#include <stddef.h>
#include <stdio.h>
+#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
-#include "libavutil/intmath.h"
+#include "libavutil/internal.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/mathematics.h"
-#include "libavutil/audioconvert.h"
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
#include "avcodec.h"
-#include "dsputil.h"
#include "fft.h"
#include "get_bits.h"
#include "put_bits.h"
#include "dcadata.h"
#include "dcahuff.h"
#include "dca.h"
-#include "dca_parser.h"
+#include "mathops.h"
#include "synth_filter.h"
#include "dcadsp.h"
#include "fmtconvert.h"
+#include "internal.h"
#if ARCH_ARM
# include "arm/dca.h"
#define DCA_BUFFER_PADDING_SIZE 1024
+#define DCA_NSYNCAUX 0x9A1105A0
+
/** Bit allocation */
typedef struct {
int offset; ///< code values offset
}
typedef struct {
+ AVClass *class; ///< class for AVOptions
AVCodecContext *avctx;
- AVFrame frame;
/* Frame header */
int frame_type; ///< type of the current frame
int samples_deficit; ///< deficit sample count
int bit_rate; ///< transmission bit rate
int bit_rate_index; ///< transmission bit rate index
- int downmix; ///< embedded downmix enabled
int dynrange; ///< embedded dynamic range flag
int timestamp; ///< embedded time stamp flag
int aux_data; ///< auxiliary data flag
/* Primary audio coding header */
int subframes; ///< number of subframes
- int is_channels_set; ///< check for if the channel number is already set
int total_channels; ///< number of channels including extensions
int prim_channels; ///< number of primary audio channels
int subband_activity[DCA_PRIM_CHANNELS_MAX]; ///< subband activity count
int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2]; ///< scale factors (2 if transient)
int joint_huff[DCA_PRIM_CHANNELS_MAX]; ///< joint subband scale factors codebook
int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors
- int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]; ///< stereo downmix coefficients
+ float downmix_coef[DCA_PRIM_CHANNELS_MAX + 1][2]; ///< stereo downmix coefficients
int dynrange_coef; ///< dynamic range coefficient
+ /* Core substream's embedded downmix coefficients (cf. ETSI TS 102 114 V1.4.1)
+ * Input: primary audio channels (incl. LFE if present)
+ * Output: downmix audio channels (up to 4, no LFE) */
+ uint8_t core_downmix; ///< embedded downmix coefficients available
+ uint8_t core_downmix_amode; ///< audio channel arrangement of embedded downmix
+ uint16_t core_downmix_codes[DCA_PRIM_CHANNELS_MAX + 1][4]; ///< embedded downmix coefficients (9-bit codes)
+
int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands
float lfe_data[2 * DCA_LFE_MAX * (DCA_BLOCKS_MAX + 4)]; ///< Low frequency effect data
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];
+ float *extra_channels[DCA_PRIM_CHANNELS_MAX + 1];
+ uint8_t *extra_channels_buffer;
+ unsigned int extra_channels_buffer_size;
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
/* XCh extension information */
int xch_present; ///< XCh extension present and valid
int xch_base_channel; ///< index of first (only) channel containing XCH data
+ int xch_disable; ///< whether the XCh extension should be decoded or not
/* ExSS header parser */
int static_fields; ///< static fields present
if (s->frame_size < 95)
return AVERROR_INVALIDDATA;
s->amode = get_bits(&s->gb, 6);
- s->sample_rate = ff_dca_sample_rates[get_bits(&s->gb, 4)];
+ s->sample_rate = avpriv_dca_sample_rates[get_bits(&s->gb, 4)];
if (!s->sample_rate)
return AVERROR_INVALIDDATA;
s->bit_rate_index = get_bits(&s->gb, 5);
if (!s->bit_rate)
return AVERROR_INVALIDDATA;
- s->downmix = get_bits(&s->gb, 1);
+ skip_bits1(&s->gb); // always 0 (reserved, cf. ETSI TS 102 114 V1.4.1)
s->dynrange = get_bits(&s->gb, 1);
s->timestamp = get_bits(&s->gb, 1);
s->aux_data = get_bits(&s->gb, 1);
s->lfe = get_bits(&s->gb, 2);
s->predictor_history = get_bits(&s->gb, 1);
+ if (s->lfe > 2) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE value: %d\n", s->lfe);
+ return AVERROR_INVALIDDATA;
+ }
+
/* TODO: check CRC */
if (s->crc_present)
s->header_crc = get_bits(&s->gb, 16);
s->sample_rate);
av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i bits/s\n",
s->bit_rate);
- av_log(s->avctx, AV_LOG_DEBUG, "downmix: %i\n", s->downmix);
av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange);
av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp);
av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data);
}
if (s->bitalloc[j][k] > 26) {
- // av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index [%i][%i] too big (%i)\n",
- // j, k, s->bitalloc[j][k]);
+ av_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n",
+ j, k, s->bitalloc[j][k]);
return AVERROR_INVALIDDATA;
}
}
}
}
- /* Stereo downmix coefficients */
- if (!base_channel && s->prim_channels > 2) {
- if (s->downmix) {
- for (j = base_channel; j < s->prim_channels; j++) {
- s->downmix_coef[j][0] = get_bits(&s->gb, 7);
- s->downmix_coef[j][1] = get_bits(&s->gb, 7);
- }
- } else {
- int am = s->amode & DCA_CHANNEL_MASK;
- if (am >= FF_ARRAY_ELEMS(dca_default_coeffs)) {
- av_log(s->avctx, AV_LOG_ERROR,
- "Invalid channel mode %d\n", am);
- return AVERROR_INVALIDDATA;
- }
- for (j = base_channel; j < s->prim_channels; j++) {
- s->downmix_coef[j][0] = dca_default_coeffs[am][j][0];
- s->downmix_coef[j][1] = dca_default_coeffs[am][j][1];
- }
- }
- }
-
/* Dynamic range coefficient */
if (!base_channel && s->dynrange)
s->dynrange_coef = get_bits(&s->gb, 8);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
}
- if (!base_channel && s->prim_channels > 2 && s->downmix) {
- av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n");
- for (j = 0; j < s->prim_channels; j++) {
- av_log(s->avctx, AV_LOG_DEBUG, "Channel 0, %d = %f\n", j,
- dca_downmix_coeffs[s->downmix_coef[j][0]]);
- av_log(s->avctx, AV_LOG_DEBUG, "Channel 1, %d = %f\n", j,
- dca_downmix_coeffs[s->downmix_coef[j][1]]);
- }
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
- }
for (j = base_channel; j < s->prim_channels; j++)
for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]);
float scale)
{
const float *prCoeff;
- int i;
int sb_act = s->subband_activity[chans];
- int subindex;
scale *= sqrt(1 / 8.0);
else /* Perfect reconstruction */
prCoeff = fir_32bands_perfect;
- for (i = sb_act; i < 32; i++)
- s->raXin[i] = 0.0;
-
- /* Reconstructed channel sample index */
- for (subindex = 0; subindex < 8; subindex++) {
- /* Load in one sample from each subband and clear inactive subbands */
- for (i = 0; i < sb_act; i++) {
- unsigned sign = (i - 1) & 2;
- uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30;
- AV_WN32A(&s->raXin[i], v);
- }
-
- s->synth.synth_filter_float(&s->imdct,
- s->subband_fir_hist[chans],
- &s->hist_index[chans],
- s->subband_fir_noidea[chans], prCoeff,
- samples_out, s->raXin, scale);
- samples_out += 32;
- }
+ s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct,
+ s->subband_fir_hist[chans],
+ &s->hist_index[chans],
+ s->subband_fir_noidea[chans], prCoeff,
+ samples_out, s->raXin, scale);
}
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++) { \
op2 \
}
-static void dca_downmix(float *samples, int srcfmt,
- int downmix_coef[DCA_PRIM_CHANNELS_MAX][2],
+static void dca_downmix(float **samples, int srcfmt, int lfe_present,
+ float coef[DCA_PRIM_CHANNELS_MAX + 1][2],
const int8_t *channel_mapping)
{
int c, l, r, sl, sr, s;
int i;
float t, u, v;
- float coef[DCA_PRIM_CHANNELS_MAX][2];
-
- for (i = 0; i < DCA_PRIM_CHANNELS_MAX; i++) {
- coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]];
- coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]];
- }
switch (srcfmt) {
case DCA_MONO:
- case DCA_CHANNEL:
- case DCA_STEREO_TOTAL:
- case DCA_STEREO_SUMDIFF:
case DCA_4F2R:
av_log(NULL, 0, "Not implemented!\n");
break;
+ case DCA_CHANNEL:
case DCA_STEREO:
+ case DCA_STEREO_TOTAL:
+ case DCA_STEREO_SUMDIFF:
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;
}
+ if (lfe_present) {
+ int lf_buf = dca_lfe_index[srcfmt];
+ int lf_idx = dca_channels [srcfmt];
+ for (i = 0; i < 256; i++) {
+ samples[0][i] += samples[lf_buf][i] * coef[lf_idx][0];
+ samples[1][i] += samples[lf_buf][i] * coef[lf_idx][1];
+ }
+ }
}
#ifndef decode_blockcodes
/* Very compact version of the block code decoder that does not use table
* look-up but is slightly slower */
-static int decode_blockcode(int code, int levels, int *values)
+static int decode_blockcode(int code, int levels, int32_t *values)
{
int i;
int offset = (levels - 1) >> 1;
return code;
}
-static int decode_blockcodes(int code1, int code2, int levels, int *values)
+static int decode_blockcodes(int code1, int code2, int levels, int32_t *values)
{
return decode_blockcode(code1, levels, values) |
decode_blockcode(code2, levels, values + 4);
static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
#ifndef int8x8_fmul_int32
-static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)
+static inline void int8x8_fmul_int32(DCADSPContext *dsp, float *dst,
+ const int8_t *src, int scale)
{
- float fscale = scale / 16.0;
- int i;
- for (i = 0; i < 8; i++)
- dst[i] = src[i] * fscale;
+ dsp->int8x8_fmul_int32(dst, src, scale);
}
#endif
/* FIXME */
float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
- LOCAL_ALIGNED_16(int, block, [8]);
+ LOCAL_ALIGNED_16(int32_t, block, [8 * DCA_SUBBANDS]);
/*
* Audio data
quant_step_table = lossy_quant_d;
for (k = base_channel; k < s->prim_channels; k++) {
+ float rscale[DCA_SUBBANDS];
+
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
* Extract bits from the bit stream
*/
if (!abits) {
- memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));
+ rscale[l] = 0;
+ memset(block + 8 * l, 0, 8 * sizeof(block[0]));
} else {
/* Deal with transients */
int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];
- float rscale = quant_step_size * s->scale_factor[k][l][sfi] *
+ rscale[l] = quant_step_size * s->scale_factor[k][l][sfi] *
s->scalefactor_adj[k][sel];
if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {
block_code1 = get_bits(&s->gb, size);
block_code2 = get_bits(&s->gb, size);
err = decode_blockcodes(block_code1, block_code2,
- levels, block);
+ levels, block + 8 * l);
if (err) {
av_log(s->avctx, AV_LOG_ERROR,
"ERROR: block code look-up failed\n");
} else {
/* no coding */
for (m = 0; m < 8; m++)
- block[m] = get_sbits(&s->gb, abits - 3);
+ block[8 * l + m] = get_sbits(&s->gb, abits - 3);
}
} else {
/* Huffman coded */
for (m = 0; m < 8; m++)
- block[m] = get_bitalloc(&s->gb,
+ block[8 * l + m] = get_bitalloc(&s->gb,
&dca_smpl_bitalloc[abits], sel);
}
- s->fmt_conv.int32_to_float_fmul_scalar(subband_samples[k][l],
- block, rscale, 8);
}
+ }
+
+ s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[k][0],
+ block, rscale, 8 * s->vq_start_subband[k]);
+ for (l = 0; l < s->vq_start_subband[k]; l++) {
+ int m;
/*
* Inverse ADPCM if in prediction mode
*/
s->debug_flag |= 0x01;
}
- int8x8_fmul_int32(subband_samples[k][l],
+ int8x8_fmul_int32(&s->dcadsp, subband_samples[k][l],
&high_freq_vq[hfvq][subsubframe * 8],
s->scale_factor[k][l][0]);
}
#endif
} else {
av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
+ return AVERROR_INVALIDDATA;
}
}
for (k = 0; k < s->prim_channels; k++) {
/* 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] */);
- }
-
- /* 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);
+ 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 /* pcm_to_double[s->source_pcm_res] */);
}
/* Generate LFE samples for this subsubframe FIXME!!! */
- if (s->output & DCA_LFE) {
+ if (s->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 */
}
+ /* Downmixing to Stereo */
+ if (s->prim_channels + !!s->lfe > 2 &&
+ s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
+ dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef,
+ s->channel_order_tab);
+ }
+
return 0;
}
static int dca_subframe_footer(DCAContext *s, int base_channel)
{
- int aux_data_count = 0, i;
+ int in, out, aux_data_count, aux_data_end, reserved;
+ uint32_t nsyncaux;
/*
* Unpack optional information
if (s->timestamp)
skip_bits_long(&s->gb, 32);
- if (s->aux_data)
+ if (s->aux_data) {
aux_data_count = get_bits(&s->gb, 6);
- for (i = 0; i < aux_data_count; i++)
- get_bits(&s->gb, 8);
+ // align (32-bit)
+ skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
+
+ aux_data_end = 8 * aux_data_count + get_bits_count(&s->gb);
+
+ if ((nsyncaux = get_bits_long(&s->gb, 32)) != DCA_NSYNCAUX) {
+ av_log(s->avctx, AV_LOG_ERROR, "nSYNCAUX mismatch %#"PRIx32"\n",
+ nsyncaux);
+ return AVERROR_INVALIDDATA;
+ }
+
+ if (get_bits1(&s->gb)) { // bAUXTimeStampFlag
+ avpriv_request_sample(s->avctx,
+ "Auxiliary Decode Time Stamp Flag");
+ // align (4-bit)
+ skip_bits(&s->gb, (-get_bits_count(&s->gb)) & 4);
+ // 44 bits: nMSByte (8), nMarker (4), nLSByte (28), nMarker (4)
+ skip_bits_long(&s->gb, 44);
+ }
+
+ if ((s->core_downmix = get_bits1(&s->gb))) {
+ int am = get_bits(&s->gb, 3);
+ switch (am) {
+ case 0:
+ s->core_downmix_amode = DCA_MONO;
+ break;
+ case 1:
+ s->core_downmix_amode = DCA_STEREO;
+ break;
+ case 2:
+ s->core_downmix_amode = DCA_STEREO_TOTAL;
+ break;
+ case 3:
+ s->core_downmix_amode = DCA_3F;
+ break;
+ case 4:
+ s->core_downmix_amode = DCA_2F1R;
+ break;
+ case 5:
+ s->core_downmix_amode = DCA_2F2R;
+ break;
+ case 6:
+ s->core_downmix_amode = DCA_3F1R;
+ break;
+ default:
+ av_log(s->avctx, AV_LOG_ERROR,
+ "Invalid mode %d for embedded downmix coefficients\n",
+ am);
+ return AVERROR_INVALIDDATA;
+ }
+ for (out = 0; out < dca_channels[s->core_downmix_amode]; out++) {
+ for (in = 0; in < s->prim_channels + !!s->lfe; in++) {
+ uint16_t tmp = get_bits(&s->gb, 9);
+ if ((tmp & 0xFF) > 241) {
+ av_log(s->avctx, AV_LOG_ERROR,
+ "Invalid downmix coefficient code %"PRIu16"\n",
+ tmp);
+ return AVERROR_INVALIDDATA;
+ }
+ s->core_downmix_codes[in][out] = tmp;
+ }
+ }
+ }
+
+ align_get_bits(&s->gb); // byte align
+ skip_bits(&s->gb, 16); // nAUXCRC16
- if (s->crc_present && (s->downmix || s->dynrange))
+ // additional data (reserved, cf. ETSI TS 102 114 V1.4.1)
+ if ((reserved = (aux_data_end - get_bits_count(&s->gb))) < 0) {
+ av_log(s->avctx, AV_LOG_ERROR,
+ "Overread auxiliary data by %d bits\n", -reserved);
+ return AVERROR_INVALIDDATA;
+ } else if (reserved) {
+ avpriv_request_sample(s->avctx,
+ "Core auxiliary data reserved content");
+ skip_bits_long(&s->gb, reserved);
+ }
+ }
+
+ if (s->crc_present && s->dynrange)
get_bits(&s->gb, 16);
}
num_audiop = get_bits(&s->gb, 3) + 1;
if (num_audiop > 1) {
- av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio presentations.");
+ avpriv_request_sample(s->avctx,
+ "Multiple DTS-HD audio presentations");
/* ignore such streams for now */
return;
}
num_assets = get_bits(&s->gb, 3) + 1;
if (num_assets > 1) {
- av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio assets.");
+ avpriv_request_sample(s->avctx, "Multiple DTS-HD audio assets");
/* ignore such streams for now */
return;
}
static int dca_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
+ AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
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 channels, full_channels;
int core_ss_end;
avctx->profile = s->profile;
- channels = s->prim_channels + !!s->lfe;
+ full_channels = channels = s->prim_channels + !!s->lfe;
if (s->amode < 16) {
avctx->channel_layout = dca_core_channel_layout[s->amode];
- if (s->xch_present && (!avctx->request_channels ||
- avctx->request_channels > num_core_channels + !!s->lfe)) {
+#if FF_API_REQUEST_CHANNELS
+FF_DISABLE_DEPRECATION_WARNINGS
+ if (s->xch_present && !s->xch_disable &&
+ (!avctx->request_channels ||
+ avctx->request_channels > num_core_channels + !!s->lfe)) {
+FF_ENABLE_DEPRECATION_WARNINGS
+#else
+ if (s->xch_present && !s->xch_disable) {
+#endif
avctx->channel_layout |= AV_CH_BACK_CENTER;
if (s->lfe) {
avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
s->channel_order_tab[channels - 1 - !!s->lfe] < 0)
return AVERROR_INVALIDDATA;
- if (avctx->request_channels == 2 && s->prim_channels > 2) {
+ if (s->prim_channels + !!s->lfe > 2 &&
+ avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
channels = 2;
- s->output = DCA_STEREO;
+ s->output = s->prim_channels == 2 ? s->amode : DCA_STEREO;
avctx->channel_layout = AV_CH_LAYOUT_STEREO;
+
+ /* Stereo downmix coefficients
+ *
+ * The decoder can only downmix to 2-channel, so we need to ensure
+ * embedded downmix coefficients are actually targeting 2-channel.
+ */
+ if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO ||
+ s->core_downmix_amode == DCA_STEREO_TOTAL)) {
+ int sign, code;
+ for (i = 0; i < s->prim_channels + !!s->lfe; i++) {
+ sign = s->core_downmix_codes[i][0] & 0x100 ? 1 : -1;
+ code = s->core_downmix_codes[i][0] & 0x0FF;
+ s->downmix_coef[i][0] = (!code ? 0.0f :
+ sign * dca_dmixtable[code - 1]);
+ sign = s->core_downmix_codes[i][1] & 0x100 ? 1 : -1;
+ code = s->core_downmix_codes[i][1] & 0x0FF;
+ s->downmix_coef[i][1] = (!code ? 0.0f :
+ sign * dca_dmixtable[code - 1]);
+ }
+ s->output = s->core_downmix_amode;
+ } else {
+ int am = s->amode & DCA_CHANNEL_MASK;
+ if (am >= FF_ARRAY_ELEMS(dca_default_coeffs)) {
+ av_log(s->avctx, AV_LOG_ERROR,
+ "Invalid channel mode %d\n", am);
+ return AVERROR_INVALIDDATA;
+ }
+ if (s->prim_channels + !!s->lfe >
+ FF_ARRAY_ELEMS(dca_default_coeffs[0])) {
+ avpriv_request_sample(s->avctx, "Downmixing %d channels",
+ s->prim_channels + !!s->lfe);
+ return AVERROR_PATCHWELCOME;
+ }
+ for (i = 0; i < s->prim_channels + !!s->lfe; i++) {
+ s->downmix_coef[i][0] = dca_default_coeffs[am][i][0];
+ s->downmix_coef[i][1] = dca_default_coeffs[am][i][1];
+ }
+ }
+ av_dlog(s->avctx, "Stereo downmix coeffs:\n");
+ for (i = 0; i < s->prim_channels + !!s->lfe; i++) {
+ av_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,
+ s->downmix_coef[i][1]);
+ }
+ av_dlog(s->avctx, "\n");
}
} else {
av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);
return AVERROR_INVALIDDATA;
}
-
-
- /* There is nothing that prevents a dts frame to change channel configuration
- but Libav doesn't support that so only set the channels if it is previously
- unset. Ideally during the first probe for channels the crc should be checked
- and only set avctx->channels when the crc is ok. Right now the decoder could
- set the channels based on a broken first frame.*/
- if (s->is_channels_set == 0) {
- s->is_channels_set = 1;
- avctx->channels = channels;
- }
- if (avctx->channels != channels) {
- av_log(avctx, AV_LOG_ERROR, "DCA decoder does not support number of "
- "channels changing in stream. Skipping frame.\n");
- return AVERROR_PATCHWELCOME;
- }
+ avctx->channels = channels;
/* get output buffer */
- s->frame.nb_samples = 256 * (s->sample_blocks / 8);
- if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+ frame->nb_samples = 256 * (s->sample_blocks / 8);
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
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 **)frame->extended_data;
+
+ /* allocate buffer for extra channels if downmixing */
+ if (avctx->channels < full_channels) {
+ ret = av_samples_get_buffer_size(NULL, full_channels - channels,
+ frame->nb_samples,
+ avctx->sample_fmt, 0);
+ if (ret < 0)
+ return ret;
+
+ av_fast_malloc(&s->extra_channels_buffer,
+ &s->extra_channels_buffer_size, ret);
+ if (!s->extra_channels_buffer)
+ return AVERROR(ENOMEM);
+
+ ret = av_samples_fill_arrays((uint8_t **)s->extra_channels, NULL,
+ s->extra_channels_buffer,
+ full_channels - channels,
+ frame->nb_samples, avctx->sample_fmt, 0);
+ if (ret < 0)
+ return ret;
+ }
/* 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;
+ for (; ch < full_channels; ch++)
+ s->samples_chanptr[ch] = s->extra_channels[ch - channels] + 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 */
for (i = 0; i < 2 * s->lfe * 4; i++)
s->lfe_data[i] = s->lfe_data[i + lfe_samples];
- *got_frame_ptr = 1;
- *(AVFrame *) data = s->frame;
+ /* AVMatrixEncoding
+ *
+ * DCA_STEREO_TOTAL (Lt/Rt) is equivalent to Dolby Surround */
+ ret = ff_side_data_update_matrix_encoding(frame,
+ (s->output & ~DCA_LFE) == DCA_STEREO_TOTAL ?
+ AV_MATRIX_ENCODING_DOLBY : AV_MATRIX_ENCODING_NONE);
+ if (ret < 0)
+ return ret;
+
+ *got_frame_ptr = 1;
return buf_size;
}
static av_cold int dca_decode_init(AVCodecContext *avctx)
{
DCAContext *s = avctx->priv_data;
- int i;
s->avctx = avctx;
dca_init_vlcs();
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 &&
- avctx->request_channels == 2) {
- avctx->channels = avctx->request_channels;
- }
-
- avcodec_get_frame_defaults(&s->frame);
- avctx->coded_frame = &s->frame;
+#if FF_API_REQUEST_CHANNELS
+FF_DISABLE_DEPRECATION_WARNINGS
+ if (avctx->request_channels == 2)
+ avctx->request_channel_layout = AV_CH_LAYOUT_STEREO;
+FF_ENABLE_DEPRECATION_WARNINGS
+#endif
+ if (avctx->channels > 2 &&
+ avctx->request_channel_layout == AV_CH_LAYOUT_STEREO)
+ avctx->channels = 2;
return 0;
}
{
DCAContext *s = avctx->priv_data;
ff_mdct_end(&s->imdct);
+ av_freep(&s->extra_channels_buffer);
return 0;
}
{ FF_PROFILE_UNKNOWN },
};
+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 },
+ { NULL },
+};
+
+static const AVClass dca_decoder_class = {
+ .class_name = "DCA decoder",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
AVCodec ff_dca_decoder = {
.name = "dca",
+ .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
.type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_DTS,
+ .id = AV_CODEC_ID_DTS,
.priv_data_size = sizeof(DCAContext),
.init = dca_decode_init,
.decode = dca_decode_frame,
.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),
+ .priv_class = &dca_decoder_class,
};
projects / ffmpeg / blobdiff