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authorNick Brereton <nick@nbrereton.net>2010-06-22 08:35:44 +0000
committerMartin Storsjö <martin@martin.st>2010-06-22 08:35:44 +0000
commitd1177cb589621016f681789dd66873832d5fb14a (patch)
tree42f71ac8df9996b60c9197860e8e5c23a57e86cf
parent774e9acfa748b929e1f288fca3f5b7f2d6746a61 (diff)
downloadffmpeg-d1177cb589621016f681789dd66873832d5fb14a.tar.gz
Support DTS-ES extension (XCh) in dca: Cosmetic cleanup
Patch by Nick Brereton, nick at nbrereton dot net Originally committed as revision 23698 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r--libavcodec/dca.c46
1 files changed, 23 insertions, 23 deletions
diff --git a/libavcodec/dca.c b/libavcodec/dca.c
index 59a3e1da49..0d4fa5aee6 100644
--- a/libavcodec/dca.c
+++ b/libavcodec/dca.c
@@ -341,9 +341,9 @@ static av_cold void dca_init_vlcs(void)
tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
}
- for(i = 0; i < 10; i++)
- for(j = 0; j < 7; j++){
- if(!bitalloc_codes[i][j]) break;
+ for (i = 0; i < 10; i++)
+ for (j = 0; j < 7; j++){
+ if (!bitalloc_codes[i][j]) break;
dca_smpl_bitalloc[i+1].offset = bitalloc_offsets[i];
dca_smpl_bitalloc[i+1].wrap = 1 + (j > 4);
dca_smpl_bitalloc[i+1].vlc[j].table = &dca_table[dca_vlc_offs[c]];
@@ -491,7 +491,7 @@ static int dca_parse_frame_header(DCAContext * s)
/* FIXME: channels mixing levels */
s->output = s->amode;
- if(s->lfe) s->output |= DCA_LFE;
+ if (s->lfe) s->output |= DCA_LFE;
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type);
@@ -543,7 +543,7 @@ static inline int get_scale(GetBitContext *gb, int level, int value)
if (level < 5) {
/* huffman encoded */
value += get_bitalloc(gb, &dca_scalefactor, level);
- } else if(level < 8)
+ } else if (level < 8)
value = get_bits(gb, level + 1);
return value;
}
@@ -672,7 +672,7 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index
/* Stereo downmix coefficients */
if (!base_channel && s->prim_channels > 2) {
- if(s->downmix) {
+ 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);
@@ -888,7 +888,7 @@ static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
samples[i+256] = t * coef[0][1] + samples[i+256] * coef[1][1] + samples[i+512] * coef[2][1];
#define DOWNMIX_TO_STEREO(op1, op2) \
- for(i = 0; i < 256; i++){ \
+ for (i = 0; i < 256; i++){ \
op1 \
op2 \
}
@@ -900,7 +900,7 @@ static void dca_downmix(float *samples, int srcfmt,
float t;
float coef[DCA_PRIM_CHANNELS_MAX][2];
- for(i=0; i<DCA_PRIM_CHANNELS_MAX; i++) {
+ 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]];
}
@@ -1000,15 +1000,15 @@ static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)
/*
* Extract bits from the bit stream
*/
- if(!abits){
+ if (!abits){
memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][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] * s->scalefactor_adj[k][sel];
- if(abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
- if(abits <= 7){
+ if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
+ if (abits <= 7){
/* Block code */
int block_code1, block_code2, size, levels;
@@ -1139,17 +1139,17 @@ static int dca_subframe_footer(DCAContext * s, int base_channel)
/* presumably optional information only appears in the core? */
if (!base_channel) {
- if (s->timestamp)
- get_bits(&s->gb, 32);
+ if (s->timestamp)
+ get_bits(&s->gb, 32);
- if (s->aux_data)
- aux_data_count = get_bits(&s->gb, 6);
+ 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);
+ for (i = 0; i < aux_data_count; i++)
+ get_bits(&s->gb, 8);
- if (s->crc_present && (s->downmix || s->dynrange))
- get_bits(&s->gb, 16);
+ if (s->crc_present && (s->downmix || s->dynrange))
+ get_bits(&s->gb, 16);
}
return 0;
@@ -1217,7 +1217,7 @@ static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * ds
uint16_t *sdst = (uint16_t *) dst;
PutBitContext pb;
- if((unsigned)src_size > (unsigned)max_size) {
+ if ((unsigned)src_size > (unsigned)max_size) {
// av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n");
// return -1;
src_size = max_size;
@@ -1357,7 +1357,7 @@ static int dca_decode_frame(AVCodecContext * avctx,
s->channel_order_tab[s->prim_channels - 1] < 0)
return -1;
- if(avctx->request_channels == 2 && s->prim_channels > 2) {
+ if (avctx->request_channels == 2 && s->prim_channels > 2) {
channels = 2;
s->output = DCA_STEREO;
avctx->channel_layout = CH_LAYOUT_STEREO;
@@ -1376,7 +1376,7 @@ static int dca_decode_frame(AVCodecContext * avctx,
if (!avctx->channels)
avctx->channels = channels;
- if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
+ 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;
@@ -1421,7 +1421,7 @@ static av_cold int dca_decode_init(AVCodecContext * avctx)
s->samples_chanptr[i] = s->samples + i * 256;
avctx->sample_fmt = SAMPLE_FMT_S16;
- if(s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
+ if (s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
s->add_bias = 385.0f;
s->scale_bias = 1.0 / 32768.0;
} else {