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.rheinha...@outlook.com>
---
libavcodec/libvorbisenc.c | 6 ++++--
1 file changed, 4 insertions(+), 2 deletions(-)
diff --git a/libavcodec/libvorbisenc.c b/libavcodec/libvorbisenc.c
index d0f4c65aac..ec674695d4 100644
--- a/libavcodec/libvorbisenc.c
+++ b/libavcodec/libvorbisenc.c
@@ -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,
--
2.27.0
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