From 04a6d1b0dbfc58b05b64c2caeced6b3b1338865c Mon Sep 17 00:00:00 2001 From: Benjamin Larsson Date: Sun, 13 Sep 2009 18:05:14 +0000 Subject: [PATCH] Cosmetics. Renames, indentation and spacing. Originally committed as revision 19832 to svn://svn.ffmpeg.org/ffmpeg/trunk --- libavcodec/atrac1.c | 48 ++++++++++++++++++++++++++++-------------------- 1 file changed, 28 insertions(+), 20 deletions(-) diff --git a/libavcodec/atrac1.c b/libavcodec/atrac1.c index 6d6abe6..5a68fbf 100644 --- a/libavcodec/atrac1.c +++ b/libavcodec/atrac1.c @@ -94,7 +94,8 @@ static const uint16_t samples_per_band[3] = {128, 128, 256}; static const uint8_t mdct_long_nbits[3] = {7, 7, 8}; -static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, int rev_spec) +static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, + int rev_spec) { MDCTContext* mdct_context; int transf_size = 1 << nbits; @@ -104,9 +105,9 @@ static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, int rev_spe if (rev_spec) { int i; for (i=0 ; idsp.vector_fmul_window(&su->spectrum[0][ref_pos+start_pos], - &su->spectrum[0][ref_pos+start_pos],q->short_buf,short_window, 0, 16); + &su->spectrum[0][ref_pos+start_pos], + q->short_buf,short_window, 0, 16); start_pos += 32; // use hardcoded block_size pos += 32; } @@ -166,8 +168,11 @@ static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q) return 0; } +/** + * Parse the block size mode byte + */ -static int at1_parse_block_size_mode(GetBitContext* gb, int log2_block_count[AT1_QMF_BANDS]) +static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS]) { int log2_block_count_tmp, i; @@ -176,21 +181,22 @@ static int at1_parse_block_size_mode(GetBitContext* gb, int log2_block_count[AT1 log2_block_count_tmp = get_bits(gb, 2); if (log2_block_count_tmp & 1) return -1; - log2_block_count[i] = 2 - log2_block_count_tmp; + log2_block_cnt[i] = 2 - log2_block_count_tmp; } /* high band */ log2_block_count_tmp = get_bits(gb, 2); if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3) return -1; - log2_block_count[IDX_HIGH_BAND] = 3 - log2_block_count_tmp; + log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp; skip_bits(gb, 2); return 0; } -static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, float spec[AT1_SU_SAMPLES]) +static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, + float spec[AT1_SU_SAMPLES]) { int bits_used, band_num, bfu_num, i; @@ -234,7 +240,7 @@ static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, float spec[AT1_SU pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num]; if (word_len) { - float max_quant = 1.0/(float)((1 << (word_len - 1)) - 1); + float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1); for (i=0 ; ibands[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); } -static int atrac1_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - AVPacket *avpkt) +static int atrac1_decode_frame(AVCodecContext *avctx, void *data, + int *data_size, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; - int buf_size = avpkt->size; - AT1Ctx *q = avctx->priv_data; + int buf_size = avpkt->size; + AT1Ctx *q = avctx->priv_data; int ch, ret, i; GetBitContext gb; float* samples = data; @@ -292,7 +297,7 @@ static int atrac1_decode_frame(AVCodecContext *avctx, init_get_bits(&gb, &buf[212*ch], 212*8); /* parse block_size_mode, 1st byte */ - ret = at1_parse_block_size_mode(&gb, su->log2_block_count); + ret = at1_parse_bsm(&gb, su->log2_block_count); if (ret < 0) return ret; @@ -309,12 +314,15 @@ static int atrac1_decode_frame(AVCodecContext *avctx, /* round, convert to 16bit and interleave */ if (q->channels == 1) { /* mono */ - q->dsp.vector_clipf(samples, q->out_samples[0], -32700./(1<<15), 32700./(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./(1<<15), 32700./(1<<15)); - samples[i*2+1] = av_clipf(q->out_samples[1][i], -32700./(1<<15), 32700./(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)); } } -- 2.7.4