[FFmpeg-cvslog] r19929 - trunk/libavcodec/atrac1.c
diego
subversion
Sun Sep 20 19:16:12 CEST 2009
Author: diego
Date: Sun Sep 20 19:16:12 2009
New Revision: 19929
Log:
K&R coding style whitespace cosmetics
Modified:
trunk/libavcodec/atrac1.c
Modified: trunk/libavcodec/atrac1.c
==============================================================================
--- trunk/libavcodec/atrac1.c Sun Sep 20 16:09:27 2009 (r19928)
+++ trunk/libavcodec/atrac1.c Sun Sep 20 19:16:12 2009 (r19929)
@@ -60,11 +60,11 @@ typedef struct {
int idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU
int idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU
float* spectrum[2];
- DECLARE_ALIGNED_16(float,spec1[AT1_SU_SAMPLES]); ///< mdct buffer
- DECLARE_ALIGNED_16(float,spec2[AT1_SU_SAMPLES]); ///< mdct buffer
- DECLARE_ALIGNED_16(float,fst_qmf_delay[46]); ///< delay line for the 1st stacked QMF filter
- DECLARE_ALIGNED_16(float,snd_qmf_delay[46]); ///< delay line for the 2nd stacked QMF filter
- DECLARE_ALIGNED_16(float,last_qmf_delay[256+23]); ///< delay line for the last stacked QMF filter
+ DECLARE_ALIGNED_16(float, spec1[AT1_SU_SAMPLES]); ///< mdct buffer
+ DECLARE_ALIGNED_16(float, spec2[AT1_SU_SAMPLES]); ///< mdct buffer
+ DECLARE_ALIGNED_16(float, fst_qmf_delay[46]); ///< delay line for the 1st stacked QMF filter
+ DECLARE_ALIGNED_16(float, snd_qmf_delay[46]); ///< delay line for the 2nd stacked QMF filter
+ DECLARE_ALIGNED_16(float, last_qmf_delay[256+23]); ///< delay line for the last stacked QMF filter
} AT1SUCtx;
/**
@@ -72,13 +72,13 @@ typedef struct {
*/
typedef struct {
AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit
- DECLARE_ALIGNED_16(float,spec[AT1_SU_SAMPLES]); ///< the mdct spectrum buffer
+ DECLARE_ALIGNED_16(float, spec[AT1_SU_SAMPLES]); ///< the mdct spectrum buffer
- DECLARE_ALIGNED_16(float, low[256]);
- DECLARE_ALIGNED_16(float, mid[256]);
- DECLARE_ALIGNED_16(float,high[512]);
+ DECLARE_ALIGNED_16(float, low[256]);
+ DECLARE_ALIGNED_16(float, mid[256]);
+ DECLARE_ALIGNED_16(float, high[512]);
float* bands[3];
- DECLARE_ALIGNED_16(float,out_samples[AT1_MAX_CHANNELS][AT1_SU_SAMPLES]);
+ DECLARE_ALIGNED_16(float, out_samples[AT1_MAX_CHANNELS][AT1_SU_SAMPLES]);
MDCTContext mdct_ctx[3];
int channels;
DSPContext dsp;
@@ -97,11 +97,11 @@ static void at1_imdct(AT1Ctx *q, float *
MDCTContext* mdct_context;
int transf_size = 1 << nbits;
- mdct_context = &q->mdct_ctx[nbits - 5 - (nbits>6)];
+ mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)];
if (rev_spec) {
int i;
- for (i=0 ; i<transf_size/2 ; i++)
+ for (i = 0; i < transf_size / 2; i++)
FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
}
ff_imdct_half(mdct_context, out, spec);
@@ -110,10 +110,10 @@ static void at1_imdct(AT1Ctx *q, float *
static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
{
- int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
- unsigned int start_pos, ref_pos=0, pos = 0;
+ int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
+ unsigned int start_pos, ref_pos = 0 pos = 0;
- for (band_num=0 ; band_num<AT1_QMF_BANDS ; band_num++) {
+ for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
band_samples = samples_per_band[band_num];
log2_block_count = su->log2_block_count[band_num];
@@ -128,7 +128,7 @@ static int at1_imdct_block(AT1SUCtx* su,
/* calc transform size in bits according to the block_size_mode */
nbits = mdct_long_nbits[band_num] - log2_block_count;
- if (nbits!=5 && nbits!=7 && nbits!=8)
+ if (nbits != 5 && nbits != 7 && nbits != 8)
return -1;
if (num_blocks == 1) {
@@ -137,23 +137,22 @@ static int at1_imdct_block(AT1SUCtx* su,
pos += block_size; // move to the next mdct block in the spectrum
/* overlap and window long blocks */
- q->dsp.vector_fmul_window(q->bands[band_num], &su->spectrum[1][ref_pos+band_samples-16],
- &su->spectrum[0][ref_pos], short_window, 0, 16);
- memcpy(q->bands[band_num]+32, &su->spectrum[0][ref_pos+16], 240 * sizeof(float));
-
+ q->dsp.vector_fmul_window(q->bands[band_num], &su->spectrum[1][ref_pos + band_samples - 16],
+ &su->spectrum[0][ref_pos], short_window, 0, 16);
+ memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
} else {
/* short blocks */
float *prev_buf;
start_pos = 0;
- prev_buf = &su->spectrum[1][ref_pos+band_samples-16];
- for (; num_blocks!=0 ; num_blocks--) {
- at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos+start_pos], 5, band_num);
+ prev_buf = &su->spectrum[1][ref_pos + band_samples - 16];
+ for (; num_blocks != 0; num_blocks--) {
+ at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], 5, band_num);
/* overlap and window between short blocks */
q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
- &su->spectrum[0][ref_pos+start_pos], short_window, 0, 16);
+ &su->spectrum[0][ref_pos + start_pos], short_window, 0, 16);
- prev_buf = &su->spectrum[0][ref_pos+start_pos+16];
+ prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
start_pos += 32; // use hardcoded block_size
pos += 32;
}
@@ -175,7 +174,7 @@ static int at1_parse_bsm(GetBitContext*
{
int log2_block_count_tmp, i;
- for(i=0 ; i<2 ; i++) {
+ for (i = 0; i < 2; i++) {
/* low and mid band */
log2_block_count_tmp = get_bits(gb, 2);
if (log2_block_count_tmp & 1)
@@ -210,11 +209,11 @@ static int at1_unpack_dequant(GetBitCont
(bfu_amount_tab3[get_bits(gb, 3)] << 1);
/* get word length index (idwl) for each BFU */
- for (i=0 ; i<su->num_bfus ; i++)
+ for (i = 0; i < su->num_bfus; i++)
su->idwls[i] = get_bits(gb, 4);
/* get scalefactor index (idsf) for each BFU */
- for (i=0 ; i<su->num_bfus ; i++)
+ for (i = 0; i < su->num_bfus; i++)
su->idsfs[i] = get_bits(gb, 6);
/* zero idwl/idsf for empty BFUs */
@@ -222,8 +221,8 @@ static int at1_unpack_dequant(GetBitCont
su->idwls[i] = su->idsfs[i] = 0;
/* read in the spectral data and reconstruct MDCT spectrum of this channel */
- for (band_num=0 ; band_num<AT1_QMF_BANDS ; band_num++) {
- for (bfu_num=bfu_bands_t[band_num] ; bfu_num<bfu_bands_t[band_num+1] ; bfu_num++) {
+ for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
+ for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) {
int pos;
int num_specs = specs_per_bfu[bfu_num];
@@ -241,14 +240,14 @@ static int at1_unpack_dequant(GetBitCont
if (word_len) {
float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
- for (i=0 ; i<num_specs ; i++) {
+ for (i = 0; i < num_specs; i++) {
/* read in a quantized spec and convert it to
* signed int and then inverse quantization
*/
spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
}
} else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */
- memset(&spec[pos], 0, num_specs*sizeof(float));
+ memset(&spec[pos], 0, num_specs * sizeof(float));
}
}
}
@@ -259,15 +258,15 @@ static int at1_unpack_dequant(GetBitCont
void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
{
- float temp[256];
- float iqmf_temp[512 + 46];
+ float temp[256];
+ float iqmf_temp[512 + 46];
/* combine low and middle bands */
atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
/* delay the signal of the high band by 23 samples */
- memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float)*23);
- memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float)*256);
+ memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 23);
+ memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float) * 256);
/* combine (low + middle) and high bands */
atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
@@ -290,10 +289,10 @@ static int atrac1_decode_frame(AVCodecCo
return -1;
}
- for (ch=0 ; ch<q->channels ; ch++) {
+ for (ch = 0; ch < q->channels; ch++) {
AT1SUCtx* su = &q->SUs[ch];
- init_get_bits(&gb, &buf[212*ch], 212*8);
+ init_get_bits(&gb, &buf[212 * ch], 212 * 8);
/* parse block_size_mode, 1st byte */
ret = at1_parse_bsm(&gb, su->log2_block_count);
@@ -313,15 +312,17 @@ static int atrac1_decode_frame(AVCodecCo
/* round, convert to 16bit and interleave */
if (q->channels == 1) {
/* mono */
- q->dsp.vector_clipf(samples, q->out_samples[0], -32700.0 / (1<<15),
- 32700.0 / (1<<15), AT1_SU_SAMPLES);
+ q->dsp.vector_clipf(samples, q->out_samples[0], -32700.0 / (1 << 15),
+ 32700.0 / (1 << 15), AT1_SU_SAMPLES);
} else {
/* stereo */
for (i = 0; i < AT1_SU_SAMPLES; i++) {
- samples[i*2] = av_clipf(q->out_samples[0][i], -32700.0 / (1<<15),
- 32700.0 / (1<<15));
- samples[i*2+1] = av_clipf(q->out_samples[1][i], -32700.0 / (1<<15),
- 32700.0 / (1<<15));
+ samples[i * 2] = av_clipf(q->out_samples[0][i],
+ -32700.0 / (1 << 15),
+ 32700.0 / (1 << 15));
+ samples[i * 2 + 1] = av_clipf(q->out_samples[1][i],
+ -32700.0 / (1 << 15),
+ 32700.0 / (1 << 15));
}
}
@@ -339,9 +340,9 @@ static av_cold int atrac1_decode_init(AV
q->channels = avctx->channels;
/* Init the mdct transforms */
- ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1<<15));
- ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1<<15));
- ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1<<15));
+ ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15));
+ ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15));
+ ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15));
ff_sine_window_init(short_window, 32);
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