libavcodec/alacenc.c - nest-android-app/ffmpeg - Git at Google

 /*
  * 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 },
 };
	/*
	* 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 },
	};