This fixes sending back RTCP RR packets if receiving RTP over
multicast.
If the multicast stream is sent on demand (set up and signalled
via RTSP), the sender might depend on getting RTCP RR packets
knowing that there are listeners, otherwise the stream can be
closed after a certain timeout.
This fixes receiving RTSP streams over multicast on unix, from
certain Axis cameras.
Signed-off-by: Martin Storsjö <martin@martin.st>
When this code was added in 36b532815c, the new code was added
between the existing comment and the existing line of code, making
the old comment seem to refer to the new code. This makes it read
correctly.
Signed-off-by: Martin Storsjö <martin@martin.st>
The ogg decoder wasn't padding the input buffer with the appropriate
FF_INPUT_BUFFER_PADDING_SIZE bytes. Which led to uninitialized reads in
various pieces of parsing code when they thought they had more data than
they actually did.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Also, do not keep trying to find and open a decoder in try_decode_frame() if
we already tried and failed once.
Fixes always searching until max_analyze_duration in
avformat_find_stream_info() when demuxing codecs without a decoder.
Also, do not give AVCodecContext.frame_size priority for muxing.
Updated 2 FATE references:
dxa-feeble - adds 1 audio frame that is still within 2 seconds as specified
by -t 2 in the FATE test
wmv8-drm-nodec - durations are not needed. previously they were estimated
using the packet size and average bit rate.
It is unnecessary. Also, for some codecs we're reading more than 1 frame per
packet. Instead we use a private context variable to calculate the bit rate,
stream duration, and packet durations.
Updated FATE seek test, which has slightly different timestamps due to a
more accurate bit rate calculation.
For encoding, frame_size is not a reliable indicator of packet duration.
Also, we don't want to have to force the demuxer to find frame_size for
stream copy to work.
Split off packet parsing into a separate function. Parse full packets at
once and store them in a queue, eliminating the need for tracking
parsing state in AVStream.
The horrible unreadable loop in read_frame_internal() now isn't weirdly
ordered and doesn't contain evil gotos, so it should be much easier to
understand.
compute_pkt_fields() now invents slightly different timestamps for two
raw vc1 tests, due to has_b_frames being set a bit later. They shouldn't
be more wrong (or right) than previous ones.
Make packet buffer a parameter, don't hardcode it to be
AVFormatContext.packet_buffer.
Also move the function higher in the file, since it will be called from
read_frame_internal().
The time base is 1 / sample_rate, not 90000.
Several more codecs encode the sample count in the first 4 bytes of the
chunk, so we set the durations accordingly. Also, we can set start_time and
packet duration instead of keeping track of the sample count in the demuxer.
Fixes timestamp calculation.
The FATE reference is updated because timestamp calculations are now more
accurate. Previous timestamps were based on average bit rate.
This allows it to be used with get_bits without the thread of overreads.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
The container has no timestamps and the framerate isn't stored in the
data either.
The decoder sets codec timebase to experimentally found value 1/15. Do
the same for the demuxer too, it should at least be better than the
default 1/90000.
The fields "Number of Bytes" and "Number of Frames" are mixed up. "Bytes"
come first, "Frames" behind.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Alex Converse <alex.converse@gmail.com>
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>
Frame sizes in ID3v2.3 are not synchsafe, they are simply 32be numbers.
In practice this bug is not noticeable unless the frame size takes more
than 7 bits (which is almost never for text frames).
Seeking back on EOF will reset the EOF flag, causing us to re-enter
the loop to find the next marker in the ASF file, thus potentially
causing an infinite loop.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Provide a way to wrap around the segment index so pseudostreaming
live through a web server and html5 browser is simpler.
Also ensure that 0 (disable) is a valid value across the options
providing wrap around.
By validating the index entries while reading, we don't need to
seek at startup to validate the entries. If the error in the
index entries is not pointing to (our definition of) the start
of packets, and there is an index entry pointing at some of the
first packets after the metadata, the invalid index can be discarded
almost immediately.
Signed-off-by: Martin Storsjö <martin@martin.st>
This returns 200 OK for OPTIONS requests and 501 Not Implemented
for all other requests.
Even though this doesn't do much actual handling of the requests,
it makes the code properly identify server requests as such, instead
of interpreting it as a reply to the client's request as it did
before.
Signed-off-by: Martin Storsjö <martin@martin.st>
For encoding, AVCodecContext.frame_size is the number of input samples to
send to the encoder and does not necessarily correspond directly to the
timestamps of the output packets.
This prevents certain tags with a default value assigned to them (as per
the EBML syntax elements) from ever being assigned a NULL value. Other
parts of the code rely on these being non-NULL (i.e. they don't check for
NULL before e.g. using the string in strcmp() or similar), and thus in
effect this prevents crashes when reading of such specific tags fails,
either because of low memory or because of targeted file corruption.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
compute_pkt_fields() is for unreliable estimates or guessing. The
keyframe information from the parser is (at least in theory) reliable,
so it should be used even when the other guessing is disabled with the
AVFMT_FLAG_NOFILLIN flag.
Therefore, move setting the packet keyframe flag based on parser
information from compute_pkt_fields() to read_frame_internal().
fixes a memleak for Vorbis and Theora, where the comment header from
avpriv_split_xiph_headers() is replaced by a buffer that must be freed
separately.