Only copy it manually in the muxers where it makes sense (rtspenc,
sapenc). Don't touch the original AVStream in movenchint, where
the original AVStream should be kept untouched.
This fixes the normal tracks in RTP hinted files after
abb810db - the hint tracks were ok while the normal media tracks
were broken, noticed by Michael Niedermayer.
This reverts abb810db but achieves the same effect for the other
muxers.
Signed-off-by: Martin Storsjö <martin@martin.st>
While it strictly isn't necessary to copy the time base (since
any use of it is scaled in ff_write_chained), it still is better
to signal the actual time base to the caller, avoiding one
unnecessary rescaling. This also lets the caller know what the
actual internal time base is, in case that is useful info
for some caller.
This reverts commit 397ffde115f4e0482a007b672f40457596cebfc4.
Signed-off-by: Martin Storsjö <martin@martin.st>
This results in DefaultDuration not being written when the framerate is
not known, but as this field is purely informative, this should not
break any sane demuxers.
Add the low overhead pipe mode and the extended broadcast mode.
Export the options as 'syncponts' since it impacts only that.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This corrects the bug that caused the checksums to change in
9767d7c092c890ecc5953452e8a951fd902dd67b.
It caused the EOS flag to be set incorrectly; the ogg spec does not
allow it to be set in the middle of a logical bitstream.
Signed-off-by: Andrew Kelley <superjoe30@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids all the ABI troubles associated with avpriv_.
Since this function is very small and does not depend on any tables,
making it inline should have no adverse effects.
Before, header information for ogg format files was sent with the
first encoded packet.
This patch makes it so that it is possible for API users to
differentiate between headers and encoded audio. This is useful, for
example, when creating an audio stream where you want to send one set
of headers for every client that connects and then the encoded stream
of audio.
Signed-off-by: Martin Storsjö <martin@martin.st>
Use it instead of checking CODEC_FLAG_BITEXACT in the first stream's
codec context.
Using codec options inside lavf is fragile and can easily break when the
muxing codec context is not the encoding context.
Support the URL scheme where the playpath is in an RTMP URL is
passed as the slist argument and the app is given infront of the
query part of the URL:
rtmp://host[:port]/[app]?slist=[playpath]
(other arguments in the query part are stripped as they are not used)
Signed-off-by: Martin Storsjö <martin@martin.st>
Also, stop using AVCodecContext.codec_name as fallback, since it will be
deprecated.
Changes the result of the lavf-asf test (and its associated seektest),
since 'msmpeg4v3' gets written instead of just 'msmpeg4'.