Functions interrupted by url_interrupt_cb should not be restarted.
Therefore using AVERROR(EINTR) was wrong, as it did not allow to distinguish
when the underlying system call was interrupted and actually needed to be
restarted.
This fixes roundup issues 2657 and 2659 (ffplay not exiting for streamed
content).
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
In the event of overflow, the JV_PADDING state will avio_skip over
any overflow bytes (using JVFrame.total_size).
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
when writing and pressing q during encoding. Instead, check url_interrupt_cb
at the end.
Note that when a protocol is interrupted by url_interrupt_cb, some data may
be silently discarded: the protocol context is not suitable for anything
anymore.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This is a substitute for the url_fskip function that was deprecated by
commit 0300db8ad7. avio_fskip is provided to
improve demuxer code readability. It distinguishes the act of skipping over
unknown or irrelevant bytes from the standard avio_seek operation.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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>
Also change the comments a bit since the FOURCCs aren't specific to Flip4Mac
and different ones are used for 720 versus 1080 lines.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This adds the AAC codec to the list of audio codecs that results
in a PES stream_id of 0xc0 (audio stream).
Signed-off-by: Mans Rullgard <mans@mansr.com>
In the name of consistency:
put_byte -> avio_w8
put_<type> -> avio_w<type>
put_buffer -> avio_write
put_nbyte will be made private
put_tag will be merged with avio_put_str
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
In the name of consistency:
get_byte -> avio_r8
get_<type> -> avio_r<type>
get_buffer -> avio_read
get_partial_buffer will be made private later
get_strz is left out becase I want to change it later to return
something useful.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Update libavformat/version.h and doc/APIChanges after renaming
init_put_byte() and ByteIOContext to ffio_init_context() (private)
and AVIOContext, (public), and deprecating the originals.
init_put_byte should never be used outside of lavf, since
sizeof(AVIOContext) isn't part of public ABI.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Allows playback of nonprimary audio streams in multiple bitrate sources,
such as mmsh://wmscr1.dr.dk/e02ch03m
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
The new av_parse_time() is created in libavutil/parseutils.h, all the
internal functions used by parse_date are moved to
libavutil/parseutils.c and made static.
Signed-off-by: Mans Rullgard <mans@mansr.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.
The current implementation has a bug, it is returning the stream index
in the found program, and not the stream index in the list of all
streams. The attached patch fixes this issue.
Signed-off-by: Mans Rullgard <mans@mansr.com>
It is pretty hopeless that other considerable projects will adopt
libavutil alone in other projects. Projects that need small footprint
are better off with more specialized libraries such as gnulib or rather
just copy the necessary parts that they need. With this in mind, nobody
is helped by having libavutil and libavcore split. In order to ease
maintenance inside and around FFmpeg and to reduce confusion where to
put common code, avcore's functionality is merged (back) to avutil.
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
There is a check for HAVE_BIGENDIAN when outputting the IEC 61937
stream. On big-endian systems the payload data is not byteswapped,
causing in effect the outputted payload data to be in a different byte
order on big-endian than on little-endian systems.
However, the IEC 61937 preamble (and the final odd byte if present) is
always outputted in the same byte order. This means that on big-endian
systems the headers have a different byte order than the payload,
preventing useful use of the output.
Fix that by outputting the data in a format suitable for sending to an
audio device in S16LE format by default. Output as big-endian (S16BE)
is added as an AVOption. This makes the muxer output the same on all
archs by default.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
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>
it's not touched anywhere in ffmpeg, the code setting it was removed
over two years ago (e9b78eeba2).
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This code will be later split out into a function which takes a 'size'
argument, so I'm keeping the name 'sizeX' here.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Keep the original corner case behaviour, where reuse is enabled
for the case where no argument is given to the reuse url option.
Signed-off-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Currently (since the data_offset fix) the ogg demuxer assumes that
after the first non-header packets in any stream no more header packets
will follow.
This is not guaranteed, so change the code back again to wait until it
has finished the headers for all streams before returning from ogg_get_headers.
This fixes issue 2428.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This validate the length of a mkv element directly after reading
it.
This has the advantage that it is easy to add new limits and makes
it less likely to forget to add checks and also avoids issues like
bits of the length value above the first 32 being ignored because
the parsing functions only takes an int.
Previously discussed in the "mkv 0-byte integer parsing" thread.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Change int64_t into a int, which caused this compiler warning:
libavformat/oggparseskeleton.c:64: warning: passing argument 2 of ‘av_reduce’ from incompatible pointer type
Use avio functions instead of bytestream ones (also drops dependency on
lavc and removes a bunch of warnings).
Drop custom version of avio_get_str16 and use that instead.
Tested on mewmew-ssa.avi sample.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Makes playing QDMC files in MPlayer work when using the libavformat demuxer.
Problem was that the extradata was not passed from demuxer to decoder.
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>
Now the first argument is URLContext *h. However, the function logs to
LOG_CONTEXT, which is #defined as 's' for new lavf major versions.
Therefore, rename h -> s.
Signed-off-by: Mans Rullgard <mans@mansr.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 bumps are for adding version.h and avio_{get/put}_str functions in
lavf and making av_dlog public in lavu.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
The first part of the metadata, the "vendor" string, is required by
libvorbis, it will refuse to play when it is not available.
Also we do not currently parse that part into metadata so it would also
be lost if we removed it as well.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Around 01/28/11 18:56, Ronald S. Bultje scribbled:
> That patch is now merged, can you submit the update to muxers.texi?
> Then we'll apply the whole thing.
See attached. I hope the documentation is enough.
--
Georgi Chorbadzhiyski
http://georgi.unixsol.org/
From c236024b8254f5c2c45934c30fff390cb0e55a5e Mon Sep 17 00:00:00 2001
From: Georgi Chorbadzhiyski <gf@unixsol.org>
Date: Tue, 25 Jan 2011 13:09:17 +0200
Subject: [PATCH] mpegts: Replace defines in with AVOptions
This patch adds support for setting transport_stream_id,
original_network_id, service_id, pmt_start_pid and start_pid
in mpegts muxer.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Avoids an assert when the sample rate is invalid and the timebase
is thus set to e.g. 1/0.
Sample file is http://samples.mplayerhq.hu/ogg/fuzzed-srate-crash.ogg
This is a quick fix for a crash, not a final solution.
Signed-off-by: Mans Rullgard <mans@mansr.com>
$subject. Have used this for loopback testing with mpegts.c.
-- Peter
(A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B)
[2. text/x-diff; 0001-mpegtsenc-support-CODEC_ID_AAC_LATM.patch]
From 0f7f9db4b7da1793996af6dda84298507703759a Mon Sep 17 00:00:00 2001
From: Peter Ross <pross@xvid.org>
Date: Sun, 9 Jan 2011 09:45:50 +1100
Subject: [PATCH] mpegtsenc: support CODEC_ID_AAC_LATM
Signed-off-by: Mans Rullgard <mans@mansr.com>
poll() is only used by networking code, so the fallback should
only be built if networking is enabled. Also remove CONFIG_FFSERVER
condition from the declarations.
This should fix building on systems without poll(), broken
by a8475bbdb6.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Add an error message in case the user requests to write more than one file
and the path does not contain a "%d" or "%0Nd" pattern.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
According to EN 300 468 section 3.1 (Definitions):
Unless otherwise specified within the present document all
"reserved_future_use" bits is set to "1".
This was not the case for SDT generation so this patch fixes it.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
The iff.h header only declared one function that is now static, the
libavformat/iff.c source file wasn't using it before. Drop the file
entirely.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Remove two variables that were not used and caused the following
warnings:
CC libavformat/mpegtsenc.o
libavformat/mpegtsenc.c: In function 'mpegts_write_section':
libavformat/mpegtsenc.c:72:18: warning: unused variable 'ts'
libavformat/mpegtsenc.c: In function 'mpegts_insert_null_packet':
libavformat/mpegtsenc.c:586:18: warning: unused variable 'ts'
Signed-off-by: Mans Rullgard <mans@mansr.com>
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>
make the initialization of put clearer
this are the differences between
[FFmpeg-devel] [PATCH 1/3] mp3enc: add support for writing UTF-16 tags
and the already applied 187e23478b
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
Set service_provider and service_name in mpegts demuxer, previously
name and provider_name were set but since the muxer uses service_provider
and service_name use them.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Apparently some broken taggers prepend a new ID3v2 tag leaving the
existing one intact. Our parser currently reads all tags and overwrites
existing values with supposedly outdated ones.
fixes issue2419
Signed-off-by: Mans Rullgard <mans@mansr.com>
This patch adds support in mpegts muxer for using service_provider and
service_name metadata to set service_provider_name and service_name
fields in SDT.
Example usage:
ffmpeg -i file.ts -f mpegts -re -acodec copy -vcodec copy -f mpegts \
-metadata service_provider="Some provider" \
-metadata service_name="Some Channel" \
"udp://239.0.70.2:5000?pkt_size=1316&ttl=1"
Signed-off-by: Mans Rullgard <mans@mansr.com>
This should improve duration accuracy slightly and avoids a warning about its
inaccuracy when accurate values are available. Idea by Frank Barchard
Originally committed as revision 26366 to svn://svn.ffmpeg.org/ffmpeg/trunk
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
close the stream.
This way the flushed packets can still reference the still unclosed
format context.
In particular this fixes a spurious error issued when closing the
video4linux2 buffer in mmap_release_buffer(), which tries to access
the file descriptor of an already closed file.
Originally committed as revision 26237 to svn://svn.ffmpeg.org/ffmpeg/trunk
This avoids having a large temporary buffer in the struct used for
storing the rtsp reply headers.
Originally committed as revision 26192 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allows ff_rtsp_parse_line to do more changes directly in RTSPState
when parsing the reply, instead of having to store large amounts of
temporary data in RTSPMessageHeader.
Originally committed as revision 26190 to svn://svn.ffmpeg.org/ffmpeg/trunk
Emitted timestamps in each stream start from 0, for the first received
RTP packet. Once an RTCP packet is received, that one is used for
sync, emitting timestamps that fit seamlessly into the earlier ones.
Originally committed as revision 26187 to svn://svn.ffmpeg.org/ffmpeg/trunk
wav. In that case, DTS can be transmitted through S/PDIF without
the IEC 61937 headers.
Patch by Anssi Hannula, anssi d hannula a iki d fi
Originally committed as revision 26160 to svn://svn.ffmpeg.org/ffmpeg/trunk
Noticed by CrystalP from XBMC.
Patch by Anssi Hannula, anssi d hannula a iki d fi
Originally committed as revision 26130 to svn://svn.ffmpeg.org/ffmpeg/trunk
For example MS-RTSP doesn't have RTPDemuxContexts for all streams.
This fixes issue 2448.
Originally committed as revision 26107 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes a crash if we requested TCP interleaved transport, but the
server replies with transport data for UDP. According to the RFC, the
server isn't allowed to respond with another transport type than the
one requested.
Originally committed as revision 26077 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes compilation with --disable-everything --enable-<component>,
for all encoders, decoders, muxers, demuxers, parsers, protocols, bsfs,
indevs, outdevs and filters at the moment. (All those that work without
any external dependencies at least.)
Originally committed as revision 26076 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes one of the issues found if building with
--disable-everything --enable-muxer=webm
Originally committed as revision 26066 to svn://svn.ffmpeg.org/ffmpeg/trunk
While not mentioned in RFC 4629, this is required for H.263 in
3GPP TS 26.234. It is in practice required for playback with
Android stagefright and on Samsung bada phones.
Originally committed as revision 26062 to svn://svn.ffmpeg.org/ffmpeg/trunk
(largest size according to spec: 64k). Fixes playback of
mmsh://a1635.v24937.c2493.g.vm.akamaistream.net/7/1635/2493/v0001/premrad.download.akamai.com/2493/premiere_rock_report/Country_Report.wma
Patch by Zhentan Feng <spyfeng gmail com>.
Originally committed as revision 26047 to svn://svn.ffmpeg.org/ffmpeg/trunk
This also reverts SVN rev 26016, which incorrectly overwrote the time base
with 90 kHz for all streams, regardless of what was set by the SDP parsing.
The stream that triggered the fix in 26016 still works after this commit.
Originally committed as revision 26022 to svn://svn.ffmpeg.org/ffmpeg/trunk
The generic default is 0/0 and that obviously triggers once the value is used.
Originally committed as revision 26016 to svn://svn.ffmpeg.org/ffmpeg/trunk
This makes it possible to abort a blocking connect call.
Patch by Thomas Guillem, thomas dot guillem at gmail
Originally committed as revision 26014 to svn://svn.ffmpeg.org/ffmpeg/trunk
of returning packets with uninitialized data.
Returning partial packets as for other demuxers is problematice due to
packet scrambling and thus is not done.
Originally committed as revision 25931 to svn://svn.ffmpeg.org/ffmpeg/trunk
ID3v2 tags which no longer works since ID3v2 handling was moved to
generic code.
In addition, in caused false-positives for all files starting with
one or more 0-bytes.
Originally committed as revision 25929 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes cases where the RTP time base and the sample rate of the stream
differ. Previously, the AVStream time_base was unconditionally set to
the sample rate (which initially was set to one value when parsing the
rtpmap field in the SDP, but later overridden by an a=SampleRate field).
Additionally, this makes the code actually use the stream time base set
in rtpmap for video codecs, instead of hardcoding it to always be 90 kHz.
Originally committed as revision 25908 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes playback of a RealRTSP/MP3 URL from the RTSP samples on
MultimediaWiki.
Originally committed as revision 25906 to svn://svn.ffmpeg.org/ffmpeg/trunk
According to the spec, this type shouldn't ever be used. Nevertheless,
passing it through enables decoding streams which otherwise aren't
decodeable.
Originally committed as revision 25897 to svn://svn.ffmpeg.org/ffmpeg/trunk
The current implementation is incompatible with the latest spec drafts,
this should be communicated clearly to the user.
Originally committed as revision 25887 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes PCR drift due to accumulating TS_PACKET_SIZE*8*90000LL/ts->mux_rate each packet, due to rounding errors when mux_rate does not evenly divide 135360000.
This patch also increases the PCR precision to 27 MHz from 90 kHz and takes the location of the PCR data into account (+11 bytes according to the spec).
Originally committed as revision 25864 to svn://svn.ffmpeg.org/ffmpeg/trunk
data packets before the first complete one.
Patch by Aaron Colwell [acolwell chromium org].
Originally committed as revision 25846 to svn://svn.ffmpeg.org/ffmpeg/trunk
This reverts snprintf->av_d2str part of r20840.
With it, track number is exported as a float, which is not
desirable.
Originally committed as revision 25845 to svn://svn.ffmpeg.org/ffmpeg/trunk
