On 2 May 2017 at 22:53, Даниил Чередник <dan.chered...@gmail.com> wrote:
> Hi. > > This patch introduces initial implementation of subband ADPCM encoding for > DCA codec. > > Some results: > > sample: > > https://yadi.sk/d/B_3sVskM3HZiWK - original > > https://yadi.sk/d/7CK47Nt63HZiWf - without adpcm > > https://yadi.sk/d/25q1JDV93HZiWq - with adpcm > > chirp tone: > > https://yadi.sk/i/tZKHoJ1d3HZk4c > > Right now this feature is disabled by default. But it is ready to try > using -dca_adpcm 1 option. > > There are some issues, should be solved before enabling this feature by > default: > > 1. Speed up: I am trying to find best filter in each subband. But with real > signal, usually only few subbands has significant prediction gain. The idea > is try to analyze FFT spectrum (which is already calculated), to check is > particular subband looks like tonal or noise. If subband is noise like - do > not try to find best LPC predictor. > > 2. Modify psychoacoustic to use prediction gain for bit allocation. Right > now ADPCM encoded block can get some extra bits. > > 3. Tuning the prediction gain threshold. > > > Thank you. > -- > Daniil Cherednik > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > >+static int64_t calc_corr(const int32_t *x, int len, int j, int k) Add inline attrib? Seems appropriate here. >+for (n = 0; n < len; n++) { >+ s += MUL64(x[n-j], x[n-k]); >+ } For loops with 1 line we leave the brackets out. >+for (i = 0; i <= DCA_ADPCM_COEFFS; i++) { >+ for (j = i; j <= DCA_ADPCM_COEFFS; j++) { >+ corr[k++] = calc_corr(in+4, len, i, j); >+ } >+ } Same >+ for (i = 0; i < len + DCA_ADPCM_COEFFS; i++) { >+ max |= FFABS(in[i]); >+ } Same >for (ch = 0; ch < c->fullband_channels; ch++) { >+ for (band = 0; band < 32; band++) { >+ if (c->prediction_mode[ch][band] >= 0) { >+ quantize_adpcm_subband(c, ch, band); >+ } >+ } >+ } Same >+ for (ch = 0; ch < c->fullband_channels; ch++) { >+ for (band = 0; band < 32; band++) { >+ if (c->prediction_mode[ch][band] == -1) { >+ for (sample = 0; sample < SUBBAND_SAMPLES; sample++) { >+ c->quantized[ch][band][sample] = quantize_value(c->subband[ch][band][sample], c->quant[ch][band]); >+ } >+ } >+ } >+ } Same, 4 whole whitespace lines added here. >+ if (c->bitrate_index == 3) { >+ step_size = ff_dca_lossless_quant[c->abits[ch][band]]; >+ } else { >+ step_size = ff_dca_lossy_quant[c->abits[ch][band]]; >+ } Same >for (;;) { while (1) { >+ if (i++ == last_pos) >+ break; Better yet remove the infinite loop and just use a normal for () loop. >+static inline void ff_dca_core_dequantize(int32_t *output, const int32_t *input, >+ int32_t step_size, int32_t scale, int residual, int len) Fix second line's alignment. >+struct premultiplied_coeffs { >+ int32_t aa[10]; >+}; I think it would be simpler to just use int32_t premultiplied_coeffs[10] instead of a struct. Apart from these style issues patch looks fine. I'll be able to test it in a day. _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
