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K&R coding style whitespace cosmetics 

Originally committed as revision 19929 to svn://svn.ffmpeg.org/ffmpeg/trunk
tags/v0.6
Diego Biurrun 16 years ago
parent
7a38c1149c
commit
1e1898c00f
1 changed files with 48 additions and 47 deletions
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  1. +48
    -47
      libavcodec/atrac1.c

+ 48
- 47
libavcodec/atrac1.c View File

@@ -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 *spec, float *out, int nbits,
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 *spec, float *out, int nbits,

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, AT1Ctx *q)
/* 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, AT1Ctx *q)
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* gb, int log2_block_cnt[AT1_QMF_BANDS])
{
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(GetBitContext* gb, AT1SUCtx* su,
(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(GetBitContext* gb, AT1SUCtx* su,
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(GetBitContext* gb, AT1SUCtx* su,
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(GetBitContext* gb, AT1SUCtx* su,

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(AVCodecContext *avctx, void *data,
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(AVCodecContext *avctx, void *data,
/* 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(AVCodecContext *avctx)
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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