FFmpeg: libavcodec/dcaenc.c Source File

00001 /*
00002  * DCA encoder
00003  * Copyright (C) 2008 Alexander E. Patrakov
00004  *               2010 Benjamin Larsson
00005  *               2011 Xiang Wang
00006  *
00007  * This file is part of FFmpeg.
00008  *
00009  * FFmpeg is free software; you can redistribute it and/or
00010  * modify it under the terms of the GNU Lesser General Public
00011  * License as published by the Free Software Foundation; either
00012  * version 2.1 of the License, or (at your option) any later version.
00013  *
00014  * FFmpeg is distributed in the hope that it will be useful,
00015  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00016  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00017  * Lesser General Public License for more details.
00018  *
00019  * You should have received a copy of the GNU Lesser General Public
00020  * License along with FFmpeg; if not, write to the Free Software
00021  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00022  */
00023 
00024 #include "libavutil/common.h"
00025 #include "libavutil/avassert.h"
00026 #include "libavutil/audioconvert.h"
00027 #include "avcodec.h"
00028 #include "get_bits.h"
00029 #include "internal.h"
00030 #include "put_bits.h"
00031 #include "dcaenc.h"
00032 #include "dcadata.h"
00033 #include "dca.h"
00034 
00035 #undef NDEBUG
00036 
00037 #define MAX_CHANNELS 6
00038 #define DCA_SUBBANDS_32 32
00039 #define DCA_MAX_FRAME_SIZE 16383
00040 #define DCA_HEADER_SIZE 13
00041 
00042 #define DCA_SUBBANDS 32 
00043 #define QUANTIZER_BITS 16
00044 #define SUBFRAMES 1
00045 #define SUBSUBFRAMES 4
00046 #define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8)
00047 #define LFE_BITS 8
00048 #define LFE_INTERPOLATION 64
00049 #define LFE_PRESENT 2
00050 #define LFE_MISSING 0
00051 
00052 static const int8_t dca_lfe_index[] = {
00053     1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
00054 };
00055 
00056 static const int8_t dca_channel_reorder_lfe[][9] = {
00057     { 0, -1, -1, -1, -1, -1, -1, -1, -1 },
00058     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
00059     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
00060     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
00061     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
00062     { 1,  2,  0, -1, -1, -1, -1, -1, -1 },
00063     { 0,  1, -1,  2, -1, -1, -1, -1, -1 },
00064     { 1,  2,  0, -1,  3, -1, -1, -1, -1 },
00065     { 0,  1, -1,  2,  3, -1, -1, -1, -1 },
00066     { 1,  2,  0, -1,  3,  4, -1, -1, -1 },
00067     { 2,  3, -1,  0,  1,  4,  5, -1, -1 },
00068     { 1,  2,  0, -1,  3,  4,  5, -1, -1 },
00069     { 0, -1,  4,  5,  2,  3,  1, -1, -1 },
00070     { 3,  4,  1, -1,  0,  2,  5,  6, -1 },
00071     { 2,  3, -1,  5,  7,  0,  1,  4,  6 },
00072     { 3,  4,  1, -1,  0,  2,  5,  7,  6 },
00073 };
00074 
00075 static const int8_t dca_channel_reorder_nolfe[][9] = {
00076     { 0, -1, -1, -1, -1, -1, -1, -1, -1 },
00077     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
00078     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
00079     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
00080     { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
00081     { 1,  2,  0, -1, -1, -1, -1, -1, -1 },
00082     { 0,  1,  2, -1, -1, -1, -1, -1, -1 },
00083     { 1,  2,  0,  3, -1, -1, -1, -1, -1 },
00084     { 0,  1,  2,  3, -1, -1, -1, -1, -1 },
00085     { 1,  2,  0,  3,  4, -1, -1, -1, -1 },
00086     { 2,  3,  0,  1,  4,  5, -1, -1, -1 },
00087     { 1,  2,  0,  3,  4,  5, -1, -1, -1 },
00088     { 0,  4,  5,  2,  3,  1, -1, -1, -1 },
00089     { 3,  4,  1,  0,  2,  5,  6, -1, -1 },
00090     { 2,  3,  5,  7,  0,  1,  4,  6, -1 },
00091     { 3,  4,  1,  0,  2,  5,  7,  6, -1 },
00092 };
00093 
00094 typedef struct {
00095     PutBitContext pb;
00096     int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
00097     int start[MAX_CHANNELS];
00098     int frame_size;
00099     int prim_channels;
00100     int lfe_channel;
00101     int sample_rate_code;
00102     int scale_factor[MAX_CHANNELS][DCA_SUBBANDS_32];
00103     int lfe_scale_factor;
00104     int lfe_data[SUBFRAMES*SUBSUBFRAMES*4];
00105 
00106     int a_mode;                         
00107     int num_channel;
00108     int lfe_state;
00109     int lfe_offset;
00110     const int8_t *channel_order_tab;    
00111 
00112     int32_t pcm[FFMAX(LFE_INTERPOLATION, DCA_SUBBANDS_32)];
00113     int32_t subband[PCM_SAMPLES][MAX_CHANNELS][DCA_SUBBANDS_32]; /* [sample][channel][subband] */
00114 } DCAContext;
00115 
00116 static int32_t cos_table[128];
00117 
00118 static inline int32_t mul32(int32_t a, int32_t b)
00119 {
00120     int64_t r = (int64_t) a * b;
00121     /* round the result before truncating - improves accuracy */
00122     return (r + 0x80000000) >> 32;
00123 }
00124 
00125 /* Integer version of the cosine modulated Pseudo QMF */
00126 
00127 static void qmf_init(void)
00128 {
00129     int i;
00130     int32_t c[17], s[17];
00131     s[0] = 0;           /* sin(index * PI / 64) * 0x7fffffff */
00132     c[0] = 0x7fffffff;  /* cos(index * PI / 64) * 0x7fffffff */
00133 
00134     for (i = 1; i <= 16; i++) {
00135         s[i] = 2 * (mul32(c[i - 1], 105372028)  + mul32(s[i - 1], 2144896908));
00136         c[i] = 2 * (mul32(c[i - 1], 2144896908) - mul32(s[i - 1], 105372028));
00137     }
00138 
00139     for (i = 0; i < 16; i++) {
00140         cos_table[i      ]  =  c[i]      >> 3; /* avoid output overflow */
00141         cos_table[i +  16]  =  s[16 - i] >> 3;
00142         cos_table[i +  32]  = -s[i]      >> 3;
00143         cos_table[i +  48]  = -c[16 - i] >> 3;
00144         cos_table[i +  64]  = -c[i]      >> 3;
00145         cos_table[i +  80]  = -s[16 - i] >> 3;
00146         cos_table[i +  96]  =  s[i]      >> 3;
00147         cos_table[i + 112]  =  c[16 - i] >> 3;
00148     }
00149 }
00150 
00151 static int32_t band_delta_factor(int band, int sample_num)
00152 {
00153     int index = band * (2 * sample_num + 1);
00154     if (band == 0)
00155         return 0x07ffffff;
00156     else
00157         return cos_table[index & 127];
00158 }
00159 
00160 static void add_new_samples(DCAContext *c, const int32_t *in,
00161                             int count, int channel)
00162 {
00163     int i;
00164 
00165     /* Place new samples into the history buffer */
00166     for (i = 0; i < count; i++) {
00167         c->history[channel][c->start[channel] + i] = in[i];
00168         av_assert0(c->start[channel] + i < 512);
00169     }
00170     c->start[channel] += count;
00171     if (c->start[channel] == 512)
00172         c->start[channel] = 0;
00173     av_assert0(c->start[channel] < 512);
00174 }
00175 
00176 static void qmf_decompose(DCAContext *c, int32_t in[32], int32_t out[32],
00177                           int channel)
00178 {
00179     int band, i, j, k;
00180     int32_t resp;
00181     int32_t accum[DCA_SUBBANDS_32] = {0};
00182 
00183     add_new_samples(c, in, DCA_SUBBANDS_32, channel);
00184 
00185     /* Calculate the dot product of the signal with the (possibly inverted)
00186        reference decoder's response to this vector:
00187        (0.0, 0.0, ..., 0.0, -1.0, 1.0, 0.0, ..., 0.0)
00188        so that -1.0 cancels 1.0 from the previous step */
00189 
00190     for (k = 48, j = 0, i = c->start[channel]; i < 512; k++, j++, i++)
00191         accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
00192     for (i = 0; i < c->start[channel]; k++, j++, i++)
00193         accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
00194 
00195     resp = 0;
00196     /* TODO: implement FFT instead of this naive calculation */
00197     for (band = 0; band < DCA_SUBBANDS_32; band++) {
00198         for (j = 0; j < 32; j++)
00199             resp += mul32(accum[j], band_delta_factor(band, j));
00200 
00201         out[band] = (band & 2) ? (-resp) : resp;
00202     }
00203 }
00204 
00205 static int32_t lfe_fir_64i[512];
00206 static int lfe_downsample(DCAContext *c, int32_t in[LFE_INTERPOLATION])
00207 {
00208     int i, j;
00209     int channel = c->prim_channels;
00210     int32_t accum = 0;
00211 
00212     add_new_samples(c, in, LFE_INTERPOLATION, channel);
00213     for (i = c->start[channel], j = 0; i < 512; i++, j++)
00214         accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
00215     for (i = 0; i < c->start[channel]; i++, j++)
00216         accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
00217     return accum;
00218 }
00219 
00220 static void init_lfe_fir(void)
00221 {
00222     static int initialized = 0;
00223     int i;
00224     if (initialized)
00225         return;
00226 
00227     for (i = 0; i < 512; i++)
00228         lfe_fir_64i[i] = lfe_fir_64[i] * (1 << 25); //float -> int32_t
00229     initialized = 1;
00230 }
00231 
00232 static void put_frame_header(DCAContext *c)
00233 {
00234     /* SYNC */
00235     put_bits(&c->pb, 16, 0x7ffe);
00236     put_bits(&c->pb, 16, 0x8001);
00237 
00238     /* Frame type: normal */
00239     put_bits(&c->pb, 1, 1);
00240 
00241     /* Deficit sample count: none */
00242     put_bits(&c->pb, 5, 31);
00243 
00244     /* CRC is not present */
00245     put_bits(&c->pb, 1, 0);
00246 
00247     /* Number of PCM sample blocks */
00248     put_bits(&c->pb, 7, PCM_SAMPLES-1);
00249 
00250     /* Primary frame byte size */
00251     put_bits(&c->pb, 14, c->frame_size-1);
00252 
00253     /* Audio channel arrangement: L + R (stereo) */
00254     put_bits(&c->pb, 6, c->num_channel);
00255 
00256     /* Core audio sampling frequency */
00257     put_bits(&c->pb, 4, c->sample_rate_code);
00258 
00259     /* Transmission bit rate: 1411.2 kbps */
00260     put_bits(&c->pb, 5, 0x16); /* FIXME: magic number */
00261 
00262     /* Embedded down mix: disabled */
00263     put_bits(&c->pb, 1, 0);
00264 
00265     /* Embedded dynamic range flag: not present */
00266     put_bits(&c->pb, 1, 0);
00267 
00268     /* Embedded time stamp flag: not present */
00269     put_bits(&c->pb, 1, 0);
00270 
00271     /* Auxiliary data flag: not present */
00272     put_bits(&c->pb, 1, 0);
00273 
00274     /* HDCD source: no */
00275     put_bits(&c->pb, 1, 0);
00276 
00277     /* Extension audio ID: N/A */
00278     put_bits(&c->pb, 3, 0);
00279 
00280     /* Extended audio data: not present */
00281     put_bits(&c->pb, 1, 0);
00282 
00283     /* Audio sync word insertion flag: after each sub-frame */
00284     put_bits(&c->pb, 1, 0);
00285 
00286     /* Low frequency effects flag: not present or interpolation factor=64 */
00287     put_bits(&c->pb, 2, c->lfe_state);
00288 
00289     /* Predictor history switch flag: on */
00290     put_bits(&c->pb, 1, 1);
00291 
00292     /* No CRC */
00293     /* Multirate interpolator switch: non-perfect reconstruction */
00294     put_bits(&c->pb, 1, 0);
00295 
00296     /* Encoder software revision: 7 */
00297     put_bits(&c->pb, 4, 7);
00298 
00299     /* Copy history: 0 */
00300     put_bits(&c->pb, 2, 0);
00301 
00302     /* Source PCM resolution: 16 bits, not DTS ES */
00303     put_bits(&c->pb, 3, 0);
00304 
00305     /* Front sum/difference coding: no */
00306     put_bits(&c->pb, 1, 0);
00307 
00308     /* Surrounds sum/difference coding: no */
00309     put_bits(&c->pb, 1, 0);
00310 
00311     /* Dialog normalization: 0 dB */
00312     put_bits(&c->pb, 4, 0);
00313 }
00314 
00315 static void put_primary_audio_header(DCAContext *c)
00316 {
00317     static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
00318     static const int thr[11]    = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
00319 
00320     int ch, i;
00321     /* Number of subframes */
00322     put_bits(&c->pb, 4, SUBFRAMES - 1);
00323 
00324     /* Number of primary audio channels */
00325     put_bits(&c->pb, 3, c->prim_channels - 1);
00326 
00327     /* Subband activity count */
00328     for (ch = 0; ch < c->prim_channels; ch++)
00329         put_bits(&c->pb, 5, DCA_SUBBANDS - 2);
00330 
00331     /* High frequency VQ start subband */
00332     for (ch = 0; ch < c->prim_channels; ch++)
00333         put_bits(&c->pb, 5, DCA_SUBBANDS - 1);
00334 
00335     /* Joint intensity coding index: 0, 0 */
00336     for (ch = 0; ch < c->prim_channels; ch++)
00337         put_bits(&c->pb, 3, 0);
00338 
00339     /* Transient mode codebook: A4, A4 (arbitrary) */
00340     for (ch = 0; ch < c->prim_channels; ch++)
00341         put_bits(&c->pb, 2, 0);
00342 
00343     /* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */
00344     for (ch = 0; ch < c->prim_channels; ch++)
00345         put_bits(&c->pb, 3, 6);
00346 
00347     /* Bit allocation quantizer select: linear 5-bit */
00348     for (ch = 0; ch < c->prim_channels; ch++)
00349         put_bits(&c->pb, 3, 6);
00350 
00351     /* Quantization index codebook select: dummy data
00352        to avoid transmission of scale factor adjustment */
00353 
00354     for (i = 1; i < 11; i++)
00355         for (ch = 0; ch < c->prim_channels; ch++)
00356             put_bits(&c->pb, bitlen[i], thr[i]);
00357 
00358     /* Scale factor adjustment index: not transmitted */
00359 }
00360 
00366 static inline uint32_t quantize(int32_t sample, int bits)
00367 {
00368     av_assert0(sample <    1 << (bits - 1));
00369     av_assert0(sample >= -(1 << (bits - 1)));
00370     return sample & ((1 << bits) - 1);
00371 }
00372 
00373 static inline int find_scale_factor7(int64_t max_value, int bits)
00374 {
00375     int i = 0, j = 128, q;
00376     max_value = ((max_value << 15) / lossy_quant[bits + 3]) >> (bits - 1);
00377     while (i < j) {
00378         q = (i + j) >> 1;
00379         if (max_value < scale_factor_quant7[q])
00380             j = q;
00381         else
00382             i = q + 1;
00383     }
00384     av_assert1(i < 128);
00385     return i;
00386 }
00387 
00388 static inline void put_sample7(DCAContext *c, int64_t sample, int bits,
00389                                int scale_factor)
00390 {
00391     sample = (sample << 15) / ((int64_t) lossy_quant[bits + 3] * scale_factor_quant7[scale_factor]);
00392     put_bits(&c->pb, bits, quantize((int) sample, bits));
00393 }
00394 
00395 static void put_subframe(DCAContext *c,
00396                          int32_t subband_data[8 * SUBSUBFRAMES][MAX_CHANNELS][32],
00397                          int subframe)
00398 {
00399     int i, sub, ss, ch, max_value;
00400     int32_t *lfe_data = c->lfe_data + 4 * SUBSUBFRAMES * subframe;
00401 
00402     /* Subsubframes count */
00403     put_bits(&c->pb, 2, SUBSUBFRAMES -1);
00404 
00405     /* Partial subsubframe sample count: dummy */
00406     put_bits(&c->pb, 3, 0);
00407 
00408     /* Prediction mode: no ADPCM, in each channel and subband */
00409     for (ch = 0; ch < c->prim_channels; ch++)
00410         for (sub = 0; sub < DCA_SUBBANDS; sub++)
00411             put_bits(&c->pb, 1, 0);
00412 
00413     /* Prediction VQ addres: not transmitted */
00414     /* Bit allocation index */
00415     for (ch = 0; ch < c->prim_channels; ch++)
00416         for (sub = 0; sub < DCA_SUBBANDS; sub++)
00417             put_bits(&c->pb, 5, QUANTIZER_BITS+3);
00418 
00419     if (SUBSUBFRAMES > 1) {
00420         /* Transition mode: none for each channel and subband */
00421         for (ch = 0; ch < c->prim_channels; ch++)
00422             for (sub = 0; sub < DCA_SUBBANDS; sub++)
00423                 put_bits(&c->pb, 1, 0); /* codebook A4 */
00424     }
00425 
00426     /* Determine scale_factor */
00427     for (ch = 0; ch < c->prim_channels; ch++)
00428         for (sub = 0; sub < DCA_SUBBANDS; sub++) {
00429             max_value = 0;
00430             for (i = 0; i < 8 * SUBSUBFRAMES; i++)
00431                 max_value = FFMAX(max_value, FFABS(subband_data[i][ch][sub]));
00432             c->scale_factor[ch][sub] = find_scale_factor7(max_value, QUANTIZER_BITS);
00433         }
00434 
00435     if (c->lfe_channel) {
00436         max_value = 0;
00437         for (i = 0; i < 4 * SUBSUBFRAMES; i++)
00438             max_value = FFMAX(max_value, FFABS(lfe_data[i]));
00439         c->lfe_scale_factor = find_scale_factor7(max_value, LFE_BITS);
00440     }
00441 
00442     /* Scale factors: the same for each channel and subband,
00443        encoded according to Table D.1.2 */
00444     for (ch = 0; ch < c->prim_channels; ch++)
00445         for (sub = 0; sub < DCA_SUBBANDS; sub++)
00446             put_bits(&c->pb, 7, c->scale_factor[ch][sub]);
00447 
00448     /* Joint subband scale factor codebook select: not transmitted */
00449     /* Scale factors for joint subband coding: not transmitted */
00450     /* Stereo down-mix coefficients: not transmitted */
00451     /* Dynamic range coefficient: not transmitted */
00452     /* Stde information CRC check word: not transmitted */
00453     /* VQ encoded high frequency subbands: not transmitted */
00454 
00455     /* LFE data */
00456     if (c->lfe_channel) {
00457         for (i = 0; i < 4 * SUBSUBFRAMES; i++)
00458             put_sample7(c, lfe_data[i], LFE_BITS, c->lfe_scale_factor);
00459         put_bits(&c->pb, 8, c->lfe_scale_factor);
00460     }
00461 
00462     /* Audio data (subsubframes) */
00463 
00464     for (ss = 0; ss < SUBSUBFRAMES ; ss++)
00465         for (ch = 0; ch < c->prim_channels; ch++)
00466             for (sub = 0; sub < DCA_SUBBANDS; sub++)
00467                 for (i = 0; i < 8; i++)
00468                     put_sample7(c, subband_data[ss * 8 + i][ch][sub], QUANTIZER_BITS, c->scale_factor[ch][sub]);
00469 
00470     /* DSYNC */
00471     put_bits(&c->pb, 16, 0xffff);
00472 }
00473 
00474 static void put_frame(DCAContext *c,
00475                       int32_t subband_data[PCM_SAMPLES][MAX_CHANNELS][32],
00476                       uint8_t *frame)
00477 {
00478     int i;
00479     init_put_bits(&c->pb, frame + DCA_HEADER_SIZE, DCA_MAX_FRAME_SIZE-DCA_HEADER_SIZE);
00480 
00481     put_primary_audio_header(c);
00482     for (i = 0; i < SUBFRAMES; i++)
00483         put_subframe(c, &subband_data[SUBSUBFRAMES * 8 * i], i);
00484 
00485     flush_put_bits(&c->pb);
00486     c->frame_size = (put_bits_count(&c->pb) >> 3) + DCA_HEADER_SIZE;
00487 
00488     init_put_bits(&c->pb, frame, DCA_HEADER_SIZE);
00489     put_frame_header(c);
00490     flush_put_bits(&c->pb);
00491 }
00492 
00493 static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
00494                         const AVFrame *frame, int *got_packet_ptr)
00495 {
00496     int i, k, channel;
00497     DCAContext *c = avctx->priv_data;
00498     const int16_t *samples;
00499     int ret, real_channel = 0;
00500 
00501     if ((ret = ff_alloc_packet2(avctx, avpkt, DCA_MAX_FRAME_SIZE + DCA_HEADER_SIZE)))
00502         return ret;
00503 
00504     samples = (const int16_t *)frame->data[0];
00505     for (i = 0; i < PCM_SAMPLES; i ++) { /* i is the decimated sample number */
00506         for (channel = 0; channel < c->prim_channels + 1; channel++) {
00507             real_channel = c->channel_order_tab[channel];
00508             if (real_channel >= 0) {
00509                 /* Get 32 PCM samples */
00510                 for (k = 0; k < 32; k++) { /* k is the sample number in a 32-sample block */
00511                     c->pcm[k] = samples[avctx->channels * (32 * i + k) + channel] << 16;
00512                 }
00513                 /* Put subband samples into the proper place */
00514                 qmf_decompose(c, c->pcm, &c->subband[i][real_channel][0], real_channel);
00515             }
00516         }
00517     }
00518 
00519     if (c->lfe_channel) {
00520         for (i = 0; i < PCM_SAMPLES / 2; i++) {
00521             for (k = 0; k < LFE_INTERPOLATION; k++) /* k is the sample number in a 32-sample block */
00522                 c->pcm[k] = samples[avctx->channels * (LFE_INTERPOLATION*i+k) + c->lfe_offset] << 16;
00523             c->lfe_data[i] = lfe_downsample(c, c->pcm);
00524         }
00525     }
00526 
00527     put_frame(c, c->subband, avpkt->data);
00528 
00529     avpkt->size     = c->frame_size;
00530     *got_packet_ptr = 1;
00531     return 0;
00532 }
00533 
00534 static int encode_init(AVCodecContext *avctx)
00535 {
00536     DCAContext *c = avctx->priv_data;
00537     int i;
00538     uint64_t layout = avctx->channel_layout;
00539 
00540     c->prim_channels = avctx->channels;
00541     c->lfe_channel   = (avctx->channels == 3 || avctx->channels == 6);
00542 
00543     if (!layout) {
00544         av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "
00545                                       "encoder will guess the layout, but it "
00546                                       "might be incorrect.\n");
00547         layout = av_get_default_channel_layout(avctx->channels);
00548     }
00549     switch (layout) {
00550     case AV_CH_LAYOUT_STEREO:       c->a_mode = 2; c->num_channel = 2; break;
00551     case AV_CH_LAYOUT_5POINT0:      c->a_mode = 9; c->num_channel = 9; break;
00552     case AV_CH_LAYOUT_5POINT1:      c->a_mode = 9; c->num_channel = 9; break;
00553     case AV_CH_LAYOUT_5POINT0_BACK: c->a_mode = 9; c->num_channel = 9; break;
00554     case AV_CH_LAYOUT_5POINT1_BACK: c->a_mode = 9; c->num_channel = 9; break;
00555     default:
00556     av_log(avctx, AV_LOG_ERROR,
00557            "Only stereo, 5.0, 5.1 channel layouts supported at the moment!\n");
00558     return AVERROR_PATCHWELCOME;
00559     }
00560 
00561     if (c->lfe_channel) {
00562         init_lfe_fir();
00563         c->prim_channels--;
00564         c->channel_order_tab = dca_channel_reorder_lfe[c->a_mode];
00565         c->lfe_state         = LFE_PRESENT;
00566         c->lfe_offset        = dca_lfe_index[c->a_mode];
00567     } else {
00568         c->channel_order_tab = dca_channel_reorder_nolfe[c->a_mode];
00569         c->lfe_state         = LFE_MISSING;
00570     }
00571 
00572     for (i = 0; i < 16; i++) {
00573         if (avpriv_dca_sample_rates[i] && (avpriv_dca_sample_rates[i] == avctx->sample_rate))
00574             break;
00575     }
00576     if (i == 16) {
00577         av_log(avctx, AV_LOG_ERROR, "Sample rate %iHz not supported, only ", avctx->sample_rate);
00578         for (i = 0; i < 16; i++)
00579             av_log(avctx, AV_LOG_ERROR, "%d, ", avpriv_dca_sample_rates[i]);
00580         av_log(avctx, AV_LOG_ERROR, "supported.\n");
00581         return -1;
00582     }
00583     c->sample_rate_code = i;
00584 
00585     avctx->frame_size = 32 * PCM_SAMPLES;
00586 
00587     if (!cos_table[127])
00588         qmf_init();
00589     return 0;
00590 }
00591 
00592 AVCodec ff_dca_encoder = {
00593     .name           = "dca",
00594     .type           = AVMEDIA_TYPE_AUDIO,
00595     .id             = AV_CODEC_ID_DTS,
00596     .priv_data_size = sizeof(DCAContext),
00597     .init           = encode_init,
00598     .encode2        = encode_frame,
00599     .capabilities   = CODEC_CAP_EXPERIMENTAL,
00600     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
00601                                                      AV_SAMPLE_FMT_NONE },
00602     .long_name      = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
00603 };