| Age | Commit message (Collapse) | Author |
|---|
|
|
|
|
|
Signed-off-by: Diego Biurrun <diego@biurrun.de>
|
|
Currently, AVStream contains an embedded AVCodecContext instance, which
is used by demuxers to export stream parameters to the caller and by
muxers to receive stream parameters from the caller. It is also used
internally as the codec context that is passed to parsers.
In addition, it is also widely used by the callers as the decoding (when
demuxer) or encoding (when muxing) context, though this has been
officially discouraged since Libav 11.
There are multiple important problems with this approach:
- the fields in AVCodecContext are in general one of
* stream parameters
* codec options
* codec state
However, it's not clear which ones are which. It is consequently
unclear which fields are a demuxer allowed to set or a muxer allowed to
read. This leads to erratic behaviour depending on whether decoding or
encoding is being performed or not (and whether it uses the AVStream
embedded codec context).
- various synchronization issues arising from the fact that the same
context is used by several different APIs (muxers/demuxers,
parsers, bitstream filters and encoders/decoders) simultaneously, with
there being no clear rules for who can modify what and the different
processes being typically delayed with respect to each other.
- avformat_find_stream_info() making it necessary to support opening
and closing a single codec context multiple times, thus
complicating the semantics of freeing various allocated objects in the
codec context.
Those problems are resolved by replacing the AVStream embedded codec
context with a newly added AVCodecParameters instance, which stores only
the stream parameters exported by the demuxers or read by the muxers.
|
|
This applies to every library where performance is not critical.
|
|
frame rate
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>
|
|
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 makes things more consistent by using the variable in the same
way as in all other packetizers.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
Intentionally keeping some conditional statements on single lines
in rtpenc_h263.c.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
This avoids having to conditionally set the default within the
packetizer function.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
They share a great deal of common structure; only a few minor
bits in the headers differ.
This also fixes an off-by-one in sending of the last fragment
of large HEVC nals (where it previously sent len+2 bytes, even
if it should have been len+RTP_HEVC_HEADERS_SIZE aka len+3).
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
The packetizer only supports splitting at GOB headers - if
such aren't available frequently enough, it splits at any
random byte offset (not at a macroblock boundary either, which
would be allowed by the spec) and sends a payload header pretend
that it starts with a GOB header.
As long as a receiver doesn't try to handle such cases cleverly
but just drops broken frames, this shouldn't matter too much
in practice.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
Instead explicitly jump to the default case in the cases where
it is wanted, and avoid fallthrough between different codecs,
which could easily introduce bugs if people editing the code
aren't careful.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
In particular, when packetizing mpegts into rtp, the input packet
timestamp may come from more than one stream, which could cause
multiple packets be written with the same timestamp.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
Signed-off-by: Diego Biurrun <diego@biurrun.de>
|
|
Fall back to 1 (which is what is used for most cases anyway) when the
framerate is not set.
|
|
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
|
|
Expose the current sequence number via an AVOption - this can
be used both for setting the initial sequence number, or for
querying the current number.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
Support multiple video/audio streams with different format in the
same session.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
|
|
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
Tested to work with the gstreamer depacketizer.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
This packetization scheme simply places the full packets into the
RTP packet without any extra header bytes.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
|
|
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
|
|
|
|
This makes usage of the macro look more natural when
used with array entries.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
|
|
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
Also return a proper error code in these cases.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
This allows the caller to set it, and allows the caller to query
what it was set to.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
Also add missing trailing commas, break long codec_tag lines and
add spaces in codec_tag declarations.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
Make the muxers/demuxers that use the field handle the default
-1 in the same way as 0.
This allows distinguishing an intentionally set 0 from the default
value where the user hasn't set it.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
The private option has not been part of any release yet (and
it is only of use in quite rare cases), so just remove it instead
of keeping it with deprecation warnings.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
The current one has a zero denominator - this is what was
intended in 14aecc50fae6.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
It is more reliable than AVCodecContext.frame_size for codecs with constant
packet duration.
|
|
This makes the packetization spec compliant for cases where one single
GOB doesn't fit into an RTP packet.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
This fixes cases where the user had specified one desired MTU
via an option, and the protocol indicates another one.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
This allows opting for a lower MTU than what the AVIOContext
indicated, and allows writing into outputs that don't indicate
an MTU at all (such as plain files, which is useful for testing).
This also allows querying for the MTU via the avoption.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
This is in preparation for exposing this via an avoption.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
Also return a proper error code.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
|
According to newer RFCs, this packetization scheme should only
be used for interfacing with legacy systems.
Implementing this packetization mode properly requires parsing
the full H263 bitstream to find macroblock boundaries (and knowing
their macroblock and gob numbers and motion vector predictors).
This implementation tries to look for GOB headers (which
can be inserted by using -ps <small number>), but if the GOBs
aren't small enough to fit into the MTU, the packetizer blindly
splits packets at any offset and claims it to be a GOB boundary
(by using Mode A from the RFC). While not correct, this seems
to work with some receivers.
Signed-off-by: Martin Storsjö <martin@martin.st>
|
