Mercurial > libavcodec.hg
view binkaudio.c @ 11776:906fdc96cdf4 libavcodec
Mark vorbis encoder as experimental.
author | cehoyos |
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date | Wed, 26 May 2010 18:54:59 +0000 |
parents | 7dd2a45249a9 |
children |
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/* * 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 * 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 "fft.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 (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 }; /** * 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 (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) { coeffs[0] /= 0.5; ff_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) ff_rdft_calc(&s->trans.rdft, coeffs); } if (s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) { for (i = 0; i < s->channels; i++) for (j = 0; j < s->frame_len; j++) s->coeffs_ptr[i][j] = 385.0 + s->coeffs_ptr[i][j]*(1.0/32767.0); } 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 (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; 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 = FFMIN(reported_size, (uint8_t*)samples - (uint8_t*)data); return buf_size; } AVCodec 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 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)") };