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d7e5aebae7
* qatar/master: (23 commits) ac3enc: correct the flipped sign in the ac3_fixed encoder Eliminate pointless '#if 1' statements without matching '#else'. Add AVX FFT implementation. Increase alignment of av_malloc() as needed by AVX ASM. Update x86inc.asm from x264 to allow AVX emulation using SSE and MMX. mjpeg: Detect overreads in mjpeg_decode_scan() and error out. documentation: extend documentation for ffmpeg -aspect option APIChanges: update commit hashes for recent additions. lavc: deprecate FF_*_TYPE macros in favor of AV_PICTURE_TYPE_* enums aac: add headers needed for log2f() lavc: remove FF_API_MB_Q cruft lavc: remove FF_API_RATE_EMU cruft lavc: remove FF_API_HURRY_UP cruft pad: make the filter parametric vsrc_movie: add key_frame and pict_type. vsrc_movie: fix leak in request_frame() lavfi: add key_frame and pict_type to AVFilterBufferRefVideo. vsrc_buffer: add sample_aspect_ratio fields to arguments. lavfi: add fieldorder filter scale: make the filter parametric ... Conflicts: Changelog doc/filters.texi ffmpeg.c libavcodec/ac3dec.h libavcodec/dsputil.c libavfilter/avfilter.h libavfilter/vf_scale.c libavfilter/vf_yadif.c libavfilter/vsrc_buffer.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
317 lines
9.3 KiB
C
317 lines
9.3 KiB
C
/*
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* Bink Audio decoder
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* Copyright (c) 2007-2011 Peter Ross (pross@xvid.org)
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* Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu)
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Bink Audio decoder
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*
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* Technical details here:
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* http://wiki.multimedia.cx/index.php?title=Bink_Audio
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*/
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#include "avcodec.h"
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#define ALT_BITSTREAM_READER_LE
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#include "get_bits.h"
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#include "dsputil.h"
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#include "dct.h"
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#include "rdft.h"
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#include "fmtconvert.h"
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#include "libavutil/intfloat_readwrite.h"
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extern const uint16_t ff_wma_critical_freqs[25];
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#define MAX_CHANNELS 2
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#define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
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typedef struct {
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GetBitContext gb;
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DSPContext dsp;
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FmtConvertContext fmt_conv;
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int version_b; ///< Bink version 'b'
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int first;
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int channels;
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int frame_len; ///< transform size (samples)
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int overlap_len; ///< overlap size (samples)
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int block_size;
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int num_bands;
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unsigned int *bands;
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float root;
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DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
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DECLARE_ALIGNED(16, short, previous)[BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block
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float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave
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union {
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RDFTContext rdft;
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DCTContext dct;
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} trans;
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} BinkAudioContext;
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static av_cold int decode_init(AVCodecContext *avctx)
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{
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BinkAudioContext *s = avctx->priv_data;
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int sample_rate = avctx->sample_rate;
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int sample_rate_half;
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int i;
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int frame_len_bits;
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dsputil_init(&s->dsp, avctx);
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ff_fmt_convert_init(&s->fmt_conv, avctx);
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/* determine frame length */
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if (avctx->sample_rate < 22050) {
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frame_len_bits = 9;
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} else if (avctx->sample_rate < 44100) {
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frame_len_bits = 10;
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} else {
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frame_len_bits = 11;
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}
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if (avctx->channels > MAX_CHANNELS) {
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av_log(avctx, AV_LOG_ERROR, "too many channels: %d\n", avctx->channels);
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return -1;
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}
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s->version_b = avctx->codec_tag == MKTAG('B','I','K','b');
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if (avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT) {
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// audio is already interleaved for the RDFT format variant
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sample_rate *= avctx->channels;
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s->channels = 1;
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if (!s->version_b)
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frame_len_bits += av_log2(avctx->channels);
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} else {
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s->channels = avctx->channels;
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}
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s->frame_len = 1 << frame_len_bits;
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s->overlap_len = s->frame_len / 16;
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s->block_size = (s->frame_len - s->overlap_len) * s->channels;
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sample_rate_half = (sample_rate + 1) / 2;
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s->root = 2.0 / sqrt(s->frame_len);
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/* calculate number of bands */
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for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
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if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])
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break;
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s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands));
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if (!s->bands)
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return AVERROR(ENOMEM);
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/* populate bands data */
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s->bands[0] = 2;
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for (i = 1; i < s->num_bands; i++)
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s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1;
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s->bands[s->num_bands] = s->frame_len;
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s->first = 1;
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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for (i = 0; i < s->channels; i++)
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s->coeffs_ptr[i] = s->coeffs + i * s->frame_len;
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if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
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ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
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else if (CONFIG_BINKAUDIO_DCT_DECODER)
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ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);
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else
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return -1;
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return 0;
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}
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static float get_float(GetBitContext *gb)
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{
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int power = get_bits(gb, 5);
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float f = ldexpf(get_bits_long(gb, 23), power - 23);
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if (get_bits1(gb))
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f = -f;
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return f;
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}
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static const uint8_t rle_length_tab[16] = {
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2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
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};
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/**
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* Decode Bink Audio block
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* @param[out] out Output buffer (must contain s->block_size elements)
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*/
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static void decode_block(BinkAudioContext *s, short *out, int use_dct)
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{
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int ch, i, j, k;
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float q, quant[25];
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int width, coeff;
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GetBitContext *gb = &s->gb;
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if (use_dct)
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skip_bits(gb, 2);
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for (ch = 0; ch < s->channels; ch++) {
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FFTSample *coeffs = s->coeffs_ptr[ch];
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if (s->version_b) {
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coeffs[0] = av_int2flt(get_bits(gb, 32)) * s->root;
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coeffs[1] = av_int2flt(get_bits(gb, 32)) * s->root;
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} else {
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coeffs[0] = get_float(gb) * s->root;
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coeffs[1] = get_float(gb) * s->root;
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}
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for (i = 0; i < s->num_bands; i++) {
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/* constant is result of 0.066399999/log10(M_E) */
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int value = get_bits(gb, 8);
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quant[i] = expf(FFMIN(value, 95) * 0.15289164787221953823f) * s->root;
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}
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k = 0;
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q = quant[0];
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// parse coefficients
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i = 2;
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while (i < s->frame_len) {
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if (s->version_b) {
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j = i + 16;
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} else if (get_bits1(gb)) {
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j = i + rle_length_tab[get_bits(gb, 4)] * 8;
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} else {
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j = i + 8;
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}
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j = FFMIN(j, s->frame_len);
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width = get_bits(gb, 4);
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if (width == 0) {
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memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
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i = j;
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while (s->bands[k] < i)
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q = quant[k++];
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} else {
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while (i < j) {
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if (s->bands[k] == i)
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q = quant[k++];
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coeff = get_bits(gb, width);
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if (coeff) {
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if (get_bits1(gb))
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coeffs[i] = -q * coeff;
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else
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coeffs[i] = q * coeff;
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} else {
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coeffs[i] = 0.0f;
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}
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i++;
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}
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}
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}
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if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
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coeffs[0] /= 0.5;
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s->trans.dct.dct_calc(&s->trans.dct, coeffs);
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s->dsp.vector_fmul_scalar(coeffs, coeffs, s->frame_len / 2, s->frame_len);
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}
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else if (CONFIG_BINKAUDIO_RDFT_DECODER)
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s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
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}
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s->fmt_conv.float_to_int16_interleave(out, (const float **)s->coeffs_ptr,
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s->frame_len, s->channels);
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if (!s->first) {
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int count = s->overlap_len * s->channels;
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int shift = av_log2(count);
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for (i = 0; i < count; i++) {
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out[i] = (s->previous[i] * (count - i) + out[i] * i) >> shift;
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}
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}
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memcpy(s->previous, out + s->block_size,
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s->overlap_len * s->channels * sizeof(*out));
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s->first = 0;
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}
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static av_cold int decode_end(AVCodecContext *avctx)
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{
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BinkAudioContext * s = avctx->priv_data;
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av_freep(&s->bands);
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if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == CODEC_ID_BINKAUDIO_RDFT)
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ff_rdft_end(&s->trans.rdft);
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else if (CONFIG_BINKAUDIO_DCT_DECODER)
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ff_dct_end(&s->trans.dct);
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return 0;
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}
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static void get_bits_align32(GetBitContext *s)
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{
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int n = (-get_bits_count(s)) & 31;
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if (n) skip_bits(s, n);
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}
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static int decode_frame(AVCodecContext *avctx,
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void *data, int *data_size,
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AVPacket *avpkt)
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{
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BinkAudioContext *s = avctx->priv_data;
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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short *samples = data;
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short *samples_end = (short*)((uint8_t*)data + *data_size);
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int reported_size;
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GetBitContext *gb = &s->gb;
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init_get_bits(gb, buf, buf_size * 8);
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reported_size = get_bits_long(gb, 32);
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while (get_bits_count(gb) / 8 < buf_size &&
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samples + s->block_size <= samples_end) {
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decode_block(s, samples, avctx->codec->id == CODEC_ID_BINKAUDIO_DCT);
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samples += s->block_size;
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get_bits_align32(gb);
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}
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*data_size = FFMIN(reported_size, (uint8_t*)samples - (uint8_t*)data);
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return buf_size;
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}
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AVCodec ff_binkaudio_rdft_decoder = {
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"binkaudio_rdft",
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AVMEDIA_TYPE_AUDIO,
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CODEC_ID_BINKAUDIO_RDFT,
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sizeof(BinkAudioContext),
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decode_init,
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NULL,
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decode_end,
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decode_frame,
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.long_name = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)")
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};
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AVCodec ff_binkaudio_dct_decoder = {
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"binkaudio_dct",
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AVMEDIA_TYPE_AUDIO,
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CODEC_ID_BINKAUDIO_DCT,
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sizeof(BinkAudioContext),
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decode_init,
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NULL,
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decode_end,
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decode_frame,
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.long_name = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)")
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};
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