Cosmetics. Renames, indentation and spacing.
Originally committed as revision 19832 to svn://svn.ffmpeg.org/ffmpeg/trunk
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@ -94,7 +94,8 @@ static const uint16_t samples_per_band[3] = {128, 128, 256};
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static const uint8_t mdct_long_nbits[3] = {7, 7, 8};
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static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, int rev_spec)
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static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
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int rev_spec)
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{
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MDCTContext* mdct_context;
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int transf_size = 1 << nbits;
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@ -104,9 +105,9 @@ static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, int rev_spe
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if (rev_spec) {
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int i;
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for (i=0 ; i<transf_size/2 ; i++)
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FFSWAP(float, spec[i], spec[transf_size-1-i]);
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FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
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}
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ff_imdct_half(mdct_context,out,spec);
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ff_imdct_half(mdct_context, out, spec);
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}
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@ -147,7 +148,8 @@ static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
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/* overlap and window between short blocks */
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q->dsp.vector_fmul_window(&su->spectrum[0][ref_pos+start_pos],
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&su->spectrum[0][ref_pos+start_pos],q->short_buf,short_window, 0, 16);
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&su->spectrum[0][ref_pos+start_pos],
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q->short_buf,short_window, 0, 16);
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start_pos += 32; // use hardcoded block_size
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pos += 32;
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}
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@ -166,8 +168,11 @@ static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
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return 0;
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}
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/**
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* Parse the block size mode byte
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*/
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static int at1_parse_block_size_mode(GetBitContext* gb, int log2_block_count[AT1_QMF_BANDS])
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static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
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{
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int log2_block_count_tmp, i;
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@ -176,21 +181,22 @@ static int at1_parse_block_size_mode(GetBitContext* gb, int log2_block_count[AT1
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log2_block_count_tmp = get_bits(gb, 2);
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if (log2_block_count_tmp & 1)
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return -1;
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log2_block_count[i] = 2 - log2_block_count_tmp;
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log2_block_cnt[i] = 2 - log2_block_count_tmp;
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}
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/* high band */
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log2_block_count_tmp = get_bits(gb, 2);
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if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
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return -1;
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log2_block_count[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
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log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
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skip_bits(gb, 2);
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return 0;
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}
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static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, float spec[AT1_SU_SAMPLES])
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static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
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float spec[AT1_SU_SAMPLES])
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{
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int bits_used, band_num, bfu_num, i;
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@ -234,7 +240,7 @@ static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, float spec[AT1_SU
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pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
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if (word_len) {
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float max_quant = 1.0/(float)((1 << (word_len - 1)) - 1);
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float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
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for (i=0 ; i<num_specs ; i++) {
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/* read in a quantized spec and convert it to
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@ -261,21 +267,20 @@ void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
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atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
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/* delay the signal of the high band by 23 samples */
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memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float)*23);
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memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float)*256);
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memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float)*23);
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memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float)*256);
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/* combine (low + middle) and high bands */
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atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
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}
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static int atrac1_decode_frame(AVCodecContext *avctx,
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void *data, int *data_size,
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AVPacket *avpkt)
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static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
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int *data_size, AVPacket *avpkt)
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{
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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AT1Ctx *q = avctx->priv_data;
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int buf_size = avpkt->size;
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AT1Ctx *q = avctx->priv_data;
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int ch, ret, i;
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GetBitContext gb;
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float* samples = data;
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@ -292,7 +297,7 @@ static int atrac1_decode_frame(AVCodecContext *avctx,
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init_get_bits(&gb, &buf[212*ch], 212*8);
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/* parse block_size_mode, 1st byte */
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ret = at1_parse_block_size_mode(&gb, su->log2_block_count);
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ret = at1_parse_bsm(&gb, su->log2_block_count);
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if (ret < 0)
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return ret;
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@ -309,12 +314,15 @@ static int atrac1_decode_frame(AVCodecContext *avctx,
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/* round, convert to 16bit and interleave */
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if (q->channels == 1) {
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/* mono */
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q->dsp.vector_clipf(samples, q->out_samples[0], -32700./(1<<15), 32700./(1<<15), AT1_SU_SAMPLES);
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q->dsp.vector_clipf(samples, q->out_samples[0], -32700.0 / (1<<15),
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32700.0 / (1<<15), AT1_SU_SAMPLES);
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} else {
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/* stereo */
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for (i = 0; i < AT1_SU_SAMPLES; i++) {
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samples[i*2] = av_clipf(q->out_samples[0][i], -32700./(1<<15), 32700./(1<<15));
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samples[i*2+1] = av_clipf(q->out_samples[1][i], -32700./(1<<15), 32700./(1<<15));
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samples[i*2] = av_clipf(q->out_samples[0][i], -32700.0 / (1<<15),
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32700.0 / (1<<15));
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samples[i*2+1] = av_clipf(q->out_samples[1][i], -32700.0 / (1<<15),
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32700.0 / (1<<15));
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}
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}
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