[FFmpeg-cvslog] r23695 - trunk/libavcodec/dca.c
mstorsjo
subversion
Tue Jun 22 10:33:00 CEST 2010
Author: mstorsjo
Date: Tue Jun 22 10:33:00 2010
New Revision: 23695
Log:
Support DTS-ES extension (XCh) in dca: move original code around to allow reused by DTS-ES code
Patch by Nick Brereton, nick at nbrereton dot net
Modified:
trunk/libavcodec/dca.c
Modified: trunk/libavcodec/dca.c
==============================================================================
--- trunk/libavcodec/dca.c Tue Jun 22 10:32:01 2010 (r23694)
+++ trunk/libavcodec/dca.c Tue Jun 22 10:33:00 2010 (r23695)
@@ -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(GetBitConte
*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(DCACon
/* 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(GetBitContex
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(DCAContex
/* 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(DCAContex
/* 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(DCAContex
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 *
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 *
}
/* 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 *
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(DCAContex
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 *
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(AVCodecConte
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(AVCodecConte
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(AVCodecConte
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(AVCod
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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