virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00
Code Issues Releases Activity

avcodec/flacenc: Avoid copying packet data, allow user-supplied buffers

Browse Source 
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.

Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This commit is contained in:
Andreas Rheinhardt 2021-04-25 01:43:26 +02:00
parent 314c086a85
commit 5abb5c0415
1 changed files with 6 additions and 3 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

9
libavcodec/flacenc.c
View File

@ -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;
}
@ -1459,11 +1461,12 @@ const AVCodec ff_flac_encoder = {
.long_name = NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_FLAC,
.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY |
AV_CODEC_CAP_SMALL_LAST_FRAME,
.priv_data_size = sizeof(FlacEncodeContext),
.init = flac_encode_init,
.encode2 = flac_encode_frame,
.close = flac_encode_close,
.capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_NONE },