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Support DTS-ES extension (XCh) in dca: move original code around to allow reused by DTS-ES code

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Patch by Nick Brereton, nick at nbrereton dot net

Originally committed as revision 23695 to svn://svn.ffmpeg.org/ffmpeg/trunk
oldabi
Nick Brereton 15 years ago committed by Martin Storsjö
parent 77b4b7c383
commit 6baef06e8f
1 changed files with 110 additions and 91 deletions
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  1. 201
      libavcodec/dca.c

201
libavcodec/dca.c
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@ -223,8 +223,7 @@ typedef struct {
int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands
float lfe_data[2 * DCA_SUBSUBFRAMES_MAX * DCA_LFE_MAX *
2 /*history */ ]; ///< Low frequency effect data
float lfe_data[2 * DCA_LFE_MAX * (DCA_BLOCKS_MAX + 4)]; ///< Low frequency effect data
int lfe_scale_factor;
/* Subband samples history (for ADPCM) */
@ -326,13 +325,85 @@ static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
*dst++ = get_bits(gb, bits);
}
static int dca_parse_frame_header(DCAContext * s)
static int dca_parse_audio_coding_header(DCAContext * s)
{
int i, j;
static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
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->subband_activity[i] = get_bits(&s->gb, 5) + 2;
if (s->subband_activity[i] > DCA_SUBBANDS)
s->subband_activity[i] = DCA_SUBBANDS;
}
for (i = 0; i < s->prim_channels; i++) {
s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
if (s->vq_start_subband[i] > DCA_SUBBANDS)
s->vq_start_subband[i] = DCA_SUBBANDS;
}
get_array(&s->gb, s->joint_intensity, s->prim_channels, 3);
get_array(&s->gb, s->transient_huffman, s->prim_channels, 2);
get_array(&s->gb, s->scalefactor_huffman, s->prim_channels, 3);
get_array(&s->gb, s->bitalloc_huffman, s->prim_channels, 3);
/* Get codebooks quantization indexes */
memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
for (j = 1; j < 11; j++)
for (i = 0; i < s->prim_channels; i++)
s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
/* Get scale factor adjustment */
for (j = 0; j < 11; j++)
for (i = 0; i < s->prim_channels; i++)
s->scalefactor_adj[i][j] = 1;
for (j = 1; j < 11; j++)
for (i = 0; i < s->prim_channels; i++)
if (s->quant_index_huffman[i][j] < thr[j])
s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
if (s->crc_present) {
/* Audio header CRC check */
get_bits(&s->gb, 16);
}
s->current_subframe = 0;
s->current_subsubframe = 0;
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);
av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);
for (i = 0; i < s->prim_channels; i++){
av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]);
av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]);
av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]);
av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]);
av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]);
av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]);
av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");
for (j = 0; j < 11; j++)
av_log(s->avctx, AV_LOG_DEBUG, " %i",
s->quant_index_huffman[i][j]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
for (j = 0; j < 11; j++)
av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
#endif
return 0;
}
static int dca_parse_frame_header(DCAContext * s)
{
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
/* Sync code */
@ -422,74 +493,8 @@ static int dca_parse_frame_header(DCAContext * s)
/* Primary audio coding header */
s->subframes = get_bits(&s->gb, 4) + 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->subband_activity[i] = get_bits(&s->gb, 5) + 2;
if (s->subband_activity[i] > DCA_SUBBANDS)
s->subband_activity[i] = DCA_SUBBANDS;
}
for (i = 0; i < s->prim_channels; i++) {
s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
if (s->vq_start_subband[i] > DCA_SUBBANDS)
s->vq_start_subband[i] = DCA_SUBBANDS;
}
get_array(&s->gb, s->joint_intensity, s->prim_channels, 3);
get_array(&s->gb, s->transient_huffman, s->prim_channels, 2);
get_array(&s->gb, s->scalefactor_huffman, s->prim_channels, 3);
get_array(&s->gb, s->bitalloc_huffman, s->prim_channels, 3);
/* Get codebooks quantization indexes */
memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
for (j = 1; j < 11; j++)
for (i = 0; i < s->prim_channels; i++)
s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
/* Get scale factor adjustment */
for (j = 0; j < 11; j++)
for (i = 0; i < s->prim_channels; i++)
s->scalefactor_adj[i][j] = 1;
for (j = 1; j < 11; j++)
for (i = 0; i < s->prim_channels; i++)
if (s->quant_index_huffman[i][j] < thr[j])
s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
if (s->crc_present) {
/* Audio header CRC check */
get_bits(&s->gb, 16);
}
s->current_subframe = 0;
s->current_subsubframe = 0;
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);
av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);
for(i = 0; i < s->prim_channels; i++){
av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]);
av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]);
av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]);
av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]);
av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]);
av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]);
av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");
for (j = 0; j < 11; j++)
av_log(s->avctx, AV_LOG_DEBUG, " %i",
s->quant_index_huffman[i][j]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
for (j = 0; j < 11; j++)
av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
#endif
return 0;
return dca_parse_audio_coding_header(s);
}
@ -503,7 +508,7 @@ static inline int get_scale(GetBitContext *gb, int level, int value)
return value;
}
static int dca_subframe_header(DCAContext * s)
static int dca_subframe_header(DCAContext * s, int block_index)
{
/* Primary audio coding side information */
int j, k;
@ -660,10 +665,11 @@ static int dca_subframe_header(DCAContext * s)
/* Low frequency effect data */
if (s->lfe) {
/* LFE samples */
int lfe_samples = 2 * s->lfe * s->subsubframes;
int lfe_samples = 2 * s->lfe * (4 + block_index);
int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes);
float lfe_scale;
for (j = lfe_samples; j < lfe_samples * 2; j++) {
for (j = lfe_samples; j < lfe_end_sample; j++) {
/* Signed 8 bits int */
s->lfe_data[j] = get_sbits(&s->gb, 8);
}
@ -674,7 +680,7 @@ static int dca_subframe_header(DCAContext * s)
/* Quantization step size * scale factor */
lfe_scale = 0.035 * s->lfe_scale_factor;
for (j = lfe_samples; j < lfe_samples * 2; j++)
for (j = lfe_samples; j < lfe_end_sample; j++)
s->lfe_data[j] *= lfe_scale;
}
@ -740,9 +746,11 @@ static int dca_subframe_header(DCAContext * s)
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]);
if(s->lfe){
int lfe_samples = 2 * s->lfe * s->subsubframes;
int lfe_samples = 2 * s->lfe * (4 + block_index);
int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);
av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n");
for (j = lfe_samples; j < lfe_samples * 2; j++)
for (j = lfe_samples; j < lfe_end_sample; j++)
av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
@ -1043,6 +1051,14 @@ static int dca_subsubframe(DCAContext * s, int block_index)
memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4],
4 * sizeof(subband_samples[0][0][0]));
return 0;
}
static int dca_filter_channels(DCAContext * s, int block_index)
{
float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
int k;
/* 32 subbands QMF */
for (k = 0; k < s->prim_channels; k++) {
/* static float pcm_to_double[8] =
@ -1053,18 +1069,14 @@ static int dca_subsubframe(DCAContext * s, int block_index)
}
/* Down mixing */
if (s->prim_channels > dca_channels[s->output & DCA_CHANNEL_MASK]) {
if (s->avctx->request_channels == 2 && s->prim_channels > 2) {
dca_downmix(s->samples, s->amode, s->downmix_coef);
}
/* Generate LFE samples for this subsubframe FIXME!!! */
if (s->output & DCA_LFE) {
int lfe_samples = 2 * s->lfe * s->subsubframes;
lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
s->lfe_data + lfe_samples +
2 * s->lfe * subsubframe,
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->add_bias);
/* Outputs 20bits pcm samples */
@ -1077,7 +1089,6 @@ static int dca_subsubframe(DCAContext * s, int block_index)
static int dca_subframe_footer(DCAContext * s)
{
int aux_data_count = 0, i;
int lfe_samples;
/*
* Unpack optional information
@ -1095,11 +1106,6 @@ static int dca_subframe_footer(DCAContext * s)
if (s->crc_present && (s->downmix || s->dynrange))
get_bits(&s->gb, 16);
lfe_samples = 2 * s->lfe * s->subsubframes;
for (i = 0; i < lfe_samples; i++) {
s->lfe_data[i] = s->lfe_data[i + lfe_samples];
}
return 0;
}
@ -1124,7 +1130,7 @@ static int dca_decode_block(DCAContext * s, int block_index)
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
#endif
/* Read subframe header */
if (dca_subframe_header(s))
if (dca_subframe_header(s, block_index))
return -1;
}
@ -1205,6 +1211,7 @@ static int dca_decode_frame(AVCodecContext * avctx,
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int lfe_samples;
int i;
int16_t *samples = data;
DCAContext *s = avctx->priv_data;
@ -1227,6 +1234,10 @@ static int dca_decode_frame(AVCodecContext * avctx,
avctx->sample_rate = s->sample_rate;
avctx->bit_rate = s->bit_rate;
for (i = 0; i < (s->sample_blocks / 8); i++) {
dca_decode_block(s, i);
}
channels = s->prim_channels + !!s->lfe;
if (s->amode<16) {
@ -1264,12 +1275,20 @@ static int dca_decode_frame(AVCodecContext * avctx,
if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
return -1;
*data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels;
/* filter to get final output */
for (i = 0; i < (s->sample_blocks / 8); i++) {
dca_decode_block(s, i);
dca_filter_channels(s, i);
s->dsp.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
samples += 256 * channels;
}
/* update lfe history */
lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);
for (i = 0; i < 2 * s->lfe * 4; i++) {
s->lfe_data[i] = s->lfe_data[i + lfe_samples];
}
return buf_size;
}
@ -1294,7 +1313,7 @@ static av_cold int dca_decode_init(AVCodecContext * avctx)
ff_synth_filter_init(&s->synth);
ff_dcadsp_init(&s->dcadsp);
for(i = 0; i < 6; i++)
for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)
s->samples_chanptr[i] = s->samples + i * 256;
avctx->sample_fmt = SAMPLE_FMT_S16;

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