/*

  * ALAC audio encoder
  * Copyright (c) 2008  Jaikrishnan Menon <realityman@gmx.net>
  *
  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
  * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
  * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 

 #include "libavutil/opt.h"
 

 #include "avcodec.h"

 #include "put_bits.h"

 #include "internal.h"

 #include "lpc.h"

 #include "mathops.h"

 #include "alac_data.h"

 
 #define DEFAULT_FRAME_SIZE        4096
 #define ALAC_EXTRADATA_SIZE       36
 #define ALAC_FRAME_HEADER_SIZE    55
 #define ALAC_FRAME_FOOTER_SIZE    3
 
 #define ALAC_ESCAPE_CODE          0x1FF
 #define ALAC_MAX_LPC_ORDER        30

 #define DEFAULT_MAX_PRED_ORDER    6
 #define DEFAULT_MIN_PRED_ORDER    4
 #define ALAC_MAX_LPC_PRECISION    9
 #define ALAC_MAX_LPC_SHIFT        9
 

 #define ALAC_CHMODE_LEFT_RIGHT    0
 #define ALAC_CHMODE_LEFT_SIDE     1
 #define ALAC_CHMODE_RIGHT_SIDE    2
 #define ALAC_CHMODE_MID_SIDE      3
 

 typedef struct RiceContext {
     int history_mult;
     int initial_history;
     int k_modifier;
     int rice_modifier;
 } RiceContext;
 

 typedef struct AlacLPCContext {

     int lpc_order;
     int lpc_coeff[ALAC_MAX_LPC_ORDER+1];
     int lpc_quant;

 } AlacLPCContext;

 
 typedef struct AlacEncodeContext {

     const AVClass *class;

     AVCodecContext *avctx;

     int frame_size;                     /**< current frame size               */

     int verbatim;                       /**< current frame verbatim mode flag */

     int compression_level;

     int min_prediction_order;
     int max_prediction_order;

     int max_coded_frame_size;
     int write_sample_size;

     int extra_bits;

     int32_t sample_buf[2][DEFAULT_FRAME_SIZE];

     int32_t predictor_buf[2][DEFAULT_FRAME_SIZE];

     int interlacing_shift;
     int interlacing_leftweight;
     PutBitContext pbctx;

     RiceContext rc;

     AlacLPCContext lpc[2];

     LPCContext lpc_ctx;

 } AlacEncodeContext;
 
 

 static void init_sample_buffers(AlacEncodeContext *s, int channels,
                                 uint8_t const *samples[2])

 {
     int ch, i;

     int shift = av_get_bytes_per_sample(s->avctx->sample_fmt) * 8 -
                 s->avctx->bits_per_raw_sample;
 
 #define COPY_SAMPLES(type) do {                             \

         for (ch = 0; ch < channels; ch++) {                 \

             int32_t       *bptr = s->sample_buf[ch];        \
             const type *sptr = (const type *)samples[ch];   \
             for (i = 0; i < s->frame_size; i++)             \
                 bptr[i] = sptr[i] >> shift;                 \
         }                                                   \
     } while (0)
 
     if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P)
         COPY_SAMPLES(int32_t);
     else
         COPY_SAMPLES(int16_t);

 }
 

 static void encode_scalar(AlacEncodeContext *s, int x,
                           int k, int write_sample_size)

 {
     int divisor, q, r;
 
     k = FFMIN(k, s->rc.k_modifier);
     divisor = (1<<k) - 1;
     q = x / divisor;
     r = x % divisor;
 

     if (q > 8) {

         // write escape code and sample value directly
         put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE);
         put_bits(&s->pbctx, write_sample_size, x);
     } else {

         if (q)

             put_bits(&s->pbctx, q, (1<<q) - 1);
         put_bits(&s->pbctx, 1, 0);
 

         if (k != 1) {
             if (r > 0)

                 put_bits(&s->pbctx, k, r+1);
             else
                 put_bits(&s->pbctx, k-1, 0);
         }
     }
 }
 

 static void write_element_header(AlacEncodeContext *s,
                                  enum AlacRawDataBlockType element,
                                  int instance)

 {

     int encode_fs = 0;
 
     if (s->frame_size < DEFAULT_FRAME_SIZE)
         encode_fs = 1;
 

     put_bits(&s->pbctx, 3,  element);               // element type
     put_bits(&s->pbctx, 4,  instance);              // element instance
     put_bits(&s->pbctx, 12, 0);                     // unused header bits

     put_bits(&s->pbctx, 1,  encode_fs);             // Sample count is in the header

     put_bits(&s->pbctx, 2,  s->extra_bits >> 3);    // Extra bytes (for 24-bit)

     put_bits(&s->pbctx, 1,  s->verbatim);           // Audio block is verbatim

     if (encode_fs)
         put_bits32(&s->pbctx, s->frame_size);       // No. of samples in the frame

 }
 

 static void calc_predictor_params(AlacEncodeContext *s, int ch)
 {
     int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER];
     int shift[MAX_LPC_ORDER];
     int opt_order;
 

     if (s->compression_level == 1) {

         s->lpc[ch].lpc_order = 6;
         s->lpc[ch].lpc_quant = 6;
         s->lpc[ch].lpc_coeff[0] =  160;
         s->lpc[ch].lpc_coeff[1] = -190;
         s->lpc[ch].lpc_coeff[2] =  170;
         s->lpc[ch].lpc_coeff[3] = -130;
         s->lpc[ch].lpc_coeff[4] =   80;
         s->lpc[ch].lpc_coeff[5] =  -25;
     } else {

         opt_order = ff_lpc_calc_coefs(&s->lpc_ctx, s->sample_buf[ch],

                                       s->frame_size,

                                       s->min_prediction_order,
                                       s->max_prediction_order,

                                       ALAC_MAX_LPC_PRECISION, coefs, shift,

                                       FF_LPC_TYPE_LEVINSON, 0,

                                       ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1);
 
         s->lpc[ch].lpc_order = opt_order;
         s->lpc[ch].lpc_quant = shift[opt_order-1];
         memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int));

     }

 }
 

 static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
 {
     int i, best;
     int32_t lt, rt;
     uint64_t sum[4];
     uint64_t score[4];
 
     /* calculate sum of 2nd order residual for each channel */
     sum[0] = sum[1] = sum[2] = sum[3] = 0;

     for (i = 2; i < n; i++) {

         lt =  left_ch[i] - 2 *  left_ch[i - 1] +  left_ch[i - 2];
         rt = right_ch[i] - 2 * right_ch[i - 1] + right_ch[i - 2];

         sum[2] += FFABS((lt + rt) >> 1);
         sum[3] += FFABS(lt - rt);
         sum[0] += FFABS(lt);
         sum[1] += FFABS(rt);
     }
 
     /* calculate score for each mode */
     score[0] = sum[0] + sum[1];
     score[1] = sum[0] + sum[3];
     score[2] = sum[1] + sum[3];
     score[3] = sum[2] + sum[3];
 
     /* return mode with lowest score */
     best = 0;

     for (i = 1; i < 4; i++) {

         if (score[i] < score[best])

             best = i;
     }

     return best;
 }
 
 static void alac_stereo_decorrelation(AlacEncodeContext *s)
 {
     int32_t *left = s->sample_buf[0], *right = s->sample_buf[1];

     int i, mode, n = s->frame_size;

     int32_t tmp;
 
     mode = estimate_stereo_mode(left, right, n);
 

     switch (mode) {
     case ALAC_CHMODE_LEFT_RIGHT:
         s->interlacing_leftweight = 0;
         s->interlacing_shift      = 0;
         break;
     case ALAC_CHMODE_LEFT_SIDE:
         for (i = 0; i < n; i++)
             right[i] = left[i] - right[i];
         s->interlacing_leftweight = 1;
         s->interlacing_shift      = 0;
         break;
     case ALAC_CHMODE_RIGHT_SIDE:
         for (i = 0; i < n; i++) {
             tmp = right[i];
             right[i] = left[i] - right[i];
             left[i]  = tmp + (right[i] >> 31);
         }
         s->interlacing_leftweight = 1;
         s->interlacing_shift      = 31;
         break;
     default:
         for (i = 0; i < n; i++) {
             tmp = left[i];
             left[i]  = (tmp + right[i]) >> 1;
             right[i] =  tmp - right[i];
         }
         s->interlacing_leftweight = 1;
         s->interlacing_shift      = 1;
         break;

     }
 }

 static void alac_linear_predictor(AlacEncodeContext *s, int ch)
 {
     int i;

     AlacLPCContext lpc = s->lpc[ch];

     int32_t *residual = s->predictor_buf[ch];

     if (lpc.lpc_order == 31) {

         residual[0] = s->sample_buf[ch][0];

         for (i = 1; i < s->frame_size; i++) {

             residual[i] = s->sample_buf[ch][i    ] -
                           s->sample_buf[ch][i - 1];

         }

 
         return;
     }
 
     // generalised linear predictor
 

     if (lpc.lpc_order > 0) {

         int32_t *samples  = s->sample_buf[ch];
 
         // generate warm-up samples
         residual[0] = samples[0];

         for (i = 1; i <= lpc.lpc_order; i++)

             residual[i] = sign_extend(samples[i] - samples[i-1], s->write_sample_size);

 
         // perform lpc on remaining samples

         for (i = lpc.lpc_order + 1; i < s->frame_size; i++) {

             int sum = 1 << (lpc.lpc_quant - 1), res_val, j;
 
             for (j = 0; j < lpc.lpc_order; j++) {
                 sum += (samples[lpc.lpc_order-j] - samples[0]) *

                        lpc.lpc_coeff[j];

             }
 
             sum >>= lpc.lpc_quant;
             sum += samples[0];

             residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum,
                                       s->write_sample_size);

             res_val = residual[i];
 

             if (res_val) {

                 int index = lpc.lpc_order - 1;
                 int neg = (res_val < 0);
 

                 while (index >= 0 && (neg ? (res_val < 0) : (res_val > 0))) {
                     int val  = samples[0] - samples[lpc.lpc_order - index];

                     int sign = (val ? FFSIGN(val) : 0);
 

                     if (neg)
                         sign *= -1;

 
                     lpc.lpc_coeff[index] -= sign;
                     val *= sign;

                     res_val -= (val >> lpc.lpc_quant) * (lpc.lpc_order - index);

                     index--;
                 }
             }
             samples++;
         }
     }
 }
 

 static void alac_entropy_coder(AlacEncodeContext *s, int ch)

 {
     unsigned int history = s->rc.initial_history;
     int sign_modifier = 0, i, k;

     int32_t *samples = s->predictor_buf[ch];

     for (i = 0; i < s->frame_size;) {

         int x;
 
         k = av_log2((history >> 9) + 3);
 

         x  = -2 * (*samples) -1;
         x ^= x >> 31;

 
         samples++;
         i++;
 
         encode_scalar(s, x - sign_modifier, k, s->write_sample_size);
 

         history += x * s->rc.history_mult -
                    ((history * s->rc.history_mult) >> 9);

 
         sign_modifier = 0;

         if (x > 0xFFFF)

             history = 0xFFFF;
 

         if (history < 128 && i < s->frame_size) {

             unsigned int block_size = 0;
 
             k = 7 - av_log2(history) + ((history + 16) >> 6);
 

             while (*samples == 0 && i < s->frame_size) {

                 samples++;
                 i++;
                 block_size++;
             }
             encode_scalar(s, block_size, k, 16);
             sign_modifier = (block_size <= 0xFFFF);
             history = 0;
         }
 
     }
 }
 

 static void write_element(AlacEncodeContext *s,
                           enum AlacRawDataBlockType element, int instance,
                           const uint8_t *samples0, const uint8_t *samples1)

 {

     uint8_t const *samples[2] = { samples0, samples1 };
     int i, j, channels;

     int prediction_type = 0;

     PutBitContext *pb = &s->pbctx;

     channels = element == TYPE_CPE ? 2 : 1;

 
     if (s->verbatim) {

         write_element_header(s, element, instance);

         /* samples are channel-interleaved in verbatim mode */

         if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P) {
             int shift = 32 - s->avctx->bits_per_raw_sample;

             int32_t const *samples_s32[2] = { (const int32_t *)samples0,
                                               (const int32_t *)samples1 };

             for (i = 0; i < s->frame_size; i++)

                 for (j = 0; j < channels; j++)

                     put_sbits(pb, s->avctx->bits_per_raw_sample,
                               samples_s32[j][i] >> shift);
         } else {

             int16_t const *samples_s16[2] = { (const int16_t *)samples0,
                                               (const int16_t *)samples1 };

             for (i = 0; i < s->frame_size; i++)

                 for (j = 0; j < channels; j++)

                     put_sbits(pb, s->avctx->bits_per_raw_sample,
                               samples_s16[j][i]);
         }

     } else {

         s->write_sample_size = s->avctx->bits_per_raw_sample - s->extra_bits +
                                channels - 1;

         init_sample_buffers(s, channels, samples);
         write_element_header(s, element, instance);
 

         // extract extra bits if needed
         if (s->extra_bits) {
             uint32_t mask = (1 << s->extra_bits) - 1;
             for (j = 0; j < channels; j++) {
                 int32_t *extra = s->predictor_buf[j];
                 int32_t *smp   = s->sample_buf[j];
                 for (i = 0; i < s->frame_size; i++) {
                     extra[i] = smp[i] & mask;
                     smp[i] >>= s->extra_bits;
                 }
             }
         }
 

         if (channels == 2)

             alac_stereo_decorrelation(s);

         else
             s->interlacing_shift = s->interlacing_leftweight = 0;

         put_bits(pb, 8, s->interlacing_shift);
         put_bits(pb, 8, s->interlacing_leftweight);

         for (i = 0; i < channels; i++) {

             calc_predictor_params(s, i);

             put_bits(pb, 4, prediction_type);
             put_bits(pb, 4, s->lpc[i].lpc_quant);

             put_bits(pb, 3, s->rc.rice_modifier);
             put_bits(pb, 5, s->lpc[i].lpc_order);
             // predictor coeff. table
             for (j = 0; j < s->lpc[i].lpc_order; j++)
                 put_sbits(pb, 16, s->lpc[i].lpc_coeff[j]);

         }
 

         // write extra bits if needed
         if (s->extra_bits) {
             for (i = 0; i < s->frame_size; i++) {

                 for (j = 0; j < channels; j++) {

                     put_bits(pb, s->extra_bits, s->predictor_buf[j][i]);

                 }
             }
         }
 

         // apply lpc and entropy coding to audio samples

         for (i = 0; i < channels; i++) {

             alac_linear_predictor(s, i);

             // TODO: determine when this will actually help. for now it's not used.
             if (prediction_type == 15) {
                 // 2nd pass 1st order filter

                 int32_t *residual = s->predictor_buf[i];

                 for (j = s->frame_size - 1; j > 0; j--)

                     residual[j] -= residual[j - 1];

             }

             alac_entropy_coder(s, i);

         }

     }

 }
 
 static int write_frame(AlacEncodeContext *s, AVPacket *avpkt,
                        uint8_t * const *samples)
 {
     PutBitContext *pb = &s->pbctx;
     const enum AlacRawDataBlockType *ch_elements = ff_alac_channel_elements[s->avctx->channels - 1];
     const uint8_t *ch_map = ff_alac_channel_layout_offsets[s->avctx->channels - 1];
     int ch, element, sce, cpe;
 
     init_put_bits(pb, avpkt->data, avpkt->size);
 
     ch = element = sce = cpe = 0;
     while (ch < s->avctx->channels) {
         if (ch_elements[element] == TYPE_CPE) {
             write_element(s, TYPE_CPE, cpe, samples[ch_map[ch]],
                           samples[ch_map[ch + 1]]);
             cpe++;
             ch += 2;
         } else {
             write_element(s, TYPE_SCE, sce, samples[ch_map[ch]], NULL);
             sce++;
             ch++;
         }
         element++;
     }
 
     put_bits(pb, 3, TYPE_END);

     flush_put_bits(pb);

     return put_bits_count(pb) >> 3;

 }
 

 static av_always_inline int get_max_frame_size(int frame_size, int ch, int bps)
 {

     int header_bits = 23 + 32 * (frame_size < DEFAULT_FRAME_SIZE);
     return FFALIGN(header_bits + bps * ch * frame_size + 3, 8) / 8;

 }
 

 static av_cold int alac_encode_close(AVCodecContext *avctx)
 {
     AlacEncodeContext *s = avctx->priv_data;
     ff_lpc_end(&s->lpc_ctx);
     av_freep(&avctx->extradata);
     avctx->extradata_size = 0;
     return 0;

 }
 
 static av_cold int alac_encode_init(AVCodecContext *avctx)
 {

     AlacEncodeContext *s = avctx->priv_data;

     int ret;

     uint8_t *alac_extradata;

     avctx->frame_size = s->frame_size = DEFAULT_FRAME_SIZE;

     if (avctx->sample_fmt == AV_SAMPLE_FMT_S32P) {
         if (avctx->bits_per_raw_sample != 24)
             av_log(avctx, AV_LOG_WARNING, "encoding as 24 bits-per-sample\n");
         avctx->bits_per_raw_sample = 24;
     } else {
         avctx->bits_per_raw_sample = 16;
         s->extra_bits              = 0;
     }
 

     // Set default compression level

     if (avctx->compression_level == FF_COMPRESSION_DEFAULT)

         s->compression_level = 2;

     else

         s->compression_level = av_clip(avctx->compression_level, 0, 2);

 
     // Initialize default Rice parameters
     s->rc.history_mult    = 40;
     s->rc.initial_history = 10;
     s->rc.k_modifier      = 14;
     s->rc.rice_modifier   = 4;
 

     s->max_coded_frame_size = get_max_frame_size(avctx->frame_size,
                                                  avctx->channels,

                                                  avctx->bits_per_raw_sample);

     avctx->extradata = av_mallocz(ALAC_EXTRADATA_SIZE + AV_INPUT_BUFFER_PADDING_SIZE);

     if (!avctx->extradata) {
         ret = AVERROR(ENOMEM);
         goto error;
     }
     avctx->extradata_size = ALAC_EXTRADATA_SIZE;
 
     alac_extradata = avctx->extradata;

     AV_WB32(alac_extradata,    ALAC_EXTRADATA_SIZE);
     AV_WB32(alac_extradata+4,  MKBETAG('a','l','a','c'));
     AV_WB32(alac_extradata+12, avctx->frame_size);

     AV_WB8 (alac_extradata+17, avctx->bits_per_raw_sample);

     AV_WB8 (alac_extradata+21, avctx->channels);

     AV_WB32(alac_extradata+24, s->max_coded_frame_size);

     AV_WB32(alac_extradata+28,

             avctx->sample_rate * avctx->channels * avctx->bits_per_raw_sample); // average bitrate

     AV_WB32(alac_extradata+32, avctx->sample_rate);

 
     // Set relevant extradata fields

     if (s->compression_level > 0) {

         AV_WB8(alac_extradata+18, s->rc.history_mult);
         AV_WB8(alac_extradata+19, s->rc.initial_history);
         AV_WB8(alac_extradata+20, s->rc.k_modifier);
     }
 

 #if FF_API_PRIVATE_OPT
 FF_DISABLE_DEPRECATION_WARNINGS

     if (avctx->min_prediction_order >= 0) {
         if (avctx->min_prediction_order < MIN_LPC_ORDER ||

            avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) {

             av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n",
                    avctx->min_prediction_order);

             ret = AVERROR(EINVAL);
             goto error;

         }
 
         s->min_prediction_order = avctx->min_prediction_order;
     }
 

     if (avctx->max_prediction_order >= 0) {
         if (avctx->max_prediction_order < MIN_LPC_ORDER ||
             avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) {
             av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n",
                    avctx->max_prediction_order);

             ret = AVERROR(EINVAL);
             goto error;

         }
 
         s->max_prediction_order = avctx->max_prediction_order;
     }

 FF_ENABLE_DEPRECATION_WARNINGS
 #endif

     if (s->max_prediction_order < s->min_prediction_order) {
         av_log(avctx, AV_LOG_ERROR,
                "invalid prediction orders: min=%d max=%d\n",

                s->min_prediction_order, s->max_prediction_order);

         ret = AVERROR(EINVAL);
         goto error;

     }
 

     s->avctx = avctx;
 

     if ((ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size,
                            s->max_prediction_order,
                            FF_LPC_TYPE_LEVINSON)) < 0) {
         goto error;
     }
 
     return 0;
 error:
     alac_encode_close(avctx);

     return ret;

 }
 

 static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                              const AVFrame *frame, int *got_packet_ptr)

 {
     AlacEncodeContext *s = avctx->priv_data;

     int out_bytes, max_frame_size, ret;

     s->frame_size = frame->nb_samples;

     if (frame->nb_samples < DEFAULT_FRAME_SIZE)

         max_frame_size = get_max_frame_size(s->frame_size, avctx->channels,

                                             avctx->bits_per_raw_sample);

     else
         max_frame_size = s->max_coded_frame_size;

     if ((ret = ff_alloc_packet2(avctx, avpkt, 2 * max_frame_size, 0)) < 0)

         return ret;

     /* use verbatim mode for compression_level 0 */

     if (s->compression_level) {
         s->verbatim   = 0;
         s->extra_bits = avctx->bits_per_raw_sample - 16;
     } else {
         s->verbatim   = 1;
         s->extra_bits = 0;
     }

     out_bytes = write_frame(s, avpkt, frame->extended_data);

     if (out_bytes > max_frame_size) {

         /* frame too large. use verbatim mode */

         s->verbatim = 1;

         s->extra_bits = 0;
         out_bytes = write_frame(s, avpkt, frame->extended_data);

     }
 

     avpkt->size = out_bytes;
     *got_packet_ptr = 1;

     return 0;
 }
 

 #define OFFSET(x) offsetof(AlacEncodeContext, x)
 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
 static const AVOption options[] = {
     { "min_prediction_order", NULL, OFFSET(min_prediction_order), AV_OPT_TYPE_INT, { .i64 = DEFAULT_MIN_PRED_ORDER }, MIN_LPC_ORDER, ALAC_MAX_LPC_ORDER, AE },
     { "max_prediction_order", NULL, OFFSET(max_prediction_order), AV_OPT_TYPE_INT, { .i64 = DEFAULT_MAX_PRED_ORDER }, MIN_LPC_ORDER, ALAC_MAX_LPC_ORDER, AE },
 
     { NULL },
 };
 
 static const AVClass alacenc_class = {
     .class_name = "alacenc",
     .item_name  = av_default_item_name,
     .option     = options,
     .version    = LIBAVUTIL_VERSION_INT,
 };
 

 AVCodec ff_alac_encoder = {

     .name           = "alac",

     .long_name      = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),

     .type           = AVMEDIA_TYPE_AUDIO,

     .id             = AV_CODEC_ID_ALAC,

     .priv_data_size = sizeof(AlacEncodeContext),

     .priv_class     = &alacenc_class,

     .init           = alac_encode_init,

     .encode2        = alac_encode_frame,

     .close          = alac_encode_close,

     .capabilities   = AV_CODEC_CAP_SMALL_LAST_FRAME,

     .channel_layouts = ff_alac_channel_layouts,

     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32P,
                                                      AV_SAMPLE_FMT_S16P,

                                                      AV_SAMPLE_FMT_NONE },

 };