Mercurial > libavcodec.hg
changeset 7584:d8717018ac03 libavcodec
Add approved chunks to AAC encoder
author | kostya |
---|---|
date | Sat, 16 Aug 2008 05:47:18 +0000 |
parents | 2a3f40605dec |
children | 12976f458c7a |
files | aacenc.c |
diffstat | 1 files changed, 58 insertions(+), 2 deletions(-) [+] |
line wrap: on
line diff
--- a/aacenc.c Sat Aug 16 01:25:12 2008 +0000 +++ b/aacenc.c Sat Aug 16 05:47:18 2008 +0000 @@ -27,8 +27,7 @@ /*********************************** * TODOs: * psy model selection with some option - * change greedy codebook search into something more optimal, like Viterbi algorithm - * determine run lengths along with codebook + * add sane pulse detection ***********************************/ #include "avcodec.h" @@ -130,6 +129,16 @@ }; /** + * 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. * @see 1.6.2.1 "Syntax - AudioSpecificConfig" */ @@ -176,6 +185,11 @@ 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]); @@ -212,6 +226,48 @@ } /** + * 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. */ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, const char *name)