On Tue, Mar 14, 2017 at 09:15:30AM +0000, Luca Barbato wrote:
> ---
> While at it.
>
> libavcodec/dcadec.c | 10 ++++++----
> 1 file changed, 6 insertions(+), 4 deletions(-)
dca: Refactor dca_filter_channels() a little
> --- a/libavcodec/dcadec.c
> +++ b/libavcodec/dcadec.c
> @@ -947,15 +947,16 @@ static int dca_filter_channels(DCAContext *s, int
> block_index, int upsample)
>
> /* 64 subbands QMF */
> for (k = 0; k < s->audio_header.prim_channels; k++) {
> + int channel = s->channel_order_tab[k];
> int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] =
> s->dca_chan[k].subband_samples[block_index];
>
> s->fmt_conv.int32_to_float(samples[0], subband_samples[0],
> DCA_SUBBANDS_X96K *
> SAMPLES_PER_SUBBAND);
>
> - if (s->channel_order_tab[k] >= 0)
> + if (channel >= 0)
> qmf_64_subbands(s, k, samples,
> - s->samples_chanptr[s->channel_order_tab[k]],
> + s->samples_chanptr[channel],
> /* Upsampling needs a factor 2 here. */
> M_SQRT2 / 32768.0);
> }
> @@ -964,15 +965,16 @@ static int dca_filter_channels(DCAContext *s, int
> block_index, int upsample)
> LOCAL_ALIGNED(32, float, samples, [DCA_SUBBANDS],
> [SAMPLES_PER_SUBBAND]);
>
> for (k = 0; k < s->audio_header.prim_channels; k++) {
> + int channel = s->channel_order_tab[k];
> int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] =
> s->dca_chan[k].subband_samples[block_index];
>
> s->fmt_conv.int32_to_float(samples[0], subband_samples[0],
> DCA_SUBBANDS * SAMPLES_PER_SUBBAND);
>
> - if (s->channel_order_tab[k] >= 0)
> + if (channel >= 0)
> qmf_32_subbands(s, k, samples,
> - s->samples_chanptr[s->channel_order_tab[k]],
> + s->samples_chanptr[channel],
> M_SQRT1_2 / 32768.0);
> }
> }
These two blocks should be completely refactored ;)
OK
Diego
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