Normally, all channel ids are between 0 and 10, while they in
uncommon cases can have values up to 64k.
This avoids allocating two arrays for up to 64k entries (at a total
of over 6 MB in size) each when most of them aren't used at all.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '8921e32f730c191543b84e61338bc9d549aa05a3':
rtmpproto: Readjust the end of the flv buffer if handle_metadata exited early
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '24fee95321c1463360ba7042d026dae021854360':
rtmpproto: Move the flv header/trailer addition to append_flv_data
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '4d6d70292e91a7ef027824d731b6b6570ceabf2f':
rtmpproto: Pass the 'live' parameter in the right unit
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'a6b361325f2bfc8d9d4e5f761d6c1a07b209c4fb':
rtmpproto: Print the error code string if there's no description
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This wasn't an issue prior to 58404738, when the whole RTMP packet
was copied at once and the length of the individual embedded flv
packets only were validated by the flv demuxer.
Prior to this patch, this could lead to reads and writes out of bound.
Signed-off-by: Martin Storsjö <martin@martin.st>
If the embedded flv packets were incomplete and we aborted the
copying loop early, make sure the flv buffer is trimmed to
only contain full packets.
Signed-off-by: Martin Storsjö <martin@martin.st>
update_offset is also called from handle_metadata, where the
packet header sizes is already included in the size.
Previously this lead to flv_data/flv_size including 15 uninitialized
bytes at the end after each call to handle_metadata, making the
flv demuxer lose sync with the stream.
Also remove leftover copying in handle_metadata. This is a leftover
from the refactoring in 5840473. (Previously this final mempcy was
the one that copied all the packets at once, while this is done
within the loop right now.) After making sure flv_size is set to
the right size, this write was out of bounds.
Signed-off-by: Martin Storsjö <martin@martin.st>
This was overlooked in d872fb0f7 since I assumed all the realloc
issues in the rtmp code was fixed already.
Signed-off-by: Martin Storsjö <martin@martin.st>
The current magic numbers passed are values in seconds, while the
parameter itself should be passed over the wire in milliseconds.
This makes (some/all?) live streams from Red5 work correctly, that
previously returned StreamNotFound even with "-rtmp_live live". After
this commit, the default 'any' also works on these streams.
Signed-off-by: Martin Storsjö <martin@martin.st>
On (certain streams/setups at least on) Red5, the description string
actually is present, but empty. Therefore, first try loading the
description, but if not found or empty, load the code string instead.
The code string is quite understandable in most cases anyway (like
"NetStream.Play.StreamNotFound").
Signed-off-by: Martin Storsjö <martin@martin.st>
When av_reallocp fails, the associated variables that keep track of
the number of elements in the array (and in some cases, the
separate number of allocated elements) need to be reset.
Not all of these might technically be needed, but it's better to
reset them if in doubt, to make sure variables don't end up
conflicting.
Signed-off-by: Martin Storsjö <martin@martin.st>
Use update_offset() as done for rtmp audio, video and notifications and
read update and write the fields instead of replacing them in the rtmp
packet and then memcpying it to the output buffer.
* commit 'b97b1adb3f807e1acd00d56319ee6cb41cc727e4':
rtmpproto: Add a comment explaining the logic in handle_notify
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'dc4acc820076b2149ef6c921bdabe05d07ca1bc6':
rtmpproto: Extend a comment to explain the prev_pkt arrays roles
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '647d655d19c38e9716328e4787199149097d6089':
rtmpproto: Consistently use the right prev_pkt array
Merged-by: Michael Niedermayer <michaelni@gmx.at>
A given packet won't always come in contiguously; sometimes
they may be broken up on chunk boundaries by packets of another
channel.
This support primarily involves tracking information about the
data that's been read, so the reader can pick up where it left
off for a given channel.
As a side effect, we no longer over-report the bytes read if
(toread = MIN(size, chunk_size)) == size
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'd4aef997809167832ecc64e89dda8cb445e5fe10':
rtmp: Follow Flash player numbering for channels.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'fe0337e89bbbe84b7274fbb0d9d56ed992937931':
rtmp: Do not send the first field twice within the handshake
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This more closely corresponds to the usage of the field.
Its usage here is unrelated to the channel ID.
Signed-off-by: Martin Storsjö <martin@martin.st>
Channel 4 is typically used by the Flash player to transmit
audio, channel 6 for video, and various stream-specific invokes
get sent over channel 8, which is designated the source channel.
This more closely matches the behavior of the Flash player,
including the transmission of play requests over channel 8.
Signed-off-by: Martin Storsjö <martin@martin.st>
Sending non-monotonic packets (e.g. when the audio and video
streams are monotonic within themselves but not muxed
monotonically) will lead to negative values the RTMP timestamp
field (where timestamps are transmitted only as deltas for each
channel), and this delta can end up being incorrectly written as
a large unsigned number.
Signed-off-by: Martin Storsjö <martin@martin.st>
When streaming to limelight, the app name is either a full
"appname/subaccount" or "appname/_definst_". In the latter case,
the app name can be simplified into simply "appname", but the
authentication hashing assumes the /_definst_ still to be present.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '8e1fe345577a42f99591caf8a06c447613449694':
rtmp: Detect and warn if the user tries to pass librtmp style parameters
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '28306e6d620c109ddd672f7243adfbc2bbb3b18f':
network: factor out bind-listening code
use my full first name instead of short one in copyrights
Conflicts:
libavformat/tcp.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b3ea76624ad1baab0b6bcc13f3f856be2f958110':
vf_aspect: use the name 's' for the pointer to the private context
Remove commented-out debug #define cruft
Conflicts:
libavcodec/4xm.c
libavcodec/dvdsubdec.c
libavcodec/ituh263dec.c
libavcodec/mpeg12.c
libavfilter/avfilter.c
libavfilter/vf_aspect.c
libavfilter/vf_fieldorder.c
libavformat/rtmpproto.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
do_adobe_auth takes the parameters in the order "opaque, challenge".
Due to the way they are treated, this didn't matter in the tested
setups though - if both are set, we only use one. In the tested
setups (Wowza and Akamai) either one of them were null or they
were both set to the same value, which is why this worked before.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
rtmp: Add support for limelight authentication
rtmp: Add support for adobe authentication
Conflicts:
Changelog
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '33f28a3be3092f642778253d9529dd66fe2a014a':
rtmp: Add a function for writing AMF strings based on two substrings
rtmp: Return a proper error code in handle_invoke_error
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Limelight is a not too uncommon CDN. The authentication scheme is
pretty similar to the adobe authentication, but is even closer to
normal http digest authentication (but not close enough to warrant
sharing code) than the adobe version.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is mostly used to authenticate the client when publishing.
Tested with wowza and akamai.
Some but not all servers support resending a new connect invoke
within the same connection, so always reconnect for sending a new
connection attempt. This matches what other applications do as well.
The authentication scheme is structurally pretty similar to http
digest authentication, but uses base64 instead of hex strings.
Signed-off-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes: [FFmpeg-devel] rtmpproto compile error
Similar patch: [FFmpeg-devel] [PATCH] call to strncat replaced with av_strlcat to avoid compile issue with systems implementing strncat via strcat.
* commit 'e002e3291e6dc7953f843abf56fc14f08f238b21':
Use the new aes/md5/sha/tree allocation functions
avutil: Add functions for allocating opaque contexts for algorithms
svq3: fix pointer type warning
svq3: replace unsafe pointer casting with intreadwrite macros
parseutils-test: various cleanups
Conflicts:
doc/APIchanges
libavcodec/svq3.c
libavutil/parseutils.c
libavutil/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '124134e42455763b28cc346fed1d07017a76e84e':
avopt: Store defaults for AV_OPT_TYPE_CONST in the i64 union member
Conflicts:
libavcodec/aacenc.c
libavcodec/libopenjpegenc.c
libavcodec/options_table.h
libavdevice/bktr.c
libavdevice/v4l2.c
libavdevice/x11grab.c
libavfilter/af_amix.c
libavfilter/vf_drawtext.c
libavformat/movenc.c
libavformat/options_table.h
libavutil/opt.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
doc/APIchanges: add an entry for codec descriptors.
vorbisenc: set AVCodecContext.bit_rate to 0
vorbisenc: fix quality parameter
FATE: add ALAC encoding tests
lpc: fix alignment of windowed samples for odd maximum LPC order
alacenc: use s16p sample format as input
alacenc: remove unneeded sample_fmt check
alacenc: fix max_frame_size calculation for the final frame
adpcm_swf: Use correct sample offsets when using trellis.
rtmp: support strict rtmp servers
mjpegdec: support AVRn interlaced
x86: remove FASTDIV inline asm
Conflicts:
doc/APIchanges
libavcodec/mjpegdec.c
libavcodec/vorbisenc.c
libavutil/x86/intmath.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In order to send or receive a stream FCPublish, FCSubscribe and _checkbw
are completely optional and often not implemented. releaseStream over a
non-existen stream might report an error instead of being silent.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* qatar/master:
libvpxenc: use the default bitrate if not set
utvideo: Rename utvideo.c to utvideodec.c
doc: Fix syntax errors in sample Emacs config
mjpegdec: more meaningful return values
configure: clean up Altivec detection
getopt: Remove an unnecessary define
rtmp: Use int instead of ssize_t
getopt: Add missing includes
rtmp: Add support for receiving incoming streams
Add missing includes for code relying on external libraries
Conflicts:
libavcodec/libopenjpegenc.c
libavcodec/libvpxenc.c
libavcodec/mjpegdec.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Not all compilers support ssize_t (MSVC doesn't), and none of these
variables need to be larger than 32 bit.
Signed-off-by: Martin Storsjö <martin@martin.st>
