/*
  * Bink Audio decoder

  * Copyright (c) 2007-2011 Peter Ross (pross@xvid.org)

  * Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu)
  *
  * 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

  * Bink Audio decoder
  *
  * Technical details here:
  *  http://wiki.multimedia.cx/index.php?title=Bink_Audio
  */
 

 #include "libavutil/channel_layout.h"

 #include "avcodec.h"

 #define BITSTREAM_READER_LE

 #include "get_bits.h"

 #include "dct.h"
 #include "rdft.h"

 #include "fmtconvert.h"

 #include "internal.h"

 #include "wma.h"

 #include "libavutil/intfloat.h"

 static float quant_table[96];

 #define MAX_CHANNELS 2
 #define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
 
 typedef struct {
     GetBitContext gb;

     int version_b;          ///< Bink version 'b'

     int first;
     int channels;
     int frame_len;          ///< transform size (samples)
     int overlap_len;        ///< overlap size (samples)
     int block_size;
     int num_bands;
     unsigned int *bands;
     float root;

     DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];

     float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16];  ///< coeffs from previous audio block

     uint8_t *packet_buffer;

     union {
         RDFTContext rdft;
         DCTContext dct;
     } trans;
 } BinkAudioContext;
 
 
 static av_cold int decode_init(AVCodecContext *avctx)
 {
     BinkAudioContext *s = avctx->priv_data;
     int sample_rate = avctx->sample_rate;
     int sample_rate_half;
     int i;
     int frame_len_bits;
 
     /* determine frame length */
     if (avctx->sample_rate < 22050) {
         frame_len_bits = 9;
     } else if (avctx->sample_rate < 44100) {
         frame_len_bits = 10;
     } else {
         frame_len_bits = 11;
     }
 

     if (avctx->channels < 1 || avctx->channels > MAX_CHANNELS) {
         av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", avctx->channels);
         return AVERROR_INVALIDDATA;

     }

     avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :
                                                    AV_CH_LAYOUT_STEREO;

     s->version_b = avctx->extradata_size >= 4 && avctx->extradata[3] == 'b';

     if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) {

         // audio is already interleaved for the RDFT format variant

         avctx->sample_fmt = AV_SAMPLE_FMT_FLT;

         sample_rate  *= avctx->channels;
         s->channels = 1;

         if (!s->version_b)
             frame_len_bits += av_log2(avctx->channels);

     } else {
         s->channels = avctx->channels;

         avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

     }
 

     s->frame_len     = 1 << frame_len_bits;

     s->overlap_len   = s->frame_len / 16;
     s->block_size    = (s->frame_len - s->overlap_len) * s->channels;
     sample_rate_half = (sample_rate + 1) / 2;

     if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)

         s->root = 2.0 / (sqrt(s->frame_len) * 32768.0);

     else

         s->root = s->frame_len / (sqrt(s->frame_len) * 32768.0);

     for (i = 0; i < 96; i++) {

         /* constant is result of 0.066399999/log10(M_E) */
         quant_table[i] = expf(i * 0.15289164787221953823f) * s->root;
     }

 
     /* calculate number of bands */
     for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
         if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])
             break;
 
     s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands));
     if (!s->bands)
         return AVERROR(ENOMEM);
 
     /* populate bands data */

     s->bands[0] = 2;

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

         s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1;
     s->bands[s->num_bands] = s->frame_len;

 
     s->first = 1;
 

     if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)

         ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);

     else if (CONFIG_BINKAUDIO_DCT_DECODER)

         ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);

     else
         return -1;

 
     return 0;
 }
 
 static float get_float(GetBitContext *gb)
 {
     int power = get_bits(gb, 5);
     float f = ldexpf(get_bits_long(gb, 23), power - 23);
     if (get_bits1(gb))
         f = -f;
     return f;
 }
 
 static const uint8_t rle_length_tab[16] = {
     2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
 };
 
 /**
  * Decode Bink Audio block
  * @param[out] out Output buffer (must contain s->block_size elements)

  * @return 0 on success, negative error code on failure

  */

 static int decode_block(BinkAudioContext *s, float **out, int use_dct)

 {
     int ch, i, j, k;
     float q, quant[25];
     int width, coeff;
     GetBitContext *gb = &s->gb;
 
     if (use_dct)
         skip_bits(gb, 2);
 
     for (ch = 0; ch < s->channels; ch++) {

         FFTSample *coeffs = out[ch];
 

         if (s->version_b) {

             if (get_bits_left(gb) < 64)
                 return AVERROR_INVALIDDATA;

             coeffs[0] = av_int2float(get_bits_long(gb, 32)) * s->root;
             coeffs[1] = av_int2float(get_bits_long(gb, 32)) * s->root;

         } else {

             if (get_bits_left(gb) < 58)
                 return AVERROR_INVALIDDATA;

             coeffs[0] = get_float(gb) * s->root;
             coeffs[1] = get_float(gb) * s->root;
         }

         if (get_bits_left(gb) < s->num_bands * 8)
             return AVERROR_INVALIDDATA;

         for (i = 0; i < s->num_bands; i++) {
             int value = get_bits(gb, 8);

             quant[i]  = quant_table[FFMIN(value, 95)];

         }
 

         k = 0;
         q = quant[0];

 
         // parse coefficients
         i = 2;
         while (i < s->frame_len) {

             if (s->version_b) {
                 j = i + 16;

             } else {

                 int v = get_bits1(gb);

                 if (v) {

                     v = get_bits(gb, 4);

                     j = i + rle_length_tab[v] * 8;
                 } else {
                     j = i + 8;
                 }

             }
 
             j = FFMIN(j, s->frame_len);
 

             width = get_bits(gb, 4);

             if (width == 0) {
                 memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
                 i = j;

                 while (s->bands[k] < i)

                     q = quant[k++];
             } else {
                 while (i < j) {

                     if (s->bands[k] == i)

                         q = quant[k++];

                     coeff = get_bits(gb, width);

                     if (coeff) {

                         int v;

                         v = get_bits1(gb);

                         if (v)

                             coeffs[i] = -q * coeff;
                         else
                             coeffs[i] =  q * coeff;
                     } else {
                         coeffs[i] = 0.0f;
                     }
                     i++;
                 }
             }
         }
 

         if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
             coeffs[0] /= 0.5;

             s->trans.dct.dct_calc(&s->trans.dct,  coeffs);

         }

         else if (CONFIG_BINKAUDIO_RDFT_DECODER)

             s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);

     }
 

     for (ch = 0; ch < s->channels; ch++) {
         int j;

         int count = s->overlap_len * s->channels;

         if (!s->first) {
             j = ch;
             for (i = 0; i < s->overlap_len; i++, j += s->channels)
                 out[ch][i] = (s->previous[ch][i] * (count - j) +
                                       out[ch][i] *          j) / count;

         }

         memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len],
                s->overlap_len * sizeof(*s->previous[ch]));

     }
 
     s->first = 0;

 
     return 0;

 }
 
 static av_cold int decode_end(AVCodecContext *avctx)
 {
     BinkAudioContext * s = avctx->priv_data;
     av_freep(&s->bands);

     av_freep(&s->packet_buffer);

     if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)

         ff_rdft_end(&s->trans.rdft);

     else if (CONFIG_BINKAUDIO_DCT_DECODER)

         ff_dct_end(&s->trans.dct);

     return 0;
 }
 
 static void get_bits_align32(GetBitContext *s)
 {
     int n = (-get_bits_count(s)) & 31;
     if (n) skip_bits(s, n);
 }
 

 static int decode_frame(AVCodecContext *avctx, void *data,
                         int *got_frame_ptr, AVPacket *avpkt)

 {
     BinkAudioContext *s = avctx->priv_data;

     AVFrame *frame      = data;

     GetBitContext *gb = &s->gb;

     int ret, consumed = 0;

 
     if (!get_bits_left(gb)) {
         uint8_t *buf;
         /* handle end-of-stream */
         if (!avpkt->size) {

             *got_frame_ptr = 0;

             return 0;
         }
         if (avpkt->size < 4) {
             av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
             return AVERROR_INVALIDDATA;
         }
         buf = av_realloc(s->packet_buffer, avpkt->size + FF_INPUT_BUFFER_PADDING_SIZE);
         if (!buf)
             return AVERROR(ENOMEM);

         memset(buf + avpkt->size, 0, FF_INPUT_BUFFER_PADDING_SIZE);

         s->packet_buffer = buf;
         memcpy(s->packet_buffer, avpkt->data, avpkt->size);

         if ((ret = init_get_bits8(gb, s->packet_buffer, avpkt->size)) < 0)
             return ret;

         consumed = avpkt->size;
 
         /* skip reported size */
         skip_bits_long(gb, 32);

     }

     /* get output buffer */

     frame->nb_samples = s->frame_len;

     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)

         return ret;

     if (decode_block(s, (float **)frame->extended_data,

                      avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) {

         av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
         return AVERROR_INVALIDDATA;

     }

     get_bits_align32(gb);

     frame->nb_samples = s->block_size / avctx->channels;
     *got_frame_ptr    = 1;

     return consumed;

 }
 

 AVCodec ff_binkaudio_rdft_decoder = {

     .name           = "binkaudio_rdft",

     .long_name      = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)"),

     .type           = AVMEDIA_TYPE_AUDIO,

     .id             = AV_CODEC_ID_BINKAUDIO_RDFT,

     .priv_data_size = sizeof(BinkAudioContext),
     .init           = decode_init,
     .close          = decode_end,
     .decode         = decode_frame,

     .capabilities   = CODEC_CAP_DELAY | CODEC_CAP_DR1,

 };
 

 AVCodec ff_binkaudio_dct_decoder = {

     .name           = "binkaudio_dct",

     .long_name      = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)"),

     .type           = AVMEDIA_TYPE_AUDIO,

     .id             = AV_CODEC_ID_BINKAUDIO_DCT,

     .priv_data_size = sizeof(BinkAudioContext),
     .init           = decode_init,
     .close          = decode_end,
     .decode         = decode_frame,

     .capabilities   = CODEC_CAP_DELAY | CODEC_CAP_DR1,

 };