/*
  * 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 "avcodec.h"

 #define BITSTREAM_READER_LE

 #include "get_bits.h"
 #include "dsputil.h"

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

 #include "fmtconvert.h"

 #include "libavutil/intfloat.h"

 extern const uint16_t ff_wma_critical_freqs[25];
 

 static float quant_table[96];

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

     AVFrame frame;

     GetBitContext gb;
     DSPContext dsp;

     FmtConvertContext fmt_conv;

     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];

     DECLARE_ALIGNED(16, int16_t, previous)[BINK_BLOCK_MAX_SIZE / 16];  ///< coeffs from previous audio block

     DECLARE_ALIGNED(16, int16_t, current)[BINK_BLOCK_MAX_SIZE / 16];

     float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave

     float *prev_ptr[MAX_CHANNELS];   ///< pointers to the overlap points in the coeffs array

     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;
 

     ff_dsputil_init(&s->dsp, avctx);

     ff_fmt_convert_init(&s->fmt_conv, avctx);

 
     /* 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 > MAX_CHANNELS) {

         av_log(avctx, AV_LOG_ERROR, "too many channels: %d\n", avctx->channels);

         return -1;
     }
 

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

     if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) {
         // audio is already interleaved for the RDFT format variant
         sample_rate  *= avctx->channels;
         s->channels = 1;

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

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

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

     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;

     avctx->sample_fmt = AV_SAMPLE_FMT_S16;

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

         s->coeffs_ptr[i] = s->coeffs + i * s->frame_len;

         s->prev_ptr[i]   = s->coeffs_ptr[i] + s->frame_len - s->overlap_len;
     }

     if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == 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;

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

     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
 };
 

 #define GET_BITS_SAFE(out, nbits) do {  \
     if (get_bits_left(gb) < nbits)      \
         return AVERROR_INVALIDDATA;     \
     out = get_bits(gb, nbits);          \
 } while (0)
 

 /**
  * 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, int16_t *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 = s->coeffs_ptr[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_BITS_SAFE(v, 1);
                 if (v) {
                     GET_BITS_SAFE(v, 4);
                     j = i + rle_length_tab[v] * 8;
                 } else {
                     j = i + 8;
                 }

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

             GET_BITS_SAFE(width, 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++];

                     GET_BITS_SAFE(coeff, width);

                     if (coeff) {

                         int v;
                         GET_BITS_SAFE(v, 1);
                         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);

             s->dsp.vector_fmul_scalar(coeffs, coeffs, s->frame_len / 2, s->frame_len);
         }

         else if (CONFIG_BINKAUDIO_RDFT_DECODER)

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

     }
 

     s->fmt_conv.float_to_int16_interleave(s->current,
                                           (const float **)s->prev_ptr,
                                           s->overlap_len, s->channels);

     s->fmt_conv.float_to_int16_interleave(out, (const float **)s->coeffs_ptr,

                                           s->frame_len - s->overlap_len,
                                           s->channels);

 
     if (!s->first) {
         int count = s->overlap_len * s->channels;
         int shift = av_log2(count);
         for (i = 0; i < count; i++) {
             out[i] = (s->previous[i] * (count - i) + out[i] * i) >> shift;
         }
     }
 

     memcpy(s->previous, s->current,
            s->overlap_len * s->channels * sizeof(*s->previous));

 
     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 == 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;

     int16_t *samples;

     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);
         s->packet_buffer = buf;
         memcpy(s->packet_buffer, avpkt->data, avpkt->size);
         init_get_bits(gb, s->packet_buffer, avpkt->size * 8);
         consumed = avpkt->size;
 
         /* skip reported size */
         skip_bits_long(gb, 32);

     }

     /* get output buffer */
     s->frame.nb_samples = s->block_size / avctx->channels;
     if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;

     }

     samples = (int16_t *)s->frame.data[0];

     if (decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT)) {
         av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
         return AVERROR_INVALIDDATA;

     }

     get_bits_align32(gb);

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

     return consumed;

 }
 

 AVCodec ff_binkaudio_rdft_decoder = {

     .name           = "binkaudio_rdft",
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = 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,

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

 };
 

 AVCodec ff_binkaudio_dct_decoder = {

     .name           = "binkaudio_dct",
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = 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,

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

 };