@@ -25,6 +25,7 @@
#include "libavutil/opt.h"
#include "avcodec.h"
#include "audio_frame_queue.h"
+#include "encode.h"
#include "internal.h"
#include "vorbis.h"
#include "vorbis_parser.h"
@@ -348,7 +349,7 @@ static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
- if ((ret = ff_alloc_packet2(avctx, avpkt, op.bytes, 0)) < 0)
+ if ((ret = ff_get_encode_buffer(avctx, avpkt, op.bytes, 0)) < 0)
return ret;
av_fifo_generic_read(s->pkt_fifo, avpkt->data, op.bytes, NULL);
@@ -378,11 +379,12 @@ const AVCodec ff_libvorbis_encoder = {
.long_name = NULL_IF_CONFIG_SMALL("libvorbis"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_VORBIS,
+ .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY |
+ AV_CODEC_CAP_SMALL_LAST_FRAME,
.priv_data_size = sizeof(LibvorbisEncContext),
.init = libvorbis_encode_init,
.encode2 = libvorbis_encode_frame,
.close = libvorbis_encode_close,
- .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.priv_class = &vorbis_class,
Here the packet size is known before allocating the packet because the encoder provides said information (and works with internal buffers itself), so one can use this information to avoid the implicit use of another intermediate buffer for the packet data; and by switching to ff_get_encode_buffer() one can also allow user-supplied buffers. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com> --- libavcodec/libvorbisenc.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-)