/* * 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 Libav. * * Libav 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. * * Libav 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 Libav; 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 ALT_BITSTREAM_READER_LE #include "get_bits.h" #include "dsputil.h" #include "dct.h" #include "rdft.h" #include "fmtconvert.h" #include "libavutil/intfloat_readwrite.h" extern const uint16_t ff_wma_critical_freqs[25]; #define MAX_CHANNELS 2 #define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11) typedef struct { 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, short, previous)[BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave 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; 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); /* 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; 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; 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, short *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_int2flt(get_bits(gb, 32)) * s->root; coeffs[1] = av_int2flt(get_bits(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++) { /* constant is result of 0.066399999/log10(M_E) */ int value = get_bits(gb, 8); quant[i] = expf(FFMIN(value, 95) * 0.15289164787221953823f) * s->root; } 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(out, (const float **)s->coeffs_ptr, s->frame_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, out + s->block_size, s->overlap_len * s->channels * sizeof(*out)); s->first = 0; return 0; } static av_cold int decode_end(AVCodecContext *avctx) { BinkAudioContext * s = avctx->priv_data; av_freep(&s->bands); 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 *data_size, AVPacket *avpkt) { BinkAudioContext *s = avctx->priv_data; const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; short *samples = data; short *samples_end = (short*)((uint8_t*)data + *data_size); int reported_size; GetBitContext *gb = &s->gb; if (buf_size < 4) { av_log(avctx, AV_LOG_ERROR, "Packet is too small\n"); return AVERROR_INVALIDDATA; } init_get_bits(gb, buf, buf_size * 8); reported_size = get_bits_long(gb, 32); while (samples + s->block_size <= samples_end) { if (decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT)) break; samples += s->block_size; get_bits_align32(gb); } *data_size = FFMIN(reported_size, (uint8_t*)samples - (uint8_t*)data); return buf_size; } 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, .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, .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)") };