Yes, now it is not easy to hear the difference. I found one track on Marantz demo cd where quantization noise is noticeable without ADPCM. About bit allocation. Current algorithm fails (av_assert0(used_abits != USED_1ABITS)) even with 250-320kbps on some tracks - probably I can fix it.
Thank you. On Mon, May 8, 2017 at 7:59 AM, Rostislav Pehlivanov <atomnu...@gmail.com> wrote: > On 6 May 2017 at 11:47, Даниил Чередник <dan.chered...@gmail.com> wrote: > > > Tiny fix: missed inline attr. > > > > On Thu, May 4, 2017 at 1:17 AM, Даниил Чередник <dan.chered...@gmail.com > > > > wrote: > > > > > 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 > > > > > > > > > > > -- > > Daniil Cherednik > > > > _______________________________________________ > > ffmpeg-devel mailing list > > ffmpeg-devel@ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > > > > Tested, there's a difference though looking at the spectrum is the only way > to see it. The encoder only supports bitrates higher than ~251kbps which > isn't enough to really make a big difference. Tried to reduce it but the > bit allocation algorithm of the encoder broke. > > Anyway, thanks, pushed the patch. > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel > -- Daniil Cherednik _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
