| Message ID | HE1PR0301MB21544CF9C254344F80904FD58F299@HE1PR0301MB2154.eurprd03.prod.outlook.com |
|---|---|
| State | Accepted |
| Commit | c59398970bff5ee6c0305a0228eb562ec5901b08 |
| Headers | show |
| Series | [FFmpeg-devel,01/39] avcodec/audiotoolboxenc: Remove AV_CODEC_CAP_DR1 | expand |
| Context | Check | Description |
|---|---|---|
| andriy/x86_make | success | Make finished |
| andriy/x86_make_fate | success | Make fate finished |
| andriy/PPC64_make | success | Make finished |
| andriy/PPC64_make_fate | success | Make fate finished |
On 5/21/2021 6:17 AM, Andreas Rheinhardt wrote: > 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.. Extra dot. > > Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com> > --- > libavcodec/libmp3lame.c | 7 ++++--- > 1 file 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, >
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,
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.. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com> --- libavcodec/libmp3lame.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-)