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Merge the AAC encoder from SoC svn. It is still considered experimental.

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Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Alex Converse 2009-07-08 20:01:31 +00:00
parent 5e039e1b4c
commit 78e65cd772
10 changed files with 1992 additions and 115 deletions
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1
Changelog
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@ -27,6 +27,7 @@ version <next>:
- Electronic Arts Madcow decoder
- DivX (XSUB) subtitle encoder
- nonfree libamr support for AMR-NB/WB decoding/encoding removed
- Experimental AAC encoder

1
libavcodec/Makefile
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@ -36,6 +36,7 @@ OBJS-$(CONFIG_VDPAU) += vdpau.o
# decoders/encoders/hardware accelerators
OBJS-$(CONFIG_AAC_DECODER) += aac.o aactab.o mpeg4audio.o aac_parser.o aac_ac3_parser.o
OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o aacpsy.o aactab.o psymodel.o iirfilter.o mdct.o fft.o mpeg4audio.o
OBJS-$(CONFIG_AASC_DECODER) += aasc.o msrledec.o
OBJS-$(CONFIG_AC3_DECODER) += eac3dec.o ac3dec.o ac3tab.o ac3dec_data.o ac3.o
OBJS-$(CONFIG_AC3_ENCODER) += ac3enc.o ac3tab.o ac3.o

19
libavcodec/aac.h
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@ -116,6 +116,12 @@ typedef struct {
#define MAX_PREDICTORS 672
#define SCALE_DIV_512 36 ///< scalefactor difference that corresponds to scale difference in 512 times
#define SCALE_ONE_POS 140 ///< scalefactor index that corresponds to scale=1.0
#define SCALE_MAX_POS 255 ///< scalefactor index maximum value
#define SCALE_MAX_DIFF 60 ///< maximum scalefactor difference allowed by standard
#define SCALE_DIFF_ZERO 60 ///< codebook index corresponding to zero scalefactor indices difference
/**
* Individual Channel Stream
*/
@ -126,6 +132,7 @@ typedef struct {
int num_window_groups;
uint8_t group_len[8];
const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window
const uint8_t *swb_sizes; ///< table of scalefactor band sizes for a particular window
int num_swb; ///< number of scalefactor window bands
int num_windows;
int tns_max_bands;
@ -165,6 +172,7 @@ typedef struct {
typedef struct {
int num_pulse;
int start;
int pos[4];
int amp[4];
} Pulse;
@ -189,11 +197,14 @@ typedef struct {
typedef struct {
IndividualChannelStream ics;
TemporalNoiseShaping tns;
enum BandType band_type[120]; ///< band types
Pulse pulse;
enum BandType band_type[128]; ///< band types
int band_type_run_end[120]; ///< band type run end points
float sf[120]; ///< scalefactors
int sf_idx[128]; ///< scalefactor indices (used by encoder)
uint8_t zeroes[128]; ///< band is not coded (used by encoder)
DECLARE_ALIGNED_16(float, coeffs[1024]); ///< coefficients for IMDCT
DECLARE_ALIGNED_16(float, saved[512]); ///< overlap
DECLARE_ALIGNED_16(float, saved[1024]); ///< overlap
DECLARE_ALIGNED_16(float, ret[1024]); ///< PCM output
PredictorState predictor_state[MAX_PREDICTORS];
} SingleChannelElement;
@ -203,7 +214,9 @@ typedef struct {
*/
typedef struct {
// CPE specific
uint8_t ms_mask[120]; ///< Set if mid/side stereo is used for each scalefactor window band
int common_window; ///< Set if channels share a common 'IndividualChannelStream' in bitstream.
int ms_mode; ///< Signals mid/side stereo flags coding mode (used by encoder)
uint8_t ms_mask[128]; ///< Set if mid/side stereo is used for each scalefactor window band
// shared
SingleChannelElement ch[2];
// CCE specific

1037
libavcodec/aaccoder.c Normal file
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File diff suppressed because it is too large Load Diff

402
libavcodec/aacenc.c
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@ -26,19 +26,20 @@
/***********************************
* TODOs:
* psy model selection with some option
* add sane pulse detection
* add temporal noise shaping
***********************************/
#include "avcodec.h"
#include "get_bits.h"
#include "put_bits.h"
#include "dsputil.h"
#include "mpeg4audio.h"
#include "aacpsy.h"
#include "aac.h"
#include "aactab.h"
#include "aacenc.h"
#include "psymodel.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,
@ -83,7 +84,7 @@ static const uint8_t swb_size_1024_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,
@ -110,7 +111,7 @@ 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[] = {
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,
@ -119,23 +120,6 @@ static const uint8_t * const swb_size_128[] = {
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
@ -146,33 +130,6 @@ static const uint8_t aac_chan_configs[6][5] = {
{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"
@ -197,6 +154,8 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
{
AACEncContext *s = avctx->priv_data;
int i;
const uint8_t *sizes[2];
int lengths[2];
avctx->frame_size = 1024;
@ -224,25 +183,90 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
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);
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];
ff_psy_init(&s->psy, avctx, 2, sizes, lengths);
s->psypp = ff_psy_preprocess_init(avctx);
s->coder = &ff_aac_coders[0];
s->lambda = avctx->global_quality ? avctx->global_quality : 120;
#if !CONFIG_HARDCODED_TABLES
for (i = 0; i < 428; i++)
ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
#endif /* CONFIG_HARDCODED_TABLES */
if (avctx->channels > 5)
av_log(avctx, AV_LOG_ERROR, "This encoder does not yet enforce the restrictions on LFEs. "
"The output will most likely be an illegal bitstream.\n");
return 0;
}
static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce, short *audio, int channel)
{
int i, j, k;
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[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;
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
memcpy(s->output, sce->saved, sizeof(float)*1024);
if(sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE){
memset(s->output, 0, sizeof(s->output[0]) * 448);
for(i = 448; i < 576; i++)
s->output[i] = sce->saved[i] * pwindow[i - 448];
for(i = 576; i < 704; i++)
s->output[i] = sce->saved[i];
}
if(sce->ics.window_sequence[0] != LONG_START_SEQUENCE){
j = channel;
for (i = 0; i < 1024; i++, j += avctx->channels){
s->output[i+1024] = audio[j] * lwindow[1024 - i - 1];
sce->saved[i] = audio[j] * lwindow[i];
}
}else{
j = channel;
for(i = 0; i < 448; i++, j += avctx->channels)
s->output[i+1024] = audio[j];
for(i = 448; i < 576; i++, j += avctx->channels)
s->output[i+1024] = audio[j] * swindow[576 - i - 1];
memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448);
j = channel;
for(i = 0; i < 1024; i++, j += avctx->channels)
sce->saved[i] = audio[j];
}
ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output);
}else{
j = channel;
for (k = 0; k < 1024; k += 128) {
for(i = 448 + k; i < 448 + k + 256; i++)
s->output[i - 448 - k] = (i < 1024)
? sce->saved[i]
: audio[channel + (i-1024)*avctx->channels];
s->dsp.vector_fmul (s->output, k ? swindow : pwindow, 128);
s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128);
ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output);
}
j = channel;
for(i = 0; i < 1024; i++, j += avctx->channels)
sce->saved[i] = audio[j];
}
}
/**
* Encode ics_info element.
* @see Table 4.6 (syntax of ics_info)
*/
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]);
@ -252,27 +276,118 @@ static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
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]);
for(w = 1; w < 8; w++){
put_bits(&s->pb, 1, !info->group_len[w]);
}
}
}
/**
* Calculate the number of bits needed to code all coefficient signs in current band.
* Encode MS data.
* @see 4.6.8.1 "Joint Coding - M/S Stereo"
*/
static int calculate_band_sign_bits(AACEncContext *s, SingleChannelElement *sce,
int group_len, int start, int size)
static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
{
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++;
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(AACEncContext *apc, ChannelElement *cpe, int chans)
{
int i, w, w2, g, ch;
int start, sum, 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++){
sum = 0;
//apply M/S
if(!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 ? 1 : 2;
}
}
/**
* Encode scalefactor band coding type.
*/
static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
{
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 < 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;
}
/**
@ -298,27 +413,43 @@ static void encode_pulses(AACEncContext *s, Pulse *pulse)
*/
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]){
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.
*/
@ -339,13 +470,130 @@ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, const ch
put_bits(&s->pb, 12 - padbits, 0);
}
static int aac_encode_frame(AVCodecContext *avctx,
uint8_t *frame, int buf_size, void *data)
{
AACEncContext *s = avctx->priv_data;
int16_t *samples = s->samples, *samples2, *la;
ChannelElement *cpe;
int i, j, chans, tag, start_ch;
const uint8_t *chan_map = aac_chan_configs[avctx->channels-1];
int chan_el_counter[4];
if(s->last_frame)
return 0;
if(data){
if(!s->psypp){
memcpy(s->samples + 1024 * avctx->channels, data, 1024 * avctx->channels * sizeof(s->samples[0]));
}else{
start_ch = 0;
samples2 = s->samples + 1024 * avctx->channels;
for(i = 0; i < chan_map[0]; i++){
tag = chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch, samples2 + start_ch, start_ch, chans);
start_ch += chans;
}
}
}
if(!avctx->frame_number){
memcpy(s->samples, s->samples + 1024 * avctx->channels, 1024 * avctx->channels * sizeof(s->samples[0]));
return 0;
}
init_put_bits(&s->pb, frame, buf_size*8);
if((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT)){
put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
}
start_ch = 0;
memset(chan_el_counter, 0, sizeof(chan_el_counter));
for(i = 0; i < chan_map[0]; i++){
FFPsyWindowInfo wi[2];
tag = chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
cpe = &s->cpe[i];
samples2 = samples + start_ch;
la = samples2 + 1024 * avctx->channels + start_ch;
if(!data) la = NULL;
for(j = 0; j < chans; j++){
IndividualChannelStream *ics = &cpe->ch[j].ics;
int k;
wi[j] = ff_psy_suggest_window(&s->psy, samples2, la, start_ch + j, ics->window_sequence[0]);
ics->window_sequence[1] = ics->window_sequence[0];
ics->window_sequence[0] = wi[j].window_type[0];
ics->use_kb_window[1] = ics->use_kb_window[0];
ics->use_kb_window[0] = wi[j].window_shape;
ics->num_windows = wi[j].num_windows;
ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
ics->num_swb = s->psy.num_bands[ics->num_windows == 8];
for(k = 0; k < ics->num_windows; k++)
ics->group_len[k] = wi[j].grouping[k];
s->cur_channel = start_ch + j;
apply_window_and_mdct(avctx, s, &cpe->ch[j], samples2, j);
s->coder->search_for_quantizers(avctx, s, &cpe->ch[j], 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(j = 0; j < wi[0].num_windows; j++){
if(wi[0].grouping[j] != wi[1].grouping[j]){
cpe->common_window = 0;
break;
}
}
}
if(cpe->common_window && s->coder->search_for_ms)
s->coder->search_for_ms(s, cpe, s->lambda);
adjust_frame_information(s, cpe, chans);
put_bits(&s->pb, 3, tag);
put_bits(&s->pb, 4, chan_el_counter[tag]++);
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(j = 0; j < chans; j++){
s->cur_channel = start_ch + j;
ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window);
}
start_ch += chans;
}
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;
}
if (avctx->frame_bits > 6144*avctx->channels) {
av_log(avctx, AV_LOG_ERROR, "input buffer violation %d > %d.\n", avctx->frame_bits, 6144*avctx->channels);
}
if(!data)
s->last_frame = 1;
memcpy(s->samples, s->samples + 1024 * avctx->channels, 1024 * avctx->channels * sizeof(s->samples[0]));
return put_bits_count(&s->pb)>>3;
}
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);
ff_psy_end(&s->psy);
ff_psy_preprocess_end(s->psypp);
av_freep(&s->samples);
av_freep(&s->cpe);
return 0;

71
libavcodec/aacenc.h Normal file
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@ -0,0 +1,71 @@
/*
* 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
*/
#ifndef AVCODEC_AACENC_H
#define AVCODEC_AACENC_H
#include "avcodec.h"
#include "put_bits.h"
#include "dsputil.h"
#include "aac.h"
#include "psymodel.h"
struct AACEncContext;
typedef struct AACCoefficientsEncoder{
void (*search_for_quantizers)(AVCodecContext *avctx, struct AACEncContext *s,
SingleChannelElement *sce, const float lambda);
void (*encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce,
int win, int group_len, const float lambda);
void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, int size,
int scale_idx, int cb, const float lambda);
void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe, const float lambda);
}AACCoefficientsEncoder;
extern AACCoefficientsEncoder ff_aac_coders[];
/**
* AAC encoder context
*/
typedef struct AACEncContext {
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
FFPsyContext psy;
struct FFPsyPreprocessContext* psypp;
AACCoefficientsEncoder *coder;
int cur_channel;
int last_frame;
float lambda;
DECLARE_ALIGNED_16(int, qcoefs[96][2]); ///< quantized coefficients
DECLARE_ALIGNED_16(float, scoefs[1024]); ///< scaled coefficients
} AACEncContext;
#endif /* AVCODEC_AACENC_H */

286
libavcodec/aacpsy.c
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@ -25,54 +25,25 @@
*/
#include "avcodec.h"
#include "aacpsy.h"
#include "aactab.h"
#include "psymodel.h"
/***********************************
* TODOs:
* General:
* better audio preprocessing (add DC highpass filter?)
* more psy models
* maybe improve coefficient quantization function in some way
*
* 3GPP-based psy model:
* thresholds linearization after their modifications for attaining given bitrate
* try other bitrate controlling mechanism (maybe use ratecontrol.c?)
* control quality for quality-based output
**********************************/
/**
* Quantize one coefficient.
* @return absolute value of the quantized coefficient
* @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
*/
static av_always_inline int quant(float coef, const float Q)
{
return av_clip((int)(pow(fabsf(coef) * Q, 0.75) + 0.4054), 0, 8191);
}
static inline float get_approximate_quant_error(float *c, int size, int scale_idx)
{
int i;
int q;
float coef, unquant, sum = 0.0f;
const float Q = ff_aac_pow2sf_tab[200 - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
const float IQ = ff_aac_pow2sf_tab[200 + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
for(i = 0; i < size; i++){
coef = fabs(c[i]);
q = quant(c[i], Q);
unquant = (q * cbrt(q)) * IQ;
sum += (coef - unquant) * (coef - unquant);
}
return sum;
}
/**
* constants for 3GPP AAC psychoacoustic model
* @{
*/
#define PSY_3GPP_SPREAD_LOW 1.5f // spreading factor for ascending threshold spreading (15 dB/Bark)
#define PSY_3GPP_SPREAD_HI 3.0f // spreading factor for descending threshold spreading (30 dB/Bark)
#define PSY_3GPP_RPEMIN 0.01f
#define PSY_3GPP_RPELEV 2.0f
/**
* @}
*/
@ -83,8 +54,24 @@ static inline float get_approximate_quant_error(float *c, int size, int scale_id
typedef struct Psy3gppBand{
float energy; ///< band energy
float ffac; ///< form factor
float thr; ///< energy threshold
float min_snr; ///< minimal SNR
float thr_quiet; ///< threshold in quiet
}Psy3gppBand;
/**
* single/pair channel context for psychoacoustic model
*/
typedef struct Psy3gppChannel{
Psy3gppBand band[128]; ///< bands information
Psy3gppBand prev_band[128]; ///< bands information from the previous frame
float win_energy; ///< sliding average of channel energy
float iir_state[2]; ///< hi-pass IIR filter state
uint8_t next_grouping; ///< stored grouping scheme for the next frame (in case of 8 short window sequence)
enum WindowSequence next_window_seq; ///< window sequence to be used in the next frame
}Psy3gppChannel;
/**
* psychoacoustic model frame type-dependent coefficients
*/
@ -95,10 +82,241 @@ typedef struct Psy3gppCoeffs{
float spread_hi [64]; ///< spreading factor for high-to-low threshold spreading in long frame
}Psy3gppCoeffs;
/**
* 3GPP TS26.403-inspired psychoacoustic model specific data
*/
typedef struct Psy3gppContext{
Psy3gppCoeffs psy_coef[2];
Psy3gppChannel *ch;
}Psy3gppContext;
/**
* Calculate Bark value for given line.
*/
static inline float calc_bark(float f)
static av_cold float calc_bark(float f)
{
return 13.3f * atanf(0.00076f * f) + 3.5f * atanf((f / 7500.0f) * (f / 7500.0f));
}
#define ATH_ADD 4
/**
* Calculate ATH value for given frequency.
* Borrowed from Lame.
*/
static av_cold float ath(float f, float add)
{
f /= 1000.0f;
return 3.64 * pow(f, -0.8)
- 6.8 * exp(-0.6 * (f - 3.4) * (f - 3.4))
+ 6.0 * exp(-0.15 * (f - 8.7) * (f - 8.7))
+ (0.6 + 0.04 * add) * 0.001 * f * f * f * f;
}
static av_cold int psy_3gpp_init(FFPsyContext *ctx){
Psy3gppContext *pctx;
float barks[1024];
int i, j, g, start;
float prev, minscale, minath;
ctx->model_priv_data = av_mallocz(sizeof(Psy3gppContext));
pctx = (Psy3gppContext*) ctx->model_priv_data;
for(i = 0; i < 1024; i++)
barks[i] = calc_bark(i * ctx->avctx->sample_rate / 2048.0);
minath = ath(3410, ATH_ADD);
for(j = 0; j < 2; j++){
Psy3gppCoeffs *coeffs = &pctx->psy_coef[j];
i = 0;
prev = 0.0;
for(g = 0; g < ctx->num_bands[j]; g++){
i += ctx->bands[j][g];
coeffs->barks[g] = (barks[i - 1] + prev) / 2.0;
prev = barks[i - 1];
}
for(g = 0; g < ctx->num_bands[j] - 1; g++){
coeffs->spread_low[g] = pow(10.0, -(coeffs->barks[g+1] - coeffs->barks[g]) * PSY_3GPP_SPREAD_LOW);
coeffs->spread_hi [g] = pow(10.0, -(coeffs->barks[g+1] - coeffs->barks[g]) * PSY_3GPP_SPREAD_HI);
}
start = 0;
for(g = 0; g < ctx->num_bands[j]; g++){
minscale = ath(ctx->avctx->sample_rate * start / 1024.0, ATH_ADD);
for(i = 1; i < ctx->bands[j][g]; i++){
minscale = fminf(minscale, ath(ctx->avctx->sample_rate * (start + i) / 1024.0 / 2.0, ATH_ADD));
}
coeffs->ath[g] = minscale - minath;
start += ctx->bands[j][g];
}
}
pctx->ch = av_mallocz(sizeof(Psy3gppChannel) * ctx->avctx->channels);
return 0;
}
/**
* IIR filter used in block switching decision
*/
static float iir_filter(int in, float state[2])
{
float ret;
ret = 0.7548f * (in - state[0]) + 0.5095f * state[1];
state[0] = in;
state[1] = ret;
return ret;
}
/**
* window grouping information stored as bits (0 - new group, 1 - group continues)
*/
static const uint8_t window_grouping[9] = {
0xB6, 0x6C, 0xD8, 0xB2, 0x66, 0xC6, 0x96, 0x36, 0x36
};
/**
* Tell encoder which window types to use.
* @see 3GPP TS26.403 5.4.1 "Blockswitching"
*/
static FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
const int16_t *audio, const int16_t *la,
int channel, int prev_type)
{
int i, j;
int br = ctx->avctx->bit_rate / ctx->avctx->channels;
int attack_ratio = br <= 16000 ? 18 : 10;
Psy3gppContext *pctx = (Psy3gppContext*) ctx->model_priv_data;
Psy3gppChannel *pch = &pctx->ch[channel];
uint8_t grouping = 0;
FFPsyWindowInfo wi;
memset(&wi, 0, sizeof(wi));
if(la){
float s[8], v;
int switch_to_eight = 0;
float sum = 0.0, sum2 = 0.0;
int attack_n = 0;
for(i = 0; i < 8; i++){
for(j = 0; j < 128; j++){
v = iir_filter(audio[(i*128+j)*ctx->avctx->channels], pch->iir_state);
sum += v*v;
}
s[i] = sum;
sum2 += sum;
}
for(i = 0; i < 8; i++){
if(s[i] > pch->win_energy * attack_ratio){
attack_n = i + 1;
switch_to_eight = 1;
break;
}
}
pch->win_energy = pch->win_energy*7/8 + sum2/64;
wi.window_type[1] = prev_type;
switch(prev_type){
case ONLY_LONG_SEQUENCE:
wi.window_type[0] = switch_to_eight ? LONG_START_SEQUENCE : ONLY_LONG_SEQUENCE;
break;
case LONG_START_SEQUENCE:
wi.window_type[0] = EIGHT_SHORT_SEQUENCE;
grouping = pch->next_grouping;
break;
case LONG_STOP_SEQUENCE:
wi.window_type[0] = ONLY_LONG_SEQUENCE;
break;
case EIGHT_SHORT_SEQUENCE:
wi.window_type[0] = switch_to_eight ? EIGHT_SHORT_SEQUENCE : LONG_STOP_SEQUENCE;
grouping = switch_to_eight ? pch->next_grouping : 0;
break;
}
pch->next_grouping = window_grouping[attack_n];
}else{
for(i = 0; i < 3; i++)
wi.window_type[i] = prev_type;
grouping = (prev_type == EIGHT_SHORT_SEQUENCE) ? window_grouping[0] : 0;
}
wi.window_shape = 1;
if(wi.window_type[0] != EIGHT_SHORT_SEQUENCE){
wi.num_windows = 1;
wi.grouping[0] = 1;
}else{
int lastgrp = 0;
wi.num_windows = 8;
for(i = 0; i < 8; i++){
if(!((grouping >> i) & 1))
lastgrp = i;
wi.grouping[lastgrp]++;
}
}
return wi;
}
/**
* Calculate band thresholds as suggested in 3GPP TS26.403
*/
static void psy_3gpp_analyze(FFPsyContext *ctx, int channel, const float *coefs,
FFPsyWindowInfo *wi)
{
Psy3gppContext *pctx = (Psy3gppContext*) ctx->model_priv_data;
Psy3gppChannel *pch = &pctx->ch[channel];
int start = 0;
int i, w, g;
const int num_bands = ctx->num_bands[wi->num_windows == 8];
const uint8_t* band_sizes = ctx->bands[wi->num_windows == 8];
Psy3gppCoeffs *coeffs = &pctx->psy_coef[wi->num_windows == 8];
//calculate energies, initial thresholds and related values - 5.4.2 "Threshold Calculation"
for(w = 0; w < wi->num_windows*16; w += 16){
for(g = 0; g < num_bands; g++){
Psy3gppBand *band = &pch->band[w+g];
band->energy = 0.0f;
for(i = 0; i < band_sizes[g]; i++)
band->energy += coefs[start+i] * coefs[start+i];
band->energy *= 1.0f / (512*512);
band->thr = band->energy * 0.001258925f;
start += band_sizes[g];
ctx->psy_bands[channel*PSY_MAX_BANDS+w+g].energy = band->energy;
}
}
//modify thresholds - spread, threshold in quiet - 5.4.3 "Spreaded Energy Calculation"
for(w = 0; w < wi->num_windows*16; w += 16){
Psy3gppBand *band = &pch->band[w];
for(g = 1; g < num_bands; g++){
band[g].thr = FFMAX(band[g].thr, band[g-1].thr * coeffs->spread_low[g-1]);
}
for(g = num_bands - 2; g >= 0; g--){
band[g].thr = FFMAX(band[g].thr, band[g+1].thr * coeffs->spread_hi [g]);
}
for(g = 0; g < num_bands; g++){
band[g].thr_quiet = FFMAX(band[g].thr, coeffs->ath[g]);
if(wi->num_windows != 8 && wi->window_type[1] != EIGHT_SHORT_SEQUENCE){
band[g].thr_quiet = fmaxf(PSY_3GPP_RPEMIN*band[g].thr_quiet,
fminf(band[g].thr_quiet,
PSY_3GPP_RPELEV*pch->prev_band[w+g].thr_quiet));
}
band[g].thr = FFMAX(band[g].thr, band[g].thr_quiet * 0.25);
ctx->psy_bands[channel*PSY_MAX_BANDS+w+g].threshold = band[g].thr;
}
}
memcpy(pch->prev_band, pch->band, sizeof(pch->band));
}
static av_cold void psy_3gpp_end(FFPsyContext *apc)
{
Psy3gppContext *pctx = (Psy3gppContext*) apc->model_priv_data;
av_freep(&pctx->ch);
av_freep(&apc->model_priv_data);
}
const FFPsyModel ff_aac_psy_model =
{
.name = "3GPP TS 26.403-inspired model",
.init = psy_3gpp_init,
.window = psy_3gpp_window,
.analyze = psy_3gpp_analyze,
.end = psy_3gpp_end,
};

2
libavcodec/allcodecs.c
View File

@ -195,7 +195,7 @@ void avcodec_register_all(void)
REGISTER_ENCDEC (ZMBV, zmbv);
/* audio codecs */
REGISTER_DECODER (AAC, aac);
REGISTER_ENCDEC (AAC, aac);
REGISTER_ENCDEC (AC3, ac3);
REGISTER_ENCDEC (ALAC, alac);
REGISTER_DECODER (APE, ape);

130
libavcodec/psymodel.c Normal file
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@ -0,0 +1,130 @@
/*
* audio encoder psychoacoustic model
* 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
*/
#include "avcodec.h"
#include "psymodel.h"
#include "iirfilter.h"
extern const FFPsyModel ff_aac_psy_model;
av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx,
int num_lens,
const uint8_t **bands, const int* num_bands)
{
ctx->avctx = avctx;
ctx->psy_bands = av_mallocz(sizeof(FFPsyBand) * PSY_MAX_BANDS * avctx->channels);
ctx->bands = av_malloc (sizeof(ctx->bands[0]) * num_lens);
ctx->num_bands = av_malloc (sizeof(ctx->num_bands[0]) * num_lens);
memcpy(ctx->bands, bands, sizeof(ctx->bands[0]) * num_lens);
memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) * num_lens);
switch(ctx->avctx->codec_id){
case CODEC_ID_AAC:
ctx->model = &ff_aac_psy_model;
break;
}
if(ctx->model->init)
return ctx->model->init(ctx);
return 0;
}
FFPsyWindowInfo ff_psy_suggest_window(FFPsyContext *ctx,
const int16_t *audio, const int16_t *la,
int channel, int prev_type)
{
return ctx->model->window(ctx, audio, la, channel, prev_type);
}
void ff_psy_set_band_info(FFPsyContext *ctx, int channel,
const float *coeffs, FFPsyWindowInfo *wi)
{
ctx->model->analyze(ctx, channel, coeffs, wi);
}
av_cold void ff_psy_end(FFPsyContext *ctx)
{
if(ctx->model->end)
ctx->model->end(ctx);
av_freep(&ctx->bands);
av_freep(&ctx->num_bands);
av_freep(&ctx->psy_bands);
}
typedef struct FFPsyPreprocessContext{
AVCodecContext *avctx;
float stereo_att;
struct FFIIRFilterCoeffs *fcoeffs;
struct FFIIRFilterState **fstate;
}FFPsyPreprocessContext;
#define FILT_ORDER 4
av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx)
{
FFPsyPreprocessContext *ctx;
int i;
float cutoff_coeff;
ctx = av_mallocz(sizeof(FFPsyPreprocessContext));
ctx->avctx = avctx;
if(avctx->flags & CODEC_FLAG_QSCALE)
cutoff_coeff = 1.0f / av_clip(1 + avctx->global_quality / FF_QUALITY_SCALE, 1, 8);
else
cutoff_coeff = avctx->bit_rate / (4.0f * avctx->sample_rate * avctx->channels);
ctx->fcoeffs = ff_iir_filter_init_coeffs(FF_FILTER_TYPE_BUTTERWORTH, FF_FILTER_MODE_LOWPASS,
FILT_ORDER, cutoff_coeff, 0.0, 0.0);
if(ctx->fcoeffs){
ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels);
for(i = 0; i < avctx->channels; i++)
ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER);
}
return ctx;
}
void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
const int16_t *audio, int16_t *dest,
int tag, int channels)
{
int ch, i;
if(ctx->fstate){
for(ch = 0; ch < channels; ch++){
ff_iir_filter(ctx->fcoeffs, ctx->fstate[tag+ch], ctx->avctx->frame_size,
audio + ch, ctx->avctx->channels,
dest + ch, ctx->avctx->channels);
}
}else{
for(ch = 0; ch < channels; ch++){
for(i = 0; i < ctx->avctx->frame_size; i++)
dest[i*ctx->avctx->channels + ch] = audio[i*ctx->avctx->channels + ch];
}
}
}
av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
{
int i;
ff_iir_filter_free_coeffs(ctx->fcoeffs);
if (ctx->fstate)
for (i = 0; i < ctx->avctx->channels; i++)
ff_iir_filter_free_state(ctx->fstate[i]);
av_freep(&ctx->fstate);
}

158
libavcodec/psymodel.h Normal file
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@ -0,0 +1,158 @@
/*
* audio encoder psychoacoustic model
* 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
*/
#ifndef AVCODEC_PSYMODEL_H
#define AVCODEC_PSYMODEL_H
#include "avcodec.h"
/** maximum possible number of bands */
#define PSY_MAX_BANDS 128
/**
* single band psychoacoustic information
*/
typedef struct FFPsyBand{
int bits;
float energy;
float threshold;
float distortion;
float perceptual_weight;
}FFPsyBand;
/**
* windowing related information
*/
typedef struct FFPsyWindowInfo{
int window_type[3]; ///< window type (short/long/transitional, etc.) - current, previous and next
int window_shape; ///< window shape (sine/KBD/whatever)
int num_windows; ///< number of windows in a frame
int grouping[8]; ///< window grouping (for e.g. AAC)
int *window_sizes; ///< sequence of window sizes inside one frame (for eg. WMA)
}FFPsyWindowInfo;
/**
* context used by psychoacoustic model
*/
typedef struct FFPsyContext{
AVCodecContext *avctx; ///< encoder context
const struct FFPsyModel *model; ///< encoder-specific model functions
FFPsyBand *psy_bands; ///< frame bands information
uint8_t **bands; ///< scalefactor band sizes for possible frame sizes
int *num_bands; ///< number of scalefactor bands for possible frame sizes
int num_lens; ///< number of scalefactor band sets
void* model_priv_data; ///< psychoacoustic model implementation private data
}FFPsyContext;
/**
* codec-specific psychoacoustic model implementation
*/
typedef struct FFPsyModel {
const char *name;
int (*init) (FFPsyContext *apc);
FFPsyWindowInfo (*window)(FFPsyContext *ctx, const int16_t *audio, const int16_t *la, int channel, int prev_type);
void (*analyze)(FFPsyContext *ctx, int channel, const float *coeffs, FFPsyWindowInfo *wi);
void (*end) (FFPsyContext *apc);
}FFPsyModel;
/**
* Initialize psychoacoustic model.
*
* @param ctx model context
* @param avctx codec context
* @param num_lens number of possible frame lengths
* @param bands scalefactor band lengths for all frame lengths
* @param num_bands number of scalefactor bands for all frame lengths
*
* @return zero if successful, a negative value if not
*/
av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx,
int num_lens,
const uint8_t **bands, const int* num_bands);
/**
* Suggest window sequence for channel.
*
* @param ctx model context
* @param audio samples for the current frame
* @param la lookahead samples (NULL when unavailable)
* @param channel number of channel element to analyze
* @param prev_type previous window type
*
* @return suggested window information in a structure
*/
FFPsyWindowInfo ff_psy_suggest_window(FFPsyContext *ctx,
const int16_t *audio, const int16_t *la,
int channel, int prev_type);
/**
* Perform psychoacoustic analysis and set band info (threshold, energy).
*
* @param ctx model context
* @param channel audio channel number
* @param coeffs pointer to the transformed coefficients
* @param wi window information
*/
void ff_psy_set_band_info(FFPsyContext *ctx, int channel, const float *coeffs,
FFPsyWindowInfo *wi);
/**
* Cleanup model context at the end.
*
* @param ctx model context
*/
av_cold void ff_psy_end(FFPsyContext *ctx);
/**************************************************************************
* Audio preprocessing stuff. *
* This should be moved into some audio filter eventually. *
**************************************************************************/
struct FFPsyPreprocessContext;
/**
* psychoacoustic model audio preprocessing initialization
*/
av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx);
/**
* Preprocess several channel in audio frame in order to compress it better.
*
* @param ctx preprocessing context
* @param audio samples to preprocess
* @param dest place to put filtered samples
* @param tag channel number
* @param channels number of channel to preprocess (some additional work may be done on stereo pair)
*/
void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
const int16_t *audio, int16_t *dest,
int tag, int channels);
/**
* Cleanup audio preprocessing module.
*/
av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx);
#endif /* AVCODEC_PSYMODEL_H */