diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index bb9004842e2e4101c183728238d79cb633620f14..1b95ebd7551ed629641ea2e4e79a9e9ca259773b 100644
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -489,7 +489,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
float **samples = s->planar_samples, *samples2, *la, *overlap;
ChannelElement *cpe;
SingleChannelElement *sce;
- int i, ch, w, chans, tag, start_ch, ret;
+ int i, its, ch, w, chans, tag, start_ch, ret, frame_bits;
int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;
int chan_el_counter[4];
FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
@@ -581,14 +581,16 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
}
if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels, 0)) < 0)
return ret;
+ frame_bits = its = 0;
do {
- int frame_bits;
+ int target_bits, too_many_bits, too_few_bits;
init_put_bits(&s->pb, avpkt->data, avpkt->size);
if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT))
put_bitstream_info(s, LIBAVCODEC_IDENT);
start_ch = 0;
+ target_bits = 0;
memset(chan_el_counter, 0, sizeof(chan_el_counter));
for (i = 0; i < s->chan_map[0]; i++) {
FFPsyWindowInfo* wi = windows + start_ch;
@@ -611,7 +613,15 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
if (sce->band_type[w] > RESERVED_BT)
sce->band_type[w] = 0;
}
+ s->psy.bitres.alloc = -1;
+ s->psy.bitres.bits = avctx->frame_bits / 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->psy.bitres.alloc /= chans;
+ }
+ s->cur_type = tag;
for (ch = 0; ch < chans; ch++) {
s->cur_channel = start_ch + ch;
s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
@@ -692,36 +702,69 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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
+ * target either the nominal bitrate, or what psy's bit reservoir says to target
+ * whichever is greatest
+ */
+
+ frame_bits = put_bits_count(&s->pb);
+ target_bits = FFMAX(target_bits, avctx->bit_rate * 1024 / avctx->sample_rate);
+ target_bits = FFMIN(target_bits, 6144 * s->channels - 3);
+
+ /* When using ABR, be strict (but only for increasing) */
+ too_many_bits = target_bits + target_bits/2;
+ too_few_bits = target_bits - target_bits/8;
+
+ 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)target_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 (s->lambda < 300.f || ratio > 0.9f) {
+ 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);
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);
- }
-
if (!frame)
s->last_frame++;
diff --git a/libavcodec/aacenc.h b/libavcodec/aacenc.h
index 54951f9f70a8527351d1a629236e17ae11bd8464..7e7609b1a8398804dd8437418bb675c432285567 100644
--- a/libavcodec/aacenc.h
+++ b/libavcodec/aacenc.h
@@ -96,10 +96,12 @@ typedef struct AACEncContext {
FFPsyContext psy;
struct FFPsyPreprocessContext* psypp;
AACCoefficientsEncoder *coder;
- int cur_channel;
+ int cur_channel; ///< current channel for coder context
int last_frame;
int random_state;
float lambda;
+ enum RawDataBlockType cur_type; ///< channel group type cur_channel belongs to
+
AudioFrameQueue afq;
DECLARE_ALIGNED(16, int, qcoefs)[96]; ///< quantized coefficients
DECLARE_ALIGNED(32, float, scoefs)[1024]; ///< scaled coefficients
diff --git a/libavcodec/aacpsy.c b/libavcodec/aacpsy.c
index 82b670d49d8c41d160ae639b63e1933a3f9aa38d..af235c758c8bd30417c8b7943ea9c1ae91e501db 100644
--- a/libavcodec/aacpsy.c
+++ b/libavcodec/aacpsy.c
@@ -87,6 +87,7 @@ enum {
};
#define PSY_3GPP_BITS_TO_PE(bits) ((bits) * 1.18f)
+#define PSY_3GPP_PE_TO_BITS(bits) ((bits) / 1.18f)
/* LAME psy model constants */
#define PSY_LAME_FIR_LEN 21 ///< LAME psy model FIR order
@@ -687,6 +688,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
desired_pe *= av_clipf(pctx->pe.previous / PSY_3GPP_BITS_TO_PE(ctx->bitres.bits),
0.85f, 1.15f);
pctx->pe.previous = PSY_3GPP_BITS_TO_PE(desired_bits);
+ ctx->bitres.alloc = desired_bits;
if (desired_pe < pe) {
/* 5.6.1.3.4 "First Estimation of the reduction value" */
@@ -788,6 +790,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
psy_band->threshold = band->thr;
psy_band->energy = band->energy;
psy_band->spread = band->active_lines * 2.0f / band_sizes[g];
+ psy_band->bits = PSY_3GPP_PE_TO_BITS(band->pe);
}
}
diff --git a/libavcodec/psymodel.h b/libavcodec/psymodel.h
index e9be1f6fa5c8898916da21a83e3ce4e749fee762..a04cc4d2260c39d6d09c9741cf48b4ddc0ded18a 100644
--- a/libavcodec/psymodel.h
+++ b/libavcodec/psymodel.h
@@ -88,6 +88,7 @@ typedef struct FFPsyContext {
struct {
int size; ///< size of the bitresevoir in bits
int bits; ///< number of bits used in the bitresevoir
+ int alloc; ///< number of bits allocated by the psy, or -1 if no allocation was done
} bitres;
void* model_priv_data; ///< psychoacoustic model implementation private data
diff --git a/tests/fate/aac.mak b/tests/fate/aac.mak
index f30d4dbbc5ba0f1bdfda971905e4c4fbe202842c..8e9c91507f91ee11d0b7be1e1b97104872beb5bf 100644
--- a/tests/fate/aac.mak
+++ b/tests/fate/aac.mak
@@ -146,7 +146,7 @@ fate-aac-aref-encode: CMD = enc_dec_pcm adts wav s16le $(REF) -strict -2 -c:a aa
fate-aac-aref-encode: CMP = stddev
fate-aac-aref-encode: REF = ./tests/data/asynth-44100-2.wav
fate-aac-aref-encode: CMP_SHIFT = -4096
-fate-aac-aref-encode: CMP_TARGET = 594
+fate-aac-aref-encode: CMP_TARGET = 584
fate-aac-aref-encode: SIZE_TOLERANCE = 2464
fate-aac-aref-encode: FUZZ = 6
@@ -172,7 +172,7 @@ fate-aac-pns-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-re
fate-aac-pns-encode: CMP = stddev
fate-aac-pns-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate-aac-pns-encode: CMP_SHIFT = -4096
-fate-aac-pns-encode: CMP_TARGET = 633.77
+fate-aac-pns-encode: CMP_TARGET = 623.77
fate-aac-pns-encode: SIZE_TOLERANCE = 3560
fate-aac-pns-encode: FUZZ = 1
@@ -181,7 +181,7 @@ fate-aac-tns-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-re
fate-aac-tns-encode: CMP = stddev
fate-aac-tns-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate-aac-tns-encode: CMP_SHIFT = -4096
-fate-aac-tns-encode: CMP_TARGET = 650.37
+fate-aac-tns-encode: CMP_TARGET = 644.50
fate-aac-tns-encode: FUZZ = 2.8
fate-aac-tns-encode: SIZE_TOLERANCE = 3560
@@ -190,7 +190,7 @@ fate-aac-is-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-ref
fate-aac-is-encode: CMP = stddev
fate-aac-is-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate-aac-is-encode: CMP_SHIFT = -4096
-fate-aac-is-encode: CMP_TARGET = 616.75
+fate-aac-is-encode: CMP_TARGET = 614.04
fate-aac-is-encode: SIZE_TOLERANCE = 3560
fate-aac-is-encode: FUZZ = 1
@@ -199,7 +199,7 @@ fate-aac-pred-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-r
fate-aac-pred-encode: CMP = stddev
fate-aac-pred-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate-aac-pred-encode: CMP_SHIFT = -4096
-fate-aac-pred-encode: CMP_TARGET = 652.60
+fate-aac-pred-encode: CMP_TARGET = 657
fate-aac-pred-encode: FUZZ = 5
fate-aac-pred-encode: SIZE_TOLERANCE = 3560