* AAC encoder
* Copyright (C) 2008 Konstantin Shishkov
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
- * @file libavcodec/aacenc.c
+ * @file
* AAC encoder
*/
/***********************************
* TODOs:
- * psy model selection with some option
* add sane pulse detection
* add temporal noise shaping
***********************************/
+#include "libavutil/float_dsp.h"
+#include "libavutil/opt.h"
#include "avcodec.h"
-#include "get_bits.h"
-#include "dsputil.h"
+#include "put_bits.h"
+#include "internal.h"
#include "mpeg4audio.h"
+#include "kbdwin.h"
+#include "sinewin.h"
-#include "aacpsy.h"
#include "aac.h"
#include "aactab.h"
+#include "aacenc.h"
+
+#include "psymodel.h"
+
+#define AAC_MAX_CHANNELS 6
+
+#define ERROR_IF(cond, ...) \
+ if (cond) { \
+ av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
+ return AVERROR(EINVAL); \
+ }
+
+float ff_aac_pow34sf_tab[428];
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,
32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
};
-static const uint8_t * const swb_size_1024[] = {
+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,
4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
};
-static const uint8_t * const swb_size_128[] = {
+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_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
};
/**
- * structure used in optimal codebook search
+ * Table to remap channels from Libav's default order to AAC order.
*/
-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;
+static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
+ { 0 },
+ { 0, 1 },
+ { 2, 0, 1 },
+ { 2, 0, 1, 3 },
+ { 2, 0, 1, 3, 4 },
+ { 2, 0, 1, 4, 5, 3 },
+};
/**
* Make AAC audio config object.
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);
+ put_bits(&pb, 4, s->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
+
+ //Explicitly Mark SBR absent
+ put_bits(&pb, 11, 0x2b7); //sync extension
+ put_bits(&pb, 5, AOT_SBR);
+ put_bits(&pb, 1, 0);
flush_put_bits(&pb);
}
-static av_cold int aac_encode_init(AVCodecContext *avctx)
+#define WINDOW_FUNC(type) \
+static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
+ SingleChannelElement *sce, \
+ const float *audio)
+
+WINDOW_FUNC(only_long)
{
- AACEncContext *s = avctx->priv_data;
- int i;
+ const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
+ const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
+ float *out = sce->ret_buf;
- avctx->frame_size = 1024;
+ fdsp->vector_fmul (out, audio, lwindow, 1024);
+ fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 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;
+WINDOW_FUNC(long_start)
+{
+ const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
+ const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
+ float *out = sce->ret_buf;
+
+ fdsp->vector_fmul(out, audio, lwindow, 1024);
+ memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
+ fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
+ memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
+}
- 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;
+WINDOW_FUNC(long_stop)
+{
+ const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
+ const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
+ float *out = sce->ret_buf;
+
+ memset(out, 0, sizeof(out[0]) * 448);
+ fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
+ memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
+ fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
+}
+
+WINDOW_FUNC(eight_short)
+{
+ const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
+ const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
+ const float *in = audio + 448;
+ float *out = sce->ret_buf;
+ int w;
+
+ for (w = 0; w < 8; w++) {
+ fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
+ out += 128;
+ in += 128;
+ fdsp->vector_fmul_reverse(out, in, swindow, 128);
+ out += 128;
}
- avctx->extradata = av_malloc(2);
- avctx->extradata_size = 2;
- put_audio_specific_config(avctx);
- return 0;
+}
+
+static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
+ SingleChannelElement *sce,
+ const float *audio) = {
+ [ONLY_LONG_SEQUENCE] = apply_only_long_window,
+ [LONG_START_SEQUENCE] = apply_long_start_window,
+ [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
+ [LONG_STOP_SEQUENCE] = apply_long_stop_window
+};
+
+static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
+ float *audio)
+{
+ int i;
+ float *output = sce->ret_buf;
+
+ apply_window[sce->ics.window_sequence[0]](&s->fdsp, sce, audio);
+
+ if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
+ s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
+ else
+ for (i = 0; i < 1024; i += 128)
+ s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2);
+ memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
}
/**
*/
static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
{
- int i;
+ int w;
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){
+ 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{
+ } 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]);
+ for (w = 1; w < 8; w++)
+ put_bits(&s->pb, 1, !info->group_len[w]);
+ }
+}
+
+/**
+ * Encode MS data.
+ * @see 4.6.8.1 "Joint Coding - M/S Stereo"
+ */
+static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
+{
+ int i, w;
+
+ put_bits(pb, 2, cpe->ms_mode);
+ if (cpe->ms_mode == 1)
+ for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
+ for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
+ put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
+}
+
+/**
+ * Produce integer coefficients from scalefactors provided by the model.
+ */
+static void adjust_frame_information(ChannelElement *cpe, int chans)
+{
+ int i, w, w2, g, ch;
+ int start, maxsfb, cmaxsfb;
+
+ for (ch = 0; ch < chans; ch++) {
+ IndividualChannelStream *ics = &cpe->ch[ch].ics;
+ start = 0;
+ maxsfb = 0;
+ cpe->ch[ch].pulse.num_pulse = 0;
+ for (w = 0; w < ics->num_windows*16; w += 16) {
+ for (g = 0; g < ics->num_swb; g++) {
+ //apply M/S
+ if (cpe->common_window && !ch && cpe->ms_mask[w + g]) {
+ for (i = 0; i < ics->swb_sizes[g]; i++) {
+ cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
+ cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
+ }
+ }
+ start += ics->swb_sizes[g];
+ }
+ for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--)
+ ;
+ maxsfb = FFMAX(maxsfb, cmaxsfb);
+ }
+ ics->max_sfb = maxsfb;
+
+ //adjust zero bands for window groups
+ for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
+ for (g = 0; g < ics->max_sfb; g++) {
+ i = 1;
+ for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
+ if (!cpe->ch[ch].zeroes[w2*16 + g]) {
+ i = 0;
+ break;
+ }
+ }
+ cpe->ch[ch].zeroes[w*16 + g] = i;
+ }
+ }
+ }
+
+ if (chans > 1 && cpe->common_window) {
+ IndividualChannelStream *ics0 = &cpe->ch[0].ics;
+ IndividualChannelStream *ics1 = &cpe->ch[1].ics;
+ int msc = 0;
+ ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
+ ics1->max_sfb = ics0->max_sfb;
+ for (w = 0; w < ics0->num_windows*16; w += 16)
+ for (i = 0; i < ics0->max_sfb; i++)
+ if (cpe->ms_mask[w+i])
+ msc++;
+ if (msc == 0 || ics0->max_sfb == 0)
+ cpe->ms_mode = 0;
+ else
+ cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
}
}
/**
- * Calculate the number of bits needed to code all coefficient signs in current band.
+ * Encode scalefactor band coding type.
*/
-static int calculate_band_sign_bits(AACEncContext *s, SingleChannelElement *sce,
- int group_len, int start, int size)
+static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
{
- int bits = 0;
+ int w;
+
+ for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
+ s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
+}
+
+/**
+ * Encode scalefactors.
+ */
+static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
+ SingleChannelElement *sce)
+{
+ int off = sce->sf_idx[0], diff;
int i, w;
- for(w = 0; w < group_len; w++){
- for(i = 0; i < size; i++){
- if(sce->icoefs[start + i])
- bits++;
+
+ for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+ for (i = 0; i < sce->ics.max_sfb; i++) {
+ if (!sce->zeroes[w*16 + i]) {
+ diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
+ if (diff < 0 || diff > 120)
+ av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n");
+ off = sce->sf_idx[w*16 + i];
+ put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
+ }
}
- start += 128;
}
- return bits;
}
/**
int i;
put_bits(&s->pb, 1, !!pulse->num_pulse);
- if(!pulse->num_pulse) return;
+ 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++){
+ for (i = 0; i < pulse->num_pulse; i++) {
put_bits(&s->pb, 5, pulse->pos[i]);
put_bits(&s->pb, 4, pulse->amp[i]);
}
*/
static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
{
- int start, i, w, w2, wg;
+ int start, i, w, w2;
- w = 0;
- for(wg = 0; wg < sce->ics.num_window_groups; wg++){
+ for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
start = 0;
- for(i = 0; i < sce->ics.max_sfb; i++){
- if(sce->zeroes[w*16 + i]){
+ 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]);
- }
+ for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)
+ s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
+ sce->ics.swb_sizes[i],
+ sce->sf_idx[w*16 + i],
+ sce->band_type[w*16 + i],
+ s->lambda);
start += sce->ics.swb_sizes[i];
}
- w += sce->ics.group_len[wg];
}
}
+/**
+ * Encode one channel of audio data.
+ */
+static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
+ SingleChannelElement *sce,
+ int common_window)
+{
+ put_bits(&s->pb, 8, sce->sf_idx[0]);
+ if (!common_window)
+ put_ics_info(s, &sce->ics);
+ encode_band_info(s, sce);
+ encode_scale_factors(avctx, s, sce);
+ encode_pulses(s, &sce->pulse);
+ put_bits(&s->pb, 1, 0); //tns
+ put_bits(&s->pb, 1, 0); //ssr
+ encode_spectral_coeffs(s, sce);
+ return 0;
+}
+
/**
* Write some auxiliary information about the created AAC file.
*/
-static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, const char *name)
+static void put_bitstream_info(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);
+ if (namelen >= 15)
+ put_bits(&s->pb, 8, namelen - 14);
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++)
+ padbits = -put_bits_count(&s->pb) & 7;
+ avpriv_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);
}
+/*
+ * Copy input samples.
+ * Channels are reordered from Libav's default order to AAC order.
+ */
+static void copy_input_samples(AACEncContext *s, const AVFrame *frame)
+{
+ int ch;
+ int end = 2048 + (frame ? frame->nb_samples : 0);
+ const uint8_t *channel_map = aac_chan_maps[s->channels - 1];
+
+ /* copy and remap input samples */
+ for (ch = 0; ch < s->channels; ch++) {
+ /* copy last 1024 samples of previous frame to the start of the current frame */
+ memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
+
+ /* copy new samples and zero any remaining samples */
+ if (frame) {
+ memcpy(&s->planar_samples[ch][2048],
+ frame->extended_data[channel_map[ch]],
+ frame->nb_samples * sizeof(s->planar_samples[0][0]));
+ }
+ memset(&s->planar_samples[ch][end], 0,
+ (3072 - end) * sizeof(s->planar_samples[0][0]));
+ }
+}
+
+static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
+{
+ AACEncContext *s = avctx->priv_data;
+ float **samples = s->planar_samples, *samples2, *la, *overlap;
+ ChannelElement *cpe;
+ int i, ch, w, g, chans, tag, start_ch, ret;
+ int chan_el_counter[4];
+ FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
+
+ if (s->last_frame == 2)
+ return 0;
+
+ /* add current frame to queue */
+ if (frame) {
+ if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
+ return ret;
+ }
+
+ copy_input_samples(s, frame);
+ if (s->psypp)
+ ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
+
+ if (!avctx->frame_number)
+ return 0;
+
+ start_ch = 0;
+ for (i = 0; i < s->chan_map[0]; i++) {
+ FFPsyWindowInfo* wi = windows + start_ch;
+ tag = s->chan_map[i+1];
+ chans = tag == TYPE_CPE ? 2 : 1;
+ cpe = &s->cpe[i];
+ for (ch = 0; ch < chans; ch++) {
+ IndividualChannelStream *ics = &cpe->ch[ch].ics;
+ int cur_channel = start_ch + ch;
+ overlap = &samples[cur_channel][0];
+ samples2 = overlap + 1024;
+ la = samples2 + (448+64);
+ if (!frame)
+ la = NULL;
+ if (tag == TYPE_LFE) {
+ wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
+ wi[ch].window_shape = 0;
+ wi[ch].num_windows = 1;
+ wi[ch].grouping[0] = 1;
+
+ /* Only the lowest 12 coefficients are used in a LFE channel.
+ * The expression below results in only the bottom 8 coefficients
+ * being used for 11.025kHz to 16kHz sample rates.
+ */
+ ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
+ } else {
+ wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel,
+ ics->window_sequence[0]);
+ }
+ ics->window_sequence[1] = ics->window_sequence[0];
+ ics->window_sequence[0] = wi[ch].window_type[0];
+ ics->use_kb_window[1] = ics->use_kb_window[0];
+ ics->use_kb_window[0] = wi[ch].window_shape;
+ ics->num_windows = wi[ch].num_windows;
+ ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
+ ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
+ for (w = 0; w < ics->num_windows; w++)
+ ics->group_len[w] = wi[ch].grouping[w];
+
+ apply_window_and_mdct(s, &cpe->ch[ch], overlap);
+ }
+ start_ch += chans;
+ }
+ if ((ret = ff_alloc_packet(avpkt, 768 * s->channels))) {
+ av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+ return ret;
+ }
+
+ do {
+ int frame_bits;
+
+ init_put_bits(&s->pb, avpkt->data, avpkt->size);
+
+ if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
+ put_bitstream_info(s, LIBAVCODEC_IDENT);
+ start_ch = 0;
+ memset(chan_el_counter, 0, sizeof(chan_el_counter));
+ for (i = 0; i < s->chan_map[0]; i++) {
+ FFPsyWindowInfo* wi = windows + start_ch;
+ const float *coeffs[2];
+ tag = s->chan_map[i+1];
+ chans = tag == TYPE_CPE ? 2 : 1;
+ cpe = &s->cpe[i];
+ put_bits(&s->pb, 3, tag);
+ put_bits(&s->pb, 4, chan_el_counter[tag]++);
+ for (ch = 0; ch < chans; ch++)
+ coeffs[ch] = cpe->ch[ch].coeffs;
+ s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
+ for (ch = 0; ch < chans; ch++) {
+ s->cur_channel = start_ch + ch;
+ s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
+ }
+ cpe->common_window = 0;
+ if (chans > 1
+ && wi[0].window_type[0] == wi[1].window_type[0]
+ && wi[0].window_shape == wi[1].window_shape) {
+
+ cpe->common_window = 1;
+ for (w = 0; w < wi[0].num_windows; w++) {
+ if (wi[0].grouping[w] != wi[1].grouping[w]) {
+ cpe->common_window = 0;
+ break;
+ }
+ }
+ }
+ s->cur_channel = start_ch;
+ if (s->options.stereo_mode && cpe->common_window) {
+ if (s->options.stereo_mode > 0) {
+ IndividualChannelStream *ics = &cpe->ch[0].ics;
+ for (w = 0; w < ics->num_windows; w += ics->group_len[w])
+ for (g = 0; g < ics->num_swb; g++)
+ cpe->ms_mask[w*16+g] = 1;
+ } else if (s->coder->search_for_ms) {
+ s->coder->search_for_ms(s, cpe, s->lambda);
+ }
+ }
+ adjust_frame_information(cpe, chans);
+ if (chans == 2) {
+ put_bits(&s->pb, 1, cpe->common_window);
+ if (cpe->common_window) {
+ put_ics_info(s, &cpe->ch[0].ics);
+ encode_ms_info(&s->pb, cpe);
+ }
+ }
+ for (ch = 0; ch < chans; ch++) {
+ s->cur_channel = start_ch + ch;
+ encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);
+ }
+ start_ch += chans;
+ }
+
+ frame_bits = put_bits_count(&s->pb);
+ if (frame_bits <= 6144 * s->channels - 3) {
+ s->psy.bitres.bits = frame_bits / s->channels;
+ break;
+ }
+
+ s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;
+
+ } while (1);
+
+ put_bits(&s->pb, 3, TYPE_END);
+ flush_put_bits(&s->pb);
+ avctx->frame_bits = put_bits_count(&s->pb);
+
+ // rate control stuff
+ if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
+ float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
+ s->lambda *= ratio;
+ s->lambda = FFMIN(s->lambda, 65536.f);
+ }
+
+ if (!frame)
+ s->last_frame++;
+
+ ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
+ &avpkt->duration);
+
+ avpkt->size = put_bits_count(&s->pb) >> 3;
+ *got_packet_ptr = 1;
+ return 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);
+ ff_psy_end(&s->psy);
+ if (s->psypp)
+ ff_psy_preprocess_end(s->psypp);
+ av_freep(&s->buffer.samples);
av_freep(&s->cpe);
+ ff_af_queue_close(&s->afq);
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"),
+static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
+{
+ int ret = 0;
+
+ avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
+
+ // 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_init_ff_sine_windows(10);
+ ff_init_ff_sine_windows(7);
+
+ if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0))
+ return ret;
+ if (ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0))
+ return ret;
+
+ return 0;
+}
+
+static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
+{
+ int ch;
+ FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 3 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail);
+ FF_ALLOCZ_OR_GOTO(avctx, s->cpe, sizeof(ChannelElement) * s->chan_map[0], alloc_fail);
+ FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
+
+ for(ch = 0; ch < s->channels; ch++)
+ s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
+
+ return 0;
+alloc_fail:
+ return AVERROR(ENOMEM);
+}
+
+static av_cold int aac_encode_init(AVCodecContext *avctx)
+{
+ AACEncContext *s = avctx->priv_data;
+ int i, ret = 0;
+ const uint8_t *sizes[2];
+ uint8_t grouping[AAC_MAX_CHANNELS];
+ int lengths[2];
+
+ avctx->frame_size = 1024;
+
+ for (i = 0; i < 16; i++)
+ if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
+ break;
+
+ s->channels = avctx->channels;
+
+ ERROR_IF(i == 16,
+ "Unsupported sample rate %d\n", avctx->sample_rate);
+ ERROR_IF(s->channels > AAC_MAX_CHANNELS,
+ "Unsupported number of channels: %d\n", s->channels);
+ ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW,
+ "Unsupported profile %d\n", avctx->profile);
+ ERROR_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
+ "Too many bits per frame requested\n");
+
+ s->samplerate_index = i;
+
+ s->chan_map = aac_chan_configs[s->channels-1];
+
+ if (ret = dsp_init(avctx, s))
+ goto fail;
+
+ if (ret = alloc_buffers(avctx, s))
+ goto fail;
+
+ avctx->extradata_size = 5;
+ put_audio_specific_config(avctx);
+
+ sizes[0] = swb_size_1024[i];
+ sizes[1] = swb_size_128[i];
+ lengths[0] = ff_aac_num_swb_1024[i];
+ lengths[1] = ff_aac_num_swb_128[i];
+ for (i = 0; i < s->chan_map[0]; i++)
+ grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
+ if (ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping))
+ goto fail;
+ s->psypp = ff_psy_preprocess_init(avctx);
+ s->coder = &ff_aac_coders[2];
+
+ s->lambda = avctx->global_quality ? avctx->global_quality : 120;
+
+ ff_aac_tableinit();
+
+ for (i = 0; i < 428; i++)
+ ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
+
+ avctx->delay = 1024;
+ ff_af_queue_init(avctx, &s->afq);
+
+ return 0;
+fail:
+ aac_encode_end(avctx);
+ return ret;
+}
+
+#define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
+static const AVOption aacenc_options[] = {
+ {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.i64 = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
+ {"auto", "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
+ {"ms_off", "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
+ {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
+ {NULL}
+};
+
+static const AVClass aacenc_class = {
+ "AAC encoder",
+ av_default_item_name,
+ aacenc_options,
+ LIBAVUTIL_VERSION_INT,
+};
+
+AVCodec ff_aac_encoder = {
+ .name = "aac",
+ .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_AAC,
+ .priv_data_size = sizeof(AACEncContext),
+ .init = aac_encode_init,
+ .encode2 = aac_encode_frame,
+ .close = aac_encode_end,
+ .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
+ CODEC_CAP_EXPERIMENTAL,
+ .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
+ .priv_class = &aacenc_class,
};