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