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| author | Andreas Rheinhardt <andreas.rheinhardt@outlook.com> | 2021-08-24 19:41:16 +0200 |
|---|---|---|
| committer | Andreas Rheinhardt <andreas.rheinhardt@outlook.com> | 2021-09-17 13:22:25 +0200 |
| commit | 40bdd8cc05d9c98a18cf2b1c2a00c8a5a7b38113 (patch) | |
| tree | 0fc408f78b9b6934ac351cd4499c07737f8f6a62 /libavformat/swfdec.c | |
| parent | 9f05b3ba604a30eeb6f5ff877b8b5b5c93a268d7 (diff) | |
avformat: Avoid allocation for AVStreamInternal
Do this by allocating AVStream together with the data that is
currently in AVStreamInternal; or rather: Put AVStream at the
beginning of a new structure called FFStream (which encompasses
more than just the internal fields and is a proper context in its own
right, hence the name) and remove AVStreamInternal altogether.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Diffstat (limited to 'libavformat/swfdec.c')
| -rw-r--r-- | libavformat/swfdec.c | 4 |
1 files changed, 2 insertions, 2 deletions
diff --git a/libavformat/swfdec.c b/libavformat/swfdec.c index 8eda80b4f6..e549e4f50b 100644 --- a/libavformat/swfdec.c +++ b/libavformat/swfdec.c @@ -204,7 +204,7 @@ static AVStream *create_new_audio_stream(AVFormatContext *s, int id, int info) } ast->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; ast->codecpar->codec_id = ff_codec_get_id(swf_audio_codec_tags, info>>4 & 15); - ast->internal->need_parsing = AVSTREAM_PARSE_FULL; + ffstream(ast)->need_parsing = AVSTREAM_PARSE_FULL; sample_rate_code = info>>2 & 3; sample_size_code = info>>1 & 1; if (!sample_size_code && ast->codecpar->codec_id == AV_CODEC_ID_PCM_S16LE) @@ -293,7 +293,7 @@ static int swf_read_packet(AVFormatContext *s, AVPacket *pkt) return AVERROR(ENOMEM); ast->duration = avio_rl32(pb); // number of samples if (((v>>4) & 15) == 2) { // MP3 sound data record - ast->internal->skip_samples = avio_rl16(pb); + ffstream(ast)->skip_samples = avio_rl16(pb); len -= 2; } len -= 7; |