bytes, verbatim_flag = 0;
00468 
00469     if(avctx->frame_size > DEFAULT_FRAME_SIZE) {
00470         av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n");
00471         return -1;
00472     }
00473 
00474     if(buf_size < 2*s->max_coded_frame_size) {
00475         av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n");
00476         return -1;
00477     }
00478 
00479 verbatim:
00480     init_put_bits(pb, frame, buf_size);
00481 
00482     if((s->compression_level == 0) || verbatim_flag) {
00483         // Verbatim mode
00484         int16_t *samples = data;
00485         write_frame_header(s, 1);
00486         for(i=0; i<avctx->frame_size*avctx->channels; i++) {
00487             put_sbits(pb, 16, *samples++);
00488         }
00489     } else {
00490         init_sample_buffers(s, data);
00491         write_frame_header(s, 0);
00492         write_compressed_frame(s);
00493     }
00494 
00495     put_bits(pb, 3, 7);
00496     flush_put_bits(pb);
00497     out_bytes = put_bits_count(pb) >> 3;
00498 
00499     if(out_bytes > s->max_coded_frame_size) {
00500         /* frame too large. use verbatim mode */
00501         if(verbatim_flag || (s->compression_level == 0)) {
00502             /* still too large. must be an error. */
00503             av_log(avctx, AV_LOG_ERROR, "error encoding frame\n");
00504             return -1;
00505         }
00506         verbatim_flag = 1;
00507         goto verbatim;
00508     }
00509 
00510     return out_bytes;
00511 }
00512 
00513 static av_cold int alac_encode_close(AVCodecContext *avctx)
00514 {
00515     av_freep(&avctx->extradata);
00516     avctx->extradata_size = 0;
00517     av_freep(&avctx->coded_frame);
00518     return 0;
00519 }
00520 
00521 AVCodec alac_encoder = {
00522     "alac",
00523     AVMEDIA_TYPE_AUDIO,
00524     CODEC_ID_ALAC,
00525     sizeof(AlacEncodeContext),
00526     alac_encode_init,
00527     alac_encode_frame,
00528     alac_encode_close,
00529     .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
00530     .sample_fmts = (const enum SampleFormat[]){ SAMPLE_FMT_S16, SAMPLE_FMT_NONE},
00531     .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
00532 };

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