lavc: support extracting audio service type from side data

[ffmpeg] / libavcodec / aacenc.c

diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index 00a6d0397d673b1c6cdacfdc42e60313a8aa6a5f..a28e1a1290ed19cdfde1b6c3a2386c1b1c665b16 100644 (file)
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -34,7 +34,6 @@
 #include "libavutil/opt.h"
 #include "avcodec.h"
 #include "put_bits.h"
-#include "dsputil.h"
 #include "internal.h"
 #include "mpeg4audio.h"
 #include "kbdwin.h"
@@ -598,7 +597,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                 coeffs[ch] = cpe->ch[ch].coeffs;
             s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
             for (ch = 0; ch < chans; ch++) {
-                s->cur_channel = start_ch * 2 + ch;
+                s->cur_channel = start_ch + ch;
                 s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
             }
             cpe->common_window = 0;
@@ -614,7 +613,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                     }
                 }
             }
-            s->cur_channel = start_ch * 2;
+            s->cur_channel = start_ch;
             if (s->options.stereo_mode && cpe->common_window) {
                 if (s->options.stereo_mode > 0) {
                     IndividualChannelStream *ics = &cpe->ch[0].ics;
@@ -684,9 +683,6 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
     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;
 }
 
@@ -720,11 +716,6 @@ 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);
@@ -759,10 +750,10 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
 
     s->chan_map = aac_chan_configs[s->channels-1];
 
-    if (ret = dsp_init(avctx, s))
+    if ((ret = dsp_init(avctx, s)) < 0)
         goto fail;
 
-    if (ret = alloc_buffers(avctx, s))
+    if ((ret = alloc_buffers(avctx, s)) < 0)
         goto fail;
 
     avctx->extradata_size = 5;
@@ -774,7 +765,8 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     lengths[1] = ff_aac_num_swb_128[i];
     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))
+    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[2];
@@ -786,7 +778,7 @@ 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;
+    avctx->initial_padding = 1024;
     ff_af_queue_init(avctx, &s->afq);
 
     return 0;
@@ -813,6 +805,7 @@ static const AVClass aacenc_class = {
 
 AVCodec ff_aac_encoder = {
     .name           = "aac",
+    .long_name      = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = AV_CODEC_ID_AAC,
     .priv_data_size = sizeof(AACEncContext),
@@ -823,6 +816,5 @@ AVCodec ff_aac_encoder = {
                       CODEC_CAP_EXPERIMENTAL,
     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
                                                      AV_SAMPLE_FMT_NONE },
-    .long_name      = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
     .priv_class     = &aacenc_class,
 };