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

Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk

Changelog

@@ -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
 
 
 

libavcodec/Makefile

@@ -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

libavcodec/aac.h

@@ -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

libavcodec/aacenc.c

@@ -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;

libavcodec/aacenc.h

@@ -0,0 +1,71 @@
+/*
+ * AAC encoder
+ * Copyright (C) 2008 Konstantin Shishkov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AACENC_H
+#define AVCODEC_AACENC_H
+
+#include "avcodec.h"
+#include "put_bits.h"
+#include "dsputil.h"
+
+#include "aac.h"
+
+#include "psymodel.h"
+
+struct AACEncContext;
+
+typedef struct AACCoefficientsEncoder{
+    void (*search_for_quantizers)(AVCodecContext *avctx, struct AACEncContext *s,
+                                  SingleChannelElement *sce, const float lambda);
+    void (*encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce,
+                                     int win, int group_len, const float lambda);
+    void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, int size,
+                                     int scale_idx, int cb, const float lambda);
+    void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe, const float lambda);
+}AACCoefficientsEncoder;
+
+extern AACCoefficientsEncoder ff_aac_coders[];
+
+/**
+ * AAC encoder context
+ */
+typedef struct AACEncContext {
+    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
+    FFPsyContext psy;
+    struct FFPsyPreprocessContext* psypp;
+    AACCoefficientsEncoder *coder;
+    int cur_channel;
+    int last_frame;
+    float lambda;
+    DECLARE_ALIGNED_16(int,   qcoefs[96][2]);    ///< quantized coefficients
+    DECLARE_ALIGNED_16(float, scoefs[1024]);     ///< scaled coefficients
+} AACEncContext;
+
+#endif /* AVCODEC_AACENC_H */

libavcodec/aacpsy.c

@@ -25,54 +25,25 @@
  */
 
 #include "avcodec.h"
-#include "aacpsy.h"
 #include "aactab.h"
+#include "psymodel.h"
 
 /***********************************
  *              TODOs:
- * General:
- * better audio preprocessing (add DC highpass filter?)
- * more psy models
- * maybe improve coefficient quantization function in some way
- *
- * 3GPP-based psy model:
  * thresholds linearization after their modifications for attaining given bitrate
  * try other bitrate controlling mechanism (maybe use ratecontrol.c?)
  * control quality for quality-based output
  **********************************/
 
-/**
- * Quantize one coefficient.
- * @return absolute value of the quantized coefficient
- * @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
- */
-static av_always_inline int quant(float coef, const float Q)
-{
-    return av_clip((int)(pow(fabsf(coef) * Q, 0.75) + 0.4054), 0, 8191);
-}
-
-static inline float get_approximate_quant_error(float *c, int size, int scale_idx)
-{
-    int i;
-    int q;
-    float coef, unquant, sum = 0.0f;
-    const float Q  = ff_aac_pow2sf_tab[200 - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
-    const float IQ = ff_aac_pow2sf_tab[200 + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
-    for(i = 0; i < size; i++){
-        coef = fabs(c[i]);
-        q = quant(c[i], Q);
-        unquant = (q * cbrt(q)) * IQ;
-        sum += (coef - unquant) * (coef - unquant);
-    }
-    return sum;
-}
-
 /**
  * constants for 3GPP AAC psychoacoustic model
  * @{
  */
 #define PSY_3GPP_SPREAD_LOW  1.5f // spreading factor for ascending threshold spreading  (15 dB/Bark)
 #define PSY_3GPP_SPREAD_HI   3.0f // spreading factor for descending threshold spreading (30 dB/Bark)
+
+#define PSY_3GPP_RPEMIN      0.01f
+#define PSY_3GPP_RPELEV      2.0f
 /**
  * @}
  */
@@ -83,8 +54,24 @@ static inline float get_approximate_quant_error(float *c, int size, int scale_id
 typedef struct Psy3gppBand{
     float energy;    ///< band energy
     float ffac;      ///< form factor
+    float thr;       ///< energy threshold
+    float min_snr;   ///< minimal SNR
+    float thr_quiet; ///< threshold in quiet
 }Psy3gppBand;
 
+/**
+ * single/pair channel context for psychoacoustic model
+ */
+typedef struct Psy3gppChannel{
+    Psy3gppBand band[128];               ///< bands information
+    Psy3gppBand prev_band[128];          ///< bands information from the previous frame
+
+    float       win_energy;              ///< sliding average of channel energy
+    float       iir_state[2];            ///< hi-pass IIR filter state
+    uint8_t     next_grouping;           ///< stored grouping scheme for the next frame (in case of 8 short window sequence)
+    enum WindowSequence next_window_seq; ///< window sequence to be used in the next frame
+}Psy3gppChannel;
+
 /**
  * psychoacoustic model frame type-dependent coefficients
  */
@@ -95,10 +82,241 @@ typedef struct Psy3gppCoeffs{
     float spread_hi [64]; ///< spreading factor for high-to-low threshold spreading in long frame
 }Psy3gppCoeffs;
 
+/**
+ * 3GPP TS26.403-inspired psychoacoustic model specific data
+ */
+typedef struct Psy3gppContext{
+    Psy3gppCoeffs psy_coef[2];
+    Psy3gppChannel *ch;
+}Psy3gppContext;
+
 /**
  * Calculate Bark value for given line.
  */
-static inline float calc_bark(float f)
+static av_cold float calc_bark(float f)
 {
     return 13.3f * atanf(0.00076f * f) + 3.5f * atanf((f / 7500.0f) * (f / 7500.0f));
 }
+
+#define ATH_ADD 4
+/**
+ * Calculate ATH value for given frequency.
+ * Borrowed from Lame.
+ */
+static av_cold float ath(float f, float add)
+{
+    f /= 1000.0f;
+    return   3.64 * pow(f, -0.8)
+            - 6.8  * exp(-0.6  * (f - 3.4) * (f - 3.4))
+            + 6.0  * exp(-0.15 * (f - 8.7) * (f - 8.7))
+            + (0.6 + 0.04 * add) * 0.001 * f * f * f * f;
+}
+
+static av_cold int psy_3gpp_init(FFPsyContext *ctx){
+    Psy3gppContext *pctx;
+    float barks[1024];
+    int i, j, g, start;
+    float prev, minscale, minath;
+
+    ctx->model_priv_data = av_mallocz(sizeof(Psy3gppContext));
+    pctx = (Psy3gppContext*) ctx->model_priv_data;
+
+    for(i = 0; i < 1024; i++)
+        barks[i] = calc_bark(i * ctx->avctx->sample_rate / 2048.0);
+    minath = ath(3410, ATH_ADD);
+    for(j = 0; j < 2; j++){
+        Psy3gppCoeffs *coeffs = &pctx->psy_coef[j];
+        i = 0;
+        prev = 0.0;
+        for(g = 0; g < ctx->num_bands[j]; g++){
+            i += ctx->bands[j][g];
+            coeffs->barks[g] = (barks[i - 1] + prev) / 2.0;
+            prev = barks[i - 1];
+        }
+        for(g = 0; g < ctx->num_bands[j] - 1; g++){
+            coeffs->spread_low[g] = pow(10.0, -(coeffs->barks[g+1] - coeffs->barks[g]) * PSY_3GPP_SPREAD_LOW);
+            coeffs->spread_hi [g] = pow(10.0, -(coeffs->barks[g+1] - coeffs->barks[g]) * PSY_3GPP_SPREAD_HI);
+        }
+        start = 0;
+        for(g = 0; g < ctx->num_bands[j]; g++){
+            minscale = ath(ctx->avctx->sample_rate * start / 1024.0, ATH_ADD);
+            for(i = 1; i < ctx->bands[j][g]; i++){
+                minscale = fminf(minscale, ath(ctx->avctx->sample_rate * (start + i) / 1024.0 / 2.0, ATH_ADD));
+            }
+            coeffs->ath[g] = minscale - minath;
+            start += ctx->bands[j][g];
+        }
+    }
+
+    pctx->ch = av_mallocz(sizeof(Psy3gppChannel) * ctx->avctx->channels);
+    return 0;
+}
+
+/**
+ * IIR filter used in block switching decision
+ */
+static float iir_filter(int in, float state[2])
+{
+    float ret;
+
+    ret = 0.7548f * (in - state[0]) + 0.5095f * state[1];
+    state[0] = in;
+    state[1] = ret;
+    return ret;
+}
+
+/**
+ * window grouping information stored as bits (0 - new group, 1 - group continues)
+ */
+static const uint8_t window_grouping[9] = {
+    0xB6, 0x6C, 0xD8, 0xB2, 0x66, 0xC6, 0x96, 0x36, 0x36
+};
+
+/**
+ * Tell encoder which window types to use.
+ * @see 3GPP TS26.403 5.4.1 "Blockswitching"
+ */
+static FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
+                                       const int16_t *audio, const int16_t *la,
+                                       int channel, int prev_type)
+{
+    int i, j;
+    int br = ctx->avctx->bit_rate / ctx->avctx->channels;
+    int attack_ratio = br <= 16000 ? 18 : 10;
+    Psy3gppContext *pctx = (Psy3gppContext*) ctx->model_priv_data;
+    Psy3gppChannel *pch = &pctx->ch[channel];
+    uint8_t grouping = 0;
+    FFPsyWindowInfo wi;
+
+    memset(&wi, 0, sizeof(wi));
+    if(la){
+        float s[8], v;
+        int switch_to_eight = 0;
+        float sum = 0.0, sum2 = 0.0;
+        int attack_n = 0;
+        for(i = 0; i < 8; i++){
+            for(j = 0; j < 128; j++){
+                v = iir_filter(audio[(i*128+j)*ctx->avctx->channels], pch->iir_state);
+                sum += v*v;
+            }
+            s[i] = sum;
+            sum2 += sum;
+        }
+        for(i = 0; i < 8; i++){
+            if(s[i] > pch->win_energy * attack_ratio){
+                attack_n = i + 1;
+                switch_to_eight = 1;
+                break;
+            }
+        }
+        pch->win_energy = pch->win_energy*7/8 + sum2/64;
+
+        wi.window_type[1] = prev_type;
+        switch(prev_type){
+        case ONLY_LONG_SEQUENCE:
+            wi.window_type[0] = switch_to_eight ? LONG_START_SEQUENCE : ONLY_LONG_SEQUENCE;
+            break;
+        case LONG_START_SEQUENCE:
+            wi.window_type[0] = EIGHT_SHORT_SEQUENCE;
+            grouping = pch->next_grouping;
+            break;
+        case LONG_STOP_SEQUENCE:
+            wi.window_type[0] = ONLY_LONG_SEQUENCE;
+            break;
+        case EIGHT_SHORT_SEQUENCE:
+            wi.window_type[0] = switch_to_eight ? EIGHT_SHORT_SEQUENCE : LONG_STOP_SEQUENCE;
+            grouping = switch_to_eight ? pch->next_grouping : 0;
+            break;
+        }
+        pch->next_grouping = window_grouping[attack_n];
+    }else{
+        for(i = 0; i < 3; i++)
+            wi.window_type[i] = prev_type;
+        grouping = (prev_type == EIGHT_SHORT_SEQUENCE) ? window_grouping[0] : 0;
+    }
+
+    wi.window_shape   = 1;
+    if(wi.window_type[0] != EIGHT_SHORT_SEQUENCE){
+        wi.num_windows = 1;
+        wi.grouping[0] = 1;
+    }else{
+        int lastgrp = 0;
+        wi.num_windows = 8;
+        for(i = 0; i < 8; i++){
+            if(!((grouping >> i) & 1))
+                lastgrp = i;
+            wi.grouping[lastgrp]++;
+        }
+    }
+
+    return wi;
+}
+
+/**
+ * Calculate band thresholds as suggested in 3GPP TS26.403
+ */
+static void psy_3gpp_analyze(FFPsyContext *ctx, int channel, const float *coefs,
+                             FFPsyWindowInfo *wi)
+{
+    Psy3gppContext *pctx = (Psy3gppContext*) ctx->model_priv_data;
+    Psy3gppChannel *pch = &pctx->ch[channel];
+    int start = 0;
+    int i, w, g;
+    const int num_bands = ctx->num_bands[wi->num_windows == 8];
+    const uint8_t* band_sizes = ctx->bands[wi->num_windows == 8];
+    Psy3gppCoeffs *coeffs = &pctx->psy_coef[wi->num_windows == 8];
+
+    //calculate energies, initial thresholds and related values - 5.4.2 "Threshold Calculation"
+    for(w = 0; w < wi->num_windows*16; w += 16){
+        for(g = 0; g < num_bands; g++){
+            Psy3gppBand *band = &pch->band[w+g];
+            band->energy = 0.0f;
+            for(i = 0; i < band_sizes[g]; i++)
+                band->energy += coefs[start+i] * coefs[start+i];
+            band->energy *= 1.0f / (512*512);
+            band->thr = band->energy * 0.001258925f;
+            start += band_sizes[g];
+
+            ctx->psy_bands[channel*PSY_MAX_BANDS+w+g].energy = band->energy;
+        }
+    }
+    //modify thresholds - spread, threshold in quiet - 5.4.3 "Spreaded Energy Calculation"
+    for(w = 0; w < wi->num_windows*16; w += 16){
+        Psy3gppBand *band = &pch->band[w];
+        for(g = 1; g < num_bands; g++){
+            band[g].thr = FFMAX(band[g].thr, band[g-1].thr * coeffs->spread_low[g-1]);
+        }
+        for(g = num_bands - 2; g >= 0; g--){
+            band[g].thr = FFMAX(band[g].thr, band[g+1].thr * coeffs->spread_hi [g]);
+        }
+        for(g = 0; g < num_bands; g++){
+            band[g].thr_quiet = FFMAX(band[g].thr, coeffs->ath[g]);
+            if(wi->num_windows != 8 && wi->window_type[1] != EIGHT_SHORT_SEQUENCE){
+                band[g].thr_quiet = fmaxf(PSY_3GPP_RPEMIN*band[g].thr_quiet,
+                                          fminf(band[g].thr_quiet,
+                                          PSY_3GPP_RPELEV*pch->prev_band[w+g].thr_quiet));
+            }
+            band[g].thr = FFMAX(band[g].thr, band[g].thr_quiet * 0.25);
+
+            ctx->psy_bands[channel*PSY_MAX_BANDS+w+g].threshold = band[g].thr;
+        }
+    }
+    memcpy(pch->prev_band, pch->band, sizeof(pch->band));
+}
+
+static av_cold void psy_3gpp_end(FFPsyContext *apc)
+{
+    Psy3gppContext *pctx = (Psy3gppContext*) apc->model_priv_data;
+    av_freep(&pctx->ch);
+    av_freep(&apc->model_priv_data);
+}
+
+
+const FFPsyModel ff_aac_psy_model =
+{
+    .name    = "3GPP TS 26.403-inspired model",
+    .init    = psy_3gpp_init,
+    .window  = psy_3gpp_window,
+    .analyze = psy_3gpp_analyze,
+    .end     = psy_3gpp_end,
+};

libavcodec/allcodecs.c

@@ -195,7 +195,7 @@ void avcodec_register_all(void)
     REGISTER_ENCDEC  (ZMBV, zmbv);
 
     /* audio codecs */
-    REGISTER_DECODER (AAC, aac);
+    REGISTER_ENCDEC  (AAC, aac);
     REGISTER_ENCDEC  (AC3, ac3);
     REGISTER_ENCDEC  (ALAC, alac);
     REGISTER_DECODER (APE, ape);

libavcodec/psymodel.c

@@ -0,0 +1,130 @@
+/*
+ * audio encoder psychoacoustic model
+ * Copyright (C) 2008 Konstantin Shishkov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "avcodec.h"
+#include "psymodel.h"
+#include "iirfilter.h"
+
+extern const FFPsyModel ff_aac_psy_model;
+
+av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx,
+                        int num_lens,
+                        const uint8_t **bands, const int* num_bands)
+{
+    ctx->avctx = avctx;
+    ctx->psy_bands = av_mallocz(sizeof(FFPsyBand) * PSY_MAX_BANDS * avctx->channels);
+    ctx->bands     = av_malloc (sizeof(ctx->bands[0])     * num_lens);
+    ctx->num_bands = av_malloc (sizeof(ctx->num_bands[0]) * num_lens);
+    memcpy(ctx->bands,     bands,     sizeof(ctx->bands[0])     *  num_lens);
+    memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) *  num_lens);
+    switch(ctx->avctx->codec_id){
+    case CODEC_ID_AAC:
+        ctx->model = &ff_aac_psy_model;
+        break;
+    }
+    if(ctx->model->init)
+        return ctx->model->init(ctx);
+    return 0;
+}
+
+FFPsyWindowInfo ff_psy_suggest_window(FFPsyContext *ctx,
+                                      const int16_t *audio, const int16_t *la,
+                                      int channel, int prev_type)
+{
+    return ctx->model->window(ctx, audio, la, channel, prev_type);
+}
+
+void ff_psy_set_band_info(FFPsyContext *ctx, int channel,
+                          const float *coeffs, FFPsyWindowInfo *wi)
+{
+    ctx->model->analyze(ctx, channel, coeffs, wi);
+}
+
+av_cold void ff_psy_end(FFPsyContext *ctx)
+{
+    if(ctx->model->end)
+        ctx->model->end(ctx);
+    av_freep(&ctx->bands);
+    av_freep(&ctx->num_bands);
+    av_freep(&ctx->psy_bands);
+}
+
+typedef struct FFPsyPreprocessContext{
+    AVCodecContext *avctx;
+    float stereo_att;
+    struct FFIIRFilterCoeffs *fcoeffs;
+    struct FFIIRFilterState **fstate;
+}FFPsyPreprocessContext;
+
+#define FILT_ORDER 4
+
+av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx)
+{
+    FFPsyPreprocessContext *ctx;
+    int i;
+    float cutoff_coeff;
+    ctx = av_mallocz(sizeof(FFPsyPreprocessContext));
+    ctx->avctx = avctx;
+
+    if(avctx->flags & CODEC_FLAG_QSCALE)
+        cutoff_coeff = 1.0f / av_clip(1 + avctx->global_quality / FF_QUALITY_SCALE, 1, 8);
+    else
+        cutoff_coeff = avctx->bit_rate / (4.0f * avctx->sample_rate * avctx->channels);
+
+    ctx->fcoeffs = ff_iir_filter_init_coeffs(FF_FILTER_TYPE_BUTTERWORTH, FF_FILTER_MODE_LOWPASS,
+                                           FILT_ORDER, cutoff_coeff, 0.0, 0.0);
+    if(ctx->fcoeffs){
+        ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels);
+        for(i = 0; i < avctx->channels; i++)
+            ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER);
+    }
+    return ctx;
+}
+
+void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
+                       const int16_t *audio, int16_t *dest,
+                       int tag, int channels)
+{
+    int ch, i;
+    if(ctx->fstate){
+        for(ch = 0; ch < channels; ch++){
+            ff_iir_filter(ctx->fcoeffs, ctx->fstate[tag+ch], ctx->avctx->frame_size,
+                          audio + ch, ctx->avctx->channels,
+                          dest  + ch, ctx->avctx->channels);
+        }
+    }else{
+        for(ch = 0; ch < channels; ch++){
+            for(i = 0; i < ctx->avctx->frame_size; i++)
+                dest[i*ctx->avctx->channels + ch] = audio[i*ctx->avctx->channels + ch];
+        }
+    }
+}
+
+av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
+{
+    int i;
+    ff_iir_filter_free_coeffs(ctx->fcoeffs);
+    if (ctx->fstate)
+        for (i = 0; i < ctx->avctx->channels; i++)
+            ff_iir_filter_free_state(ctx->fstate[i]);
+    av_freep(&ctx->fstate);
+}
+

libavcodec/psymodel.h

@@ -0,0 +1,158 @@
+/*
+ * audio encoder psychoacoustic model
+ * Copyright (C) 2008 Konstantin Shishkov
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_PSYMODEL_H
+#define AVCODEC_PSYMODEL_H
+
+#include "avcodec.h"
+
+/** maximum possible number of bands */
+#define PSY_MAX_BANDS 128
+
+/**
+ * single band psychoacoustic information
+ */
+typedef struct FFPsyBand{
+    int   bits;
+    float energy;
+    float threshold;
+    float distortion;
+    float perceptual_weight;
+}FFPsyBand;
+
+/**
+ * windowing related information
+ */
+typedef struct FFPsyWindowInfo{
+    int window_type[3];               ///< window type (short/long/transitional, etc.) - current, previous and next
+    int window_shape;                 ///< window shape (sine/KBD/whatever)
+    int num_windows;                  ///< number of windows in a frame
+    int grouping[8];                  ///< window grouping (for e.g. AAC)
+    int *window_sizes;                ///< sequence of window sizes inside one frame (for eg. WMA)
+}FFPsyWindowInfo;
+
+/**
+ * context used by psychoacoustic model
+ */
+typedef struct FFPsyContext{
+    AVCodecContext *avctx;            ///< encoder context
+    const struct FFPsyModel *model;   ///< encoder-specific model functions
+
+    FFPsyBand *psy_bands;             ///< frame bands information
+
+    uint8_t **bands;                  ///< scalefactor band sizes for possible frame sizes
+    int     *num_bands;               ///< number of scalefactor bands for possible frame sizes
+    int num_lens;                     ///< number of scalefactor band sets
+
+    void* model_priv_data;            ///< psychoacoustic model implementation private data
+}FFPsyContext;
+
+/**
+ * codec-specific psychoacoustic model implementation
+ */
+typedef struct FFPsyModel {
+    const char *name;
+    int  (*init)   (FFPsyContext *apc);
+    FFPsyWindowInfo (*window)(FFPsyContext *ctx, const int16_t *audio, const int16_t *la, int channel, int prev_type);
+    void (*analyze)(FFPsyContext *ctx, int channel, const float *coeffs, FFPsyWindowInfo *wi);
+    void (*end)    (FFPsyContext *apc);
+}FFPsyModel;
+
+/**
+ * Initialize psychoacoustic model.
+ *
+ * @param ctx        model context
+ * @param avctx      codec context
+ * @param num_lens   number of possible frame lengths
+ * @param bands      scalefactor band lengths for all frame lengths
+ * @param num_bands  number of scalefactor bands for all frame lengths
+ *
+ * @return zero if successful, a negative value if not
+ */
+av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx,
+                        int num_lens,
+                        const uint8_t **bands, const int* num_bands);
+
+/**
+ * Suggest window sequence for channel.
+ *
+ * @param ctx       model context
+ * @param audio     samples for the current frame
+ * @param la        lookahead samples (NULL when unavailable)
+ * @param channel   number of channel element to analyze
+ * @param prev_type previous window type
+ *
+ * @return suggested window information in a structure
+ */
+FFPsyWindowInfo ff_psy_suggest_window(FFPsyContext *ctx,
+                                      const int16_t *audio, const int16_t *la,
+                                      int channel, int prev_type);
+
+
+/**
+ * Perform psychoacoustic analysis and set band info (threshold, energy).
+ *
+ * @param ctx     model context
+ * @param channel audio channel number
+ * @param coeffs  pointer to the transformed coefficients
+ * @param wi      window information
+ */
+void ff_psy_set_band_info(FFPsyContext *ctx, int channel, const float *coeffs,
+                          FFPsyWindowInfo *wi);
+
+/**
+ * Cleanup model context at the end.
+ *
+ * @param ctx model context
+ */
+av_cold void ff_psy_end(FFPsyContext *ctx);
+
+
+/**************************************************************************
+ *                       Audio preprocessing stuff.                       *
+ *       This should be moved into some audio filter eventually.          *
+ **************************************************************************/
+struct FFPsyPreprocessContext;
+
+/**
+ * psychoacoustic model audio preprocessing initialization
+ */
+av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx);
+
+/**
+ * Preprocess several channel in audio frame in order to compress it better.
+ *
+ * @param ctx      preprocessing context
+ * @param audio    samples to preprocess
+ * @param dest     place to put filtered samples
+ * @param tag      channel number
+ * @param channels number of channel to preprocess (some additional work may be done on stereo pair)
+ */
+void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
+                       const int16_t *audio, int16_t *dest,
+                       int tag, int channels);
+
+/**
+ * Cleanup audio preprocessing module.
+ */
+av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx);
+
+#endif /* AVCODEC_PSYMODEL_H */