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https://github.com/librempeg/librempeg
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Merge remote-tracking branch 'qatar/master'
* qatar/master: mpegaudioenc: Move some static tables to MpegAudioContext Conflicts: libavcodec/mpegaudioenc.c libavcodec/mpegaudiotab.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
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commit
43bf4297e4
@ -64,6 +64,16 @@ typedef struct MpegAudioContext {
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unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
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int sblimit; /* number of used subbands */
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const unsigned char *alloc_table;
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int16_t filter_bank[512];
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int scale_factor_table[64];
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unsigned char scale_diff_table[128];
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#ifdef USE_FLOATS
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float scale_factor_inv_table[64];
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#else
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int8_t scale_factor_shift[64];
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unsigned short scale_factor_mult[64];
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#endif
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unsigned short total_quant_bits[17]; /* total number of bits per allocation group */
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} MpegAudioContext;
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static av_cold int MPA_encode_init(AVCodecContext *avctx)
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@ -139,24 +149,24 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx)
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#if WFRAC_BITS != 16
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v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
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#endif
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filter_bank[i] = v;
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s->filter_bank[i] = v;
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if ((i & 63) != 0)
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v = -v;
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if (i != 0)
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filter_bank[512 - i] = v;
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s->filter_bank[512 - i] = v;
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}
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for(i=0;i<64;i++) {
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v = (int)(exp2((3 - i) / 3.0) * (1 << 20));
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if (v <= 0)
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v = 1;
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scale_factor_table[i] = v;
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s->scale_factor_table[i] = v;
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#ifdef USE_FLOATS
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scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20);
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s->scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20);
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#else
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#define P 15
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scale_factor_shift[i] = 21 - P - (i / 3);
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scale_factor_mult[i] = (1 << P) * exp2((i % 3) / 3.0);
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s->scale_factor_shift[i] = 21 - P - (i / 3);
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s->scale_factor_mult[i] = (1 << P) * exp2((i % 3) / 3.0);
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#endif
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}
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for(i=0;i<128;i++) {
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@ -171,7 +181,7 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx)
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v = 3;
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else
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v = 4;
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scale_diff_table[i] = v;
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s->scale_diff_table[i] = v;
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}
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for(i=0;i<17;i++) {
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@ -180,7 +190,7 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx)
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v = -v;
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else
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v = v * 3;
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total_quant_bits[i] = 12 * v;
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s->total_quant_bits[i] = 12 * v;
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}
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return 0;
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@ -327,7 +337,7 @@ static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
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/* filter */
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p = s->samples_buf[ch] + offset;
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q = filter_bank;
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q = s->filter_bank;
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/* maxsum = 23169 */
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for(i=0;i<64;i++) {
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sum = p[0*64] * q[0*64];
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@ -361,7 +371,8 @@ static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
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s->samples_offset[ch] = offset;
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}
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static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
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static void compute_scale_factors(MpegAudioContext *s,
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unsigned char scale_code[SBLIMIT],
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unsigned char scale_factors[SBLIMIT][3],
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int sb_samples[3][12][SBLIMIT],
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int sblimit)
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@ -388,7 +399,7 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
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use at most 2 compares to find the index */
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index = (21 - n) * 3 - 3;
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if (index >= 0) {
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while (vmax <= scale_factor_table[index+1])
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while (vmax <= s->scale_factor_table[index+1])
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index++;
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} else {
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index = 0; /* very unlikely case of overflow */
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@ -398,7 +409,7 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
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}
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av_dlog(NULL, "%2d:%d in=%x %x %d\n",
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j, i, vmax, scale_factor_table[index], index);
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j, i, vmax, s->scale_factor_table[index], index);
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/* store the scale factor */
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av_assert2(index >=0 && index <= 63);
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sf[i] = index;
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@ -406,8 +417,8 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
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/* compute the transmission factor : look if the scale factors
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are close enough to each other */
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d1 = scale_diff_table[sf[0] - sf[1] + 64];
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d2 = scale_diff_table[sf[1] - sf[2] + 64];
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d1 = s->scale_diff_table[sf[0] - sf[1] + 64];
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d2 = s->scale_diff_table[sf[1] - sf[2] + 64];
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/* handle the 25 cases */
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switch(d1 * 5 + d2) {
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@ -557,12 +568,12 @@ static void compute_bit_allocation(MpegAudioContext *s,
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if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
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/* nothing was coded for this band: add the necessary bits */
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incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
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incr += total_quant_bits[alloc[1]];
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incr += s->total_quant_bits[alloc[1]];
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} else {
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/* increments bit allocation */
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b = bit_alloc[max_ch][max_sb];
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incr = total_quant_bits[alloc[b + 1]] -
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total_quant_bits[alloc[b]];
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incr = s->total_quant_bits[alloc[b + 1]] -
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s->total_quant_bits[alloc[b]];
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}
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if (current_frame_size + incr <= max_frame_size) {
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@ -676,15 +687,15 @@ static void encode_frame(MpegAudioContext *s,
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#ifdef USE_FLOATS
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{
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float a;
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a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
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a = (float)sample * s->scale_factor_inv_table[s->scale_factors[ch][i][k]];
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q[m] = (int)((a + 1.0) * steps * 0.5);
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}
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#else
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{
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int q1, e, shift, mult;
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e = s->scale_factors[ch][i][k];
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shift = scale_factor_shift[e];
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mult = scale_factor_mult[e];
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shift = s->scale_factor_shift[e];
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mult = s->scale_factor_mult[e];
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/* normalize to P bits */
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if (shift < 0)
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@ -739,7 +750,7 @@ static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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}
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for(i=0;i<s->nb_channels;i++) {
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compute_scale_factors(s->scale_code[i], s->scale_factors[i],
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compute_scale_factors(s, s->scale_code[i], s->scale_factors[i],
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s->sb_samples[i], s->sblimit);
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}
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for(i=0;i<s->nb_channels;i++) {
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@ -79,20 +79,6 @@ static const int bitinv32[32] = {
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};
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static int16_t filter_bank[512];
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static int scale_factor_table[64];
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#ifdef USE_FLOATS
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static float scale_factor_inv_table[64];
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#else
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static int8_t scale_factor_shift[64];
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static unsigned short scale_factor_mult[64];
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#endif
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static unsigned char scale_diff_table[128];
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/* total number of bits per allocation group */
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static unsigned short total_quant_bits[17];
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/* signal to noise ratio of each quantification step (could be
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computed from quant_steps[]). The values are dB multiplied by 10
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*/
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