[FFmpeg-devel] [PATCH] avcodec/dcaenc: Initial implementation of ADPCM encoding for DCA encoder

[Даниил Чередник]

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 <atomnuker at gmail.com>
wrote:

> On 2 May 2017 at 22:53, Даниил Чередник <dan.cherednik at 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 at 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 at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>



-- 
Daniil Cherednik
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