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| author | Andreas Rheinhardt <andreas.rheinhardt@outlook.com> | 2021-02-10 19:37:37 +0100 |
|---|---|---|
| committer | Andreas Rheinhardt <andreas.rheinhardt@outlook.com> | 2022-01-04 13:16:50 +0100 |
| commit | 49bf94536f059340eacd5430592e4216b29d0d20 (patch) | |
| tree | b1a59398a83aa4c2a8bdd01af8498dda08a48a34 /libavformat/matroskadec.c | |
| parent | d82c91ba2f889bc5e0ee7c976219f91e62dc9f8b (diff) | |
| download | ffmpeg-49bf94536f059340eacd5430592e4216b29d0d20.tar.gz | |
avcodec/mpeg4audio: Unavpriv and deduplicate mpeg4audio_sample_rates
avpriv_mpeg4audio_sample_rates has a size of 64B and it is currently
avpriv; a clone of it exists in aacenctab.h and from there it is inlined
in aacenc.c (which also uses the avpriv version) and in the FLV muxer.
This means that despite it being avpriv both libavformat as well as
libavcodec have copies already.
This situation is clearly suboptimal. Given the overhead of exporting
symbols (for x64 Elf/Linux/GNU: 2x2B version, 2x24B .dynsym, 24B .rela.dyn,
8B .got, 4B hash + twice the size of the name (here 31B)) the object is
unavprived, i.e. duplicated into libavformat when creating a shared
build; but the duplicates in the AAC encoder and FLV muxer are removed.
This involves splitting of the sample rate table into a file of its own;
this allowed to break some spurious dependencies (e.g. both the AAC
encoder as well as the Matroska demuxer actually don't need the
mpeg4audio_get_config stuff).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Diffstat (limited to 'libavformat/matroskadec.c')
| -rw-r--r-- | libavformat/matroskadec.c | 4 |
1 files changed, 2 insertions, 2 deletions
diff --git a/libavformat/matroskadec.c b/libavformat/matroskadec.c index 6ce553205d..e271916bf1 100644 --- a/libavformat/matroskadec.c +++ b/libavformat/matroskadec.c @@ -2029,8 +2029,8 @@ static int matroska_aac_sri(int samplerate) { int sri; - for (sri = 0; sri < FF_ARRAY_ELEMS(avpriv_mpeg4audio_sample_rates); sri++) - if (avpriv_mpeg4audio_sample_rates[sri] == samplerate) + for (sri = 0; sri < FF_ARRAY_ELEMS(ff_mpeg4audio_sample_rates); sri++) + if (ff_mpeg4audio_sample_rates[sri] == samplerate) break; return sri; } |