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  1. /*
  2.  * Bink Audio decoder
  3.  * Copyright (c) 2007-2011 Peter Ross (pross@xvid.org)
  4.  * Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu)
  5.  *
  6.  * This file is part of FFmpeg.
  7.  *
  8.  * FFmpeg is free software; you can redistribute it and/or
  9.  * modify it under the terms of the GNU Lesser General Public
  10.  * License as published by the Free Software Foundation; either
  11.  * version 2.1 of the License, or (at your option) any later version.
  12.  *
  13.  * FFmpeg is distributed in the hope that it will be useful,
  14.  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  15.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  16.  * Lesser General Public License for more details.
  17.  *
  18.  * You should have received a copy of the GNU Lesser General Public
  19.  * License along with FFmpeg; if not, write to the Free Software
  20.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  21.  */
  22.  
  23. /**
  24.  * @file
  25.  * Bink Audio decoder
  26.  *
  27.  * Technical details here:
  28.  *  http://wiki.multimedia.cx/index.php?title=Bink_Audio
  29.  */
  30.  
  31. #include "libavutil/channel_layout.h"
  32. #include "avcodec.h"
  33. #define BITSTREAM_READER_LE
  34. #include "get_bits.h"
  35. #include "dct.h"
  36. #include "rdft.h"
  37. #include "fmtconvert.h"
  38. #include "internal.h"
  39. #include "wma.h"
  40. #include "libavutil/intfloat.h"
  41.  
  42. static float quant_table[96];
  43.  
  44. #define MAX_CHANNELS 2
  45. #define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
  46.  
  47. typedef struct {
  48.     GetBitContext gb;
  49.     int version_b;          ///< Bink version 'b'
  50.     int first;
  51.     int channels;
  52.     int frame_len;          ///< transform size (samples)
  53.     int overlap_len;        ///< overlap size (samples)
  54.     int block_size;
  55.     int num_bands;
  56.     unsigned int *bands;
  57.     float root;
  58.     DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
  59.     float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16];  ///< coeffs from previous audio block
  60.     uint8_t *packet_buffer;
  61.     union {
  62.         RDFTContext rdft;
  63.         DCTContext dct;
  64.     } trans;
  65. } BinkAudioContext;
  66.  
  67.  
  68. static av_cold int decode_init(AVCodecContext *avctx)
  69. {
  70.     BinkAudioContext *s = avctx->priv_data;
  71.     int sample_rate = avctx->sample_rate;
  72.     int sample_rate_half;
  73.     int i;
  74.     int frame_len_bits;
  75.  
  76.     /* determine frame length */
  77.     if (avctx->sample_rate < 22050) {
  78.         frame_len_bits = 9;
  79.     } else if (avctx->sample_rate < 44100) {
  80.         frame_len_bits = 10;
  81.     } else {
  82.         frame_len_bits = 11;
  83.     }
  84.  
  85.     if (avctx->channels < 1 || avctx->channels > MAX_CHANNELS) {
  86.         av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", avctx->channels);
  87.         return AVERROR_INVALIDDATA;
  88.     }
  89.     avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :
  90.                                                    AV_CH_LAYOUT_STEREO;
  91.  
  92.     s->version_b = avctx->extradata_size >= 4 && avctx->extradata[3] == 'b';
  93.  
  94.     if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) {
  95.         // audio is already interleaved for the RDFT format variant
  96.         avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
  97.         sample_rate  *= avctx->channels;
  98.         s->channels = 1;
  99.         if (!s->version_b)
  100.             frame_len_bits += av_log2(avctx->channels);
  101.     } else {
  102.         s->channels = avctx->channels;
  103.         avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
  104.     }
  105.  
  106.     s->frame_len     = 1 << frame_len_bits;
  107.     s->overlap_len   = s->frame_len / 16;
  108.     s->block_size    = (s->frame_len - s->overlap_len) * s->channels;
  109.     sample_rate_half = (sample_rate + 1) / 2;
  110.     if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
  111.         s->root = 2.0 / (sqrt(s->frame_len) * 32768.0);
  112.     else
  113.         s->root = s->frame_len / (sqrt(s->frame_len) * 32768.0);
  114.     for (i = 0; i < 96; i++) {
  115.         /* constant is result of 0.066399999/log10(M_E) */
  116.         quant_table[i] = expf(i * 0.15289164787221953823f) * s->root;
  117.     }
  118.  
  119.     /* calculate number of bands */
  120.     for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
  121.         if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])
  122.             break;
  123.  
  124.     s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands));
  125.     if (!s->bands)
  126.         return AVERROR(ENOMEM);
  127.  
  128.     /* populate bands data */
  129.     s->bands[0] = 2;
  130.     for (i = 1; i < s->num_bands; i++)
  131.         s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1;
  132.     s->bands[s->num_bands] = s->frame_len;
  133.  
  134.     s->first = 1;
  135.  
  136.     if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
  137.         ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
  138.     else if (CONFIG_BINKAUDIO_DCT_DECODER)
  139.         ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);
  140.     else
  141.         return -1;
  142.  
  143.     return 0;
  144. }
  145.  
  146. static float get_float(GetBitContext *gb)
  147. {
  148.     int power = get_bits(gb, 5);
  149.     float f = ldexpf(get_bits_long(gb, 23), power - 23);
  150.     if (get_bits1(gb))
  151.         f = -f;
  152.     return f;
  153. }
  154.  
  155. static const uint8_t rle_length_tab[16] = {
  156.     2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
  157. };
  158.  
  159. /**
  160.  * Decode Bink Audio block
  161.  * @param[out] out Output buffer (must contain s->block_size elements)
  162.  * @return 0 on success, negative error code on failure
  163.  */
  164. static int decode_block(BinkAudioContext *s, float **out, int use_dct)
  165. {
  166.     int ch, i, j, k;
  167.     float q, quant[25];
  168.     int width, coeff;
  169.     GetBitContext *gb = &s->gb;
  170.  
  171.     if (use_dct)
  172.         skip_bits(gb, 2);
  173.  
  174.     for (ch = 0; ch < s->channels; ch++) {
  175.         FFTSample *coeffs = out[ch];
  176.  
  177.         if (s->version_b) {
  178.             if (get_bits_left(gb) < 64)
  179.                 return AVERROR_INVALIDDATA;
  180.             coeffs[0] = av_int2float(get_bits_long(gb, 32)) * s->root;
  181.             coeffs[1] = av_int2float(get_bits_long(gb, 32)) * s->root;
  182.         } else {
  183.             if (get_bits_left(gb) < 58)
  184.                 return AVERROR_INVALIDDATA;
  185.             coeffs[0] = get_float(gb) * s->root;
  186.             coeffs[1] = get_float(gb) * s->root;
  187.         }
  188.  
  189.         if (get_bits_left(gb) < s->num_bands * 8)
  190.             return AVERROR_INVALIDDATA;
  191.         for (i = 0; i < s->num_bands; i++) {
  192.             int value = get_bits(gb, 8);
  193.             quant[i]  = quant_table[FFMIN(value, 95)];
  194.         }
  195.  
  196.         k = 0;
  197.         q = quant[0];
  198.  
  199.         // parse coefficients
  200.         i = 2;
  201.         while (i < s->frame_len) {
  202.             if (s->version_b) {
  203.                 j = i + 16;
  204.             } else {
  205.                 int v = get_bits1(gb);
  206.                 if (v) {
  207.                     v = get_bits(gb, 4);
  208.                     j = i + rle_length_tab[v] * 8;
  209.                 } else {
  210.                     j = i + 8;
  211.                 }
  212.             }
  213.  
  214.             j = FFMIN(j, s->frame_len);
  215.  
  216.             width = get_bits(gb, 4);
  217.             if (width == 0) {
  218.                 memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
  219.                 i = j;
  220.                 while (s->bands[k] < i)
  221.                     q = quant[k++];
  222.             } else {
  223.                 while (i < j) {
  224.                     if (s->bands[k] == i)
  225.                         q = quant[k++];
  226.                     coeff = get_bits(gb, width);
  227.                     if (coeff) {
  228.                         int v;
  229.                         v = get_bits1(gb);
  230.                         if (v)
  231.                             coeffs[i] = -q * coeff;
  232.                         else
  233.                             coeffs[i] =  q * coeff;
  234.                     } else {
  235.                         coeffs[i] = 0.0f;
  236.                     }
  237.                     i++;
  238.                 }
  239.             }
  240.         }
  241.  
  242.         if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
  243.             coeffs[0] /= 0.5;
  244.             s->trans.dct.dct_calc(&s->trans.dct,  coeffs);
  245.         }
  246.         else if (CONFIG_BINKAUDIO_RDFT_DECODER)
  247.             s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
  248.     }
  249.  
  250.     for (ch = 0; ch < s->channels; ch++) {
  251.         int j;
  252.         int count = s->overlap_len * s->channels;
  253.         if (!s->first) {
  254.             j = ch;
  255.             for (i = 0; i < s->overlap_len; i++, j += s->channels)
  256.                 out[ch][i] = (s->previous[ch][i] * (count - j) +
  257.                                       out[ch][i] *          j) / count;
  258.         }
  259.         memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len],
  260.                s->overlap_len * sizeof(*s->previous[ch]));
  261.     }
  262.  
  263.     s->first = 0;
  264.  
  265.     return 0;
  266. }
  267.  
  268. static av_cold int decode_end(AVCodecContext *avctx)
  269. {
  270.     BinkAudioContext * s = avctx->priv_data;
  271.     av_freep(&s->bands);
  272.     av_freep(&s->packet_buffer);
  273.     if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
  274.         ff_rdft_end(&s->trans.rdft);
  275.     else if (CONFIG_BINKAUDIO_DCT_DECODER)
  276.         ff_dct_end(&s->trans.dct);
  277.  
  278.     return 0;
  279. }
  280.  
  281. static void get_bits_align32(GetBitContext *s)
  282. {
  283.     int n = (-get_bits_count(s)) & 31;
  284.     if (n) skip_bits(s, n);
  285. }
  286.  
  287. static int decode_frame(AVCodecContext *avctx, void *data,
  288.                         int *got_frame_ptr, AVPacket *avpkt)
  289. {
  290.     BinkAudioContext *s = avctx->priv_data;
  291.     AVFrame *frame      = data;
  292.     GetBitContext *gb = &s->gb;
  293.     int ret, consumed = 0;
  294.  
  295.     if (!get_bits_left(gb)) {
  296.         uint8_t *buf;
  297.         /* handle end-of-stream */
  298.         if (!avpkt->size) {
  299.             *got_frame_ptr = 0;
  300.             return 0;
  301.         }
  302.         if (avpkt->size < 4) {
  303.             av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
  304.             return AVERROR_INVALIDDATA;
  305.         }
  306.         buf = av_realloc(s->packet_buffer, avpkt->size + FF_INPUT_BUFFER_PADDING_SIZE);
  307.         if (!buf)
  308.             return AVERROR(ENOMEM);
  309.         s->packet_buffer = buf;
  310.         memcpy(s->packet_buffer, avpkt->data, avpkt->size);
  311.         init_get_bits(gb, s->packet_buffer, avpkt->size * 8);
  312.         consumed = avpkt->size;
  313.  
  314.         /* skip reported size */
  315.         skip_bits_long(gb, 32);
  316.     }
  317.  
  318.     /* get output buffer */
  319.     frame->nb_samples = s->frame_len;
  320.     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
  321.         return ret;
  322.  
  323.     if (decode_block(s, (float **)frame->extended_data,
  324.                      avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) {
  325.         av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
  326.         return AVERROR_INVALIDDATA;
  327.     }
  328.     get_bits_align32(gb);
  329.  
  330.     frame->nb_samples = s->block_size / avctx->channels;
  331.     *got_frame_ptr    = 1;
  332.  
  333.     return consumed;
  334. }
  335.  
  336. AVCodec ff_binkaudio_rdft_decoder = {
  337.     .name           = "binkaudio_rdft",
  338.     .long_name      = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)"),
  339.     .type           = AVMEDIA_TYPE_AUDIO,
  340.     .id             = AV_CODEC_ID_BINKAUDIO_RDFT,
  341.     .priv_data_size = sizeof(BinkAudioContext),
  342.     .init           = decode_init,
  343.     .close          = decode_end,
  344.     .decode         = decode_frame,
  345.     .capabilities   = CODEC_CAP_DELAY | CODEC_CAP_DR1,
  346. };
  347.  
  348. AVCodec ff_binkaudio_dct_decoder = {
  349.     .name           = "binkaudio_dct",
  350.     .long_name      = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)"),
  351.     .type           = AVMEDIA_TYPE_AUDIO,
  352.     .id             = AV_CODEC_ID_BINKAUDIO_DCT,
  353.     .priv_data_size = sizeof(BinkAudioContext),
  354.     .init           = decode_init,
  355.     .close          = decode_end,
  356.     .decode         = decode_frame,
  357.     .capabilities   = CODEC_CAP_DELAY | CODEC_CAP_DR1,
  358. };
  359.