From 115329f16062074e11ccf3b89ead6176606c9696 Mon Sep 17 00:00:00 2001 From: Diego Biurrun Date: Sat, 17 Dec 2005 18:14:38 +0000 Subject: COSMETICS: Remove all trailing whitespace. Originally committed as revision 4749 to svn://svn.ffmpeg.org/ffmpeg/trunk --- libavcodec/mpegaudio.c | 106 ++++++++++++++++++++++++------------------------- 1 file changed, 53 insertions(+), 53 deletions(-) (limited to 'libavcodec/mpegaudio.c') diff --git a/libavcodec/mpegaudio.c b/libavcodec/mpegaudio.c index 7a0b0a31ce..c673ebc67c 100644 --- a/libavcodec/mpegaudio.c +++ b/libavcodec/mpegaudio.c @@ -16,12 +16,12 @@ * License along with this library; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ - + /** * @file mpegaudio.c * The simplest mpeg audio layer 2 encoder. */ - + #include "avcodec.h" #include "bitstream.h" #include "mpegaudio.h" @@ -49,7 +49,7 @@ typedef struct MpegAudioContext { int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT]; unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */ /* code to group 3 scale factors */ - unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT]; + unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT]; int sblimit; /* number of used subbands */ const unsigned char *alloc_table; } MpegAudioContext; @@ -79,7 +79,7 @@ static int MPA_encode_init(AVCodecContext *avctx) /* encoding freq */ s->lsf = 0; for(i=0;i<3;i++) { - if (mpa_freq_tab[i] == freq) + if (mpa_freq_tab[i] == freq) break; if ((mpa_freq_tab[i] / 2) == freq) { s->lsf = 1; @@ -94,7 +94,7 @@ static int MPA_encode_init(AVCodecContext *avctx) /* encoding bitrate & frequency */ for(i=0;i<15;i++) { - if (mpa_bitrate_tab[s->lsf][1][i] == bitrate) + if (mpa_bitrate_tab[s->lsf][1][i] == bitrate) break; } if (i == 15){ @@ -104,14 +104,14 @@ static int MPA_encode_init(AVCodecContext *avctx) s->bitrate_index = i; /* compute total header size & pad bit */ - + a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0); s->frame_size = ((int)a) * 8; /* frame fractional size to compute padding */ s->frame_frac = 0; s->frame_frac_incr = (int)((a - floor(a)) * 65536.0); - + /* select the right allocation table */ table = l2_select_table(bitrate, s->nb_channels, freq, s->lsf); @@ -120,7 +120,7 @@ static int MPA_encode_init(AVCodecContext *avctx) s->alloc_table = alloc_tables[table]; #ifdef DEBUG - av_log(avctx, AV_LOG_DEBUG, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n", + av_log(avctx, AV_LOG_DEBUG, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n", bitrate, freq, s->frame_size, table, s->frame_frac_incr); #endif @@ -163,14 +163,14 @@ static int MPA_encode_init(AVCodecContext *avctx) v = 2; else if (v < 3) v = 3; - else + else v = 4; scale_diff_table[i] = v; } for(i=0;i<17;i++) { v = quant_bits[i]; - if (v < 0) + if (v < 0) v = -v; else v = v * 3; @@ -191,7 +191,7 @@ static void idct32(int *out, int *tab) const int *xp = costab32; for(j=31;j>=3;j-=2) tab[j] += tab[j - 2]; - + t = tab + 30; t1 = tab + 2; do { @@ -209,30 +209,30 @@ static void idct32(int *out, int *tab) t[3] += t[3-8]; t -= 8; } while (t != t1); - + t = tab; t1 = tab + 32; do { - t[ 3] = -t[ 3]; - t[ 6] = -t[ 6]; - - t[11] = -t[11]; - t[12] = -t[12]; - t[13] = -t[13]; - t[15] = -t[15]; + t[ 3] = -t[ 3]; + t[ 6] = -t[ 6]; + + t[11] = -t[11]; + t[12] = -t[12]; + t[13] = -t[13]; + t[15] = -t[15]; t += 16; } while (t != t1); - + t = tab; t1 = tab + 8; do { int x1, x2, x3, x4; - + x3 = MUL(t[16], FIX(SQRT2*0.5)); x4 = t[0] - x3; x3 = t[0] + x3; - + x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5)); x1 = MUL((t[8] - x2), xp[0]); x2 = MUL((t[8] + x2), xp[1]); @@ -255,11 +255,11 @@ static void idct32(int *out, int *tab) xr = MUL(t[4],xp[1]); t[ 4] = (t[24] - xr); t[24] = (t[24] + xr); - + xr = MUL(t[20],xp[2]); t[20] = (t[8] - xr); t[ 8] = (t[8] + xr); - + xr = MUL(t[12],xp[3]); t[12] = (t[16] - xr); t[16] = (t[16] + xr); @@ -271,19 +271,19 @@ static void idct32(int *out, int *tab) xr = MUL(tab[30-i*4],xp[0]); tab[30-i*4] = (tab[i*4] - xr); tab[ i*4] = (tab[i*4] + xr); - + xr = MUL(tab[ 2+i*4],xp[1]); tab[ 2+i*4] = (tab[28-i*4] - xr); tab[28-i*4] = (tab[28-i*4] + xr); - + xr = MUL(tab[31-i*4],xp[0]); tab[31-i*4] = (tab[1+i*4] - xr); tab[ 1+i*4] = (tab[1+i*4] + xr); - + xr = MUL(tab[ 3+i*4],xp[1]); tab[ 3+i*4] = (tab[29-i*4] - xr); tab[29-i*4] = (tab[29-i*4] + xr); - + xp += 2; } @@ -352,7 +352,7 @@ static void filter(MpegAudioContext *s, int ch, short *samples, int incr) out += 32; /* handle the wrap around */ if (offset < 0) { - memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32), + memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32), s->samples_buf[ch], (512 - 32) * 2); offset = SAMPLES_BUF_SIZE - 512; } @@ -363,14 +363,14 @@ static void filter(MpegAudioContext *s, int ch, short *samples, int incr) } static void compute_scale_factors(unsigned char scale_code[SBLIMIT], - unsigned char scale_factors[SBLIMIT][3], + unsigned char scale_factors[SBLIMIT][3], int sb_samples[3][12][SBLIMIT], int sblimit) { int *p, vmax, v, n, i, j, k, code; int index, d1, d2; unsigned char *sf = &scale_factors[0][0]; - + for(j=0;j 0) { n = av_log2(vmax); - /* n is the position of the MSB of vmax. now + /* n is the position of the MSB of vmax. now use at most 2 compares to find the index */ index = (21 - n) * 3 - 3; if (index >= 0) { @@ -399,7 +399,7 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT], } #if 0 - printf("%2d:%d in=%x %x %d\n", + printf("%2d:%d in=%x %x %d\n", j, i, vmax, scale_factor_table[index], index); #endif /* store the scale factor */ @@ -411,7 +411,7 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT], are close enough to each other */ d1 = scale_diff_table[sf[0] - sf[1] + 64]; d2 = scale_diff_table[sf[1] - sf[2] + 64]; - + /* handle the 25 cases */ switch(d1 * 5 + d2) { case 0*5+0: @@ -468,9 +468,9 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT], assert(0); //cant happen code = 0; /* kill warning */ } - + #if 0 - printf("%d: %2d %2d %2d %d %d -> %d\n", j, + printf("%d: %2d %2d %2d %d %d -> %d\n", j, sf[0], sf[1], sf[2], d1, d2, code); #endif scale_code[j] = code; @@ -498,7 +498,7 @@ static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT]) /* Try to maximize the smr while using a number of bits inferior to the frame size. I tried to make the code simpler, faster and smaller than other encoders :-) */ -static void compute_bit_allocation(MpegAudioContext *s, +static void compute_bit_allocation(MpegAudioContext *s, short smr1[MPA_MAX_CHANNELS][SBLIMIT], unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], int *padding) @@ -512,7 +512,7 @@ static void compute_bit_allocation(MpegAudioContext *s, memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT); memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT); memset(bit_alloc, 0, s->nb_channels * SBLIMIT); - + /* compute frame size and padding */ max_frame_size = s->frame_size; s->frame_frac += s->frame_frac_incr; @@ -547,13 +547,13 @@ static void compute_bit_allocation(MpegAudioContext *s, } } #if 0 - printf("current=%d max=%d max_sb=%d alloc=%d\n", + printf("current=%d max=%d max_sb=%d alloc=%d\n", current_frame_size, max_frame_size, max_sb, bit_alloc[max_sb]); -#endif +#endif if (max_sb < 0) break; - + /* find alloc table entry (XXX: not optimal, should use pointer table) */ alloc = s->alloc_table; @@ -568,7 +568,7 @@ static void compute_bit_allocation(MpegAudioContext *s, } else { /* increments bit allocation */ b = bit_alloc[max_ch][max_sb]; - incr = total_quant_bits[alloc[b + 1]] - + incr = total_quant_bits[alloc[b + 1]] - total_quant_bits[alloc[b]]; } @@ -637,11 +637,11 @@ static void encode_frame(MpegAudioContext *s, } j += 1 << bit_alloc_bits; } - + /* scale codes */ for(i=0;isblimit;i++) { for(ch=0;chnb_channels;ch++) { - if (bit_alloc[ch][i]) + if (bit_alloc[ch][i]) put_bits(p, 2, s->scale_code[ch][i]); } } @@ -669,7 +669,7 @@ static void encode_frame(MpegAudioContext *s, } } } - + /* quantization & write sub band samples */ for(k=0;k<3;k++) { @@ -699,7 +699,7 @@ static void encode_frame(MpegAudioContext *s, e = s->scale_factors[ch][i][k]; shift = scale_factor_shift[e]; mult = scale_factor_mult[e]; - + /* normalize to P bits */ if (shift < 0) q1 = sample << (-shift); @@ -716,17 +716,17 @@ static void encode_frame(MpegAudioContext *s, bits = quant_bits[qindex]; if (bits < 0) { /* group the 3 values to save bits */ - put_bits(p, -bits, + put_bits(p, -bits, q[0] + steps * (q[1] + steps * q[2])); #if 0 - printf("%d: gr1 %d\n", + printf("%d: gr1 %d\n", i, q[0] + steps * (q[1] + steps * q[2])); #endif } else { #if 0 - printf("%d: gr3 %d %d %d\n", + printf("%d: gr3 %d %d %d\n", i, q[0], q[1], q[2]); -#endif +#endif put_bits(p, bits, q[0]); put_bits(p, bits, q[1]); put_bits(p, bits, q[2]); @@ -734,7 +734,7 @@ static void encode_frame(MpegAudioContext *s, } } /* next subband in alloc table */ - j += 1 << bit_alloc_bits; + j += 1 << bit_alloc_bits; } } } @@ -761,7 +761,7 @@ static int MPA_encode_frame(AVCodecContext *avctx, } for(i=0;inb_channels;i++) { - compute_scale_factors(s->scale_code[i], s->scale_factors[i], + compute_scale_factors(s->scale_code[i], s->scale_factors[i], s->sb_samples[i], s->sblimit); } for(i=0;inb_channels;i++) { @@ -772,7 +772,7 @@ static int MPA_encode_frame(AVCodecContext *avctx, init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE); encode_frame(s, bit_alloc, padding); - + s->nb_samples += MPA_FRAME_SIZE; return pbBufPtr(&s->pb) - s->pb.buf; } -- cgit v1.2.3