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More okayed chunks of AAC encoder 

Originally committed as revision 14799 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Kostya Shishkov 2008-08-17 07:04:11 +00:00
parent 6b93147684
commit 720d52c678
1 changed files with 30 additions and 2 deletions
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32
libavcodec/aacenc.c
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@ -118,6 +118,23 @@ static const uint8_t *swb_size_128[] = {
swb_size_128_16, swb_size_128_16, swb_size_128_8 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* run_value_bits[2] = {
run_value_bits_long, run_value_bits_short
};
/** default channel configurations */ /** default channel configurations */
static const uint8_t aac_chan_configs[6][5] = { static const uint8_t aac_chan_configs[6][5] = {
{1, TYPE_SCE}, // 1 channel - single channel element {1, TYPE_SCE}, // 1 channel - single channel element
@ -128,6 +145,15 @@ 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 {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 * AAC encoder context
*/ */
@ -136,6 +162,8 @@ typedef struct {
MDCTContext mdct1024; ///< long (1024 samples) frame transform context MDCTContext mdct1024; ///< long (1024 samples) frame transform context
MDCTContext mdct128; ///< short (128 samples) frame transform context MDCTContext mdct128; ///< short (128 samples) frame transform context
DSPContext dsp; DSPContext dsp;
AACPsyContext psy; ///< psychoacoustic model context
int last_frame;
} AACEncContext; } AACEncContext;
/** /**
@ -254,12 +282,12 @@ static void encode_spectral_coeffs(AVCodecContext *avctx, AACEncContext *s, Chan
for(wg = 0; wg < cpe->ch[channel].ics.num_window_groups; wg++){ for(wg = 0; wg < cpe->ch[channel].ics.num_window_groups; wg++){
start = 0; start = 0;
for(i = 0; i < cpe->ch[channel].ics.max_sfb; i++){ for(i = 0; i < cpe->ch[channel].ics.max_sfb; i++){
if(cpe->ch[channel].zeroes[w][i]){ if(cpe->ch[channel].zeroes[w*16 + i]){
start += cpe->ch[channel].ics.swb_sizes[i]; start += cpe->ch[channel].ics.swb_sizes[i];
continue; continue;
} }
for(w2 = w; w2 < w + cpe->ch[channel].ics.group_len[wg]; w2++){ 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]); encode_band_coeffs(s, cpe, channel, start + w2*128, cpe->ch[channel].ics.swb_sizes[i], cpe->ch[channel].band_type[w*16 + i]);
} }
start += cpe->ch[channel].ics.swb_sizes[i]; start += cpe->ch[channel].ics.swb_sizes[i];
} }