X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fra288.c;h=215786c803689d986a56b258ab6ba759ff038c09;hb=8dd0a2c5cf40a8a49faae985adc11750b6429132;hp=86b0ea93ed9361c077e76a2825dd3eafaf6abfc6;hpb=111734de09e5075d5f2694f13ba02d37a84ac18a;p=ffmpeg diff --git a/libavcodec/ra288.c b/libavcodec/ra288.c index 86b0ea93ed9..215786c8036 100644 --- a/libavcodec/ra288.c +++ b/libavcodec/ra288.c @@ -2,32 +2,45 @@ * RealAudio 2.0 (28.8K) * Copyright (c) 2003 the ffmpeg project * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav 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, + * Libav 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 + * License along with Libav; 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" -#define ALT_BITSTREAM_READER_LE -#include "bitstream.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 { - float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A) - float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB) + 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. @@ -48,30 +61,27 @@ typedef struct { static av_cold int ra288_decode_init(AVCodecContext *avctx) { - avctx->sample_fmt = SAMPLE_FMT_S16; - return 0; -} + RA288Context *ractx = avctx->priv_data; -static inline float scalar_product_float(const float * v1, const float * v2, - int size) -{ - float res = 0.; + avctx->channels = 1; + avctx->channel_layout = AV_CH_LAYOUT_MONO; + avctx->sample_fmt = AV_SAMPLE_FMT_FLT; - while (size--) - res += *v1++ * *v2++; + avpriv_float_dsp_init(&ractx->fdsp, avctx->flags & CODEC_FLAG_BITEXACT); - return res; + return 0; } -static void apply_window(float *tgt, const float *m1, const float *m2, int n) +static void convolve(float *tgt, const float *src, int len, int n) { - while (n--) - *tgt++ = *m1++ * *m2++; + 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, j; + int i; double sumsum; float sum, buffer[5]; float *block = ractx->sp_hist + 70 + 36; // current block @@ -79,14 +89,8 @@ static void decode(RA288Context *ractx, float gain, int cb_coef) memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block)); - for (i=0; i < 5; i++) { - block[i] = 0.; - for (j=0; j < 36; j++) - block[i] -= block[i-1-j]*ractx->sp_lpc[j]; - } - /* block 46 of G.728 spec */ - sum = 32.; + sum = 32.0; for (i=0; i < 10; i++) sum -= gain_block[9-i] * ractx->gain_lpc[i]; @@ -94,12 +98,13 @@ static void decode(RA288Context *ractx, float gain, int cb_coef) sum = av_clipf(sum, 0, 60); /* block 48 of G.728 spec */ - sumsum = exp(sum * 0.1151292546497) * gain; /* pow(10.0,sum/20)*gain */ + /* 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 * (1./2048.); + buffer[i] = codetable[cb_coef][i] * sumsum; - sum = scalar_product_float(buffer, buffer, 5) / 5; + sum = avpriv_scalarproduct_float_c(buffer, buffer, 5) * ((1 << 24) / 5.0); sum = FFMAX(sum, 1); @@ -108,20 +113,7 @@ static void decode(RA288Context *ractx, float gain, int cb_coef) gain_block[9] = 10 * log10(sum) - 32; - for (i=1; i < 5; i++) - for (j=i-1; j >= 0; j--) - buffer[i] -= ractx->sp_lpc[i-j-1] * buffer[j]; - - /* output */ - for (i=0; i < 5; i++) - block[i] = av_clipf(block[i] + buffer[i], -4095, 4095); -} - -static void convolve(float *tgt, const float *src, int len, int n) -{ - for (; n >= 0; n--) - tgt[n] = scalar_product_float(src, src - n, len); - + ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36); } /** @@ -136,16 +128,18 @@ static void convolve(float *tgt, const float *src, int len, int n) * @param out2 pointer to the recursive part of the output * @param window pointer to the windowing function table */ -static void do_hybrid_window(int order, int n, int non_rec, - float *out, float *hist, float *out2, - const float *window) +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[order + 1]; - float buffer2[order + 1]; - float work[order + n + non_rec]; + 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)]); - apply_window(work, window, hist, order + n + non_rec); + 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); @@ -156,42 +150,35 @@ static void do_hybrid_window(int order, int n, int non_rec, } /* Multiply by the white noise correcting factor (WNCF). */ - *out *= 257./256.; + *out *= 257.0 / 256.0; } /** * Backward synthesis filter, find the LPC coefficients from past speech data. */ -static void backward_filter(RA288Context *ractx) +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 temp1[37]; // RTMP in the spec - float temp2[11]; // GPTPMP in the spec + float temp[MAX_BACKWARD_FILTER_ORDER+1]; - do_hybrid_window(36, 40, 35, temp1, ractx->sp_hist, - ractx->sp_rec, syn_window); + do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window); - if (!compute_lpc_coefs(temp1, 36, ractx->sp_lpc, 0, 1, 1)) - apply_window(ractx->sp_lpc, ractx->sp_lpc, syn_bw_tab, 36); + if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1)) + ractx->fdsp.vector_fmul(lpc, lpc, tab, FFALIGN(order, 16)); - do_hybrid_window(10, 8, 20, temp2, ractx->gain_hist, - ractx->gain_rec, gain_window); - - if (!compute_lpc_coefs(temp2, 10, ractx->gain_lpc, 0, 1, 1)) - apply_window(ractx->gain_lpc, ractx->gain_lpc, gain_bw_tab, 10); - - memmove(ractx->gain_hist, ractx->gain_hist + 8, - 28*sizeof(*ractx->gain_hist)); - - memmove(ractx->sp_hist , ractx->sp_hist + 40, - 70*sizeof(*ractx->sp_hist )); + memmove(hist, hist + n, move_size*sizeof(*hist)); } static int ra288_decode_frame(AVCodecContext * avctx, void *data, - int *data_size, const uint8_t * buf, - int buf_size) + int *got_frame_ptr, AVPacket *avpkt) { - int16_t *out = data; - int i, j; + 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; @@ -199,40 +186,49 @@ static int ra288_decode_frame(AVCodecContext * avctx, void *data, av_log(avctx, AV_LOG_ERROR, "Error! Input buffer is too small [%d<%d]\n", buf_size, avctx->block_align); - return 0; + return AVERROR_INVALIDDATA; } - if (*data_size < 32*5*2) - return -1; + /* get output buffer */ + frame->nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME; + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + return ret; + } + out = (float *)frame->data[0]; init_get_bits(&gb, buf, avctx->block_align * 8); - for (i=0; i < 32; i++) { + 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); - for (j=0; j < 5; j++) - *(out++) = 8 * ractx->sp_hist[70 + 36 + j]; + memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out)); + out += RA288_BLOCK_SIZE; - if ((i & 7) == 3) - backward_filter(ractx); + 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); + } } - *data_size = (char *)out - (char *)data; + *got_frame_ptr = 1; + return avctx->block_align; } -AVCodec ra_288_decoder = -{ - "real_288", - CODEC_TYPE_AUDIO, - CODEC_ID_RA_288, - sizeof(RA288Context), - ra288_decode_init, - NULL, - NULL, - ra288_decode_frame, - .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"), +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, + .capabilities = CODEC_CAP_DR1, };