[FFmpeg-cvslog] avcodec/flacenc: Avoid copying packet data, allow user-supplied buffers

[Andreas Rheinhardt]

ffmpeg | branch: master | Andreas Rheinhardt <andreas.rheinhardt at outlook.com> | Sun Apr 25 01:43:26 2021 +0200| [5abb5c04155b536f26fc88311ac3132890111360] | committer: Andreas Rheinhardt

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

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 at gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt at outlook.com>

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=5abb5c04155b536f26fc88311ac3132890111360
---

 libavcodec/flacenc.c | 9 ++++++---
 1 file changed, 6 insertions(+), 3 deletions(-)

diff --git a/libavcodec/flacenc.c b/libavcodec/flacenc.c
index 37ed1e4cce..de36d33333 100644
--- a/libavcodec/flacenc.c
+++ b/libavcodec/flacenc.c
@@ -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 },