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Diffstat (limited to 'ffmpeg1/libavcodec/ra288.c')
| -rw-r--r-- | ffmpeg1/libavcodec/ra288.c | 239 |
1 files changed, 239 insertions, 0 deletions
diff --git a/ffmpeg1/libavcodec/ra288.c b/ffmpeg1/libavcodec/ra288.c new file mode 100644 index 0000000..393ea0b --- /dev/null +++ b/ffmpeg1/libavcodec/ra288.c @@ -0,0 +1,239 @@ +/* + * RealAudio 2.0 (28.8K) + * Copyright (c) 2003 the ffmpeg project + * + * 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 + */ + +#include "libavutil/channel_layout.h" +#include "libavutil/float_dsp.h" +#include "libavutil/internal.h" +#include "avcodec.h" +#include "internal.h" +#define BITSTREAM_READER_LE +#include "get_bits.h" +#include "ra288.h" +#include "lpc.h" +#include "celp_filters.h" + +#define MAX_BACKWARD_FILTER_ORDER 36 +#define MAX_BACKWARD_FILTER_LEN 40 +#define MAX_BACKWARD_FILTER_NONREC 35 + +#define RA288_BLOCK_SIZE 5 +#define RA288_BLOCKS_PER_FRAME 32 + +typedef struct { + AVFloatDSPContext fdsp; + DECLARE_ALIGNED(32, float, sp_lpc)[FFALIGN(36, 16)]; ///< LPC coefficients for speech data (spec: A) + DECLARE_ALIGNED(32, float, gain_lpc)[FFALIGN(10, 16)]; ///< LPC coefficients for gain (spec: GB) + + /** speech data history (spec: SB). + * Its first 70 coefficients are updated only at backward filtering. + */ + float sp_hist[111]; + + /// speech part of the gain autocorrelation (spec: REXP) + float sp_rec[37]; + + /** log-gain history (spec: SBLG). + * Its first 28 coefficients are updated only at backward filtering. + */ + float gain_hist[38]; + + /// recursive part of the gain autocorrelation (spec: REXPLG) + float gain_rec[11]; +} RA288Context; + +static av_cold int ra288_decode_init(AVCodecContext *avctx) +{ + RA288Context *ractx = avctx->priv_data; + + avctx->channels = 1; + avctx->channel_layout = AV_CH_LAYOUT_MONO; + avctx->sample_fmt = AV_SAMPLE_FMT_FLT; + + if (avctx->block_align <= 0) { + av_log(avctx, AV_LOG_ERROR, "unsupported block align\n"); + return AVERROR_PATCHWELCOME; + } + + avpriv_float_dsp_init(&ractx->fdsp, avctx->flags & CODEC_FLAG_BITEXACT); + + return 0; +} + +static void convolve(float *tgt, const float *src, int len, int n) +{ + for (; n >= 0; n--) + tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len); + +} + +static void decode(RA288Context *ractx, float gain, int cb_coef) +{ + int i; + double sumsum; + float sum, buffer[5]; + float *block = ractx->sp_hist + 70 + 36; // current block + float *gain_block = ractx->gain_hist + 28; + + memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block)); + + /* block 46 of G.728 spec */ + sum = 32.; + for (i=0; i < 10; i++) + sum -= gain_block[9-i] * ractx->gain_lpc[i]; + + /* block 47 of G.728 spec */ + sum = av_clipf(sum, 0, 60); + + /* block 48 of G.728 spec */ + /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */ + sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23)); + + for (i=0; i < 5; i++) + buffer[i] = codetable[cb_coef][i] * sumsum; + + sum = avpriv_scalarproduct_float_c(buffer, buffer, 5); + + sum = FFMAX(sum, 5. / (1<<24)); + + /* shift and store */ + memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block)); + + gain_block[9] = 10 * log10(sum) + (10*log10(((1<<24)/5.)) - 32); + + ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36); +} + +/** + * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification. + * + * @param order filter order + * @param n input length + * @param non_rec number of non-recursive samples + * @param out filter output + * @param hist pointer to the input history of the filter + * @param out pointer to the non-recursive part of the output + * @param out2 pointer to the recursive part of the output + * @param window pointer to the windowing function table + */ +static void do_hybrid_window(RA288Context *ractx, + int order, int n, int non_rec, float *out, + float *hist, float *out2, const float *window) +{ + int i; + float buffer1[MAX_BACKWARD_FILTER_ORDER + 1]; + float buffer2[MAX_BACKWARD_FILTER_ORDER + 1]; + LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER + + MAX_BACKWARD_FILTER_LEN + + MAX_BACKWARD_FILTER_NONREC, 16)]); + + av_assert2(order>=0); + + ractx->fdsp.vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16)); + + convolve(buffer1, work + order , n , order); + convolve(buffer2, work + order + n, non_rec, order); + + for (i=0; i <= order; i++) { + out2[i] = out2[i] * 0.5625 + buffer1[i]; + out [i] = out2[i] + buffer2[i]; + } + + /* Multiply by the white noise correcting factor (WNCF). */ + *out *= 257./256.; +} + +/** + * Backward synthesis filter, find the LPC coefficients from past speech data. + */ +static void backward_filter(RA288Context *ractx, + float *hist, float *rec, const float *window, + float *lpc, const float *tab, + int order, int n, int non_rec, int move_size) +{ + float temp[MAX_BACKWARD_FILTER_ORDER+1]; + + do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window); + + if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1)) + ractx->fdsp.vector_fmul(lpc, lpc, tab, FFALIGN(order, 16)); + + memmove(hist, hist + n, move_size*sizeof(*hist)); +} + +static int ra288_decode_frame(AVCodecContext * avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) +{ + AVFrame *frame = data; + const uint8_t *buf = avpkt->data; + int buf_size = avpkt->size; + float *out; + int i, ret; + RA288Context *ractx = avctx->priv_data; + GetBitContext gb; + + if (buf_size < avctx->block_align) { + av_log(avctx, AV_LOG_ERROR, + "Error! Input buffer is too small [%d<%d]\n", + buf_size, avctx->block_align); + return AVERROR_INVALIDDATA; + } + + /* get output buffer */ + frame->nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME; + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) + return ret; + out = (float *)frame->data[0]; + + init_get_bits(&gb, buf, avctx->block_align * 8); + + for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) { + float gain = amptable[get_bits(&gb, 3)]; + int cb_coef = get_bits(&gb, 6 + (i&1)); + + decode(ractx, gain, cb_coef); + + memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out)); + out += RA288_BLOCK_SIZE; + + if ((i & 7) == 3) { + backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window, + ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70); + + backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window, + ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28); + } + } + + *got_frame_ptr = 1; + + return avctx->block_align; +} + +AVCodec ff_ra_288_decoder = { + .name = "real_288", + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_RA_288, + .priv_data_size = sizeof(RA288Context), + .init = ra288_decode_init, + .decode = ra288_decode_frame, + .capabilities = CODEC_CAP_DR1, + .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"), +}; |