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ffmpeg.wasm-core
Commit 64c312aa authored by Justin Ruggles's avatar Justin Ruggles
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dcadec: use float planar sample format

parent cbf6ee78
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Showing with 49 additions and 70 deletions
libavcodec/dcadec.c
View file @ 64c312aa
......@@ -354,11 +354,9 @@ typedef struct {
DECLARE_ALIGNED(32, float, raXin)[32];
int output; ///< type of output
float scale_bias; ///< output scale
DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
DECLARE_ALIGNED(32, float, samples)[(DCA_PRIM_CHANNELS_MAX + 1) * 256];
const float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];
float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];
uint8_t dca_buffer[DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE + DCA_BUFFER_PADDING_SIZE];
int dca_buffer_size; ///< how much data is in the dca_buffer
......@@ -1007,20 +1005,20 @@ static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
}
/* downmixing routines */
#define MIX_REAR1(samples, si1, rs, coef) \
samples[i] += samples[si1] * coef[rs][0]; \
samples[i+256] += samples[si1] * coef[rs][1];
#define MIX_REAR1(samples, s1, rs, coef) \
samples[0][i] += samples[s1][i] * coef[rs][0]; \
samples[1][i] += samples[s1][i] * coef[rs][1];
#define MIX_REAR2(samples, si1, si2, rs, coef) \
samples[i] += samples[si1] * coef[rs][0] + samples[si2] * coef[rs + 1][0]; \
samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs + 1][1];
#define MIX_REAR2(samples, s1, s2, rs, coef) \
samples[0][i] += samples[s1][i] * coef[rs][0] + samples[s2][i] * coef[rs + 1][0]; \
samples[1][i] += samples[s1][i] * coef[rs][1] + samples[s2][i] * coef[rs + 1][1];
#define MIX_FRONT3(samples, coef) \
t = samples[i + c]; \
u = samples[i + l]; \
v = samples[i + r]; \
samples[i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \
samples[i+256] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];
t = samples[c][i]; \
u = samples[l][i]; \
v = samples[r][i]; \
samples[0][i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \
samples[1][i] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];
#define DOWNMIX_TO_STEREO(op1, op2) \
for (i = 0; i < 256; i++) { \
......@@ -1028,7 +1026,7 @@ static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
op2 \
}
static void dca_downmix(float *samples, int srcfmt,
static void dca_downmix(float **samples, int srcfmt,
int downmix_coef[DCA_PRIM_CHANNELS_MAX][2],
const int8_t *channel_mapping)
{
......@@ -1053,36 +1051,36 @@ static void dca_downmix(float *samples, int srcfmt,
case DCA_STEREO:
break;
case DCA_3F:
c = channel_mapping[0] * 256;
l = channel_mapping[1] * 256;
r = channel_mapping[2] * 256;
c = channel_mapping[0];
l = channel_mapping[1];
r = channel_mapping[2];
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );
break;
case DCA_2F1R:
s = channel_mapping[2] * 256;
DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + s, 2, coef), );
s = channel_mapping[2];
DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), );
break;
case DCA_3F1R:
c = channel_mapping[0] * 256;
l = channel_mapping[1] * 256;
r = channel_mapping[2] * 256;
s = channel_mapping[3] * 256;
c = channel_mapping[0];
l = channel_mapping[1];
r = channel_mapping[2];
s = channel_mapping[3];
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
MIX_REAR1(samples, i + s, 3, coef));
MIX_REAR1(samples, s, 3, coef));
break;
case DCA_2F2R:
sl = channel_mapping[2] * 256;
sr = channel_mapping[3] * 256;
DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + sl, i + sr, 2, coef), );
sl = channel_mapping[2];
sr = channel_mapping[3];
DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), );
break;
case DCA_3F2R:
c = channel_mapping[0] * 256;
l = channel_mapping[1] * 256;
r = channel_mapping[2] * 256;
sl = channel_mapping[3] * 256;
sr = channel_mapping[4] * 256;
c = channel_mapping[0];
l = channel_mapping[1];
r = channel_mapping[2];
sl = channel_mapping[3];
sr = channel_mapping[4];
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
MIX_REAR2(samples, i + sl, i + sr, 3, coef));
MIX_REAR2(samples, sl, sr, 3, coef));
break;
}
}
......@@ -1279,21 +1277,21 @@ static int dca_filter_channels(DCAContext *s, int block_index)
/* static float pcm_to_double[8] = { 32768.0, 32768.0, 524288.0, 524288.0,
0, 8388608.0, 8388608.0 };*/
qmf_32_subbands(s, k, subband_samples[k],
&s->samples[256 * s->channel_order_tab[k]],
M_SQRT1_2 * s->scale_bias /* pcm_to_double[s->source_pcm_res] */);
s->samples_chanptr[s->channel_order_tab[k]],
M_SQRT1_2 / 32768.0 /* pcm_to_double[s->source_pcm_res] */);
}
/* Down mixing */
if (s->avctx->request_channels == 2 && s->prim_channels > 2) {
dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab);
dca_downmix(s->samples_chanptr, s->amode, s->downmix_coef, s->channel_order_tab);
}
/* Generate LFE samples for this subsubframe FIXME!!! */
if (s->output & DCA_LFE) {
lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
s->lfe_data + 2 * s->lfe * (block_index + 4),
&s->samples[256 * dca_lfe_index[s->amode]],
(1.0 / 256.0) * s->scale_bias);
s->samples_chanptr[dca_lfe_index[s->amode]],
1.0 / (256.0 * 32768.0));
/* Outputs 20bits pcm samples */
}
......@@ -1656,8 +1654,7 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
int lfe_samples;
int num_core_channels = 0;
int i, ret;
float *samples_flt;
int16_t *samples_s16;
float **samples_flt;
DCAContext *s = avctx->priv_data;
int channels;
int core_ss_end;
......@@ -1853,33 +1850,26 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples_flt = (float *) s->frame.data[0];
samples_s16 = (int16_t *) s->frame.data[0];
samples_flt = (float **) s->frame.extended_data;
/* filter to get final output */
for (i = 0; i < (s->sample_blocks / 8); i++) {
int ch;
for (ch = 0; ch < channels; ch++)
s->samples_chanptr[ch] = samples_flt[ch] + i * 256;
dca_filter_channels(s, i);
/* If this was marked as a DTS-ES stream we need to subtract back- */
/* channel from SL & SR to remove matrixed back-channel signal */
if ((s->source_pcm_res & 1) && s->xch_present) {
float *back_chan = s->samples + s->channel_order_tab[s->xch_base_channel] * 256;
float *lt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;
float *rt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;
float *back_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel]];
float *lt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]];
float *rt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]];
s->fdsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
s->fdsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
}
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
s->fmt_conv.float_interleave(samples_flt, s->samples_chanptr, 256,
channels);
samples_flt += 256 * channels;
} else {
s->fmt_conv.float_to_int16_interleave(samples_s16,
s->samples_chanptr, 256,
channels);
samples_s16 += 256 * channels;
}
}
/* update lfe history */
......@@ -1904,7 +1894,6 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
static av_cold int dca_decode_init(AVCodecContext *avctx)
{
DCAContext *s = avctx->priv_data;
int i;
s->avctx = avctx;
dca_init_vlcs();
......@@ -1915,16 +1904,7 @@ static av_cold int dca_decode_init(AVCodecContext *avctx)
ff_dcadsp_init(&s->dcadsp);
ff_fmt_convert_init(&s->fmt_conv, avctx);
for (i = 0; i < DCA_PRIM_CHANNELS_MAX + 1; i++)
s->samples_chanptr[i] = s->samples + i * 256;
if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
s->scale_bias = 1.0 / 32768.0;
} else {
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
s->scale_bias = 1.0;
}
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
/* allow downmixing to stereo */
if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
......@@ -1964,8 +1944,7 @@ AVCodec ff_dca_decoder = {
.close = dca_decode_end,
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_S16,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.profiles = NULL_IF_CONFIG_SMALL(profiles),
};
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