ood-ports-lbsd/graphics/ffmpeg/patches/patch-libavcodec_aacenc_c

$OpenBSD: patch-libavcodec_aacenc_c,v 1.19 2016/12/05 09:02:29 ajacoutot Exp $

aacenc: copy PRNG from the decoder

avcodec/aacenc: use AV_OPT_TYPE_BOOL

AAC encoder: tweak rate-distortion logic

AAC encoder: Extensive improvements

AAC encoder: memoize quantize_band_cost

aacenc: add support for encoding 7.1 channel audio

aacenc: add support for changing options based on a profile

aacenc: shorten name of ff_aac_adjust_common_prediction

aacenc: indicate that TNS is off by default

aacenc: slightly simplify and remove a redundant variable

aacenc: correctly zero prediction_used array

aacenc: (re)enable Mid/Side coding by default

aacenc: add support for encoding files using Long Term Prediction

aacenc: partially revert previous commits to set options via a profile

aacenc_tns: enable Temporal Noise Shaping by default

avcodec/aacenc: Fix "libavcodec/aacenc.c:540:13: warning: ISO C90 forbids mixed declarations and code"

AAC encoder: Fix application of M/S with PNS

AAC encoder: improve SF range utilization

aac: Provide more information on the failure message

aacenc: mark the "faac"-like coder for removal

aacenc: mark coders other than twoloop as experimental

aacenc: remove the experimental flag

aacenc: fix aac_pred option triggering an error

aacenc: move the TNS search and filtering before PNS

aacenc: switch to using the RNG from libavutil

AAC encoder: don't apply MS on special bands

acenc: remove deprecated avctx->frame_bits use

avcodec/aacenc: Fix NAN check

avcodec/aacenc: mark output as const as its not written to

avcodec/aacenc: Check for +-Inf too

lavc/aacenc: use isfinite to simplify isnan/isinf logic

aacenc: mark LTP mode as experimental

aacenc: remove FAAC-like coder

avcodec/aacenc: Check all coefficients for finiteness

aacenc: make a better estimate for the audio bitrate if not provided

aacenc: temporarily disable Mid/Side coding with multichannel files

aacenc: use generational cache instead of resetting.

AAC encoder: fix valgrind errors

aacenc: unmark the fast coder as experimental

aacenc: fix various typos and an error message

aacenc: use the decoder's lcg PRNG

aacenc: quit when the audio queue reaches 0 rather than keeping track of empty frames

--- libavcodec/aacenc.c.orig	Sat Aug 27 22:51:29 2016
+++ libavcodec/aacenc.c	Thu Nov 10 19:22:09 2016
@@ -29,6 +29,7 @@
  * add sane pulse detection
  ***********************************/
 
+#include "libavutil/libm.h"
 #include "libavutil/float_dsp.h"
 #include "libavutil/opt.h"
 #include "avcodec.h"
@@ -54,11 +55,12 @@ static void put_audio_specific_config(AVCodecContext *
 {
     PutBitContext pb;
     AACEncContext *s = avctx->priv_data;
+    int channels = s->channels - (s->channels == 8 ? 1 : 0);
 
     init_put_bits(&pb, avctx->extradata, avctx->extradata_size);
     put_bits(&pb, 5, s->profile+1); //profile
     put_bits(&pb, 4, s->samplerate_index); //sample rate index
-    put_bits(&pb, 4, s->channels);
+    put_bits(&pb, 4, channels);
     //GASpecificConfig
     put_bits(&pb, 1, 0); //frame length - 1024 samples
     put_bits(&pb, 1, 0); //does not depend on core coder
@@ -71,6 +73,15 @@ static void put_audio_specific_config(AVCodecContext *
     flush_put_bits(&pb);
 }
 
+void ff_quantize_band_cost_cache_init(struct AACEncContext *s)
+{
+    ++s->quantize_band_cost_cache_generation;
+    if (s->quantize_band_cost_cache_generation == 0) {
+        memset(s->quantize_band_cost_cache, 0, sizeof(s->quantize_band_cost_cache));
+        s->quantize_band_cost_cache_generation = 1;
+    }
+}
+
 #define WINDOW_FUNC(type) \
 static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
                                     SingleChannelElement *sce, \
@@ -140,7 +151,7 @@ static void apply_window_and_mdct(AACEncContext *s, Si
                                   float *audio)
 {
     int i;
-    float *output = sce->ret_buf;
+    const float *output = sce->ret_buf;
 
     apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, audio);
 
@@ -258,6 +269,8 @@ static void apply_intensity_stereo(ChannelElement *cpe
                     start += ics->swb_sizes[g];
                     continue;
                 }
+                if (cpe->ms_mask[w*16 + g])
+                    p *= -1;
                 for (i = 0; i < ics->swb_sizes[g]; i++) {
                     float sum = (cpe->ch[0].coeffs[start+i] + p*cpe->ch[1].coeffs[start+i])*scale;
                     cpe->ch[0].coeffs[start+i] = sum;
@@ -279,7 +292,13 @@ static void apply_mid_side_stereo(ChannelElement *cpe)
         for (w2 =  0; w2 < ics->group_len[w]; w2++) {
             int start = (w+w2) * 128;
             for (g = 0; g < ics->num_swb; g++) {
-                if (!cpe->ms_mask[w*16 + g]) {
+                /* ms_mask can be used for other purposes in PNS and I/S,
+                 * so must not apply M/S if any band uses either, even if
+                 * ms_mask is set.
+                 */
+                if (!cpe->ms_mask[w*16 + g] || cpe->is_mask[w*16 + g]
+                    || cpe->ch[0].band_type[w*16 + g] >= NOISE_BT
+                    || cpe->ch[1].band_type[w*16 + g] >= NOISE_BT) {
                     start += ics->swb_sizes[g];
                     continue;
                 }
@@ -424,6 +443,8 @@ static int encode_individual_channel(AVCodecContext *a
         put_ics_info(s, &sce->ics);
         if (s->coder->encode_main_pred)
             s->coder->encode_main_pred(s, sce);
+        if (s->coder->encode_ltp_info)
+            s->coder->encode_ltp_info(s, sce, 0);
     }
     encode_band_info(s, sce);
     encode_scale_factors(avctx, s, sce);
@@ -489,19 +510,21 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
     float **samples = s->planar_samples, *samples2, *la, *overlap;
     ChannelElement *cpe;
     SingleChannelElement *sce;
-    int i, ch, w, chans, tag, start_ch, ret;
+    IndividualChannelStream *ics;
+    int i, its, ch, w, chans, tag, start_ch, ret, frame_bits;
+    int target_bits, rate_bits, too_many_bits, too_few_bits;
     int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;
     int chan_el_counter[4];
     FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
     int k;
 
-    if (s->last_frame == 2)
-        return 0;
-
     /* add current frame to queue */
     if (frame) {
         if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
             return ret;
+    } else {
+        if (!s->afq.remaining_samples || (!s->afq.frame_alloc && !s->afq.frame_count))
+            return 0;
     }
 
     copy_input_samples(s, frame);
@@ -518,19 +541,22 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
         chans    = tag == TYPE_CPE ? 2 : 1;
         cpe      = &s->cpe[i];
         for (ch = 0; ch < chans; ch++) {
-            IndividualChannelStream *ics = &cpe->ch[ch].ics;
-            int cur_channel = start_ch + ch;
+            int k;
             float clip_avoidance_factor;
-            overlap  = &samples[cur_channel][0];
+            sce = &cpe->ch[ch];
+            ics = &sce->ics;
+            s->cur_channel = start_ch + ch;
+            overlap  = &samples[s->cur_channel][0];
             samples2 = overlap + 1024;
             la       = samples2 + (448+64);
             if (!frame)
                 la = NULL;
             if (tag == TYPE_LFE) {
-                wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
+                wi[ch].window_type[0] = wi[ch].window_type[1] = ONLY_LONG_SEQUENCE;
                 wi[ch].window_shape   = 0;
                 wi[ch].num_windows    = 1;
                 wi[ch].grouping[0]    = 1;
+                wi[ch].clipping[0]    = 0;
 
                 /* Only the lowest 12 coefficients are used in a LFE channel.
                  * The expression below results in only the bottom 8 coefficients
@@ -538,7 +564,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
                  */
                 ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
             } else {
-                wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel,
+                wi[ch] = s->psy.model->window(&s->psy, samples2, la, s->cur_channel,
                                               ics->window_sequence[0]);
             }
             ics->window_sequence[1] = ics->window_sequence[0];
@@ -555,10 +581,23 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
             ics->tns_max_bands      = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
                                         ff_tns_max_bands_128 [s->samplerate_index]:
                                         ff_tns_max_bands_1024[s->samplerate_index];
-            clip_avoidance_factor = 0.0f;
+
             for (w = 0; w < ics->num_windows; w++)
                 ics->group_len[w] = wi[ch].grouping[w];
+
+            /* Calculate input sample maximums and evaluate clipping risk */
+            clip_avoidance_factor = 0.0f;
             for (w = 0; w < ics->num_windows; w++) {
+                const float *wbuf = overlap + w * 128;
+                const int wlen = 2048 / ics->num_windows;
+                float max = 0;
+                int j;
+                /* mdct input is 2 * output */
+                for (j = 0; j < wlen; j++)
+                    max = FFMAX(max, fabsf(wbuf[j]));
+                wi[ch].clipping[w] = max;
+            }
+            for (w = 0; w < ics->num_windows; w++) {
                 if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) {
                     ics->window_clipping[w] = 1;
                     clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]);
@@ -610,15 +649,28 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
                 sce = &cpe->ch[ch];
                 coeffs[ch] = sce->coeffs;
                 sce->ics.predictor_present = 0;
-                memset(&sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
+                sce->ics.ltp.present = 0;
+                memset(sce->ics.ltp.used, 0, sizeof(sce->ics.ltp.used));
+                memset(sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
                 memset(&sce->tns, 0, sizeof(TemporalNoiseShaping));
                 for (w = 0; w < 128; w++)
                     if (sce->band_type[w] > RESERVED_BT)
                         sce->band_type[w] = 0;
             }
+            s->psy.bitres.alloc = -1;
+            s->psy.bitres.bits = s->last_frame_pb_count / s->channels;
             s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
+            if (s->psy.bitres.alloc > 0) {
+                /* Lambda unused here on purpose, we need to take psy's unscaled allocation */
+                target_bits += s->psy.bitres.alloc
+                    * (s->lambda / (avctx->global_quality ? avctx->global_quality : 120));
+                s->psy.bitres.alloc /= chans;
+            }
+            s->cur_type = tag;
             for (ch = 0; ch < chans; ch++) {
                 s->cur_channel = start_ch + ch;
+                if (s->options.pns && s->coder->mark_pns)
+                    s->coder->mark_pns(s, avctx, &cpe->ch[ch]);
                 s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
             }
             if (chans > 1
@@ -636,14 +688,14 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
             for (ch = 0; ch < chans; ch++) { /* TNS and PNS */
                 sce = &cpe->ch[ch];
                 s->cur_channel = start_ch + ch;
-                if (s->options.pns && s->coder->search_for_pns)
-                    s->coder->search_for_pns(s, avctx, sce);
                 if (s->options.tns && s->coder->search_for_tns)
                     s->coder->search_for_tns(s, sce);
                 if (s->options.tns && s->coder->apply_tns_filt)
                     s->coder->apply_tns_filt(s, sce);
                 if (sce->tns.present)
                     tns_mode = 1;
+                if (s->options.pns && s->coder->search_for_pns)
+                    s->coder->search_for_pns(s, avctx, sce);
             }
             s->cur_channel = start_ch;
             if (s->options.intensity_stereo) { /* Intensity Stereo */
@@ -660,8 +712,8 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
                         s->coder->search_for_pred(s, sce);
                     if (cpe->ch[ch].ics.predictor_present) pred_mode = 1;
                 }
-                if (s->coder->adjust_common_prediction)
-                    s->coder->adjust_common_prediction(s, cpe);
+                if (s->coder->adjust_common_pred)
+                    s->coder->adjust_common_pred(s, cpe);
                 for (ch = 0; ch < chans; ch++) {
                     sce = &cpe->ch[ch];
                     s->cur_channel = start_ch + ch;
@@ -670,22 +722,34 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
                 }
                 s->cur_channel = start_ch;
             }
-            if (s->options.stereo_mode) { /* Mid/Side stereo */
-                if (s->options.stereo_mode == -1 && s->coder->search_for_ms)
+            if (s->options.mid_side) { /* Mid/Side stereo */
+                if (s->options.mid_side == -1 && s->coder->search_for_ms)
                     s->coder->search_for_ms(s, cpe);
                 else if (cpe->common_window)
                     memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask));
-                for (w = 0; w < 128; w++)
-                    cpe->ms_mask[w] = cpe->is_mask[w] ? 0 : cpe->ms_mask[w];
                 apply_mid_side_stereo(cpe);
             }
             adjust_frame_information(cpe, chans);
+            if (s->options.ltp) { /* LTP */
+                for (ch = 0; ch < chans; ch++) {
+                    sce = &cpe->ch[ch];
+                    s->cur_channel = start_ch + ch;
+                    if (s->coder->search_for_ltp)
+                        s->coder->search_for_ltp(s, sce, cpe->common_window);
+                    if (sce->ics.ltp.present) pred_mode = 1;
+                }
+                s->cur_channel = start_ch;
+                if (s->coder->adjust_common_ltp)
+                    s->coder->adjust_common_ltp(s, cpe);
+            }
             if (chans == 2) {
                 put_bits(&s->pb, 1, cpe->common_window);
                 if (cpe->common_window) {
                     put_ics_info(s, &cpe->ch[0].ics);
                     if (s->coder->encode_main_pred)
                         s->coder->encode_main_pred(s, &cpe->ch[0]);
+                    if (s->coder->encode_ltp_info)
+                        s->coder->encode_ltp_info(s, &cpe->ch[0], 1);
                     encode_ms_info(&s->pb, cpe);
                     if (cpe->ms_mode) ms_mode = 1;
                 }
@@ -697,38 +761,77 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
             start_ch += chans;
         }
 
-        frame_bits = put_bits_count(&s->pb);
-        if (frame_bits <= 6144 * s->channels - 3) {
-            s->psy.bitres.bits = frame_bits / s->channels;
+        if (avctx->flags & CODEC_FLAG_QSCALE) {
+            /* When using a constant Q-scale, don't mess with lambda */
             break;
         }
-        if (is_mode || ms_mode || tns_mode || pred_mode) {
-            for (i = 0; i < s->chan_map[0]; i++) {
-                // Must restore coeffs
-                chans = tag == TYPE_CPE ? 2 : 1;
-                cpe = &s->cpe[i];
-                for (ch = 0; ch < chans; ch++)
-                    memcpy(cpe->ch[ch].coeffs, cpe->ch[ch].pcoeffs, sizeof(cpe->ch[ch].coeffs));
-            }
-        }
 
-        s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;
+        /* rate control stuff
+         * allow between the nominal bitrate, and what psy's bit reservoir says to target
+         * but drift towards the nominal bitrate always
+         */
+        frame_bits = put_bits_count(&s->pb);
+        rate_bits = avctx->bit_rate * 1024 / avctx->sample_rate;
+        rate_bits = FFMIN(rate_bits, 6144 * s->channels - 3);
+        too_many_bits = FFMAX(target_bits, rate_bits);
+        too_many_bits = FFMIN(too_many_bits, 6144 * s->channels - 3);
+        too_few_bits = FFMIN(FFMAX(rate_bits - rate_bits/4, target_bits), too_many_bits);
 
+        /* When using ABR, be strict (but only for increasing) */
+        too_few_bits = too_few_bits - too_few_bits/8;
+        too_many_bits = too_many_bits + too_many_bits/2;
+
+        if (   its == 0 /* for steady-state Q-scale tracking */
+            || (its < 5 && (frame_bits < too_few_bits || frame_bits > too_many_bits))
+            || frame_bits >= 6144 * s->channels - 3  )
+        {
+            float ratio = ((float)rate_bits) / frame_bits;
+
+            if (frame_bits >= too_few_bits && frame_bits <= too_many_bits) {
+                /*
+                 * This path is for steady-state Q-scale tracking
+                 * When frame bits fall within the stable range, we still need to adjust
+                 * lambda to maintain it like so in a stable fashion (large jumps in lambda
+                 * create artifacts and should be avoided), but slowly
+                 */
+                ratio = sqrtf(sqrtf(ratio));
+                ratio = av_clipf(ratio, 0.9f, 1.1f);
+            } else {
+                /* Not so fast though */
+                ratio = sqrtf(ratio);
+            }
+            s->lambda = FFMIN(s->lambda * ratio, 65536.f);
+
+            /* Keep iterating if we must reduce and lambda is in the sky */
+            if (ratio > 0.9f && ratio < 1.1f) {
+                break;
+            } else {
+                if (is_mode || ms_mode || tns_mode || pred_mode) {
+                    for (i = 0; i < s->chan_map[0]; i++) {
+                        // Must restore coeffs
+                        chans = tag == TYPE_CPE ? 2 : 1;
+                        cpe = &s->cpe[i];
+                        for (ch = 0; ch < chans; ch++)
+                            memcpy(cpe->ch[ch].coeffs, cpe->ch[ch].pcoeffs, sizeof(cpe->ch[ch].coeffs));
+                    }
+                }
+                its++;
+            }
+        } else {
+            break;
+        }
     } while (1);
 
+    if (s->options.ltp && s->coder->ltp_insert_new_frame)
+        s->coder->ltp_insert_new_frame(s);
+
     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 & AV_CODEC_FLAG_QSCALE)) {
-        float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
-        s->lambda *= ratio;
-        s->lambda = FFMIN(s->lambda, 65536.f);
-    }
+    s->last_frame_pb_count = put_bits_count(&s->pb);
 
-    if (!frame)
-        s->last_frame++;
+    s->lambda_sum += s->lambda;
+    s->lambda_count++;
 
     ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
                        &avpkt->duration);
@@ -742,6 +845,8 @@ static av_cold int aac_encode_end(AVCodecContext *avct
 {
     AACEncContext *s = avctx->priv_data;
 
+    av_log(avctx, AV_LOG_INFO, "Qavg: %.3f\n", s->lambda_sum / s->lambda_count);
+
     ff_mdct_end(&s->mdct1024);
     ff_mdct_end(&s->mdct128);
     ff_psy_end(&s->psy);
@@ -800,76 +905,123 @@ static av_cold int aac_encode_init(AVCodecContext *avc
     uint8_t grouping[AAC_MAX_CHANNELS];
     int lengths[2];
 
+    /* Constants */
+    s->last_frame_pb_count = 0;
+    avctx->extradata_size = 5;
     avctx->frame_size = 1024;
+    avctx->initial_padding = 1024;
+    s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
 
+    /* Channel map and unspecified bitrate guessing */
+    s->channels = avctx->channels;
+    ERROR_IF(s->channels > AAC_MAX_CHANNELS || s->channels == 7,
+             "Unsupported number of channels: %d\n", s->channels);
+    s->chan_map = aac_chan_configs[s->channels-1];
+    if (!avctx->bit_rate) {
+        for (i = 1; i <= s->chan_map[0]; i++) {
+            avctx->bit_rate += s->chan_map[i] == TYPE_CPE ? 128000 : /* Pair */
+                               s->chan_map[i] == TYPE_LFE ? 16000  : /* LFE  */
+                                                            69000  ; /* SCE  */
+        }
+    }
+ 
+    /* Samplerate */
     for (i = 0; i < 16; i++)
         if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
             break;
-
-    s->channels = avctx->channels;
-
-    ERROR_IF(i == 16 || i >= ff_aac_swb_size_1024_len || i >= ff_aac_swb_size_128_len,
+    s->samplerate_index = i;
+    ERROR_IF(s->samplerate_index == 16 ||
+             s->samplerate_index >= ff_aac_swb_size_1024_len ||
+             s->samplerate_index >= ff_aac_swb_size_128_len,
              "Unsupported sample rate %d\n", avctx->sample_rate);
-    ERROR_IF(s->channels > AAC_MAX_CHANNELS,
-             "Unsupported number of channels: %d\n", s->channels);
+
+    /* Bitrate limiting */
     WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
-             "Too many bits per frame requested, clamping to max\n");
-    if (avctx->profile == FF_PROFILE_AAC_MAIN) {
+             "Too many bits %f > %d per frame requested, clamping to max\n",
+             1024.0 * avctx->bit_rate / avctx->sample_rate,
+             6144 * s->channels);
+    avctx->bit_rate = (int64_t)FFMIN(6144 * s->channels / 1024.0 * avctx->sample_rate,
+                                     avctx->bit_rate);
+ 
+    /* Profile and option setting */
+    avctx->profile = avctx->profile == FF_PROFILE_UNKNOWN ? FF_PROFILE_AAC_LOW :
+                     avctx->profile;
+    for (i = 0; i < FF_ARRAY_ELEMS(aacenc_profiles); i++)
+        if (avctx->profile == aacenc_profiles[i])
+            break;
+    if (avctx->profile == FF_PROFILE_MPEG2_AAC_LOW) {
+        avctx->profile = FF_PROFILE_AAC_LOW;
+        ERROR_IF(s->options.pred,
+                 "Main prediction unavailable in the \"mpeg2_aac_low\" profile\n");
+        ERROR_IF(s->options.ltp,
+                 "LTP prediction unavailable in the \"mpeg2_aac_low\" profile\n");
+        WARN_IF(s->options.pns,
+                "PNS unavailable in the \"mpeg2_aac_low\" profile, turning off\n");
+        s->options.pns = 0;
+    } else if (avctx->profile == FF_PROFILE_AAC_LTP) {
+        s->options.ltp = 1;
+        ERROR_IF(s->options.pred,
+                 "Main prediction unavailable in the \"aac_ltp\" profile\n");
+    } else if (avctx->profile == FF_PROFILE_AAC_MAIN) {
         s->options.pred = 1;
-    } else if ((avctx->profile == FF_PROFILE_AAC_LOW ||
-                avctx->profile == FF_PROFILE_UNKNOWN) && s->options.pred) {
-        s->profile = 0; /* Main */
-        WARN_IF(1, "Prediction requested, changing profile to AAC-Main\n");
-    } else if (avctx->profile == FF_PROFILE_AAC_LOW ||
-               avctx->profile == FF_PROFILE_UNKNOWN) {
-        s->profile = 1; /* Low */
-    } else {
-        ERROR_IF(1, "Unsupported profile %d\n", avctx->profile);
+        ERROR_IF(s->options.ltp,
+                 "LTP prediction unavailable in the \"aac_main\" profile\n");
+    } else if (s->options.ltp) {
+        avctx->profile = FF_PROFILE_AAC_LTP;
+        WARN_IF(1,
+                "Chainging profile to \"aac_ltp\"\n");
+        ERROR_IF(s->options.pred,
+                 "Main prediction unavailable in the \"aac_ltp\" profile\n");
+    } else if (s->options.pred) {
+        avctx->profile = FF_PROFILE_AAC_MAIN;
+        WARN_IF(1,
+                "Chainging profile to \"aac_main\"\n");
+        ERROR_IF(s->options.ltp,
+                 "LTP prediction unavailable in the \"aac_main\" profile\n");
     }
+    s->profile = avctx->profile;
 
-    if (s->options.aac_coder != AAC_CODER_TWOLOOP) {
+    /* Coder limitations */
+    s->coder = &ff_aac_coders[s->options.coder];
+    if (s->options.coder == AAC_CODER_ANMR) {
+        ERROR_IF(avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL,
+                 "The ANMR coder is considered experimental, add -strict -2 to enable!\n");
         s->options.intensity_stereo = 0;
         s->options.pns = 0;
     }
+    ERROR_IF(s->options.ltp && avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL,
+             "The LPT profile requires experimental compliance, add -strict -2 to enable!\n");
 
-    avctx->bit_rate = (int)FFMIN(
-        6144 * s->channels / 1024.0 * avctx->sample_rate,
-        avctx->bit_rate);
+    /* M/S introduces horrible artifacts with multichannel files, this is temporary */
+    if (s->channels > 3)
+        s->options.mid_side = 0;
 
-    s->samplerate_index = i;
-
-    s->chan_map = aac_chan_configs[s->channels-1];
-
     if ((ret = dsp_init(avctx, s)) < 0)
         goto fail;
 
     if ((ret = alloc_buffers(avctx, s)) < 0)
         goto fail;
 
-    avctx->extradata_size = 5;
     put_audio_specific_config(avctx);
 
-    sizes[0]   = ff_aac_swb_size_1024[i];
-    sizes[1]   = ff_aac_swb_size_128[i];
-    lengths[0] = ff_aac_num_swb_1024[i];
-    lengths[1] = ff_aac_num_swb_128[i];
+    sizes[0]   = ff_aac_swb_size_1024[s->samplerate_index];
+    sizes[1]   = ff_aac_swb_size_128[s->samplerate_index];
+    lengths[0] = ff_aac_num_swb_1024[s->samplerate_index];
+    lengths[1] = ff_aac_num_swb_128[s->samplerate_index];
     for (i = 0; i < s->chan_map[0]; i++)
         grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
     if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
                            s->chan_map[0], grouping)) < 0)
         goto fail;
     s->psypp = ff_psy_preprocess_init(avctx);
-    s->coder = &ff_aac_coders[s->options.aac_coder];
     ff_lpc_init(&s->lpc, 2*avctx->frame_size, TNS_MAX_ORDER, FF_LPC_TYPE_LEVINSON);
+    s->random_state = 0x1f2e3d4c;
 
     if (HAVE_MIPSDSPR1)
         ff_aac_coder_init_mips(s);
 
-    s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
-
     ff_aac_tableinit();
 
-    avctx->initial_padding = 1024;
     ff_af_queue_init(avctx, &s->afq);
 
     return 0;
@@ -880,27 +1032,16 @@ fail:
 
 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
 static const AVOption aacenc_options[] = {
-    {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.i64 = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
-        {"auto",     "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
-        {"ms_off",   "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
-        {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 =  1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
-    {"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "aac_coder"},
-        {"faac",     "FAAC-inspired method",      0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAAC},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
-        {"anmr",     "ANMR method",               0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
-        {"twoloop",  "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
-        {"fast",     "Constant quantizer",        0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
-    {"aac_pns", "Perceptual Noise Substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_INT, {.i64 = 1}, 0, 1, AACENC_FLAGS, "aac_pns"},
-        {"disable",  "Disable perceptual noise substitution", 0, AV_OPT_TYPE_CONST, {.i64 =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
-        {"enable",   "Enable perceptual noise substitution",  0, AV_OPT_TYPE_CONST, {.i64 =  1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
-    {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_INT, {.i64 = 1}, 0, 1, AACENC_FLAGS, "intensity_stereo"},
-        {"disable",  "Disable intensity stereo coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "intensity_stereo"},
-        {"enable",   "Enable intensity stereo coding", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "intensity_stereo"},
-    {"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_tns"},
-        {"disable",  "Disable temporal noise shaping", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_tns"},
-        {"enable",   "Enable temporal noise shaping", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_tns"},
-    {"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_pred"},
-        {"disable",  "Disable AAC-Main prediction", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pred"},
-        {"enable",   "Enable AAC-Main prediction", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pred"},
+    {"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "coder"},
+        {"anmr",     "ANMR method",               0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR},    INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
+        {"twoloop",  "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
+        {"fast",     "Constant quantizer",        0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST},    INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
+    {"aac_ms", "Force M/S stereo coding", offsetof(AACEncContext, options.mid_side), AV_OPT_TYPE_BOOL, {.i64 = -1}, -1, 1, AACENC_FLAGS},
+    {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
+    {"aac_pns", "Perceptual noise substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
+    {"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
+    {"aac_ltp", "Long term prediction", offsetof(AACEncContext, options.ltp), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
+    {"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
     {NULL}
 };
 
@@ -911,6 +1052,11 @@ static const AVClass aacenc_class = {
     LIBAVUTIL_VERSION_INT,
 };
 
+static const AVCodecDefault aac_encode_defaults[] = {
+    { "b", "0" },
+    { NULL }
+};
+
 AVCodec ff_aac_encoder = {
     .name           = "aac",
     .long_name      = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
@@ -920,9 +1066,9 @@ AVCodec ff_aac_encoder = {
     .init           = aac_encode_init,
     .encode2        = aac_encode_frame,
     .close          = aac_encode_end,
+    .defaults       = aac_encode_defaults,
     .supported_samplerates = mpeg4audio_sample_rates,
-    .capabilities   = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY |
-                      AV_CODEC_CAP_EXPERIMENTAL,
+    .capabilities   = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
                                                      AV_SAMPLE_FMT_NONE },
     .priv_class     = &aacenc_class,