diff options
author | Peter Ross <pross@xvid.org> | 2010-01-31 15:51:15 +0300 |
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committer | Peter Ross <pross@xvid.org> | 2010-01-31 15:51:15 +0300 |
commit | c0d3f516cb18f87995153e46acbb62563b3f2969 (patch) | |
tree | 69c5d0bd8665cc44926746da1e0cdbda61df2a51 /libavcodec/binkaudio.c | |
parent | 2e375df5b257cb2c74bb80ec5cd46f6957c7ecae (diff) |
Bink Audio decoder
Originally committed as revision 21570 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/binkaudio.c')
-rw-r--r-- | libavcodec/binkaudio.c | 303 |
1 files changed, 303 insertions, 0 deletions
diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c new file mode 100644 index 0000000000..8704af7358 --- /dev/null +++ b/libavcodec/binkaudio.c @@ -0,0 +1,303 @@ +/* + * Bink Audio decoder + * Copyright (c) 2007-2010 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 libavcodec/binkaudio.c + * 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" +extern const uint16_t ff_wma_critical_freqs[25]; + +#define MAX_CHANNELS 2 +#define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11) + +typedef struct { + AVCodecContext *avctx; + GetBitContext gb; + DSPContext dsp; + 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_16(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; + + s->avctx = avctx; + dsputil_init(&s->dsp, 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; + } + s->frame_len = 1 << frame_len_bits; + + if (s->channels > MAX_CHANNELS) { + av_log(s->avctx, AV_LOG_ERROR, "too many channels: %d\n", s->channels); + return -1; + } + + if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) { + // audio is already interleaved for the RDFT format variant + sample_rate *= avctx->channels; + s->frame_len *= avctx->channels; + s->channels = 1; + if (avctx->channels == 2) + frame_len_bits++; + } else { + s->channels = avctx->channels; + } + + 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] = 1; + for (i = 1; i < s->num_bands; i++) + s->bands[i] = ff_wma_critical_freqs[i - 1] * (s->frame_len / 2) / sample_rate_half; + s->bands[s->num_bands] = s->frame_len / 2; + + s->first = 1; + avctx->sample_fmt = SAMPLE_FMT_S16; + + for (i = 0; i < s->channels; i++) + s->coeffs_ptr[i] = s->coeffs + i * s->frame_len; + + if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) + ff_rdft_init(&s->trans.rdft, frame_len_bits, IRIDFT); + else + ff_dct_init(&s->trans.dct, frame_len_bits, 0); + + 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) + */ +static void 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]; + q = 0.0f; + coeffs[0] = get_float(gb) * s->root; + coeffs[1] = get_float(gb) * s->root; + + 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; + } + + // find band (k) + for (k = 0; s->bands[k] < 1; k++) { + q = quant[k]; + } + + // parse coefficients + i = 2; + while (i < s->frame_len) { + if (get_bits1(gb)) { + j = i + rle_length_tab[get_bits(gb, 4)] * 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] * 2 < i) + q = quant[k++]; + } else { + while (i < j) { + if (s->bands[k] * 2 == i) + q = quant[k++]; + coeff = get_bits(gb, width); + if (coeff) { + if (get_bits1(gb)) + coeffs[i] = -q * coeff; + else + coeffs[i] = q * coeff; + } else { + coeffs[i] = 0.0f; + } + i++; + } + } + } + + if (use_dct) + ff_dct_calc (&s->trans.dct, coeffs); + else + ff_rdft_calc(&s->trans.rdft, coeffs); + } + + s->dsp.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; +} + +static av_cold int decode_end(AVCodecContext *avctx) +{ + BinkAudioContext * s = avctx->priv_data; + av_freep(&s->bands); + if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) + ff_rdft_end(&s->trans.rdft); + else + 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; + + init_get_bits(gb, buf, buf_size * 8); + + reported_size = get_bits_long(gb, 32); + while (get_bits_count(gb) / 8 < buf_size && + samples + s->block_size <= samples_end) { + decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT); + samples += s->block_size; + get_bits_align32(gb); + } + + *data_size = (uint8_t*)samples - (uint8_t*)data; + if (reported_size != *data_size) { + av_log(avctx, AV_LOG_WARNING, "reported data size (%d) does not match output data size (%d)\n", + reported_size, *data_size); + } + return buf_size; +} + +AVCodec binkaudio_rdft_decoder = { + "binkaudio_rdft", + CODEC_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 binkaudio_dct_decoder = { + "binkaudio_dct", + CODEC_TYPE_AUDIO, + CODEC_ID_BINKAUDIO_DCT, + sizeof(BinkAudioContext), + decode_init, + NULL, + decode_end, + decode_frame, + .long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)") +}; |