@@ -27,6 +27,7 @@
#include "avcodec.h"
#include "bswapdsp.h"
+#include "encode.h"
#include "put_bits.h"
#include "golomb.h"
#include "internal.h"
@@ -1378,7 +1379,7 @@ static int flac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
}
}
- if ((ret = ff_alloc_packet2(avctx, avpkt, frame_bytes, 0)) < 0)
+ if ((ret = ff_get_encode_buffer(avctx, avpkt, frame_bytes, 0)) < 0)
return ret;
out_bytes = write_frame(s, avpkt);
@@ -1396,10 +1397,11 @@ static int flac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
avpkt->pts = frame->pts;
avpkt->duration = ff_samples_to_time_base(avctx, frame->nb_samples);
- avpkt->size = out_bytes;
s->next_pts = avpkt->pts + avpkt->duration;
+ av_shrink_packet(avpkt, out_bytes);
+
*got_packet_ptr = 1;
return 0;
}
@@ -1463,7 +1465,8 @@ const AVCodec ff_flac_encoder = {
.init = flac_encode_init,
.encode2 = flac_encode_frame,
.close = flac_encode_close,
- .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
+ .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY |
+ AV_CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_NONE },
The FLAC encoder calculates the size in advance, so one can avoid an intermediate buffer for the packet data by using ff_get_encode_buffer() and also set AV_CODEC_CAP_DR1 at the same time. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com> --- The size was actually exact in all my tests; it seems the only way for which the size could be inexact were if the limit in set_sr_golomb_flac() (or rather set_ur_golomb_jpegls()) were triggered, but I am not sure whether this is even possible (it better be not, because if it were possible, the put_bits(pb, esc_len, i - 1) (with esc_len == 0) would run into the av_assert2() in put_bits()). libavcodec/flacenc.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-)