| /* |
| * 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 RA288Context { |
| 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_close(AVCodecContext *avctx) |
| { |
| RA288Context *ractx = avctx->priv_data; |
| |
| av_freep(&ractx->fdsp); |
| |
| return 0; |
| } |
| |
| 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; |
| } |
| |
| ractx->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT); |
| if (!ractx->fdsp) |
| return AVERROR(ENOMEM); |
| |
| 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 = 32.0; |
| 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, 5.0 / (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 *= 257.0 / 256.0; |
| } |
| |
| /** |
| * 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; |
| } |
| |
| ret = init_get_bits8(&gb, buf, avctx->block_align); |
| if (ret < 0) |
| return ret; |
| |
| /* 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]; |
| |
| 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", |
| .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"), |
| .type = AVMEDIA_TYPE_AUDIO, |
| .id = AV_CODEC_ID_RA_288, |
| .priv_data_size = sizeof(RA288Context), |
| .init = ra288_decode_init, |
| .decode = ra288_decode_frame, |
| .close = ra288_decode_close, |
| .capabilities = AV_CODEC_CAP_DR1, |
| }; |
