This allows callers with avio write callbacks to get the bytestream
positions that correspond to keyframes, suitable for live streaming.
In the simplest form, a caller could expect that a header is written
to the bytestream during the avformat_write_header, and the data
output to the avio context during e.g. av_write_frame corresponds
exactly to the current packet passed in.
When combined with av_interleaved_write_frame, and with muxers that
do buffering (most muxers that do some sort of fragmenting or
clustering), the mapping from input data to bytestream positions
is nontrivial.
This allows callers to get directly information about what part
of the bytestream is what, without having to resort to assumptions
about the muxer behaviour.
One keyframe/fragment/block can still be split into multiple (if
they are larger than the aviocontext buffer), which would call
the callback with e.g. AVIO_DATA_MARKER_SYNC_POINT, followed by
AVIO_DATA_MARKER_UNKNOWN for the second time it is called with
the following data.
Signed-off-by: Martin Storsjö <martin@martin.st>
Use the newly created vlc.h directly instead of including get_bits when needed.
The VLC and RL_VLC_ELEM structures are independent from the bitreader.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Allows emulation to work when dst is equal to src2 as long as the
instruction is commutative, e.g. `addps m0, m1, m0`.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The yasm/nasm preprocessor only checks the first token, which means that
parameters such as `dword [rax]` are treated as identifiers, which is
generally not what we want.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Those instructions are not commutative since they only change the first
element in the vector and leave the rest unmodified.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
When feeding input RTP packets to the depacketizer via custom IO,
it needs to pick the right stream using the payload type for
RTP packets, and using the SSRC for RTCP packets. If the first
packet is an RTCP packet, we don't (currently) know the SSRC
yet and thus can't pick the right RTP depacketizer to handle it.
By parsing the SSRC attribute in the SDP, we can map initial
RTCP packets to the right stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
It doesn't matter what the actual reason for not returning
an AVPacket was - if we didn't return any packet and we have
the next one queued, parse it immediately. (rtp_parse_queued_packet
always consumes a queued packet if one exists, so there's no risk
for infinite loops.)
Signed-off-by: Martin Storsjö <martin@martin.st>
Widen the values from limited to full range and use BT.709 where it
should be used according to the video resolution:
SD is BT.601, HD is BT.709
Default to BT.709 due to most observed HDMV content being HD.
BT.709 coefficients were gathered from the first two parts of BT.709
to BT.2020 conversion guide in ARIB STD-B62 (Pt. 1, Chapter 6.2.2).
They were additionally confirmed by manually calculating values.
The declarations that this comment referred to were removed
in 2439f2ca8 - there is no unbuffered IO in this header now.
Signed-off-by: Martin Storsjö <martin@martin.st>
We still only support one single layer though, but this allows
receiving streams that have this structure present even for
single layer streams.
Signed-off-by: Martin Storsjö <martin@martin.st>
