* Copyright (C) 2004 Benjamin Zores
* Copyright (C) 2006 Benjamin Larsson
* Copyright (C) 2007 Konstantin Shishkov
+ * Copyright (C) 2012 Paul B Mahol
+ * Copyright (C) 2014 Niels Möller
*
* This file is part of Libav.
*
#include <stddef.h>
#include <stdio.h>
+#include "libavutil/attributes.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "avcodec.h"
#include "dca.h"
+#include "dca_syncwords.h"
#include "dcadata.h"
#include "dcadsp.h"
#include "dcahuff.h"
-#include "dca_exss.h"
#include "fft.h"
#include "fmtconvert.h"
#include "get_bits.h"
}
if (s->bitalloc[j][k] > 26) {
- av_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n",
+ ff_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n",
j, k, s->bitalloc[j][k]);
return AVERROR_INVALIDDATA;
}
samples_out, s->raXin, scale);
}
-static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
- int num_deci_sample, float *samples_in,
+static QMF64_table *qmf64_precompute(void)
+{
+ unsigned i, j;
+ QMF64_table *table = av_malloc(sizeof(*table));
+ if (!table)
+ return NULL;
+
+ for (i = 0; i < 32; i++)
+ for (j = 0; j < 32; j++)
+ table->dct4_coeff[i][j] = cos((2 * i + 1) * (2 * j + 1) * M_PI / 128);
+ for (i = 0; i < 32; i++)
+ for (j = 0; j < 32; j++)
+ table->dct2_coeff[i][j] = cos((2 * i + 1) * j * M_PI / 64);
+
+ /* FIXME: Is the factor 0.125 = 1/8 right? */
+ for (i = 0; i < 32; i++)
+ table->rcos[i] = 0.125 / cos((2 * i + 1) * M_PI / 256);
+ for (i = 0; i < 32; i++)
+ table->rsin[i] = -0.125 / sin((2 * i + 1) * M_PI / 256);
+
+ return table;
+}
+
+/* FIXME: Totally unoptimized. Based on the reference code and
+ * http://multimedia.cx/mirror/dca-transform.pdf, with guessed tweaks
+ * for doubling the size. */
+static void qmf_64_subbands(DCAContext *s, int chans, float samples_in[64][8],
+ float *samples_out, float scale)
+{
+ float raXin[64];
+ float A[32], B[32];
+ float *raX = s->subband_fir_hist[chans];
+ float *raZ = s->subband_fir_noidea[chans];
+ unsigned i, j, k, subindex;
+
+ for (i = s->subband_activity[chans]; i < 64; i++)
+ raXin[i] = 0.0;
+ for (subindex = 0; subindex < 8; subindex++) {
+ for (i = 0; i < s->subband_activity[chans]; i++)
+ raXin[i] = samples_in[i][subindex];
+
+ for (k = 0; k < 32; k++) {
+ A[k] = 0.0;
+ for (i = 0; i < 32; i++)
+ A[k] += (raXin[2 * i] + raXin[2 * i + 1]) * s->qmf64_table->dct4_coeff[k][i];
+ }
+ for (k = 0; k < 32; k++) {
+ B[k] = raXin[0] * s->qmf64_table->dct2_coeff[k][0];
+ for (i = 1; i < 32; i++)
+ B[k] += (raXin[2 * i] + raXin[2 * i - 1]) * s->qmf64_table->dct2_coeff[k][i];
+ }
+ for (k = 0; k < 32; k++) {
+ raX[k] = s->qmf64_table->rcos[k] * (A[k] + B[k]);
+ raX[63 - k] = s->qmf64_table->rsin[k] * (A[k] - B[k]);
+ }
+
+ for (i = 0; i < 64; i++) {
+ float out = raZ[i];
+ for (j = 0; j < 1024; j += 128)
+ out += ff_dca_fir_64bands[j + i] * (raX[j + i] - raX[j + 63 - i]);
+ *samples_out++ = out * scale;
+ }
+
+ for (i = 0; i < 64; i++) {
+ float hist = 0.0;
+ for (j = 0; j < 1024; j += 128)
+ hist += ff_dca_fir_64bands[64 + j + i] * (-raX[i + j] - raX[j + 63 - i]);
+
+ raZ[i] = hist;
+ }
+
+ /* FIXME: Make buffer circular, to avoid this move. */
+ memmove(raX + 64, raX, (1024 - 64) * sizeof(*raX));
+ }
+}
+
+static void lfe_interpolation_fir(DCAContext *s, const float *samples_in,
float *samples_out)
{
/* samples_in: An array holding decimated samples.
int deciindex;
/* Select decimation filter */
- if (decimation_select == 1) {
+ if (s->lfe == 1) {
idx = 1;
prCoeff = ff_dca_lfe_fir_128;
} else {
- idx = 0;
- prCoeff = ff_dca_lfe_fir_64;
+ idx = 0;
+ if (s->exss_ext_mask & DCA_EXT_EXSS_XLL)
+ prCoeff = ff_dca_lfe_xll_fir_64;
+ else
+ prCoeff = ff_dca_lfe_fir_64;
}
/* Interpolation */
- for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
+ for (deciindex = 0; deciindex < 2 * s->lfe; deciindex++) {
s->dcadsp.lfe_fir[idx](samples_out, samples_in, prCoeff);
samples_in++;
samples_out += 2 * 32 * (1 + idx);
return 0;
}
-static int dca_filter_channels(DCAContext *s, int block_index)
+static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
{
float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
int k;
- /* 32 subbands QMF */
- for (k = 0; k < s->prim_channels; k++) {
- if (s->channel_order_tab[k] >= 0)
- qmf_32_subbands(s, k, subband_samples[k],
- s->samples_chanptr[s->channel_order_tab[k]],
- M_SQRT1_2 / 32768.0);
+ if (upsample) {
+ if (!s->qmf64_table) {
+ s->qmf64_table = qmf64_precompute();
+ if (!s->qmf64_table)
+ return AVERROR(ENOMEM);
+ }
+
+ /* 64 subbands QMF */
+ for (k = 0; k < s->prim_channels; k++) {
+ if (s->channel_order_tab[k] >= 0)
+ qmf_64_subbands(s, k, subband_samples[k],
+ s->samples_chanptr[s->channel_order_tab[k]],
+ /* Upsampling needs a factor 2 here. */
+ M_SQRT2 / 32768.0);
+ }
+ } else {
+ /* 32 subbands QMF */
+ for (k = 0; k < s->prim_channels; k++) {
+ if (s->channel_order_tab[k] >= 0)
+ qmf_32_subbands(s, k, subband_samples[k],
+ s->samples_chanptr[s->channel_order_tab[k]],
+ M_SQRT1_2 / 32768.0);
+ }
}
/* Generate LFE samples for this subsubframe FIXME!!! */
if (s->lfe) {
- lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
+ float *samples = s->samples_chanptr[ff_dca_lfe_index[s->amode]];
+ lfe_interpolation_fir(s,
s->lfe_data + 2 * s->lfe * (block_index + 4),
- s->samples_chanptr[ff_dca_lfe_index[s->amode]]);
- /* Outputs 20bits pcm samples */
+ samples);
+ if (upsample) {
+ unsigned i;
+ /* Should apply the filter in Table 6-11 when upsampling. For
+ * now, just duplicate. */
+ for (i = 511; i > 0; i--) {
+ samples[2 * i] =
+ samples[2 * i + 1] = samples[i];
+ }
+ samples[1] = samples[0];
+ }
}
+ /* FIXME: This downmixing is probably broken with upsample.
+ * Probably totally broken also with XLL in general. */
/* Downmixing to Stereo */
if (s->prim_channels + !!s->lfe > 2 &&
s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
DCAContext *s = avctx->priv_data;
int channels, full_channels;
int core_ss_end;
+ int upsample = 0;
- s->xch_present = 0;
+ s->exss_ext_mask = 0;
+ s->xch_present = 0;
s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer,
DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
uint32_t bits = get_bits_long(&s->gb, 32);
switch (bits) {
- case 0x5a5a5a5a: {
+ case DCA_SYNCWORD_XCH: {
int ext_amode, xch_fsize;
s->xch_base_channel = s->prim_channels;
s->xch_present = 1;
break;
}
- case 0x47004a03:
+ case DCA_SYNCWORD_XXCH:
/* XXCh: extended channels */
/* usually found either in core or HD part in DTS-HD HRA streams,
* but not in DTS-ES which contains XCh extensions instead */
/* check for ExSS (HD part) */
if (s->dca_buffer_size - s->frame_size > 32 &&
- get_bits_long(&s->gb, 32) == DCA_HD_MARKER)
+ get_bits_long(&s->gb, 32) == DCA_SYNCWORD_SUBSTREAM)
ff_dca_exss_parse_header(s);
avctx->profile = s->profile;
s->downmix_coef[i][1] = ff_dca_default_coeffs[am][i][1];
}
}
- av_dlog(s->avctx, "Stereo downmix coeffs:\n");
+ ff_dlog(s->avctx, "Stereo downmix coeffs:\n");
for (i = 0; i < num_core_channels + !!s->lfe; i++) {
- av_dlog(s->avctx, "L, input channel %d = %f\n", i,
+ ff_dlog(s->avctx, "L, input channel %d = %f\n", i,
s->downmix_coef[i][0]);
- av_dlog(s->avctx, "R, input channel %d = %f\n", i,
+ ff_dlog(s->avctx, "R, input channel %d = %f\n", i,
s->downmix_coef[i][1]);
}
- av_dlog(s->avctx, "\n");
+ ff_dlog(s->avctx, "\n");
}
} else {
av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);
/* get output buffer */
frame->nb_samples = 256 * (s->sample_blocks / 8);
+ if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) {
+ int xll_nb_samples = s->xll_segments * s->xll_smpl_in_seg;
+ /* Check for invalid/unsupported conditions first */
+ if (s->xll_residual_channels > channels) {
+ av_log(s->avctx, AV_LOG_WARNING,
+ "DCA: too many residual channels (%d, core channels %d). Disabling XLL\n",
+ s->xll_residual_channels, channels);
+ s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL;
+ } else if (xll_nb_samples != frame->nb_samples &&
+ 2 * frame->nb_samples != xll_nb_samples) {
+ av_log(s->avctx, AV_LOG_WARNING,
+ "DCA: unsupported upsampling (%d XLL samples, %d core samples). Disabling XLL\n",
+ xll_nb_samples, frame->nb_samples);
+ s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL;
+ } else {
+ if (2 * frame->nb_samples == xll_nb_samples) {
+ av_log(s->avctx, AV_LOG_INFO,
+ "XLL: upsampling core channels by a factor of 2\n");
+ upsample = 1;
+
+ frame->nb_samples = xll_nb_samples;
+ // FIXME: Is it good enough to copy from the first channel set?
+ avctx->sample_rate = s->xll_chsets[0].sampling_frequency;
+ }
+ /* If downmixing to stereo, don't decode additional channels.
+ * FIXME: Using the xch_disable flag for this doesn't seem right. */
+ if (!s->xch_disable)
+ avctx->channels += s->xll_channels - s->xll_residual_channels;
+ }
+ }
+
+ /* FIXME: This is an ugly hack, to just revert to the default
+ * layout if we have additional channels. Need to convert the XLL
+ * channel masks to libav channel_layout mask. */
+ if (av_get_channel_layout_nb_channels(avctx->channel_layout) != avctx->channels)
+ avctx->channel_layout = 0;
+
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
/* filter to get final output */
for (i = 0; i < (s->sample_blocks / 8); i++) {
int ch;
-
+ unsigned block = upsample ? 512 : 256;
for (ch = 0; ch < channels; ch++)
- s->samples_chanptr[ch] = samples_flt[ch] + i * 256;
+ s->samples_chanptr[ch] = samples_flt[ch] + i * block;
for (; ch < full_channels; ch++)
- s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * 256;
+ s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * block;
- dca_filter_channels(s, i);
+ dca_filter_channels(s, i, upsample);
/* If this was marked as a DTS-ES stream we need to subtract back- */
/* channel from SL & SR to remove matrixed back-channel signal */
for (i = 0; i < 2 * s->lfe * 4; i++)
s->lfe_data[i] = s->lfe_data[i + lfe_samples];
+ if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) {
+ ret = ff_dca_xll_decode_audio(s, frame);
+ if (ret < 0)
+ return ret;
+ }
/* AVMatrixEncoding
*
* DCA_STEREO_TOTAL (Lt/Rt) is equivalent to Dolby Surround */
DCAContext *s = avctx->priv_data;
ff_mdct_end(&s->imdct);
av_freep(&s->extra_channels_buffer);
+ av_freep(&s->xll_sample_buf);
+ av_freep(&s->qmf64_table);
return 0;
}
static const AVOption options[] = {
{ "disable_xch", "disable decoding of the XCh extension", offsetof(DCAContext, xch_disable), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
+ { "disable_xll", "disable decoding of the XLL extension", offsetof(DCAContext, xll_disable), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
{ NULL },
};
projects / ffmpeg / blobdiff