#define HEADER_SIZE 14
#define CONVERT_BIAS 384
-#define DCA_MAX_FRAME_SIZE 16383
+#define DCA_MAX_FRAME_SIZE 16384
/** Bit allocation */
typedef struct {
static BitAlloc dca_scalefactor; ///< scalefactor VLCs
static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs
-/** Pre-calculated cosine modulation coefs for the QMF */
-static float cos_mod[544];
-
static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx)
{
return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset;
/* Primary audio coding header */
int subframes; ///< number of subframes
+ 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 vq_start_subband[DCA_PRIM_CHANNELS_MAX]; ///< high frequency vq start subband
/* Subband samples history (for ADPCM) */
float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
- float subband_fir_hist[DCA_PRIM_CHANNELS_MAX][512];
- float subband_fir_noidea[DCA_PRIM_CHANNELS_MAX][64];
+ DECLARE_ALIGNED_16(float, subband_fir_hist[DCA_PRIM_CHANNELS_MAX][512]);
+ float subband_fir_noidea[DCA_PRIM_CHANNELS_MAX][32];
+ int hist_index[DCA_PRIM_CHANNELS_MAX];
int output; ///< type of output
int bias; ///< output bias
DECLARE_ALIGNED_16(float, samples[1536]); /* 6 * 256 = 1536, might only need 5 */
- DECLARE_ALIGNED_16(int16_t, tsamples[1536]);
+ const float *samples_chanptr[6];
uint8_t dca_buffer[DCA_MAX_FRAME_SIZE];
int dca_buffer_size; ///< how much data is in the dca_buffer
int debug_flag; ///< used for suppressing repeated error messages output
DSPContext dsp;
+ MDCTContext imdct;
} DCAContext;
-static void dca_init_vlcs(void)
+static av_cold void dca_init_vlcs(void)
{
- static int vlcs_inited = 0;
+ static int vlcs_initialized = 0;
int i, j;
- if (vlcs_inited)
+ if (vlcs_initialized)
return;
dca_bitalloc_index.offset = 1;
bitalloc_bits[i][j], 1, 1,
bitalloc_codes[i][j], 2, 2, 1);
}
- vlcs_inited = 1;
+ vlcs_initialized = 1;
}
static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
/* Primary audio coding header */
s->subframes = get_bits(&s->gb, 4) + 1;
- s->prim_channels = get_bits(&s->gb, 3) + 1;
+ s->total_channels = get_bits(&s->gb, 3) + 1;
+ s->prim_channels = s->total_channels;
+ if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
+ s->prim_channels = DCA_PRIM_CHANNELS_MAX; /* We only support DTS core */
for (i = 0; i < s->prim_channels; i++) {
s->bitalloc[j][k] = get_bits(&s->gb, 5);
else if (s->bitalloc_huffman[j] == 5)
s->bitalloc[j][k] = get_bits(&s->gb, 4);
- else {
+ else if (s->bitalloc_huffman[j] == 7) {
+ av_log(s->avctx, AV_LOG_ERROR,
+ "Invalid bit allocation index\n");
+ return -1;
+ } else {
s->bitalloc[j][k] =
get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
}
}
for (j = 0; j < s->prim_channels; j++) {
- uint32_t *scale_table;
+ const uint32_t *scale_table;
int scale_sum;
memset(s->scale_factor[j], 0, s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
if (s->scalefactor_huffman[j] == 6)
- scale_table = (uint32_t *) scale_factor_quant7;
+ scale_table = scale_factor_quant7;
else
- scale_table = (uint32_t *) scale_factor_quant6;
+ scale_table = scale_factor_quant6;
/* When huffman coded, only the difference is encoded */
scale_sum = 0;
float scale, float bias)
{
const float *prCoeff;
- int i, j, k;
- float praXin[33], *raXin = &praXin[1];
+ int i, j;
+ DECLARE_ALIGNED_16(float, raXin[32]);
- float *subband_fir_hist = s->subband_fir_hist[chans];
+ int hist_index= s->hist_index[chans];
float *subband_fir_hist2 = s->subband_fir_noidea[chans];
- int chindex = 0, subindex;
+ int subindex;
- praXin[0] = 0.0;
+ scale *= sqrt(1/8.0);
/* Select filter */
if (!s->multirate_inter) /* Non-perfect reconstruction */
/* Reconstructed channel sample index */
for (subindex = 0; subindex < 8; subindex++) {
- float t1, t2, sum[16], diff[16];
-
+ float *subband_fir_hist = s->subband_fir_hist[chans] + hist_index;
/* Load in one sample from each subband and clear inactive subbands */
- for (i = 0; i < s->subband_activity[chans]; i++)
- raXin[i] = samples_in[i][subindex];
+ for (i = 0; i < s->subband_activity[chans]; i++){
+ if((i-1)&2) raXin[i] = -samples_in[i][subindex];
+ else raXin[i] = samples_in[i][subindex];
+ }
for (; i < 32; i++)
raXin[i] = 0.0;
- /* Multiply by cosine modulation coefficients and
- * create temporary arrays SUM and DIFF */
- for (j = 0, k = 0; k < 16; k++) {
- t1 = 0.0;
- t2 = 0.0;
- for (i = 0; i < 16; i++, j++){
- t1 += (raXin[2 * i] + raXin[2 * i + 1]) * cos_mod[j];
- t2 += (raXin[2 * i] + raXin[2 * i - 1]) * cos_mod[j + 256];
- }
- sum[k] = t1 + t2;
- diff[k] = t1 - t2;
- }
-
- j = 512;
- /* Store history */
- for (k = 0; k < 16; k++)
- subband_fir_hist[k] = cos_mod[j++] * sum[k];
- for (k = 0; k < 16; k++)
- subband_fir_hist[32-k-1] = cos_mod[j++] * diff[k];
+ ff_imdct_half(&s->imdct, subband_fir_hist, raXin);
/* Multiply by filter coefficients */
- for (k = 31, i = 0; i < 32; i++, k--)
- for (j = 0; j < 512; j += 64){
- subband_fir_hist2[i] += prCoeff[i+j] * ( subband_fir_hist[i+j] - subband_fir_hist[j+k]);
- subband_fir_hist2[i+32] += prCoeff[i+j+32]*(-subband_fir_hist[i+j] - subband_fir_hist[j+k]);
+ for (i = 0; i < 16; i++){
+ float a= subband_fir_hist2[i ];
+ float b= subband_fir_hist2[i+16];
+ float c= 0;
+ float d= 0;
+ for (j = 0; j < 512-hist_index; j += 64){
+ a += prCoeff[i+j ]*(-subband_fir_hist[15-i+j]);
+ b += prCoeff[i+j+16]*( subband_fir_hist[ i+j]);
+ c += prCoeff[i+j+32]*( subband_fir_hist[16+i+j]);
+ d += prCoeff[i+j+48]*( subband_fir_hist[31-i+j]);
}
+ for ( ; j < 512; j += 64){
+ a += prCoeff[i+j ]*(-subband_fir_hist[15-i+j-512]);
+ b += prCoeff[i+j+16]*( subband_fir_hist[ i+j-512]);
+ c += prCoeff[i+j+32]*( subband_fir_hist[16+i+j-512]);
+ d += prCoeff[i+j+48]*( subband_fir_hist[31-i+j-512]);
+ }
+ samples_out[i ] = a * scale + bias;
+ samples_out[i+16] = b * scale + bias;
+ subband_fir_hist2[i ] = c;
+ subband_fir_hist2[i+16] = d;
+ }
+ samples_out+= 32;
- /* Create 32 PCM output samples */
- for (i = 0; i < 32; i++)
- samples_out[chindex++] = subband_fir_hist2[i] * scale + bias;
-
- /* Update working arrays */
- memmove(&subband_fir_hist[32], &subband_fir_hist[0], (512 - 32) * sizeof(float));
- memmove(&subband_fir_hist2[0], &subband_fir_hist2[32], 32 * sizeof(float));
- memset(&subband_fir_hist2[32], 0, 32 * sizeof(float));
+ hist_index = (hist_index-32)&511;
}
+ s->hist_index[chans]= hist_index;
}
static void lfe_interpolation_fir(int decimation_select,
int k, l;
int subsubframe = s->current_subsubframe;
- float *quant_step_table;
+ const float *quant_step_table;
/* FIXME */
float subband_samples[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
/* Select quantization step size table */
if (s->bit_rate == 0x1f)
- quant_step_table = (float *) lossless_quant_d;
+ quant_step_table = lossless_quant_d;
else
- quant_step_table = (float *) lossy_quant_d;
+ quant_step_table = lossy_quant_d;
for (k = 0; k < s->prim_channels; k++) {
for (l = 0; l < s->vq_start_subband[k]; l++) {
/* 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 * k],
- 2.0 / 3 /*pcm_to_double[s->source_pcm_res] */ ,
+ M_SQRT1_2 /*pcm_to_double[s->source_pcm_res] */ ,
0 /*s->bias */ );
}
s->lfe_data + lfe_samples +
2 * s->lfe * subsubframe,
&s->samples[256 * i_channels],
- 8388608.0, s->bias);
+ 256.0, 0 /* s->bias */);
/* Outputs 20bits pcm samples */
}
/**
* Convert bitstream to one representation based on sync marker
*/
-static int dca_convert_bitstream(uint8_t * src, int src_size, uint8_t * dst,
+static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * dst,
int max_size)
{
uint32_t mrk;
int i, tmp;
- uint16_t *ssrc = (uint16_t *) src, *sdst = (uint16_t *) dst;
+ const uint16_t *ssrc = (const uint16_t *) src;
+ uint16_t *sdst = (uint16_t *) dst;
PutBitContext pb;
if((unsigned)src_size > (unsigned)max_size) {
mrk = AV_RB32(src);
switch (mrk) {
case DCA_MARKER_RAW_BE:
- memcpy(dst, src, FFMIN(src_size, max_size));
- return FFMIN(src_size, max_size);
+ memcpy(dst, src, src_size);
+ return src_size;
case DCA_MARKER_RAW_LE:
- for (i = 0; i < (FFMIN(src_size, max_size) + 1) >> 1; i++)
+ for (i = 0; i < (src_size + 1) >> 1; i++)
*sdst++ = bswap_16(*ssrc++);
- return FFMIN(src_size, max_size);
+ return src_size;
case DCA_MARKER_14B_BE:
case DCA_MARKER_14B_LE:
init_put_bits(&pb, dst, max_size);
*/
static int dca_decode_frame(AVCodecContext * avctx,
void *data, int *data_size,
- uint8_t * buf, int buf_size)
+ const uint8_t * buf, int buf_size)
{
- int i, j, k;
+ int i;
int16_t *samples = data;
DCAContext *s = avctx->priv_data;
int channels;
avctx->bit_rate = s->bit_rate;
channels = s->prim_channels + !!s->lfe;
- avctx->channels = avctx->request_channels;
- if(avctx->channels == 0) {
- avctx->channels = channels;
- } else if(channels < avctx->channels) {
- av_log(avctx, AV_LOG_WARNING, "DTS source channels are less than "
- "specified: output to %d channels.\n", channels);
- avctx->channels = channels;
- }
- if(avctx->channels == 2) {
+ if(avctx->request_channels == 2 && s->prim_channels > 2) {
+ channels = 2;
s->output = DCA_STEREO;
- } else if(avctx->channels != channels) {
- av_log(avctx, AV_LOG_ERROR, "Cannot downmix DTS to %d channels.\n",
- avctx->channels);
- return -1;
}
- channels = avctx->channels;
+ /* There is nothing that prevents a dts frame to change channel configuration
+ but FFmpeg 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 (!avctx->channels)
+ avctx->channels = channels;
+
if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
return -1;
- *data_size = 0;
+ *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels;
for (i = 0; i < (s->sample_blocks / 8); i++) {
dca_decode_block(s);
- s->dsp.float_to_int16(s->tsamples, s->samples, 256 * channels);
- /* interleave samples */
- for (j = 0; j < 256; j++) {
- for (k = 0; k < channels; k++)
- samples[k] = s->tsamples[j + k * 256];
- samples += channels;
- }
- *data_size += 256 * sizeof(int16_t) * channels;
+ s->dsp.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
+ samples += 256 * channels;
}
return buf_size;
-/**
- * Build the cosine modulation tables for the QMF
- *
- * @param s pointer to the DCAContext
- */
-
-static void pre_calc_cosmod(DCAContext * s)
-{
- int i, j, k;
- static int cosmod_inited = 0;
-
- if(cosmod_inited) return;
- for (j = 0, k = 0; k < 16; k++)
- for (i = 0; i < 16; i++)
- cos_mod[j++] = cos((2 * i + 1) * (2 * k + 1) * M_PI / 64);
-
- for (k = 0; k < 16; k++)
- for (i = 0; i < 16; i++)
- cos_mod[j++] = cos((i) * (2 * k + 1) * M_PI / 32);
-
- for (k = 0; k < 16; k++)
- cos_mod[j++] = 0.25 / (2 * cos((2 * k + 1) * M_PI / 128));
-
- for (k = 0; k < 16; k++)
- cos_mod[j++] = -0.25 / (2.0 * sin((2 * k + 1) * M_PI / 128));
-
- cosmod_inited = 1;
-}
-
-
/**
* DCA initialization
*
* @param avctx pointer to the AVCodecContext
*/
-static int dca_decode_init(AVCodecContext * avctx)
+static av_cold int dca_decode_init(AVCodecContext * avctx)
{
DCAContext *s = avctx->priv_data;
+ int i;
s->avctx = avctx;
dca_init_vlcs();
- pre_calc_cosmod(s);
dsputil_init(&s->dsp, avctx);
+ ff_mdct_init(&s->imdct, 6, 1);
+
+ /* allow downmixing to stereo */
+ if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
+ avctx->request_channels == 2) {
+ avctx->channels = avctx->request_channels;
+ }
+ for(i = 0; i < 6; i++)
+ s->samples_chanptr[i] = s->samples + i * 256;
+ avctx->sample_fmt = SAMPLE_FMT_S16;
return 0;
}
+static av_cold int dca_decode_end(AVCodecContext * avctx)
+{
+ DCAContext *s = avctx->priv_data;
+ ff_mdct_end(&s->imdct);
+ return 0;
+}
AVCodec dca_decoder = {
.name = "dca",
.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)"),
};
projects / ffmpeg / blobdiff