| /* |
| * 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 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; |
| } |
| |
| if (avctx->extradata && avctx->extradata_size > 0) |
| s->version_b = avctx->extradata[0]; |
| |
| 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 = { |
| "binkaudio_rdft", |
| AVMEDIA_TYPE_AUDIO, |
| CODEC_ID_BINKAUDIO_RDFT, |
| sizeof(BinkAudioContext), |
| decode_init, |
| NULL, |
| decode_end, |
| decode_frame, |
| .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)") |
| }; |
| |
| AVCodec ff_binkaudio_dct_decoder = { |
| "binkaudio_dct", |
| AVMEDIA_TYPE_AUDIO, |
| CODEC_ID_BINKAUDIO_DCT, |
| sizeof(BinkAudioContext), |
| decode_init, |
| NULL, |
| decode_end, |
| decode_frame, |
| .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)") |
| }; |
