diff options
author | Nick Brereton <nick@nbrereton.net> | 2010-06-22 08:35:44 +0000 |
---|---|---|
committer | Martin Storsjö <martin@martin.st> | 2010-06-22 08:35:44 +0000 |
commit | d1177cb589621016f681789dd66873832d5fb14a (patch) | |
tree | 42f71ac8df9996b60c9197860e8e5c23a57e86cf | |
parent | 774e9acfa748b929e1f288fca3f5b7f2d6746a61 (diff) | |
download | ffmpeg-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.c | 46 |
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 { |