This reverts commit 26524e3581.
If we want the T.140 codec to have the AV_CODEC_ID_TEXT codec id,
its type needs to be AVMEDIA_TYPE_SUBTITLE, so, keep interpreting
the text media type as AVMEDIA_TYPE_SUBTITLE.
Signed-off-by: Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This makes more sense than mapping to AV_CODEC_ID_SUBRIP. Nothing
indicates that a T.140 track contains subrip sub-titles.
Signed-off-by: Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
When precision is fixed and volume is 0, filter_frame does not
perform any operation on the output buffer. This works if the
output buffer has been allocated and zeroed with ff_get_audio_buffer
but not if the input buffer is used as output buffer.
Fix this by not using the input buffer as output buffer if
precision is fixed and volume is 0.
Signed-off-by: Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '4f6cd883f06f7893a2b60a41e7a4f8ae633dac2f':
rtpenc: Don't set max_frames_per_packet based on the packet frame size or frame rate
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'bde2bba45c2f2df27a8534028bda09a6e7f835e2':
rtpenc: Restructure if statements in packetizers to simplify adding more conditions
Conflicts:
libavformat/rtpenc_xiph.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'f8c01257f93ceda3e03bc4e540a51022d1e2bff2':
rtpenc: Always do the default initialization regardless of codecs
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'd16c8d28d4e2fca3af1054ffbf635c8cee755fc8':
rtpenc_aac: Use AV_WB16 instead of manual bitshifts
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '9c9b0218e85fcd969308632f75af48a4ce229541':
rtpenc_aac: Merge a definition with a declaration
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '1fc64e2e07787bbca82a72c146588e850e6d098a':
rtpenc: Write conditional statements on separate lines
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '0662440b991361fdb5e732712d997a73e4692e34':
rtpenc_aac: Set a default value for max_frames_per_packet at init
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '12b3459979f5ea6481660cd2c99a0381e2b5ba37':
rtpenc_amr: Use s->num_frames instead of s->buf_ptr - s->buf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b9d2d6843a49f9df1d1ae1afe817d9b48c445919':
tls: Pass AVOptions dictionaries through to the chained protocol
Conflicts:
libavformat/tls.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'e14f98c62fdf8744b07419314095d1b3248cce75':
tcp: Clarify the units for the timeout avoptions
Conflicts:
libavformat/tcp.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
These could trigger assert failures previously
Found-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Instead check the timestamps while muxing, to avoid buffering a
too long timestamp range into one single packet.
This makes the AMR and AAC packetization slightly less efficient,
since we set a possibly unnecessarily high max_frames_per_packet.
(These packetizers end up doing a memmove of the TOC bytes if
sending a packet before max_frames_per_packet is achieved, and
we end up setting max_frames_per_packet to a value that should
be high enough for most uses.)
All packetizers that use max_frames_per_packet now set it either
to a default value, or to a value calculated based on other
parameters, so none of them rely on the previous default setting.
For iLBC, copy one frame at a time, to allow checking the timestamp
range for each of them - basically doing potentially multiple
loops to simplify the code instead of trying to calculate the
number of frames to buffer while honoring s1->max_delay.
This is in preparation for reducing the coupling between libavformat
and libavcodec, by not having the muxers use the encoder field
frame_size (which may not be available during e.g. stream copy).
Signed-off-by: Martin Storsjö <martin@martin.st>
Factorize out the s->num_frames check at the start of the if statements,
simplifying adding more alternative causes for sending the buffered
frames.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids having to jump to the defaultcase in the switch. Manually
override the stream time base back to 90 kHz for the few audio codecs
that don't use the sample rate as time base (mp2, mp3).
Signed-off-by: Martin Storsjö <martin@martin.st>
This doesn't fix any bug, but makes the code simpler for later
patches, and more straightforward to read as is.
Signed-off-by: Martin Storsjö <martin@martin.st>
After sending a fragmented frame, len (s->buf_ptr - s->buf) isn't
zero, while s->num_frames is zero as intended. Using s->num_frames
makes it work as intended, and is less convoluted than keeping track
of (resetting) s->buf_ptr.
This avoids sending stray data after sending a fragmented aac packet.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Don't prefix them ffio_url, which is misleading, sounding too
much like the urlprotocol layer (like ffurl_*).
Signed-off-by: Martin Storsjö <martin@martin.st>
