annotate mpegaudiodecheader.c @ 10874:bcfe2acbf190 libavcodec

AAC: Compress codebook tables and optimise sign bit handling The codebooks each consist of small number of values repeated in groups of 2 or 4. Storing the codebooks as a packed list of 2- or 4-bit indexes into a table reduces their size substantially (from 7.5k to 1.5k), resulting in less cache pressure. For the band types with sign bits in the bitstream, storing the number and position of non-zero codebook values using a few bits avoids multiple get_bits() calls and floating-point comparisons which gcc handles miserably. Some float/int type punning also avoids gcc brain damage. Overall speedup 20-35% on Cortex-A8, 20% on Core i7.

/*
 * MPEG Audio header decoder
 * Copyright (c) 2001, 2002 Fabrice Bellard
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

/**
 * @file libavcodec/mpegaudiodecheader.c
 * MPEG Audio header decoder.
 */

//#define DEBUG
#include "avcodec.h"
#include "mpegaudio.h"
#include "mpegaudiodata.h"


int ff_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header)
{
    int sample_rate, frame_size, mpeg25, padding;
    int sample_rate_index, bitrate_index;
    if (header & (1<<20)) {
        s->lsf = (header & (1<<19)) ? 0 : 1;
        mpeg25 = 0;
    } else {
        s->lsf = 1;
        mpeg25 = 1;
    }

    s->layer = 4 - ((header >> 17) & 3);
    /* extract frequency */
    sample_rate_index = (header >> 10) & 3;
    sample_rate = ff_mpa_freq_tab[sample_rate_index] >> (s->lsf + mpeg25);
    sample_rate_index += 3 * (s->lsf + mpeg25);
    s->sample_rate_index = sample_rate_index;
    s->error_protection = ((header >> 16) & 1) ^ 1;
    s->sample_rate = sample_rate;

    bitrate_index = (header >> 12) & 0xf;
    padding = (header >> 9) & 1;
    //extension = (header >> 8) & 1;
    s->mode = (header >> 6) & 3;
    s->mode_ext = (header >> 4) & 3;
    //copyright = (header >> 3) & 1;
    //original = (header >> 2) & 1;
    //emphasis = header & 3;

    if (s->mode == MPA_MONO)
        s->nb_channels = 1;
    else
        s->nb_channels = 2;

    if (bitrate_index != 0) {
        frame_size = ff_mpa_bitrate_tab[s->lsf][s->layer - 1][bitrate_index];
        s->bit_rate = frame_size * 1000;
        switch(s->layer) {
        case 1:
            frame_size = (frame_size * 12000) / sample_rate;
            frame_size = (frame_size + padding) * 4;
            break;
        case 2:
            frame_size = (frame_size * 144000) / sample_rate;
            frame_size += padding;
            break;
        default:
        case 3:
            frame_size = (frame_size * 144000) / (sample_rate << s->lsf);
            frame_size += padding;
            break;
        }
        s->frame_size = frame_size;
    } else {
        /* if no frame size computed, signal it */
        return 1;
    }

#if defined(DEBUG)
    dprintf(NULL, "layer%d, %d Hz, %d kbits/s, ",
           s->layer, s->sample_rate, s->bit_rate);
    if (s->nb_channels == 2) {
        if (s->layer == 3) {
            if (s->mode_ext & MODE_EXT_MS_STEREO)
                dprintf(NULL, "ms-");
            if (s->mode_ext & MODE_EXT_I_STEREO)
                dprintf(NULL, "i-");
        }
        dprintf(NULL, "stereo");
    } else {
        dprintf(NULL, "mono");
    }
    dprintf(NULL, "\n");
#endif
    return 0;
}