/***********************************
* 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
+ * add temporal noise shaping
***********************************/
#include "avcodec.h"
32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
};
-static const uint8_t *swb_size_1024[] = {
+static const uint8_t * const 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,
4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
};
-static const uint8_t *swb_size_128[] = {
+static const uint8_t * const 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,
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, ID_SCE}, // 1 channel - single channel element
- {1, ID_CPE}, // 2 channels - channel pair
- {2, ID_SCE, ID_CPE}, // 3 channels - center + stereo
- {3, ID_SCE, ID_CPE, ID_SCE}, // 4 channels - front center + stereo + back center
- {3, ID_SCE, ID_CPE, ID_CPE}, // 5 channels - front center + stereo + back stereo
- {4, ID_SCE, ID_CPE, ID_CPE, ID_LFE}, // 6 channels - front center + stereo + back stereo + LFE
+ {1, TYPE_SCE}, // 1 channel - single channel element
+ {1, TYPE_CPE}, // 2 channels - channel pair
+ {2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo
+ {3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center
+ {3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo
+ {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"
return -1;
}
s->samplerate_index = i;
- s->swb_sizes1024 = swb_size_1024[i];
- s->swb_num1024 = ff_aac_num_swb_1024[i];
- s->swb_sizes128 = swb_size_128[i];
- s->swb_num128 = ff_aac_num_swb_128[i];
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]);
- if(ff_aac_psy_init(&s->psy, avctx, AAC_PSY_3GPP, aac_chan_configs[avctx->channels-1][0], 0, s->swb_sizes1024, s->swb_num1024, s->swb_sizes128, s->swb_num128) < 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;
}
* Encode ics_info element.
* @see Table 4.6 (syntax of ics_info)
*/
-static void put_ics_info(AVCodecContext *avctx, IndividualChannelStream *info)
+static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
{
- AACEncContext *s = avctx->priv_data;
int i;
put_bits(&s->pb, 1, 0); // ics_reserved bit
}
}
+/**
+ * Calculate the number of bits needed to code all coefficient signs in current band.
+ */
+static int calculate_band_sign_bits(AACEncContext *s, SingleChannelElement *sce,
+ int group_len, int start, int size)
+{
+ 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++;
+ }
+ start += 128;
+ }
+ return bits;
+}
+
+/**
+ * Encode pulse data.
+ */
+static void encode_pulses(AACEncContext *s, Pulse *pulse)
+{
+ 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->pos[i]);
+ put_bits(&s->pb, 4, pulse->amp[i]);
+ }
+}
+
+/**
+ * Encode spectral coefficients processed by psychoacoustic model.
+ */
+static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
+{
+ int start, i, w, w2, wg;
+
+ w = 0;
+ for(wg = 0; wg < sce->ics.num_window_groups; wg++){
+ 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]);
+ }
+ start += sce->ics.swb_sizes[i];
+ }
+ w += sce->ics.group_len[wg];
+ }
+}
+
/**
* Write some auxiliary information about the created AAC file.
*/
int i, namelen, padbits;
namelen = strlen(name) + 2;
- put_bits(&s->pb, 3, ID_FIL);
+ put_bits(&s->pb, 3, TYPE_FIL);
put_bits(&s->pb, 4, FFMIN(namelen, 15));
if(namelen >= 15)
put_bits(&s->pb, 8, namelen - 16);
projects / ffmpeg / blobdiff