The asumption is that DCE will remove references to those functions.
However some compilers with certain instrumentation enabled doesn't DCE
those at all, resulting in linking failure. Tested with cl.exe -RTCu -RTCs.
Signed-off-by: Kacper Michajłow <kasper93@gmail.com>
In libavformat/rtmpproto.c:gen_connect(), if check on string length
or check on codec fourcc failed, ff_rtmp_packet_create() allocated
data in pkt would leak. Add ff_rtmp_packet_destory before return error
code.
Signed-off-by: Lidong Yan <502024330056@smail.nju.edu.cn>
Reviewed-by: Zhao Zhili <quinkblack@foxmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
rtmp_write is used both for writing outputs as a server. The
rt->listen flag determines which mode we're running in.
Previously, when running as a server, the message stream id would
always be set to 0 for media/metadata messages. This is surprising
given that we have both responded to "createStream()" with a value
of 1 and sent a "Stream Begin 1" to the client. Furthermore, some
client libraries (Red5) seem to trip up on receiving
"@setDataFrame" on stream 0 (and may be correct to assume that
this message would be sent on stream 1).
When running the cleanup in rtmp_close on failures in rtmp_open,
we can in rare cases end up using rt->playpath, assuming that it
is still set.
The crash could happen if we hit the fail codepath in rtmp_open
while publishing (rt->is_input == 0) with rt->state set to
a value > STATE_FCPUBLISH.
This would normally not happen while publishing; either we have
an error (and rt->state <= STATE_FCPUBLISH) or we reach
rt->state = STATE_PUBLISHING, and then we also return successfully
from rtmp_open.
The unexpected combination of states could happen if the server
responds with e.g. "NetStream.Play.Stop" while expecting
"NetStream.Publish.Start"; this sets rt->state to STATE_STOPPED,
which also fulfills the condition "> STATE_FCPUBLISH".
We don't need to free the rt->playpath/tcurl/flashver strings here;
they're handled via AVOption, and thus are freed automatically when
the protocol instance is freed (that's why they aren't freed
manually within the rtmp_close function either).
We also don't need to free the AVDictionary with options; it's
owned by the caller.
A smaller fix would be to just call rtmp_close before freeing
the strings and dictionary, but as we don't need to free them
at all, let's remove that redundant code.
Signed-off-by: Martin Storsjö <martin@martin.st>
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.
Keep it for external users in order to not cause breakages.
Also improve the other headers a bit while just at it.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
It is undefined behaviour to use a different type for a call
than the actual type of the function.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Makes it robust against adding fields before it, which will be useful in
following commits.
Majority of the patch generated by the following Coccinelle script:
@@
typedef AVOption;
identifier arr_name;
initializer list il;
initializer list[8] il1;
expression tail;
@@
AVOption arr_name[] = { il, { il1,
- tail
+ .unit = tail
}, ... };
with some manual changes, as the script:
* has trouble with options defined inside macros
* sometimes does not handle options under an #else branch
* sometimes swallows whitespace
Unnecessary since acf63d5350adeae551d412db699f8ca03f7e76b9;
also avoids relocations.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This reverts commit bec6dfcd5c.
The patch is NOP since ffurl_open_whitelist copy options from parent
automatically.
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
add option named rtmp_enhanced_codec,
it would support hvc1,av01,vp09 now,
the fourcc is using Array of strings.
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
This avoids unnecessary rebuilds of most source files if only the
list of enabled components has changed, but not the other properties
of the build, set in config.h.
Signed-off-by: Martin Storsjö <martin@martin.st>
It avoids leaving dangling pointers behind in memory.
Also remove redundant checks for whether the URLContext to be closed is
already NULL.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Replicates lavf/librtmp.c behavior in L149-156 and rtmpdump's
behavior with "--swfVfy <url>" passing the url to swfUrl.
Fixes trac ticket #5549.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit 'c541a44e029e8a4f21db028c34fee3ad1c10a409':
Revert "rtmpproto: Don't include a client version in the unencrypted C1 handshake"
Merged-by: Clément Bœsch <u@pkh.me>
* commit '7d8d726be7dc46343ab1c98c339c1ed44bcb07c1':
rtmpproto: Don't include a client version in the unencrypted C1 handshake
Merged-by: Clément Bœsch <u@pkh.me>
* commit '9f23f77a532ca9c2b7dc4b5328bc413e4f6f5b56':
rtmpproto: Don't include the libavformat version as "clientid"
Merged-by: Clément Bœsch <u@pkh.me>
This swaps which field is set when the Window Acknowledgement Size
and Set Peer BW packets are received, renames the fields in
order to clarify their role further and adds verbose comments
explaining their respective roles and how well the code currently
does what it is supposed to.
The Set Peer BW packet tells the receiver of the packet (which
can be either client or server) that it should not send more data
if it already has sent more data than the specified number of bytes,
without receiving acknowledgement for them. Actually checking this
limit is currently not implemented.
In order to be able to check that properly, one can send the
Window Acknowledgement Size packet, which tells the receiver of the
packet that it needs to send Acknowledgement packets
(RTMP_PT_BYTES_READ) at least after receiving a given number of bytes
since the last Acknowledgement.
Therefore, when we receive a Window Acknowledgement Size packet,
this sets the maximum number of bytes we can receive without sending
an Acknowledgement; therefore when handling this packet we should set
the receive_report_size field (previously client_report_size).
Signed-off-by: Martin Storsjö <martin@martin.st>
Also rename comments and log messages accordingly,
and add clarifying comments for some hardcoded values.
The previous names were taken from older, reverse engineered
references.
These names match the official public rtmp specification, and
matches the names used by wirecast in annotating captured
streams. These names also avoid hardcoding the roles of server
and client, since the handling of them is irrelevant of whether
we act as server or client.
The RTMP_PT_PING type maps to RTMP_PT_USER_CONTROL.
The SERVER_BW and CLIENT_BW types are a bit more intertwined;
RTMP_PT_SERVER_BW maps to RTMP_PT_WINDOW_ACK_SIZE and
RTMP_PT_CLIENT_BW maps to RTMP_PT_SET_PEER_BW.
Signed-off-by: Martin Storsjö <martin@martin.st>
Servers seem to be happy to receive the wrapped-around value as long
as they receive a report, otherwise they timeout.
Initially reported and analyzed by Thomas Bernhard.
When bytes_read overflowed, last_bytes_read did not yet overflow
and no bytes-read report was created leading to a timeout.
Analyzed-by: Thomas Bernhard
Fixes ticket #5836.
