If there are no variants, the total bitrate of the single
stream isn't known, and exporting variant_bitrate = 0 does
look weird, since there really aren't any variants.
Signed-off-by: Martin Storsjö <martin@martin.st>
This should hopefully fix roundup issue 2586.
This commit only implements it in the demuxer, not in the
protocol handler. If desired, some of the code could be
refactored to be shared by both implementations.
Signed-off-by: Martin Storsjö <martin@martin.st>
This can later be extended to support other AES bit sizes,
encryption, other crypto algorithms, reading the key from a URL, etc.
In order to use it, the key and initialization vector has to be
passed via AVOptions. Since such options can't be passed to
protocols from the command line, the protocol is currently
only for libavformat internal use.
Signed-off-by: Martin Storsjö <martin@martin.st>
Setting this should fix the FATE build failure on DragonFly BSD,
since inet_aton only is visible if __BSD_VISIBLE is set.
Alternatively, a line defining __BSD_VISIBLE=1 could be
added at the top of os_support.c. For FreeBSD, similar lines
are required in libavdevice/bktr.c and libavdevice/oss_audio.c, too.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes seek_test to be rebuilt when its dependencies
has changed. The changes to the dependencies didn't usually matter
in practice, but the introduction of side data in AVPacket required
a recompilation.
Signed-off-by: Martin Storsjö <martin@martin.st>
After switching this from a statically allocated array to a
dynamically allocated one in the major bump, this needs explicit
freeing.
Signed-off-by: Martin Storsjö <martin@martin.st>
DTX, discontinuous transmission, allows emitting frames with
comfort noise when no voice is detected in the input audio.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also rename the incorrectly named enc_bitrate to enc_mode, use the
enc_bitrate variable for storing the last chosen bitrate.
This avoids continuous warning log messages if not using an
exactly matching bitrate, while still allowing changing bitrate
at any point.
Signed-off-by: Martin Storsjö <martin@martin.st>
Dynamically print the supported bitrates from the local table,
instead of using a hardcoded log message.
Signed-off-by: Martin Storsjö <martin@martin.st>
Include VisualOn to clarify the codec name, but remove the
actual library name from the user-friendly description.
Also mention Android, to clarify which VisualOn implementation
this refers to, since they do sell other variants of the same
code, too.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids warnings of this kind, everywhere priv_class
is initialized:
warning: initialization discards qualifiers from pointer target type
Signed-off-by: Martin Storsjö <martin@martin.st>
This code could be executed if the demuxer reads more than one
segment before returning from av_open_input_stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
Stop the avio input at a point where amf_parse_object can
continue parsing the end of the object seamlessly, when all
data is available.
If unsupported data is encountered within the keyframes object,
try seeking to the start of the keyframes object - if the seek
back was successful, the caller can continue parsing the rest
of the AMF data.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes a failing assert in ff_raw_read_header (in
fate-g722dec-1), where bits_per_coded_sample is set using this
function and is required to have a positive value.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids issues where EOF at the end of the segment is given
the variant demuxer. Now the demuxers only see one single data
stream (as when using the applehttp protocol handler).
According to the RFC, GET_PARAMETER should be used for
this, and according to a report from Tim Ouellette,
OPTIONS doesn't work for keeping the connection alive for some
servers. Also, live555 uses GET_PARAMETER for this purpose.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This allows libavformat to guess an estimated duration for
amr files.
For streams with varying bit rates (or with silence descriptors
or "no frame" blocks) the guess is, of course, inaccurate.
According to the RFC, the default is multicast if nothing is
specified, which doesn't make sense for TCP.
According to a bug report, some Axis camera models give a
"400 Bad Request" error if this is omitted.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
A similar variable for the total stream duration was changed to
int64_t in b79c3df088, due to overflows in some odd
streams.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This fixes crashes on exit when plaing some RealRTSP streams
(among other), e.g.
rtsp://dl.lib.brown.edu:554/areserves/1093545294660883.mp3.
These crashes have been present since 3e68b3ba7b.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Add an APIchanges entry for the av_pkt_dump2 and av_pkt_dump_log2
functions, and a changelog entry for the apple http live streaming
protocol handler.
Since neither of them got a minor bump at commit time, but were
applied before the jv demuxer, they all can be considered added
in this minor version.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
If this flag is set, the protocol can handle URLs where the
scheme is a nested scheme such as applehttp+file: - the protocol
can handle any URL where the first segment of the nested scheme
belongs to this protocol.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
In most cases, s->buf_ptr will be equal to s->buf_end when
fill_buffer is called, but this may not always be the case, if
we're seeking forward by reading (permitted by the short seek
threshold).
If fill_buffer is writing to s->buf_ptr instead of s->buf_end (when
they aren't equal and s->buf_ptr is ahead of s->buffer), the data
between s->buf_ptr and s->buf_end is overwritten, leading to
inconsistent buffer content. This could return incorrect data if
later seeking back into the area before the current s->buf_ptr.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Map EAGAIN and EINTR from ff_neterrno to the normal AVERROR()
error codes. Provide fallback definitions of other errno.h network
errors, mapping them to the corresponding winsock errors.
This eases catching these error codes in common code, without having
to distinguish between FF_NETERRNO(EAGAIN) and AVERROR(EAGAIN).
This fixes roundup issue 2614, unbreaking blocking network IO on
windows.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
If udp_read_packet returns 0, rtsp_st isn't set and we shouldn't
treat it as a successfully received packet (which is counted and
possibly triggers a RTCP receiver report).
This fixes issue 2612.
This is used for mapping AVStreams back to their corresponding
RTSPStream. Since d9c0510, the RTSPStream pointer isn't stored in
AVStream->priv_data any longer, breaking this mapping from AVStreams
to RTSPStreams.
Also, we don't need to clear the priv_data in rdt cleanup any longer,
since it isn't set to duplicate pointers.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
If required, the caller can do this itself. ff_write_chained rescales
timestamps as necessary, and all current callers of rtpenc_chain
use ff_write_chained, making this timebase copy unnecessary.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This function is useful for freeing data structures allocated by
muxers, which currently have to be freed manually by the caller.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This avoids having the chained AVStream->codec point to the same
AVCodecContext owned by the outer AVStream. The downside is that
changes to the AVCodecContext made after calling av_write_header
cannot be detected automatically within the chained muxer.
This avoids having to manually unlink the chained AVStream->codec
by setting it to null before freeing the chained muxer via generic
freeing functions.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This fixes memory leaks in the RTSP muxer and RTP hinting in the
mov muxer present since SVN rev 25418.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
For mpegts in RTP, there isn't a direct mapping between RTSPStreams
and AVStreams, and the RTSPStream isn't ever stored in
AVStream->priv_data, which was earlier leaked. The fix for this
leak, in ea7f080749, lead to
double frees for other, normal RTP streams.
This patch avoids storing RTSPStreams in AVStream->priv_data, thus
avoiding the double free. The RTSPStreams are always available via
RTSPState->rtsp_streams anyway.
Tested with MS-RTSP, RealRTSP, DSS and mpegts/RTP.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The key string is supposed to contain the equals character,
too. Since the checked string was wrong, and the return value
check was wrong too, it incorrectly seemed to work right before.
Signed-off-by: Mans Rullgard <mans@mansr.com>
If filtered, only packets from the right source address and port
are received.
To test, play back e.g. some mpeg4 video RTSP stream (where the
video stream is the first stream in the presentation) over UDP.
While receiving this stream, send another stream to the same port:
ffmpeg -re -i <whatever> -vcodec mpeg4 -an -f rtp
rtp://127.0.0.1:5000?localport=1234
Normally, the RTSP playback reports lots of errors at this point.
If the RTSP stream has the ?filter_src option enabled, these
interferring packets are ignored.
Originally committed as revision 26246 to svn://svn.ffmpeg.org/ffmpeg/trunk
If the remote address is updated later with this function, the caller
shouldn't set the connect option until in this call.
Originally committed as revision 26245 to svn://svn.ffmpeg.org/ffmpeg/trunk
By calling connect on the UDP socket, only packets from the chosen
peer address and port are received on the socket. This is one
solution to issue 1688.
Originally committed as revision 26244 to svn://svn.ffmpeg.org/ffmpeg/trunk
For MS-RTSP, we don't always get RTCP packets (never?), so the earlier
timestamping code never wrote anything into pkt->pts. The rtpdec_asf
depacketizer just sets the dts of the packet, so if the generic RTP
timestamping is used, too, we get inconsistent timestamps.
Therefore, skip the generic RTP timestamp algorithm if the depacketizer
already has set something.
This fixes "Invalid timestamps" warnings, present since SVN rev 26187.
Originally committed as revision 26241 to svn://svn.ffmpeg.org/ffmpeg/trunk
Martin Storsjö
