Okayed parts of AAC encoder

Originally committed as revision 14752 to svn://svn.ffmpeg.org/ffmpeg/trunk

libavcodec/aacenc.c

+/*
+ * 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
+ * change greedy codebook search into something more optimal, like Viterbi algorithm
+ * determine run lengths along with codebook
+ ***********************************/
+
+#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 *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 *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
+};
+
+#define CB_UNSIGNED 0x01    ///< coefficients are coded as absolute values
+#define CB_PAIRS    0x02    ///< coefficients are grouped into pairs before coding (quads by default)
+#define CB_ESCAPE   0x04    ///< codebook allows escapes
+
+/** spectral coefficients codebook information */
+static const struct {
+    int16_t maxval;         ///< maximum possible value
+     int8_t cb_num;         ///< codebook number
+    uint8_t flags;          ///< codebook features
+} aac_cb_info[] = {
+    {    0, -1, CB_UNSIGNED }, // zero codebook
+    {    1,  0, 0 },
+    {    1,  1, 0 },
+    {    2,  2, CB_UNSIGNED },
+    {    2,  3, CB_UNSIGNED },
+    {    4,  4, CB_PAIRS },
+    {    4,  5, CB_PAIRS },
+    {    7,  6, CB_PAIRS | CB_UNSIGNED },
+    {    7,  7, CB_PAIRS | CB_UNSIGNED },
+    {   12,  8, CB_PAIRS | CB_UNSIGNED },
+    {   12,  9, CB_PAIRS | CB_UNSIGNED },
+    { 8191, 10, CB_PAIRS | CB_UNSIGNED | CB_ESCAPE },
+    {   -1, -1, 0 }, // reserved
+    {   -1, -1, 0 }, // perceptual noise substitution
+    {   -1, -1, 0 }, // intensity out-of-phase
+    {   -1, -1, 0 }, // intensity in-phase
+};
+
+/** default channel configurations */
+static const uint8_t aac_chan_configs[6][5] = {
+ {1, ID_SCE},                         // 1 channel  - single channel element
+ {1, ID_CPE},                         // 2 channels - channel pair
+ {2, ID_SCE, ID_CPE},                 // 3 channels - center + stereo
+ {3, ID_SCE, ID_CPE, ID_SCE},         // 4 channels - front center + stereo + back center
+ {3, ID_SCE, ID_CPE, ID_CPE},         // 5 channels - front center + stereo + back stereo
+ {4, ID_SCE, ID_CPE, ID_CPE, ID_LFE}, // 6 channels - front center + stereo + back stereo + LFE
+};
+
+/**
+ * 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
+    DECLARE_ALIGNED_16(FFTSample, tmp[1024]);    ///< temporary buffer used by MDCT
+    int16_t* samples;                            ///< saved preprocessed input
+
+    int samplerate_index;                        ///< MPEG-4 samplerate index
+    const uint8_t *swb_sizes1024;                ///< scalefactor band sizes for long frame
+    int swb_num1024;                             ///< number of scalefactor bands for long frame
+    const uint8_t *swb_sizes128;                 ///< scalefactor band sizes for short frame
+    int swb_num128;                              ///< number of scalefactor bands for short frame
+
+    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;
+    s->swb_sizes1024 = swb_size_1024[i];
+    s->swb_num1024   = ff_aac_num_swb_1024[i];
+    s->swb_sizes128  = swb_size_128[i];
+    s->swb_num128    = ff_aac_num_swb_128[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_aac_sine_long_1024, 1024);
+    ff_sine_window_init(ff_aac_sine_short_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, s->swb_sizes1024, s->swb_num1024, s->swb_sizes128, s->swb_num128) < 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(AVCodecContext *avctx, IndividualChannelStream *info)
+{
+    AACEncContext *s = avctx->priv_data;
+    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]);
+    }
+}
+
+/**
+ * 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, ID_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"),
+};