ffmpeg/tests
Marton Balint 9c2c0c37f8 avformat/pcm: factorize and improve determining the default packet size
- Remove the 1024 cap on the number of samples, for high sample rate audio it
  was suboptimal, calculate the low neighbour power of two for the number of
  samples (audio blocks) instead.
- Make the function work correctly also for non-pcm codecs by using the stream
  bitrate to estimate the target packet size. A previous version of this patch
  used av_get_audio_frame_duration2() the estimate the desired packet size, but
  for some codecs that returns the duration of a single audio frame regardless
  of frame_bytes.
- Fallback to 4096/block_align*block_align if bitrate is not available.

Signed-off-by: Marton Balint <cus@passwd.hu>
2024-03-16 19:19:42 +01:00
..
api
checkasm configure: Remove av_restrict 2024-03-15 12:51:15 +01:00
fate fate/wmavoice: add missing aresample filter dependency 2024-03-15 12:44:59 -03:00
filtergraphs
maps
ref avformat/pcm: factorize and improve determining the default packet size 2024-03-16 19:19:42 +01:00
streamgroups
.gitignore
Makefile fate/ffprobe: add missing aresample filter dependency 2024-03-15 12:44:59 -03:00
audiogen.c
audiomatch.c
base64.c
copycooker.sh
extended.ffconcat
fate-run.sh fate: fix generating references when sh=dash 2024-03-12 23:01:46 +01:00
fate-valgrind.supp
fate.sh
md5.sh
refcmp-metadata.awk
reference.pnm
rotozoom.c
simple1.ffconcat
simple2.ffconcat
test.ffmeta
tiny_psnr.c
tiny_ssim.c
utils.c
videogen.c