X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fra288.c;h=f2b97cca975c72b57ae752bb4c337f177d090122;hb=022f8d27dd0a61bfaae729d53d133b17418ea16b;hp=9445e7088e2ca3f0b8988c0bc8dbde37416aa60d;hpb=83f9bc8aeece0d7c5b1fadaf91118ee6a97a0390;p=ffmpeg diff --git a/libavcodec/ra288.c b/libavcodec/ra288.c index 9445e7088e2..a91a06cfa1c 100644 --- a/libavcodec/ra288.c +++ b/libavcodec/ra288.c @@ -26,14 +26,19 @@ #include "lpc.h" #include "celp_math.h" #include "celp_filters.h" +#include "dsputil.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) + DSPContext dsp; + DECLARE_ALIGNED(16, float, sp_lpc)[FFALIGN(36, 8)]; ///< LPC coefficients for speech data (spec: A) + DECLARE_ALIGNED(16, float, gain_lpc)[FFALIGN(10, 8)]; ///< LPC coefficients for gain (spec: GB) /** speech data history (spec: SB). * Its first 70 coefficients are updated only at backward filtering. @@ -54,16 +59,12 @@ typedef struct { static av_cold int ra288_decode_init(AVCodecContext *avctx) { + RA288Context *ractx = avctx->priv_data; avctx->sample_fmt = AV_SAMPLE_FMT_FLT; + dsputil_init(&ractx->dsp, avctx); return 0; } -static void apply_window(float *tgt, const float *m1, const float *m2, int n) -{ - while (n--) - *tgt++ = *m1++ * *m2++; -} - static void convolve(float *tgt, const float *src, int len, int n) { for (; n >= 0; n--) @@ -96,14 +97,14 @@ static void decode(RA288Context *ractx, float gain, int cb_coef) for (i=0; i < 5; i++) buffer[i] = codetable[cb_coef][i] * sumsum; - sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.); + sum = ff_dot_productf(buffer, buffer, 5); - sum = FFMAX(sum, 1); + 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) - 32; + gain_block[9] = 10 * log10(sum) + (10*log10(((1<<24)/5.)) - 32); ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36); } @@ -120,15 +121,18 @@ static void decode(RA288Context *ractx, float gain, int cb_coef) * @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, +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]; - float work[MAX_BACKWARD_FILTER_ORDER + MAX_BACKWARD_FILTER_LEN + MAX_BACKWARD_FILTER_NONREC]; + LOCAL_ALIGNED_16(float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER + + MAX_BACKWARD_FILTER_LEN + + MAX_BACKWARD_FILTER_NONREC, 8)]); - apply_window(work, window, hist, order + n + non_rec); + ractx->dsp.vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 8)); convolve(buffer1, work + order , n , order); convolve(buffer2, work + order + n, non_rec, order); @@ -145,16 +149,17 @@ static void do_hybrid_window(int order, int n, int non_rec, float *out, /** * Backward synthesis filter, find the LPC coefficients from past speech data. */ -static void backward_filter(float *hist, float *rec, const float *window, +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(order, n, non_rec, temp, hist, rec, window); + do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window); if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1)) - apply_window(lpc, lpc, tab, order); + ractx->dsp.vector_fmul(lpc, lpc, tab, FFALIGN(order, 8)); memmove(hist, hist + n, move_size*sizeof(*hist)); } @@ -165,7 +170,7 @@ static int ra288_decode_frame(AVCodecContext * avctx, void *data, const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; float *out = data; - int i, j; + int i, out_size; RA288Context *ractx = avctx->priv_data; GetBitContext gb; @@ -173,45 +178,46 @@ 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*4) - return -1; + out_size = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME * + av_get_bytes_per_sample(avctx->sample_fmt); + if (*data_size < out_size) { + av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n"); + return AVERROR(EINVAL); + } 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++) = 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->sp_hist, ractx->sp_rec, syn_window, + backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window, ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70); - backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window, + 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; + *data_size = out_size; return avctx->block_align; } -AVCodec ff_ra_288_decoder = -{ - "real_288", - AVMEDIA_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", + .type = AVMEDIA_TYPE_AUDIO, + .id = CODEC_ID_RA_288, + .priv_data_size = sizeof(RA288Context), + .init = ra288_decode_init, + .decode = ra288_decode_frame, + .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"), };