view aacenc.c @ 8006:c7c1e85d14bc libavcodec

Rename variables to clarify the channel coupling element and corresponding target channel element. Patch by Alex Converse (alex converse gmail com)
author superdump
date Mon, 06 Oct 2008 16:22:11 +0000
parents dc1a7a6ec58d
children e9d9d946f213
line wrap: on
line source

/*
 * AAC encoder
 * Copyright (C) 2008 Konstantin Shishkov
 *
 * 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 aacenc.c
 * AAC encoder
 */

/***********************************
 *              TODOs:
 * psy model selection with some option
 * add sane pulse detection
 * add temporal noise shaping
 ***********************************/

#include "avcodec.h"
#include "bitstream.h"
#include "dsputil.h"
#include "mpeg4audio.h"

#include "aacpsy.h"
#include "aac.h"
#include "aactab.h"

static const uint8_t swb_size_1024_96[] = {
    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
    12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
    64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
};

static const uint8_t swb_size_1024_64[] = {
    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
    12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
    40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
};

static const uint8_t swb_size_1024_48[] = {
    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
    12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
    32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
    96
};

static const uint8_t swb_size_1024_32[] = {
    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
    12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
    32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
};

static const uint8_t swb_size_1024_24[] = {
    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
    12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
    32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
};

static const uint8_t swb_size_1024_16[] = {
    8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
    12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
    32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
};

static const uint8_t swb_size_1024_8[] = {
    12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
    16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
    32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
};

static const uint8_t * const swb_size_1024[] = {
    swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
    swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
    swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
    swb_size_1024_16, swb_size_1024_16, swb_size_1024_8
};

static const uint8_t swb_size_128_96[] = {
    4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
};

static const uint8_t swb_size_128_48[] = {
    4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
};

static const uint8_t swb_size_128_24[] = {
    4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
};

static const uint8_t swb_size_128_16[] = {
    4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
};

static const uint8_t swb_size_128_8[] = {
    4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
};

static const uint8_t * const swb_size_128[] = {
    /* the last entry on the following row is swb_size_128_64 but is a
       duplicate of swb_size_128_96 */
    swb_size_128_96, swb_size_128_96, swb_size_128_96,
    swb_size_128_48, swb_size_128_48, swb_size_128_48,
    swb_size_128_24, swb_size_128_24, swb_size_128_16,
    swb_size_128_16, swb_size_128_16, swb_size_128_8
};

/** bits needed to code codebook run value for long windows */
static const uint8_t run_value_bits_long[64] = {
     5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,
     5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5, 10,
    10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
    10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
};

/** bits needed to code codebook run value for short windows */
static const uint8_t run_value_bits_short[16] = {
    3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
};

static const uint8_t* const run_value_bits[2] = {
    run_value_bits_long, run_value_bits_short
};

/** default channel configurations */
static const uint8_t aac_chan_configs[6][5] = {
 {1, TYPE_SCE},                               // 1 channel  - single channel element
 {1, TYPE_CPE},                               // 2 channels - channel pair
 {2, TYPE_SCE, TYPE_CPE},                     // 3 channels - center + stereo
 {3, TYPE_SCE, TYPE_CPE, TYPE_SCE},           // 4 channels - front center + stereo + back center
 {3, TYPE_SCE, TYPE_CPE, TYPE_CPE},           // 5 channels - front center + stereo + back stereo
 {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
};

/**
 * structure used in optimal codebook search
 */
typedef struct BandCodingPath {
    int prev_idx; ///< pointer to the previous path point
    int codebook; ///< codebook for coding band run
    int bits;     ///< number of bit needed to code given number of bands
} BandCodingPath;

/**
 * AAC encoder context
 */
typedef struct {
    PutBitContext pb;
    MDCTContext mdct1024;                        ///< long (1024 samples) frame transform context
    MDCTContext mdct128;                         ///< short (128 samples) frame transform context
    DSPContext  dsp;
    DECLARE_ALIGNED_16(FFTSample, output[2048]); ///< temporary buffer for MDCT input coefficients
    int16_t* samples;                            ///< saved preprocessed input

    int samplerate_index;                        ///< MPEG-4 samplerate index

    ChannelElement *cpe;                         ///< channel elements
    AACPsyContext psy;                           ///< psychoacoustic model context
    int last_frame;
} AACEncContext;

/**
 * Make AAC audio config object.
 * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
 */
static void put_audio_specific_config(AVCodecContext *avctx)
{
    PutBitContext pb;
    AACEncContext *s = avctx->priv_data;

    init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
    put_bits(&pb, 5, 2); //object type - AAC-LC
    put_bits(&pb, 4, s->samplerate_index); //sample rate index
    put_bits(&pb, 4, avctx->channels);
    //GASpecificConfig
    put_bits(&pb, 1, 0); //frame length - 1024 samples
    put_bits(&pb, 1, 0); //does not depend on core coder
    put_bits(&pb, 1, 0); //is not extension
    flush_put_bits(&pb);
}

static av_cold int aac_encode_init(AVCodecContext *avctx)
{
    AACEncContext *s = avctx->priv_data;
    int i;

    avctx->frame_size = 1024;

    for(i = 0; i < 16; i++)
        if(avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
            break;
    if(i == 16){
        av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
        return -1;
    }
    if(avctx->channels > 6){
        av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
        return -1;
    }
    s->samplerate_index = i;

    dsputil_init(&s->dsp, avctx);
    ff_mdct_init(&s->mdct1024, 11, 0);
    ff_mdct_init(&s->mdct128,   8, 0);
    // window init
    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
    ff_sine_window_init(ff_sine_1024, 1024);
    ff_sine_window_init(ff_sine_128, 128);

    s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
    s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
    if(ff_aac_psy_init(&s->psy, avctx, AAC_PSY_3GPP,
                       aac_chan_configs[avctx->channels-1][0], 0,
                       swb_size_1024[i], ff_aac_num_swb_1024[i], swb_size_128[i], ff_aac_num_swb_128[i]) < 0){
        av_log(avctx, AV_LOG_ERROR, "Cannot initialize selected model.\n");
        return -1;
    }
    avctx->extradata = av_malloc(2);
    avctx->extradata_size = 2;
    put_audio_specific_config(avctx);
    return 0;
}

/**
 * Encode ics_info element.
 * @see Table 4.6 (syntax of ics_info)
 */
static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
{
    int i;

    put_bits(&s->pb, 1, 0);                // ics_reserved bit
    put_bits(&s->pb, 2, info->window_sequence[0]);
    put_bits(&s->pb, 1, info->use_kb_window[0]);
    if(info->window_sequence[0] != EIGHT_SHORT_SEQUENCE){
        put_bits(&s->pb, 6, info->max_sfb);
        put_bits(&s->pb, 1, 0);            // no prediction
    }else{
        put_bits(&s->pb, 4, info->max_sfb);
        for(i = 1; i < info->num_windows; i++)
            put_bits(&s->pb, 1, info->group_len[i]);
    }
}

/**
 * Calculate the number of bits needed to code all coefficient signs in current band.
 */
static int calculate_band_sign_bits(AACEncContext *s, SingleChannelElement *sce,
                                    int group_len, int start, int size)
{
    int bits = 0;
    int i, w;
    for(w = 0; w < group_len; w++){
        for(i = 0; i < size; i++){
            if(sce->icoefs[start + i])
                bits++;
        }
        start += 128;
    }
    return bits;
}

/**
 * Encode pulse data.
 */
static void encode_pulses(AACEncContext *s, Pulse *pulse)
{
    int i;

    put_bits(&s->pb, 1, !!pulse->num_pulse);
    if(!pulse->num_pulse) return;

    put_bits(&s->pb, 2, pulse->num_pulse - 1);
    put_bits(&s->pb, 6, pulse->start);
    for(i = 0; i < pulse->num_pulse; i++){
        put_bits(&s->pb, 5, pulse->pos[i]);
        put_bits(&s->pb, 4, pulse->amp[i]);
    }
}

/**
 * Encode spectral coefficients processed by psychoacoustic model.
 */
static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
{
    int start, i, w, w2, wg;

    w = 0;
    for(wg = 0; wg < sce->ics.num_window_groups; wg++){
        start = 0;
        for(i = 0; i < sce->ics.max_sfb; i++){
            if(sce->zeroes[w*16 + i]){
                start += sce->ics.swb_sizes[i];
                continue;
            }
            for(w2 = w; w2 < w + sce->ics.group_len[wg]; w2++){
                encode_band_coeffs(s, sce, start + w2*128,
                                   sce->ics.swb_sizes[i],
                                   sce->band_type[w*16 + i]);
            }
            start += sce->ics.swb_sizes[i];
        }
        w += sce->ics.group_len[wg];
    }
}

/**
 * Write some auxiliary information about the created AAC file.
 */
static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, const char *name)
{
    int i, namelen, padbits;

    namelen = strlen(name) + 2;
    put_bits(&s->pb, 3, TYPE_FIL);
    put_bits(&s->pb, 4, FFMIN(namelen, 15));
    if(namelen >= 15)
        put_bits(&s->pb, 8, namelen - 16);
    put_bits(&s->pb, 4, 0); //extension type - filler
    padbits = 8 - (put_bits_count(&s->pb) & 7);
    align_put_bits(&s->pb);
    for(i = 0; i < namelen - 2; i++)
        put_bits(&s->pb, 8, name[i]);
    put_bits(&s->pb, 12 - padbits, 0);
}

static av_cold int aac_encode_end(AVCodecContext *avctx)
{
    AACEncContext *s = avctx->priv_data;

    ff_mdct_end(&s->mdct1024);
    ff_mdct_end(&s->mdct128);
    ff_aac_psy_end(&s->psy);
    av_freep(&s->samples);
    av_freep(&s->cpe);
    return 0;
}

AVCodec aac_encoder = {
    "aac",
    CODEC_TYPE_AUDIO,
    CODEC_ID_AAC,
    sizeof(AACEncContext),
    aac_encode_init,
    aac_encode_frame,
    aac_encode_end,
    .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
};