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Add approved chunks to AAC encoder

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Originally committed as revision 14785 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Kostya Shishkov 2008-08-16 05:47:18 +00:00
parent 38a1c7f2be
commit 817015e4e2
1 changed files with 58 additions and 2 deletions
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60
libavcodec/aacenc.c
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@ -27,8 +27,7 @@
/*********************************** /***********************************
* TODOs: * TODOs:
* psy model selection with some option * psy model selection with some option
* change greedy codebook search into something more optimal, like Viterbi algorithm * add sane pulse detection
* determine run lengths along with codebook
***********************************/ ***********************************/
#include "avcodec.h" #include "avcodec.h"
@ -129,6 +128,16 @@ static const uint8_t aac_chan_configs[6][5] = {
{4, ID_SCE, ID_CPE, ID_CPE, ID_LFE}, // 6 channels - front center + stereo + back stereo + LFE {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;
} AACEncContext;
/** /**
* Make AAC audio config object. * Make AAC audio config object.
* @see 1.6.2.1 "Syntax - AudioSpecificConfig" * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
@ -176,6 +185,11 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
dsputil_init(&s->dsp, avctx); dsputil_init(&s->dsp, avctx);
ff_mdct_init(&s->mdct1024, 11, 0); ff_mdct_init(&s->mdct1024, 11, 0);
ff_mdct_init(&s->mdct128, 8, 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->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]); s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
@ -211,6 +225,48 @@ static void put_ics_info(AVCodecContext *avctx, IndividualChannelStream *info)
} }
} }
/**
* Encode pulse data.
*/
static void encode_pulses(AVCodecContext *avctx, AACEncContext *s, Pulse *pulse, int channel)
{
int i;
put_bits(&s->pb, 1, !!pulse->num_pulse);
if(!pulse->num_pulse) return;
put_bits(&s->pb, 2, pulse->num_pulse - 1);
put_bits(&s->pb, 6, pulse->start);
for(i = 0; i < pulse->num_pulse; i++){
put_bits(&s->pb, 5, pulse->offset[i]);
put_bits(&s->pb, 4, pulse->amp[i]);
}
}
/**
* Encode spectral coefficients processed by psychoacoustic model.
*/
static void encode_spectral_coeffs(AVCodecContext *avctx, AACEncContext *s, ChannelElement *cpe, int channel)
{
int start, i, w, w2, wg;
w = 0;
for(wg = 0; wg < cpe->ch[channel].ics.num_window_groups; wg++){
start = 0;
for(i = 0; i < cpe->ch[channel].ics.max_sfb; i++){
if(cpe->ch[channel].zeroes[w][i]){
start += cpe->ch[channel].ics.swb_sizes[i];
continue;
}
for(w2 = w; w2 < w + cpe->ch[channel].ics.group_len[wg]; w2++){
encode_band_coeffs(s, cpe, channel, start + w2*128, cpe->ch[channel].ics.swb_sizes[i], cpe->ch[channel].band_type[w][i]);
}
start += cpe->ch[channel].ics.swb_sizes[i];
}
w += cpe->ch[channel].ics.group_len[wg];
}
}
/** /**
* Write some auxiliary information about the created AAC file. * Write some auxiliary information about the created AAC file.
*/ */