/*
  * ALAC (Apple Lossless Audio Codec) decoder
  * Copyright (c) 2005 David Hammerton
  *

  * 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

  */
 
 /**

  * @file

  * ALAC (Apple Lossless Audio Codec) decoder
  * @author 2005 David Hammerton

  * @see http://crazney.net/programs/itunes/alac.html

  *

  * Note: This decoder expects a 36-byte QuickTime atom to be

  * passed through the extradata[_size] fields. This atom is tacked onto
  * the end of an 'alac' stsd atom and has the following format:

  *

  * 32bit  atom size
  * 32bit  tag                  ("alac")
  * 32bit  tag version          (0)
  * 32bit  samples per frame    (used when not set explicitly in the frames)
  *  8bit  compatible version   (0)

  *  8bit  sample size

  *  8bit  history mult         (40)
  *  8bit  initial history      (14)

  *  8bit  rice param limit     (10)

  *  8bit  channels
  * 16bit  maxRun               (255)
  * 32bit  max coded frame size (0 means unknown)
  * 32bit  average bitrate      (0 means unknown)

  * 32bit  samplerate

  */
 

 #include "libavutil/channel_layout.h"

 #include "avcodec.h"

 #include "get_bits.h"

 #include "bytestream.h"

 #include "internal.h"

 #include "unary.h"

 #include "mathops.h"

 #include "alac_data.h"

 
 #define ALAC_EXTRADATA_SIZE 36
 

 typedef struct {
     AVCodecContext *avctx;

     AVFrame frame;

     GetBitContext gb;

     int channels;

     int32_t *predict_error_buffer[2];
     int32_t *output_samples_buffer[2];
     int32_t *extra_bits_buffer[2];

     uint32_t max_samples_per_frame;
     uint8_t  sample_size;
     uint8_t  rice_history_mult;
     uint8_t  rice_initial_history;
     uint8_t  rice_limit;

     int extra_bits;     /**< number of extra bits beyond 16-bit */
     int nb_samples;     /**< number of samples in the current frame */

 
     int direct_output;

 } ALACContext;
 

 static inline unsigned int decode_scalar(GetBitContext *gb, int k, int bps)

 {

     unsigned int x = get_unary_0_9(gb);

 
     if (x > 8) { /* RICE THRESHOLD */
         /* use alternative encoding */

         x = get_bits_long(gb, bps);

     } else if (k != 1) {

         int extrabits = show_bits(gb, k);

         /* multiply x by 2^k - 1, as part of their strange algorithm */
         x = (x << k) - x;

         if (extrabits > 1) {
             x += extrabits - 1;
             skip_bits(gb, k);
         } else
             skip_bits(gb, k - 1);

     }

     return x;
 }
 

 static int rice_decompress(ALACContext *alac, int32_t *output_buffer,

                             int nb_samples, int bps, int rice_history_mult)

 {

     int i;

     unsigned int history = alac->rice_initial_history;

     int sign_modifier = 0;
 

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

         int k;
         unsigned int x;

         if(get_bits_left(&alac->gb) <= 0)
             return -1;
 

         /* calculate rice param and decode next value */

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

         k = FFMIN(k, alac->rice_limit);

         x = decode_scalar(&alac->gb, k, bps);

         x += sign_modifier;

         sign_modifier = 0;

         output_buffer[i] = (x >> 1) ^ -(x & 1);

         /* update the history */

         if (x > 0xffff)

             history = 0xffff;

         else

             history +=         x * rice_history_mult -
                        ((history * rice_history_mult) >> 9);

 
         /* special case: there may be compressed blocks of 0 */

         if ((history < 128) && (i + 1 < nb_samples)) {

             int block_size;

             /* calculate rice param and decode block size */
             k = 7 - av_log2(history) + ((history + 16) >> 6);

             k = FFMIN(k, alac->rice_limit);
             block_size = decode_scalar(&alac->gb, k, 16);

 
             if (block_size > 0) {

                 if (block_size >= nb_samples - i) {
                     av_log(alac->avctx, AV_LOG_ERROR,
                            "invalid zero block size of %d %d %d\n", block_size,
                            nb_samples, i);
                     block_size = nb_samples - i - 1;

                 }

                 memset(&output_buffer[i + 1], 0,

                        block_size * sizeof(*output_buffer));

                 i += block_size;

             }

             if (block_size <= 0xffff)
                 sign_modifier = 1;

             history = 0;
         }
     }

     return 0;

 }
 

 static inline int sign_only(int v)
 {
     return v ? FFSIGN(v) : 0;
 }

 static void lpc_prediction(int32_t *error_buffer, int32_t *buffer_out,
                            int nb_samples, int bps, int16_t *lpc_coefs,
                            int lpc_order, int lpc_quant)

 {
     int i;

     int32_t *pred = buffer_out;

 
     /* first sample always copies */
     *buffer_out = *error_buffer;
 

     if (nb_samples <= 1)

         return;

     if (!lpc_order) {

         memcpy(&buffer_out[1], &error_buffer[1],

                (nb_samples - 1) * sizeof(*buffer_out));

         return;
     }
 

     if (lpc_order == 31) {

         /* simple 1st-order prediction */

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

             buffer_out[i] = sign_extend(buffer_out[i - 1] + error_buffer[i],

                                         bps);

         }
         return;
     }
 
     /* read warm-up samples */

     for (i = 1; i <= lpc_order && i < nb_samples; i++)

         buffer_out[i] = sign_extend(buffer_out[i - 1] + error_buffer[i], bps);

     /* NOTE: 4 and 8 are very common cases that could be optimized. */

     for (; i < nb_samples; i++) {

         int j;

         int val = 0;

         int error_val = error_buffer[i];

         int error_sign;

         int d = *pred++;

         /* LPC prediction */
         for (j = 0; j < lpc_order; j++)

             val += (pred[j] - d) * lpc_coefs[j];

         val = (val + (1 << (lpc_quant - 1))) >> lpc_quant;

         val += d + error_val;

         buffer_out[i] = sign_extend(val, bps);

         /* adapt LPC coefficients */
         error_sign = sign_only(error_val);
         if (error_sign) {

             for (j = 0; j < lpc_order && error_val * error_sign > 0; j++) {

                 int sign;

                 val  = d - pred[j];

                 sign = sign_only(val) * error_sign;

                 lpc_coefs[j] -= sign;

                 val *= sign;

                 error_val -= (val >> lpc_quant) * (j + 1);

             }
         }
     }
 }
 

 static void decorrelate_stereo(int32_t *buffer[2], int nb_samples,
                                int decorr_shift, int decorr_left_weight)

 {

     int i;
 

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

         int32_t a, b;

         a = buffer[0][i];
         b = buffer[1][i];

         a -= (b * decorr_left_weight) >> decorr_shift;

         b += a;

         buffer[0][i] = b;
         buffer[1][i] = a;

     }

 }

 static void append_extra_bits(int32_t *buffer[2], int32_t *extra_bits_buffer[2],

                               int extra_bits, int channels, int nb_samples)

 {
     int i, ch;

     for (ch = 0; ch < channels; ch++)
         for (i = 0; i < nb_samples; i++)

             buffer[ch][i] = (buffer[ch][i] << extra_bits) | extra_bits_buffer[ch][i];

 }
 

 static int decode_element(AVCodecContext *avctx, void *data, int ch_index,
                           int channels)

 {

     ALACContext *alac = avctx->priv_data;

     int has_size, bps, is_compressed, decorr_shift, decorr_left_weight, ret;

     uint32_t output_samples;

     int i, ch;

     skip_bits(&alac->gb, 4);  /* element instance tag */
     skip_bits(&alac->gb, 12); /* unused header bits */

     /* the number of output samples is stored in the frame */

     has_size = get_bits1(&alac->gb);

     alac->extra_bits = get_bits(&alac->gb, 2) << 3;

     bps = alac->sample_size - alac->extra_bits + channels - 1;

     if (bps > 32U) {

         av_log(avctx, AV_LOG_ERROR, "bps is unsupported: %d\n", bps);

         return AVERROR_PATCHWELCOME;
     }

     /* whether the frame is compressed */

     is_compressed = !get_bits1(&alac->gb);

     if (has_size)

         output_samples = get_bits_long(&alac->gb, 32);
     else
         output_samples = alac->max_samples_per_frame;
     if (!output_samples || output_samples > alac->max_samples_per_frame) {
         av_log(avctx, AV_LOG_ERROR, "invalid samples per frame: %d\n",
                output_samples);

         return AVERROR_INVALIDDATA;
     }

     if (!alac->nb_samples) {
         /* get output buffer */
         alac->frame.nb_samples = output_samples;

         if ((ret = ff_get_buffer(avctx, &alac->frame)) < 0) {

             av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
             return ret;
         }
     } else if (output_samples != alac->nb_samples) {
         av_log(avctx, AV_LOG_ERROR, "sample count mismatch: %u != %d\n",
                output_samples, alac->nb_samples);
         return AVERROR_INVALIDDATA;

     }

     alac->nb_samples = output_samples;

     if (alac->direct_output) {

         for (ch = 0; ch < channels; ch++)
             alac->output_samples_buffer[ch] = (int32_t *)alac->frame.extended_data[ch_index + ch];
     }

     if (is_compressed) {

         int16_t lpc_coefs[2][32];
         int lpc_order[2];

         int prediction_type[2];

         int lpc_quant[2];
         int rice_history_mult[2];

         decorr_shift       = get_bits(&alac->gb, 8);
         decorr_left_weight = get_bits(&alac->gb, 8);

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

             prediction_type[ch]   = get_bits(&alac->gb, 4);
             lpc_quant[ch]         = get_bits(&alac->gb, 4);
             rice_history_mult[ch] = get_bits(&alac->gb, 3);
             lpc_order[ch]         = get_bits(&alac->gb, 5);

             if (lpc_order[ch] >= alac->max_samples_per_frame)
                 return AVERROR_INVALIDDATA;
 

             /* read the predictor table */

             for (i = lpc_order[ch] - 1; i >= 0; i--)

                 lpc_coefs[ch][i] = get_sbits(&alac->gb, 16);

         }

         if (alac->extra_bits) {

             for (i = 0; i < alac->nb_samples; i++) {

                 if(get_bits_left(&alac->gb) <= 0)
                     return -1;

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

                     alac->extra_bits_buffer[ch][i] = get_bits(&alac->gb, alac->extra_bits);

             }
         }

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

             int ret=rice_decompress(alac, alac->predict_error_buffer[ch],

                             alac->nb_samples, bps,
                             rice_history_mult[ch] * alac->rice_history_mult / 4);

             if(ret<0)
                 return ret;

             /* adaptive FIR filter */
             if (prediction_type[ch] == 15) {
                 /* Prediction type 15 runs the adaptive FIR twice.
                  * The first pass uses the special-case coef_num = 31, while
                  * the second pass uses the coefs from the bitstream.
                  *
                  * However, this prediction type is not currently used by the
                  * reference encoder.

                  */

                 lpc_prediction(alac->predict_error_buffer[ch],
                                alac->predict_error_buffer[ch],
                                alac->nb_samples, bps, NULL, 31, 0);

             } else if (prediction_type[ch] > 0) {
                 av_log(avctx, AV_LOG_WARNING, "unknown prediction type: %i\n",
                        prediction_type[ch]);

             }

             lpc_prediction(alac->predict_error_buffer[ch],
                            alac->output_samples_buffer[ch], alac->nb_samples,
                            bps, lpc_coefs[ch], lpc_order[ch], lpc_quant[ch]);

         }
     } else {
         /* not compressed, easy case */

         for (i = 0; i < alac->nb_samples; i++) {

             if(get_bits_left(&alac->gb) <= 0)
                 return -1;

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

                 alac->output_samples_buffer[ch][i] =
                          get_sbits_long(&alac->gb, alac->sample_size);

             }
         }

         alac->extra_bits   = 0;

         decorr_shift       = 0;
         decorr_left_weight = 0;

     }

     if (channels == 2 && decorr_left_weight) {

         decorrelate_stereo(alac->output_samples_buffer, alac->nb_samples,

                            decorr_shift, decorr_left_weight);

     }
 

     if (alac->extra_bits) {

         append_extra_bits(alac->output_samples_buffer, alac->extra_bits_buffer,

                           alac->extra_bits, channels, alac->nb_samples);

     }
 

     if(av_sample_fmt_is_planar(avctx->sample_fmt)) {

     switch(alac->sample_size) {

     case 16: {

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

             int16_t *outbuffer = (int16_t *)alac->frame.extended_data[ch_index + ch];

             for (i = 0; i < alac->nb_samples; i++)
                 *outbuffer++ = alac->output_samples_buffer[ch][i];

         }}

         break;

     case 24: {

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

             for (i = 0; i < alac->nb_samples; i++)
                 alac->output_samples_buffer[ch][i] <<= 8;

         }}

         break;
     }

     }else{
         switch(alac->sample_size) {
         case 16: {
             int16_t *outbuffer = ((int16_t *)alac->frame.extended_data[0]) + ch_index;

             for (i = 0; i < alac->nb_samples; i++) {

                 for (ch = 0; ch < channels; ch++)
                     *outbuffer++ = alac->output_samples_buffer[ch][i];

                 outbuffer += alac->channels - channels;
             }

             }
             break;
         case 24: {
             int32_t *outbuffer = ((int32_t *)alac->frame.extended_data[0]) + ch_index;

             for (i = 0; i < alac->nb_samples; i++) {

                 for (ch = 0; ch < channels; ch++)
                     *outbuffer++ = alac->output_samples_buffer[ch][i] << 8;

                 outbuffer += alac->channels - channels;
             }

             }
             break;
         case 32: {
             int32_t *outbuffer = ((int32_t *)alac->frame.extended_data[0]) + ch_index;

             for (i = 0; i < alac->nb_samples; i++) {

                 for (ch = 0; ch < channels; ch++)
                     *outbuffer++ = alac->output_samples_buffer[ch][i];

                 outbuffer += alac->channels - channels;
             }

             }
             break;
         }
     }

     return 0;
 }
 
 static int alac_decode_frame(AVCodecContext *avctx, void *data,
                              int *got_frame_ptr, AVPacket *avpkt)
 {
     ALACContext *alac = avctx->priv_data;

     enum AlacRawDataBlockType element;

     int channels;

     int ch, ret, got_end;

 
     init_get_bits(&alac->gb, avpkt->data, avpkt->size * 8);
 

     got_end = 0;

     alac->nb_samples = 0;
     ch = 0;

     while (get_bits_left(&alac->gb) >= 3) {

         element = get_bits(&alac->gb, 3);

         if (element == TYPE_END) {
             got_end = 1;

             break;

         }

         if (element > TYPE_CPE && element != TYPE_LFE) {

             av_log(avctx, AV_LOG_ERROR, "syntax element unsupported: %d\n", element);

             return AVERROR_PATCHWELCOME;

         }

 
         channels = (element == TYPE_CPE) ? 2 : 1;

         if (ch + channels > alac->channels ||
             ff_alac_channel_layout_offsets[alac->channels - 1][ch] + channels > alac->channels) {

             av_log(avctx, AV_LOG_ERROR, "invalid element channel count\n");
             return AVERROR_INVALIDDATA;
         }
 

         ret = decode_element(avctx, data,

                              ff_alac_channel_layout_offsets[alac->channels - 1][ch],

                              channels);

         if (ret < 0 && get_bits_left(&alac->gb))

             return ret;
 
         ch += channels;

     }

     if (!got_end) {
         av_log(avctx, AV_LOG_ERROR, "no end tag found. incomplete packet.\n");
         return AVERROR_INVALIDDATA;
     }

     if (avpkt->size * 8 - get_bits_count(&alac->gb) > 8) {

         av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n",
                avpkt->size * 8 - get_bits_count(&alac->gb));

     }

     *got_frame_ptr   = 1;
     *(AVFrame *)data = alac->frame;
 

     return avpkt->size;

 }
 

 static av_cold int alac_decode_close(AVCodecContext *avctx)
 {
     ALACContext *alac = avctx->priv_data;
 

     int ch;

     for (ch = 0; ch < FFMIN(alac->channels, 2); ch++) {

         av_freep(&alac->predict_error_buffer[ch]);

         if (!alac->direct_output)

             av_freep(&alac->output_samples_buffer[ch]);

         av_freep(&alac->extra_bits_buffer[ch]);

     }
 
     return 0;
 }
 
 static int allocate_buffers(ALACContext *alac)
 {

     int ch;

     int buf_size;
 
     if (alac->max_samples_per_frame > INT_MAX / sizeof(int32_t))
         goto buf_alloc_fail;
     buf_size = alac->max_samples_per_frame * sizeof(int32_t);

     for (ch = 0; ch < FFMIN(alac->channels, 2); ch++) {

         FF_ALLOC_OR_GOTO(alac->avctx, alac->predict_error_buffer[ch],

                          buf_size, buf_alloc_fail);

         alac->direct_output = alac->sample_size > 16 && av_sample_fmt_is_planar(alac->avctx->sample_fmt);
         if (!alac->direct_output) {

             FF_ALLOC_OR_GOTO(alac->avctx, alac->output_samples_buffer[ch],
                              buf_size, buf_alloc_fail);
         }

         FF_ALLOC_OR_GOTO(alac->avctx, alac->extra_bits_buffer[ch],
                          buf_size, buf_alloc_fail);

     }
     return 0;

 buf_alloc_fail:
     alac_decode_close(alac->avctx);
     return AVERROR(ENOMEM);

 }
 
 static int alac_set_info(ALACContext *alac)
 {

     GetByteContext gb;

     bytestream2_init(&gb, alac->avctx->extradata,
                      alac->avctx->extradata_size);

     bytestream2_skipu(&gb, 12); // size:4, alac:4, version:4

     alac->max_samples_per_frame = bytestream2_get_be32u(&gb);

     if (!alac->max_samples_per_frame ||
         alac->max_samples_per_frame > INT_MAX / sizeof(int32_t)) {

         av_log(alac->avctx, AV_LOG_ERROR, "max samples per frame invalid: %u\n",
                alac->max_samples_per_frame);

         return AVERROR_INVALIDDATA;
     }
     bytestream2_skipu(&gb, 1);  // compatible version

     alac->sample_size          = bytestream2_get_byteu(&gb);
     alac->rice_history_mult    = bytestream2_get_byteu(&gb);
     alac->rice_initial_history = bytestream2_get_byteu(&gb);
     alac->rice_limit           = bytestream2_get_byteu(&gb);
     alac->channels             = bytestream2_get_byteu(&gb);

     bytestream2_get_be16u(&gb); // maxRun
     bytestream2_get_be32u(&gb); // max coded frame size
     bytestream2_get_be32u(&gb); // average bitrate
     bytestream2_get_be32u(&gb); // samplerate

 
     return 0;
 }
 

 static av_cold int alac_decode_init(AVCodecContext * avctx)

 {

     int ret;

     int req_packed;

     ALACContext *alac = avctx->priv_data;
     alac->avctx = avctx;

     /* initialize from the extradata */

     if (alac->avctx->extradata_size < ALAC_EXTRADATA_SIZE) {

         av_log(avctx, AV_LOG_ERROR, "extradata is too small\n");

         return AVERROR_INVALIDDATA;

     }
     if (alac_set_info(alac)) {

         av_log(avctx, AV_LOG_ERROR, "set_info failed\n");

         return -1;
     }
 

     req_packed = LIBAVCODEC_VERSION_MAJOR < 55 && !av_sample_fmt_is_planar(avctx->request_sample_fmt);

     switch (alac->sample_size) {

     case 16: avctx->sample_fmt = req_packed ? AV_SAMPLE_FMT_S16 : AV_SAMPLE_FMT_S16P;

              break;

     case 24:

     case 32: avctx->sample_fmt = req_packed ? AV_SAMPLE_FMT_S32 : AV_SAMPLE_FMT_S32P;

              break;

     default: av_log_ask_for_sample(avctx, "Sample depth %d is not supported.\n",

                                    alac->sample_size);

              return AVERROR_PATCHWELCOME;

     }

     avctx->bits_per_raw_sample = alac->sample_size;

     if (alac->channels < 1) {

         av_log(avctx, AV_LOG_WARNING, "Invalid channel count\n");

         alac->channels = avctx->channels;

     } else {

         if (alac->channels > ALAC_MAX_CHANNELS)

             alac->channels = avctx->channels;

         else

             avctx->channels = alac->channels;

     }

     if (avctx->channels > ALAC_MAX_CHANNELS || avctx->channels <= 0 ) {

         av_log(avctx, AV_LOG_ERROR, "Unsupported channel count: %d\n",
                avctx->channels);
         return AVERROR_PATCHWELCOME;
     }

     avctx->channel_layout = ff_alac_channel_layouts[alac->channels - 1];

     if ((ret = allocate_buffers(alac)) < 0) {
         av_log(avctx, AV_LOG_ERROR, "Error allocating buffers\n");
         return ret;

     }

     avcodec_get_frame_defaults(&alac->frame);
     avctx->coded_frame = &alac->frame;
 

     return 0;
 }
 

 AVCodec ff_alac_decoder = {

     .name           = "alac",
     .type           = AVMEDIA_TYPE_AUDIO,

     .id             = AV_CODEC_ID_ALAC,

     .priv_data_size = sizeof(ALACContext),
     .init           = alac_decode_init,
     .close          = alac_decode_close,
     .decode         = alac_decode_frame,

     .capabilities   = CODEC_CAP_DR1,

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

 };