@@ -57,7 +57,8 @@ OBJS-$(CONFIG_AAC_DECODER)             += aacdec.o aactab.o aacsbr.o aacps.o \

 OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o    \

                                           aacpsy.o aactab.o      \

                                           psymodel.o iirfilter.o \

-                                          mpeg4audio.o kbdwin.o

+                                          mpeg4audio.o kbdwin.o  \

+                                          audio_frame_queue.o

 OBJS-$(CONFIG_AASC_DECODER)            += aasc.o msrledec.o

 OBJS-$(CONFIG_AC3_DECODER)             += ac3dec.o ac3dec_data.o ac3.o kbdwin.o

 OBJS-$(CONFIG_AC3_ENCODER)             += ac3enc_float.o ac3enc.o ac3tab.o \

@@ -34,6 +34,7 @@

 #include "avcodec.h"

 #include "put_bits.h"

 #include "dsputil.h"

+#include "internal.h"

 #include "mpeg4audio.h"

 #include "kbdwin.h"

 #include "sinewin.h"

@@ -476,8 +477,7 @@ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,

  * Deinterleave input samples.

  * Channels are reordered from Libav's default order to AAC order.

  */

-static void deinterleave_input_samples(AACEncContext *s,

-                                       const float *samples, int nb_samples)

+static void deinterleave_input_samples(AACEncContext *s, AVFrame *frame)

 {

     int ch, i;

     const int sinc = s->channels;

@@ -485,35 +485,43 @@ static void deinterleave_input_samples(AACEncContext *s,

     /* deinterleave and remap input samples */

     for (ch = 0; ch < sinc; ch++) {

-        const float *sptr = samples + channel_map[ch];

-

         /* copy last 1024 samples of previous frame to the start of the current frame */

         memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));

         /* deinterleave */

-        for (i = 2048; i < 2048 + nb_samples; i++) {

-            s->planar_samples[ch][i] = *sptr;

-            sptr += sinc;

+        i = 2048;

+        if (frame) {

+            const float *sptr = ((const float *)frame->data[0]) + channel_map[ch];

+            for (; i < 2048 + frame->nb_samples; i++) {

+                s->planar_samples[ch][i] = *sptr;

+                sptr += sinc;

+            }

         }

         memset(&s->planar_samples[ch][i], 0,

                (3072 - i) * sizeof(s->planar_samples[0][0]));

     }

 }

-static int aac_encode_frame(AVCodecContext *avctx,

-                            uint8_t *frame, int buf_size, void *data)

+static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,

+                            const AVFrame *frame, int *got_packet_ptr)

 {

     AACEncContext *s = avctx->priv_data;

     float **samples = s->planar_samples, *samples2, *la, *overlap;

     ChannelElement *cpe;

-    int i, ch, w, g, chans, tag, start_ch;

+    int i, ch, w, g, chans, tag, start_ch, ret;

     int chan_el_counter[4];

     FFPsyWindowInfo windows[AAC_MAX_CHANNELS];

     if (s->last_frame == 2)

         return 0;

-    deinterleave_input_samples(s, data, data ? avctx->frame_size : 0);

+    /* add current frame to queue */

+    if (frame) {

+        if ((ret = ff_af_queue_add(&s->afq, frame) < 0))

+            return ret;

+    }

+

+    deinterleave_input_samples(s, frame);

     if (s->psypp)

         ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);

@@ -532,7 +540,7 @@ static int aac_encode_frame(AVCodecContext *avctx,

             overlap  = &samples[cur_channel][0];

             samples2 = overlap + 1024;

             la       = samples2 + (448+64);

-            if (!data)

+            if (!frame)

                 la = NULL;

             if (tag == TYPE_LFE) {

                 wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;

@@ -565,7 +573,13 @@ static int aac_encode_frame(AVCodecContext *avctx,

     }

     do {

         int frame_bits;

-        init_put_bits(&s->pb, frame, buf_size*8);

+

+        if ((ret = ff_alloc_packet(avpkt, 768 * s->channels))) {

+            av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");

+            return ret;

+        }

+        init_put_bits(&s->pb, avpkt->data, avpkt->size);

+

         if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))

             put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);

         start_ch = 0;

@@ -645,10 +659,15 @@ static int aac_encode_frame(AVCodecContext *avctx,

         s->lambda = FFMIN(s->lambda, 65536.f);

     }

-    if (!data)

+    if (!frame)

         s->last_frame++;

-    return put_bits_count(&s->pb)>>3;

+    ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,

+                       &avpkt->duration);

+

+    avpkt->size = put_bits_count(&s->pb) >> 3;

+    *got_packet_ptr = 1;

+    return 0;

 }

 static av_cold int aac_encode_end(AVCodecContext *avctx)

@@ -662,6 +681,10 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)

         ff_psy_preprocess_end(s->psypp);

     av_freep(&s->buffer.samples);

     av_freep(&s->cpe);

+    ff_af_queue_close(&s->afq);

+#if FF_API_OLD_ENCODE_AUDIO

+    av_freep(&avctx->coded_frame);

+#endif

     return 0;

 }

@@ -695,6 +718,11 @@ static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)

     for(ch = 0; ch < s->channels; ch++)

         s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;

+#if FF_API_OLD_ENCODE_AUDIO

+    if (!(avctx->coded_frame = avcodec_alloc_frame()))

+        goto alloc_fail;

+#endif

+

     return 0;

 alloc_fail:

     return AVERROR(ENOMEM);

@@ -756,6 +784,9 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)

     for (i = 0; i < 428; i++)

         ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));

+    avctx->delay = 1024;

+    ff_af_queue_init(avctx, &s->afq);

+

     return 0;

 fail:

     aac_encode_end(avctx);

@@ -784,7 +815,7 @@ AVCodec ff_aac_encoder = {

     .id             = CODEC_ID_AAC,

     .priv_data_size = sizeof(AACEncContext),

     .init           = aac_encode_init,

-    .encode         = aac_encode_frame,

+    .encode2        = aac_encode_frame,

     .close          = aac_encode_end,

     .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,

     .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},

@@ -27,7 +27,7 @@

 #include "dsputil.h"

 #include "aac.h"

-

+#include "audio_frame_queue.h"

 #include "psymodel.h"

 typedef struct AACEncOptions {

@@ -71,6 +71,7 @@ typedef struct AACEncContext {

     int cur_channel;

     int last_frame;

     float lambda;

+    AudioFrameQueue afq;

     DECLARE_ALIGNED(16, int,   qcoefs)[96];      ///< quantized coefficients

     DECLARE_ALIGNED(32, float, scoefs)[1024];    ///< scaled coefficients