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 
00034 #undef NDEBUG
00035 
00036 #define MAX_CHANNELS 6
00037 #define DCA_SUBBANDS_32 32
00038 #define DCA_MAX_FRAME_SIZE 16383
00039 #define DCA_HEADER_SIZE 13
00040 
00041 #define DCA_SUBBANDS 32 
00042 #define QUANTIZER_BITS 16
00043 #define SUBFRAMES 1
00044 #define SUBSUBFRAMES 4
00045 #define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8)
00046 #define LFE_BITS 8
00047 #define LFE_INTERPOLATION 64
00048 #define LFE_PRESENT 2
00049 #define LFE_MISSING 0
00050 
00051 static const int8_t dca_lfe_index[] = {
00052     1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
00053 };
00054 
00055 static const int8_t dca_channel_reorder_lfe[][9] = {
00056     { 0, -1, -1, -1, -1, -1, -1, -1, -1 },
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     { 1,  2,  0, -1, -1, -1, -1, -1, -1 },
00062     { 0,  1, -1,  2, -1, -1, -1, -1, -1 },
00063     { 1,  2,  0, -1,  3, -1, -1, -1, -1 },
00064     { 0,  1, -1,  2,  3, -1, -1, -1, -1 },
00065     { 1,  2,  0, -1,  3,  4, -1, -1, -1 },
00066     { 2,  3, -1,  0,  1,  4,  5, -1, -1 },
00067     { 1,  2,  0, -1,  3,  4,  5, -1, -1 },
00068     { 0, -1,  4,  5,  2,  3,  1, -1, -1 },
00069     { 3,  4,  1, -1,  0,  2,  5,  6, -1 },
00070     { 2,  3, -1,  5,  7,  0,  1,  4,  6 },
00071     { 3,  4,  1, -1,  0,  2,  5,  7,  6 },
00072 };
00073 
00074 static const int8_t dca_channel_reorder_nolfe[][9] = {
00075     { 0, -1, -1, -1, -1, -1, -1, -1, -1 },
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     { 1,  2,  0, -1, -1, -1, -1, -1, -1 },
00081     { 0,  1,  2, -1, -1, -1, -1, -1, -1 },
00082     { 1,  2,  0,  3, -1, -1, -1, -1, -1 },
00083     { 0,  1,  2,  3, -1, -1, -1, -1, -1 },
00084     { 1,  2,  0,  3,  4, -1, -1, -1, -1 },
00085     { 2,  3,  0,  1,  4,  5, -1, -1, -1 },
00086     { 1,  2,  0,  3,  4,  5, -1, -1, -1 },
00087     { 0,  4,  5,  2,  3,  1, -1, -1, -1 },
00088     { 3,  4,  1,  0,  2,  5,  6, -1, -1 },
00089     { 2,  3,  5,  7,  0,  1,  4,  6, -1 },
00090     { 3,  4,  1,  0,  2,  5,  7,  6, -1 },
00091 };
00092 
00093 typedef struct {
00094     PutBitContext pb;
00095     int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
00096     int start[MAX_CHANNELS];
00097     int frame_size;
00098     int prim_channels;
00099     int lfe_channel;
00100     int sample_rate_code;
00101     int scale_factor[MAX_CHANNELS][DCA_SUBBANDS_32];
00102     int lfe_scale_factor;
00103     int lfe_data[SUBFRAMES*SUBSUBFRAMES*4];
00104 
00105     int a_mode;                         
00106     int num_channel;
00107     int lfe_state;
00108     int lfe_offset;
00109     const int8_t *channel_order_tab;    
00110 
00111     int32_t pcm[FFMAX(LFE_INTERPOLATION, DCA_SUBBANDS_32)];
00112     int32_t subband[PCM_SAMPLES][MAX_CHANNELS][DCA_SUBBANDS_32]; /* [sample][channel][subband] */
00113 } DCAContext;
00114 
00115 static int32_t cos_table[128];
00116 
00117 static inline int32_t mul32(int32_t a, int32_t b)
00118 {
00119     int64_t r = (int64_t) a * b;
00120     /* round the result before truncating - improves accuracy */
00121     return (r + 0x80000000) >> 32;
00122 }
00123 
00124 /* Integer version of the cosine modulated Pseudo QMF */
00125 
00126 static void qmf_init(void)
00127 {
00128     int i;
00129     int32_t c[17], s[17];
00130     s[0] = 0;           /* sin(index * PI / 64) * 0x7fffffff */
00131     c[0] = 0x7fffffff;  /* cos(index * PI / 64) * 0x7fffffff */
00132 
00133     for (i = 1; i <= 16; i++) {
00134         s[i] = 2 * (mul32(c[i - 1], 105372028)  + mul32(s[i - 1], 2144896908));
00135         c[i] = 2 * (mul32(c[i - 1], 2144896908) - mul32(s[i - 1], 105372028));
00136     }
00137 
00138     for (i = 0; i < 16; i++) {
00139         cos_table[i      ]  =  c[i]      >> 3; /* avoid output overflow */
00140         cos_table[i +  16]  =  s[16 - i] >> 3;
00141         cos_table[i +  32]  = -s[i]      >> 3;
00142         cos_table[i +  48]  = -c[16 - i] >> 3;
00143         cos_table[i +  64]  = -c[i]      >> 3;
00144         cos_table[i +  80]  = -s[16 - i] >> 3;
00145         cos_table[i +  96]  =  s[i]      >> 3;
00146         cos_table[i + 112]  =  c[16 - i] >> 3;
00147     }
00148 }
00149 
00150 static int32_t band_delta_factor(int band, int sample_num)
00151 {
00152     int index = band * (2 * sample_num + 1);
00153     if (band == 0)
00154         return 0x07ffffff;
00155     else
00156         return cos_table[index & 127];
00157 }
00158 
00159 static void add_new_samples(DCAContext *c, const int32_t *in,
00160                             int count, int channel)
00161 {
00162     int i;
00163 
00164     /* Place new samples into the history buffer */
00165     for (i = 0; i < count; i++) {
00166         c->history[channel][c->start[channel] + i] = in[i];
00167         av_assert0(c->start[channel] + i < 512);
00168     }
00169     c->start[channel] += count;
00170     if (c->start[channel] == 512)
00171         c->start[channel] = 0;
00172     av_assert0(c->start[channel] < 512);
00173 }
00174 
00175 static void qmf_decompose(DCAContext *c, int32_t in[32], int32_t out[32],
00176                           int channel)
00177 {
00178     int band, i, j, k;
00179     int32_t resp;
00180     int32_t accum[DCA_SUBBANDS_32] = {0};
00181 
00182     add_new_samples(c, in, DCA_SUBBANDS_32, channel);
00183 
00184     /* Calculate the dot product of the signal with the (possibly inverted)
00185        reference decoder's response to this vector:
00186        (0.0, 0.0, ..., 0.0, -1.0, 1.0, 0.0, ..., 0.0)
00187        so that -1.0 cancels 1.0 from the previous step */
00188 
00189     for (k = 48, j = 0, i = c->start[channel]; i < 512; k++, j++, i++)
00190         accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
00191     for (i = 0; i < c->start[channel]; k++, j++, i++)
00192         accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
00193 
00194     resp = 0;
00195     /* TODO: implement FFT instead of this naive calculation */
00196     for (band = 0; band < DCA_SUBBANDS_32; band++) {
00197         for (j = 0; j < 32; j++)
00198             resp += mul32(accum[j], band_delta_factor(band, j));
00199 
00200         out[band] = (band & 2) ? (-resp) : resp;
00201     }
00202 }
00203 
00204 static int32_t lfe_fir_64i[512];
00205 static int lfe_downsample(DCAContext *c, int32_t in[LFE_INTERPOLATION])
00206 {
00207     int i, j;
00208     int channel = c->prim_channels;
00209     int32_t accum = 0;
00210 
00211     add_new_samples(c, in, LFE_INTERPOLATION, channel);
00212     for (i = c->start[channel], j = 0; i < 512; i++, j++)
00213         accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
00214     for (i = 0; i < c->start[channel]; i++, j++)
00215         accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
00216     return accum;
00217 }
00218 
00219 static void init_lfe_fir(void)
00220 {
00221     static int initialized = 0;
00222     int i;
00223     if (initialized)
00224         return;
00225 
00226     for (i = 0; i < 512; i++)
00227         lfe_fir_64i[i] = lfe_fir_64[i] * (1 << 25); //float -> int32_t
00228     initialized = 1;
00229 }
00230 
00231 static void put_frame_header(DCAContext *c)
00232 {
00233     /* SYNC */
00234     put_bits(&c->pb, 16, 0x7ffe);
00235     put_bits(&c->pb, 16, 0x8001);
00236 
00237     /* Frame type: normal */
00238     put_bits(&c->pb, 1, 1);
00239 
00240     /* Deficit sample count: none */
00241     put_bits(&c->pb, 5, 31);
00242 
00243     /* CRC is not present */
00244     put_bits(&c->pb, 1, 0);
00245 
00246     /* Number of PCM sample blocks */
00247     put_bits(&c->pb, 7, PCM_SAMPLES-1);
00248 
00249     /* Primary frame byte size */
00250     put_bits(&c->pb, 14, c->frame_size-1);
00251 
00252     /* Audio channel arrangement: L + R (stereo) */
00253     put_bits(&c->pb, 6, c->num_channel);
00254 
00255     /* Core audio sampling frequency */
00256     put_bits(&c->pb, 4, c->sample_rate_code);
00257 
00258     /* Transmission bit rate: 1411.2 kbps */
00259     put_bits(&c->pb, 5, 0x16); /* FIXME: magic number */
00260 
00261     /* Embedded down mix: disabled */
00262     put_bits(&c->pb, 1, 0);
00263 
00264     /* Embedded dynamic range flag: not present */
00265     put_bits(&c->pb, 1, 0);
00266 
00267     /* Embedded time stamp flag: not present */
00268     put_bits(&c->pb, 1, 0);
00269 
00270     /* Auxiliary data flag: not present */
00271     put_bits(&c->pb, 1, 0);
00272 
00273     /* HDCD source: no */
00274     put_bits(&c->pb, 1, 0);
00275 
00276     /* Extension audio ID: N/A */
00277     put_bits(&c->pb, 3, 0);
00278 
00279     /* Extended audio data: not present */
00280     put_bits(&c->pb, 1, 0);
00281 
00282     /* Audio sync word insertion flag: after each sub-frame */
00283     put_bits(&c->pb, 1, 0);
00284 
00285     /* Low frequency effects flag: not present or interpolation factor=64 */
00286     put_bits(&c->pb, 2, c->lfe_state);
00287 
00288     /* Predictor history switch flag: on */
00289     put_bits(&c->pb, 1, 1);
00290 
00291     /* No CRC */
00292     /* Multirate interpolator switch: non-perfect reconstruction */
00293     put_bits(&c->pb, 1, 0);
00294 
00295     /* Encoder software revision: 7 */
00296     put_bits(&c->pb, 4, 7);
00297 
00298     /* Copy history: 0 */
00299     put_bits(&c->pb, 2, 0);
00300 
00301     /* Source PCM resolution: 16 bits, not DTS ES */
00302     put_bits(&c->pb, 3, 0);
00303 
00304     /* Front sum/difference coding: no */
00305     put_bits(&c->pb, 1, 0);
00306 
00307     /* Surrounds sum/difference coding: no */
00308     put_bits(&c->pb, 1, 0);
00309 
00310     /* Dialog normalization: 0 dB */
00311     put_bits(&c->pb, 4, 0);
00312 }
00313 
00314 static void put_primary_audio_header(DCAContext *c)
00315 {
00316     static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
00317     static const int thr[11]    = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
00318 
00319     int ch, i;
00320     /* Number of subframes */
00321     put_bits(&c->pb, 4, SUBFRAMES - 1);
00322 
00323     /* Number of primary audio channels */
00324     put_bits(&c->pb, 3, c->prim_channels - 1);
00325 
00326     /* Subband activity count */
00327     for (ch = 0; ch < c->prim_channels; ch++)
00328         put_bits(&c->pb, 5, DCA_SUBBANDS - 2);
00329 
00330     /* High frequency VQ start subband */
00331     for (ch = 0; ch < c->prim_channels; ch++)
00332         put_bits(&c->pb, 5, DCA_SUBBANDS - 1);
00333 
00334     /* Joint intensity coding index: 0, 0 */
00335     for (ch = 0; ch < c->prim_channels; ch++)
00336         put_bits(&c->pb, 3, 0);
00337 
00338     /* Transient mode codebook: A4, A4 (arbitrary) */
00339     for (ch = 0; ch < c->prim_channels; ch++)
00340         put_bits(&c->pb, 2, 0);
00341 
00342     /* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */
00343     for (ch = 0; ch < c->prim_channels; ch++)
00344         put_bits(&c->pb, 3, 6);
00345 
00346     /* Bit allocation quantizer select: linear 5-bit */
00347     for (ch = 0; ch < c->prim_channels; ch++)
00348         put_bits(&c->pb, 3, 6);
00349 
00350     /* Quantization index codebook select: dummy data
00351        to avoid transmission of scale factor adjustment */
00352 
00353     for (i = 1; i < 11; i++)
00354         for (ch = 0; ch < c->prim_channels; ch++)
00355             put_bits(&c->pb, bitlen[i], thr[i]);
00356 
00357     /* Scale factor adjustment index: not transmitted */
00358 }
00359 
00365 static inline uint32_t quantize(int32_t sample, int bits)
00366 {
00367     av_assert0(sample <    1 << (bits - 1));
00368     av_assert0(sample >= -(1 << (bits - 1)));
00369     return sample & ((1 << bits) - 1);
00370 }
00371 
00372 static inline int find_scale_factor7(int64_t max_value, int bits)
00373 {
00374     int i = 0, j = 128, q;
00375     max_value = ((max_value << 15) / lossy_quant[bits + 3]) >> (bits - 1);
00376     while (i < j) {
00377         q = (i + j) >> 1;
00378         if (max_value < scale_factor_quant7[q])
00379             j = q;
00380         else
00381             i = q + 1;
00382     }
00383     av_assert1(i < 128);
00384     return i;
00385 }
00386 
00387 static inline void put_sample7(DCAContext *c, int64_t sample, int bits,
00388                                int scale_factor)
00389 {
00390     sample = (sample << 15) / ((int64_t) lossy_quant[bits + 3] * scale_factor_quant7[scale_factor]);
00391     put_bits(&c->pb, bits, quantize((int) sample, bits));
00392 }
00393 
00394 static void put_subframe(DCAContext *c,
00395                          int32_t subband_data[8 * SUBSUBFRAMES][MAX_CHANNELS][32],
00396                          int subframe)
00397 {
00398     int i, sub, ss, ch, max_value;
00399     int32_t *lfe_data = c->lfe_data + 4 * SUBSUBFRAMES * subframe;
00400 
00401     /* Subsubframes count */
00402     put_bits(&c->pb, 2, SUBSUBFRAMES -1);
00403 
00404     /* Partial subsubframe sample count: dummy */
00405     put_bits(&c->pb, 3, 0);
00406 
00407     /* Prediction mode: no ADPCM, in each channel and subband */
00408     for (ch = 0; ch < c->prim_channels; ch++)
00409         for (sub = 0; sub < DCA_SUBBANDS; sub++)
00410             put_bits(&c->pb, 1, 0);
00411 
00412     /* Prediction VQ addres: not transmitted */
00413     /* Bit allocation index */
00414     for (ch = 0; ch < c->prim_channels; ch++)
00415         for (sub = 0; sub < DCA_SUBBANDS; sub++)
00416             put_bits(&c->pb, 5, QUANTIZER_BITS+3);
00417 
00418     if (SUBSUBFRAMES > 1) {
00419         /* Transition mode: none for each channel and subband */
00420         for (ch = 0; ch < c->prim_channels; ch++)
00421             for (sub = 0; sub < DCA_SUBBANDS; sub++)
00422                 put_bits(&c->pb, 1, 0); /* codebook A4 */
00423     }
00424 
00425     /* Determine scale_factor */
00426     for (ch = 0; ch < c->prim_channels; ch++)
00427         for (sub = 0; sub < DCA_SUBBANDS; sub++) {
00428             max_value = 0;
00429             for (i = 0; i < 8 * SUBSUBFRAMES; i++)
00430                 max_value = FFMAX(max_value, FFABS(subband_data[i][ch][sub]));
00431             c->scale_factor[ch][sub] = find_scale_factor7(max_value, QUANTIZER_BITS);
00432         }
00433 
00434     if (c->lfe_channel) {
00435         max_value = 0;
00436         for (i = 0; i < 4 * SUBSUBFRAMES; i++)
00437             max_value = FFMAX(max_value, FFABS(lfe_data[i]));
00438         c->lfe_scale_factor = find_scale_factor7(max_value, LFE_BITS);
00439     }
00440 
00441     /* Scale factors: the same for each channel and subband,
00442        encoded according to Table D.1.2 */
00443     for (ch = 0; ch < c->prim_channels; ch++)
00444         for (sub = 0; sub < DCA_SUBBANDS; sub++)
00445             put_bits(&c->pb, 7, c->scale_factor[ch][sub]);
00446 
00447     /* Joint subband scale factor codebook select: not transmitted */
00448     /* Scale factors for joint subband coding: not transmitted */
00449     /* Stereo down-mix coefficients: not transmitted */
00450     /* Dynamic range coefficient: not transmitted */
00451     /* Stde information CRC check word: not transmitted */
00452     /* VQ encoded high frequency subbands: not transmitted */
00453 
00454     /* LFE data */
00455     if (c->lfe_channel) {
00456         for (i = 0; i < 4 * SUBSUBFRAMES; i++)
00457             put_sample7(c, lfe_data[i], LFE_BITS, c->lfe_scale_factor);
00458         put_bits(&c->pb, 8, c->lfe_scale_factor);
00459     }
00460 
00461     /* Audio data (subsubframes) */
00462 
00463     for (ss = 0; ss < SUBSUBFRAMES ; ss++)
00464         for (ch = 0; ch < c->prim_channels; ch++)
00465             for (sub = 0; sub < DCA_SUBBANDS; sub++)
00466                 for (i = 0; i < 8; i++)
00467                     put_sample7(c, subband_data[ss * 8 + i][ch][sub], QUANTIZER_BITS, c->scale_factor[ch][sub]);
00468 
00469     /* DSYNC */
00470     put_bits(&c->pb, 16, 0xffff);
00471 }
00472 
00473 static void put_frame(DCAContext *c,
00474                       int32_t subband_data[PCM_SAMPLES][MAX_CHANNELS][32],
00475                       uint8_t *frame)
00476 {
00477     int i;
00478     init_put_bits(&c->pb, frame + DCA_HEADER_SIZE, DCA_MAX_FRAME_SIZE-DCA_HEADER_SIZE);
00479 
00480     put_primary_audio_header(c);
00481     for (i = 0; i < SUBFRAMES; i++)
00482         put_subframe(c, &subband_data[SUBSUBFRAMES * 8 * i], i);
00483 
00484     flush_put_bits(&c->pb);
00485     c->frame_size = (put_bits_count(&c->pb) >> 3) + DCA_HEADER_SIZE;
00486 
00487     init_put_bits(&c->pb, frame, DCA_HEADER_SIZE);
00488     put_frame_header(c);
00489     flush_put_bits(&c->pb);
00490 }
00491 
00492 static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
00493                         const AVFrame *frame, int *got_packet_ptr)
00494 {
00495     int i, k, channel;
00496     DCAContext *c = avctx->priv_data;
00497     const int16_t *samples;
00498     int ret, real_channel = 0;
00499 
00500     if ((ret = ff_alloc_packet2(avctx, avpkt, DCA_MAX_FRAME_SIZE + DCA_HEADER_SIZE)))
00501         return ret;
00502 
00503     samples = (const int16_t *)frame->data[0];
00504     for (i = 0; i < PCM_SAMPLES; i ++) { /* i is the decimated sample number */
00505         for (channel = 0; channel < c->prim_channels + 1; channel++) {
00506             real_channel = c->channel_order_tab[channel];
00507             if (real_channel >= 0) {
00508                 /* Get 32 PCM samples */
00509                 for (k = 0; k < 32; k++) { /* k is the sample number in a 32-sample block */
00510                     c->pcm[k] = samples[avctx->channels * (32 * i + k) + channel] << 16;
00511                 }
00512                 /* Put subband samples into the proper place */
00513                 qmf_decompose(c, c->pcm, &c->subband[i][real_channel][0], real_channel);
00514             }
00515         }
00516     }
00517 
00518     if (c->lfe_channel) {
00519         for (i = 0; i < PCM_SAMPLES / 2; i++) {
00520             for (k = 0; k < LFE_INTERPOLATION; k++) /* k is the sample number in a 32-sample block */
00521                 c->pcm[k] = samples[avctx->channels * (LFE_INTERPOLATION*i+k) + c->lfe_offset] << 16;
00522             c->lfe_data[i] = lfe_downsample(c, c->pcm);
00523         }
00524     }
00525 
00526     put_frame(c, c->subband, avpkt->data);
00527 
00528     avpkt->size     = c->frame_size;
00529     *got_packet_ptr = 1;
00530     return 0;
00531 }
00532 
00533 static int encode_init(AVCodecContext *avctx)
00534 {
00535     DCAContext *c = avctx->priv_data;
00536     int i;
00537     uint64_t layout = avctx->channel_layout;
00538 
00539     c->prim_channels = avctx->channels;
00540     c->lfe_channel   = (avctx->channels == 3 || avctx->channels == 6);
00541 
00542     if (!layout) {
00543         av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "
00544                                       "encoder will guess the layout, but it "
00545                                       "might be incorrect.\n");
00546         layout = av_get_default_channel_layout(avctx->channels);
00547     }
00548     switch (layout) {
00549     case AV_CH_LAYOUT_STEREO:       c->a_mode = 2; c->num_channel = 2; break;
00550     case AV_CH_LAYOUT_5POINT0:      c->a_mode = 9; c->num_channel = 9; break;
00551     case AV_CH_LAYOUT_5POINT1:      c->a_mode = 9; c->num_channel = 9; break;
00552     case AV_CH_LAYOUT_5POINT0_BACK: c->a_mode = 9; c->num_channel = 9; break;
00553     case AV_CH_LAYOUT_5POINT1_BACK: c->a_mode = 9; c->num_channel = 9; break;
00554     default:
00555     av_log(avctx, AV_LOG_ERROR,
00556            "Only stereo, 5.0, 5.1 channel layouts supported at the moment!\n");
00557     return AVERROR_PATCHWELCOME;
00558     }
00559 
00560     if (c->lfe_channel) {
00561         init_lfe_fir();
00562         c->prim_channels--;
00563         c->channel_order_tab = dca_channel_reorder_lfe[c->a_mode];
00564         c->lfe_state         = LFE_PRESENT;
00565         c->lfe_offset        = dca_lfe_index[c->a_mode];
00566     } else {
00567         c->channel_order_tab = dca_channel_reorder_nolfe[c->a_mode];
00568         c->lfe_state         = LFE_MISSING;
00569     }
00570 
00571     for (i = 0; i < 16; i++) {
00572         if (dca_sample_rates[i] && (dca_sample_rates[i] == avctx->sample_rate))
00573             break;
00574     }
00575     if (i == 16) {
00576         av_log(avctx, AV_LOG_ERROR, "Sample rate %iHz not supported, only ", avctx->sample_rate);
00577         for (i = 0; i < 16; i++)
00578             av_log(avctx, AV_LOG_ERROR, "%d, ", dca_sample_rates[i]);
00579         av_log(avctx, AV_LOG_ERROR, "supported.\n");
00580         return -1;
00581     }
00582     c->sample_rate_code = i;
00583 
00584     avctx->frame_size = 32 * PCM_SAMPLES;
00585 
00586     if (!cos_table[127])
00587         qmf_init();
00588     return 0;
00589 }
00590 
00591 AVCodec ff_dca_encoder = {
00592     .name           = "dca",
00593     .type           = AVMEDIA_TYPE_AUDIO,
00594     .id             = CODEC_ID_DTS,
00595     .priv_data_size = sizeof(DCAContext),
00596     .init           = encode_init,
00597     .encode2        = encode_frame,
00598     .capabilities   = CODEC_CAP_EXPERIMENTAL,
00599     .sample_fmts    = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
00600     .long_name      = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
00601 };