All notices were fixed. Also I found issue with uninitialized subband buffer - fixed. New patch attached.
On Wed, May 3, 2017 at 7:49 AM, Rostislav Pehlivanov <atomnu...@gmail.com> wrote: > 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 > -- Daniil Cherednik
0001-avcodec-dcaenc-Initial-implementation-of-ADPCM-encod_v2.patch
Description: Binary data
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