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author | Andreas Rheinhardt <andreas.rheinhardt@outlook.com> | 2021-04-25 01:43:26 +0200 |
---|---|---|
committer | Andreas Rheinhardt <andreas.rheinhardt@outlook.com> | 2021-05-23 14:42:54 +0200 |
commit | c59398970bff5ee6c0305a0228eb562ec5901b08 (patch) | |
tree | f3f8ee6c2548ff132c5bd0e0196a2c168431fa93 | |
parent | 95c8a859d97e9902388f5acf5c735b16417a2609 (diff) | |
download | ffmpeg-c59398970bff5ee6c0305a0228eb562ec5901b08.tar.gz |
avcodec/libmp3lame: Avoid copying data, allow user-supplied buffer
The libmp3lame encoder already uses an internal buffer, so that the
packet size is already known before allocating the packet; therefore
one can avoid another (implicit) intermediate buffer by switching
to ff_get_encode_buffer(), thereby also supporting user-supplied
buffers.
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
-rw-r--r-- | libavcodec/libmp3lame.c | 7 |
1 files changed, 4 insertions, 3 deletions
diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c index fe4b50257f..5675864bb2 100644 --- a/libavcodec/libmp3lame.c +++ b/libavcodec/libmp3lame.c @@ -34,6 +34,7 @@ #include "libavutil/opt.h" #include "avcodec.h" #include "audio_frame_queue.h" +#include "encode.h" #include "internal.h" #include "mpegaudio.h" #include "mpegaudiodecheader.h" @@ -264,7 +265,7 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, ff_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index); if (len <= s->buffer_index) { - if ((ret = ff_alloc_packet2(avctx, avpkt, len, 0)) < 0) + if ((ret = ff_get_encode_buffer(avctx, avpkt, len, 0)) < 0) return ret; memcpy(avpkt->data, s->buffer, len); s->buffer_index -= len; @@ -296,7 +297,6 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, AV_WL32(side_data + 4, discard_padding); } - avpkt->size = len; *got_packet_ptr = 1; } return 0; @@ -332,11 +332,12 @@ const AVCodec ff_libmp3lame_encoder = { .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_MP3, + .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY | + AV_CODEC_CAP_SMALL_LAST_FRAME, .priv_data_size = sizeof(LAMEContext), .init = mp3lame_encode_init, .encode2 = mp3lame_encode_frame, .close = mp3lame_encode_close, - .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_SMALL_LAST_FRAME, .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16P, |