Commit 78e65cd7 authored by Alex Converse's avatar Alex Converse
Browse files

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

Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent 5e039e1b
......@@ -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
......
......@@ -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
......
......@@ -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
......
This diff is collapsed.
......@@ -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;
......