All current usages of it are incompatible with localization.
For example strcasecmp("i", "I") != 0 is possible, but would
break many of the places where it is used.
Instead use our own implementations that always treat the data
as ASCII.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master: (44 commits)
replacement Indeo 3 decoder
gsm demuxer: do not allocate packet twice.
flvenc: use first packet delay as global delay.
ac3enc: doxygen update.
imc: return error codes instead of 0 for error conditions.
imc: return meaningful error codes instead of -1
imc: do not set channel layout for stereo
imc: validate channel count
imc: check for ff_fft_init() failure
imc: check output buffer size before decoding
imc: use DSPContext.bswap16_buf() to byte-swap packet data
rtsp: add allowed_media_types option
libgsm: add flush function to reset the decoder state when seeking
libgsm: simplify decoding by using a loop
gsm: log error message when packet is too small
libgsmdec: do not needlessly set *data_size to 0
gsmdec: do not needlessly set *data_size to 0
gsmdec: add flush function to reset the decoder state when seeking
libgsmdec: check output buffer size before decoding
gsmdec: log error message when output buffer is too small.
...
Conflicts:
Changelog
ffplay.c
libavcodec/indeo3.c
libavcodec/mjpeg_parser.c
libavcodec/vp3.c
libavformat/cutils.c
libavformat/id3v2.c
libavutil/parseutils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Streams from RTSP or SDP that do not match an allowed type will
be skipped entirely, which allows video-only or audio-only
streaming from servers that provide both.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
presets: rename presets directory
lavc: make avcodec_get_context_defaults3 "officially" public
lavf: replace av_new_stream->avformat_new_stream part II.
lavf,lavd: replace av_new_stream->avformat_new_stream part I.
lavf: add avformat_new_stream as a replacement for av_new_stream.
Use correct scaling table for bwd-pred MVs in second B-field
Ut Video decoder
Makefile: change presets extension to .avpreset
lavfi: add rgbtestsrc source, ported from MPlayer libmpcodecs
lavfi: add testsrc source
AVOptions: add documentation.
presets: update libx264 ffpresets
Conflicts:
Changelog
doc/APIchanges
doc/ffmpeg.texi
ffpresets/libx264-ipod320.ffpreset
ffpresets/libx264-ipod640.ffpreset
ffserver.c
libavcodec/avcodec.h
libavcodec/options.c
libavcodec/version.h
libavdevice/libdc1394.c
libavfilter/avfilter.h
libavfilter/vsrc_testsrc.c
libavformat/flvdec.c
libavformat/riff.c
libavformat/version.h
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Manual replacements are done in this commit.
In many cases, the id is some constant made up number (e.g. 0 for video
and 1 for audio), which is then not used in the demuxer for anything.
Those ids are removed.
* qatar/master:
avconv: add presets
rtsp: Expose the flag options via private AVOptions for sdp and rtp, too
rtsp: Make the rtsp flags avoptions set via a define
rtpenc: Set a default video codec
avoptions: Fix av_opt_flag_is_set
rtp: Fix ff_rtp_get_payload_type
doc: Update the documentation on setting options for RTSP
rtsp: Remove the separate filter_source variable
rtsp: Accept options via private avoptions instead of URL options
rtsp: Simplify AVOption definitions
rtsp: Merge the AVOption lists
lavfi: port libmpcodecs delogo filter
lavfi: port boxblur filter from libmpcodecs
lavfi: add negate filter
lavfi: add LUT (LookUp Table) generic filters
AVOptions: don't segfault on NULL parameter in av_set_options_string()
avio: Check for invalid buffer length.
mpegenc/mpegtsenc: add muxrate private options.
lavf: deprecate AVFormatContext.file_size
mov: add support for TV metadata atoms tves, tvsn and stik
Conflicts:
Changelog
doc/filters.texi
doc/protocols.texi
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/avfilter.h
libavfilter/formats.c
libavfilter/internal.h
libavfilter/vf_boxblur.c
libavfilter/vf_delogo.c
libavfilter/vf_lut.c
libavformat/mpegtsenc.c
libavformat/utils.c
libavformat/version.h
libavutil/opt.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows setting the filter_src option for these demuxers, too,
which wasn't possible at all before (where the option only was set
via URL parameters for RTSP).
Signed-off-by: Martin Storsjö <martin@martin.st>
Eventually, the old way of passing options by adding
stuff to the URL can be dropped.
This avoids having to tamper with the user-specified URL to
pass options on the transport mode. This also works better
with redirects, since the options don't need to be parsed out
from the URL.
Signed-off-by: Martin Storsjö <martin@martin.st>
Use defines for shortening common parts, omit the .dbl named
initializer (since it's the first element in the union).
Signed-off-by: Martin Storsjö <martin@martin.st>
This eases adding options that are common for both. The
AV_OPT_FLAG_EN/DECODING_PARAM still indicates whether they belong
to the muxer or demuxer.
Signed-off-by: Martin Storsjö <martin@martin.st>
DSS enables this automatically if streaming VOD over TCP. If
enabled, the server feeds packets faster than realtime, screwing
up RTCP NTP based timestamps.
Also, DSS doesn't indicate that this was indicated, if it was
enabled automatically (although if it was requested to be enabled,
a header saying that it was enabled is added, but this isn't
added if it is enabled automatically), making it even harder
to detect and work around properly without explicitly asking
for it to be disabled(/enabled, if we were able to support it).
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
simple_idct: simplify some ifdeffery
simple_idct: remove code for DCTELEM != int16
Remove VLAs in ff_amrwb_lsp2lpc()
fate: make vsynth tests depend on only the relevant vref
rtsp: remove disabled code
dsputil: restore mistakenly removed hunk of disabled code
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
ffmpeg: fix some indentation
ffmpeg: fix operation with --disable-avfilter
simple_idct: remove disabled code
motion_est: remove disabled code
vc1: remove disabled code
fate: separate lavf-mxf_d10 test from lavf-mxf
cabac: Move code only used in the cabac test program to cabac.c.
ffplay: warn that -pix_fmt is no longer working, suggest alternative
ffplay: warn that -s is no longer working, suggest alternative
lavf: rename enc variable in utils.c:has_codec_parameters()
lavf: use designated initialisers for all (de)muxers.
wav: remove a use of deprecated AV_METADATA_ macro
rmdec: remove useless ap parameter from rm_read_header_old()
dct-test: remove write-only variable
des: fix #if conditional around P_shuffle
Use LOCAL_ALIGNED in ff_check_alignment()
Conflicts:
ffmpeg.c
libavformat/avidec.c
libavformat/matroskaenc.c
libavformat/mp3enc.c
libavformat/oggenc.c
libavformat/utils.c
tests/ref/lavf/mxf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (40 commits)
H.264: template left MB handling
H.264: faster fill_decode_caches
H.264: faster write_back_*
H.264: faster fill_filter_caches
H.264: make filter_mb_fast support the case of unavailable top mb
Do not include log.h in avutil.h
Do not include pixfmt.h in avutil.h
Do not include rational.h in avutil.h
Do not include mathematics.h in avutil.h
Do not include intfloat_readwrite.h in avutil.h
Remove return statements following infinite loops without break
RTSP: Doxygen comment cleanup
doxygen: Escape '\' in Doxygen documentation.
md5: cosmetics
md5: use AV_WL32 to write result
md5: add fate test
md5: include correct headers
md5: fix test program
doxygen: Drop array size declarations from Doxygen parameter names.
doxygen: Fix parameter names to match the function prototypes.
...
Conflicts:
libavcodec/x86/dsputil_mmx.c
libavformat/flvenc.c
libavformat/oggenc.c
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In this case, the string that was passed couldn't contain
user-defined data and thus there was no risk for injection
bugs, but it's safer this way, if we later change the
content of the options string.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
crypto: Use av_freep instead of av_free
lavf: don't try to free private options if priv_data is NULL.
swscale: fix types of assembly arguments.
swscale: move two macros that are only used once into caller.
swscale: remove unused function.
options: Add missing braces around struct initializer.
mov: Remove leftover crufty debug statement with references to a local file.
dvbsubdec: Fix compilation of debug code.
Remove all uses of now deprecated metadata functions.
Move metadata API from lavf to lavu.
Conflicts:
doc/APIchanges
libavformat/aiffdec.c
libavformat/asfdec.c
libavformat/avformat.h
libavformat/avidec.c
libavformat/cafdec.c
libavformat/matroskaenc.c
libavformat/mov.c
libavformat/mp3enc.c
libavformat/wtv.c
libavutil/avutil.h
libavutil/internal.h
libswscale/swscale.c
libswscale/x86/swscale_template.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
strtol could return negative values, leading to various error messages,
mainly "non-monotonically increasing dts".
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master: (32 commits)
10-bit H.264 x86 chroma v loopfilter asm
Port SMPTE S302M audio decoder from FFmbc 0.3. [Copyright headers corrected]
Fix crash of interlaced MPEG2 decoding
h264pred: fix one more aliasing violation.
doc/APIchanges: fill in missing hashes and dates.
flacenc: use proper initializers for AVOption default values.
lavc: deprecate named constants for deprecated antialias_algo.
aac: workaround for compilation on cygwin
swscale: extend YUV422p support to 10bits depth
tiff: add support for inverted FillOrder for uncompressed data
Remove unused softfloat implementation.
h264pred: fix aliasing violations.
rotozoom: Eliminate French variable name.
rotozoom: Check return value of fread().
rotozoom: Return an error value instead of calling exit().
rotozoom: Make init_demo() return int and check for errors on invocation.
rotozoom: Drop silly UINT8 typedef.
rotozoom: Drop some unnecessary parentheses.
rotozoom: K&R coding style cosmetics
rtsp: Only do keepalive using GET_PARAMETER if the server supports it
...
Conflicts:
Changelog
cmdutils.c
doc/APIchanges
doc/general.texi
ffmpeg.c
ffplay.c
libavcodec/h264pred_template.c
libavcodec/resample.c
libavutil/pixfmt.h
libavutil/softfloat.c
libavutil/softfloat.h
tests/rotozoom.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is more like what VLC does. If the server doesn't mention
supporting GET_PARAMETER in response to an OPTIONS request,
VLC doesn't send any keepalive requests at all. After this patch,
libavformat will still send OPTIONS keepalives if GET_PARAMETER
isn't explicitly said to be supported.
Some RTSP cameras don't support GET_PARAMETER, and will
close the connection if this is sent as keepalive request
(but support OPTIONS just fine, but probably don't need any
keepalive at all). Some other cameras don't support using
OPTIONS as keepalive, but require GET_PARAMETER instead.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
Handle unicode file names on windows
rtp: Rename the open/close functions to alloc/free
Lowercase all ff* program names.
Refer to ff* tools by their lowercase names.
NOT Pulled Replace more FFmpeg instances by Libav or ffmpeg.
Replace `` by $() syntax in shell scripts.
patcheck: Allow overiding grep program(s) through environment variables.
NOT Pulled Remove stray libavcore and _g binary references.
vorbis: Rename decoder/encoder files to follow general file naming scheme.
aacenc: Fix whitespace after last commit.
cook: Fix small typo in av_log_ask_for_sample message.
aacenc: Finish 3GPP psymodel analysis for non mid/side cases.
Remove RDFT dependency from AAC decoder.
Add some debug log messages to AAC extradata
Fix mov debug (u)int64_t format strings.
bswap: use native types for av_bwap16().
doc: FLV muxing is supported.
applehttp: Handle AES-128 encrypted streams
Add a protocol handler for AES CBC decryption with PKCS7 padding
doc: Mention that DragonFly BSD requires __BSD_VISIBLE set
Conflicts:
ffplay.c
ffprobe.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (37 commits)
In avcodec_open(), set return code to an error value only when an error occurs instead of unconditionally at the start of the function.
lavc: remove reference to opt.h from Makefile.
prefer avio_check() over url_exist()
avio: remove AVIO_* access symbols in favor of new AVIO_FLAG_* symbols
lavu: remove misc disabled cruft
lavu: remove FF_API_OLD_IMAGE_NAMES cruft
NOT PULLED lavu: remove FF_API_OLD_EVAL_NAMES cruft
lavc: remove misc disabled cruft.
lavc: remove the FF_API_INOFFICIAL cruft.
lavc: remove the FF_API_SET_STRING_OLD cruft.
lavc: remove the FF_API_USE_LPC cruft.
lavc: remove the FF_API_SUBTITLE_OLD cruft.
lavc: remove the FF_API_VIDEO_OLD cruft.
lavc: remove the FF_API_AUDIO_OLD cruft.
lavc: remove the FF_API_OPT_SHOW cruft.
lavc: remove the FF_API_MM_FLAGS cruft.
lavf: remove misc disabled cruft.
lavf: remove FF_API_INDEX_BUILT cruft
lavf: remove FF_API_URL_CLASS cruft.
lavf: remove FF_API_SYMVER cruft
...
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Make AVIO_FLAG_ access constants work as flags, and in particular fix
the behavior of functions (such as avio_check()) which expect them to
be flags rather than modes.
This breaks API.
* qatar/master:
proto: include os_support.h in network.h
matroskaenc: don't write an empty Cues element.
lavc: add a FF_API_REQUEST_CHANNELS deprecation macro
avio: move extern url_interrupt_cb declaration from avio.h to url.h
avio: make av_register_protocol2 internal.
avio: avio_ prefix for url_set_interrupt_cb.
avio: AVIO_ prefixes for URL_ open flags.
proto: introduce listen option in tcp
doc: clarify configure features
proto: factor ff_network_wait_fd and use it on udp
Conflicts:
ffmpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (22 commits)
ac3enc: move extract_exponents inner loop to ac3dsp
avio: deprecate url_get_filename().
avio: deprecate url_max_packet_size().
avio: make url_get_file_handle() internal.
avio: make url_filesize() internal.
avio: make url_close() internal.
avio: make url_seek() internal.
avio: cosmetics, move AVSEEK_SIZE/FORCE declarations together
avio: make url_write() internal.
avio: make url_read_complete() internal.
avio: make url_read() internal.
avio: make url_open() internal.
avio: make url_connect internal.
avio: make url_alloc internal.
applehttp: Merge two for loops
applehttp: Restructure the demuxer to use a custom AVIOContext
applehttp: Move finished and target_duration to the variant struct
aacenc: reduce the number of loop index variables
avio: deprecate url_open_protocol
avio: deprecate url_poll and URLPollEntry
...
Conflicts:
libavformat/applehttp.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fate: fix partial run when no samples path is specified
ARM: NEON fixed-point forward MDCT
ARM: NEON fixed-point FFT
lavf: bump minor version and add an APIChanges entry for avio changes
avio: simplify url_open_dyn_buf_internal by using avio_alloc_context()
avio: make url_fdopen internal.
avio: make url_open_dyn_packet_buf internal.
avio: avio_ prefix for url_close_dyn_buf
avio: avio_ prefix for url_open_dyn_buf
avio: introduce an AVIOContext.seekable field
ac3enc: use generic fixed-point mdct
lavfi: add fade filter
Change yadif to not use out of picture lines.
lavc: deprecate AVCodecContext.antialias_algo
lavc: mark mb_qmin/mb_qmax for removal on next major bump.
Conflicts:
doc/filters.texi
libavcodec/ac3enc_fixed.h
libavcodec/ac3enc_float.h
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/vf_fade.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* newdev/master: (33 commits)
Fix an infinite loop when RoQ encoded generated a frame with a size greater than the maximum valid size.
Add kbdwin.o to AC3 decoder
Detect byte-swapped AC-3 and support decoding it directly.
cosmetics: indentation
Always copy input data for AC3 decoder.
ac3enc: make sym_quant() branch-free
cosmetics: indentation
Add a CPU flag for the Atom processor.
id3v2: skip broken tags with invalid size
id3v2: don't explicitly skip padding
Make sure kbhit() is in conio.h
fate: update wmv8-drm reference
vc1: make P-frame deblock filter bit-exact.
configure: Add the -D parameter to the dlltool command
amr: Set the AVFMT_GENERIC_INDEX flag
amr: Set the pkt->pos field properly to the start of the packet
amr: Set the codec->bit_rate field based on the last packet
rtsp: Specify unicast for TCP interleaved streams, too
Set the correct target for mingw64 dlltool
applehttp: Change the variable for stream position in seconds into int64_t
...
Conflicts:
ffmpeg.c
ffplay.c
libavcodec/ac3dec.c
libavformat/avio.h
libavformat/id3v2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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>
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>
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>
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>
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>
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>
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>
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
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
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 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
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 may be needed to avoid calls to implicitly defined functions
(that will be removed by dead code elimination later anyway).
Originally committed as revision 25585 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allows compilation of one of them without requiring the others'
dependencies to be present.
Originally committed as revision 25535 to svn://svn.ffmpeg.org/ffmpeg/trunk
The demuxer inspects the payload type of a received RTP packet and
handles the cases where the content is fully described by the payload type.
Originally committed as revision 25527 to svn://svn.ffmpeg.org/ffmpeg/trunk
The new object file is added to the SDP demuxer in the makefile, since it
is needed in both the RTSP muxer and demuxer and in the SDP demuxer, due
to the current code coupling.
Originally committed as revision 25410 to svn://svn.ffmpeg.org/ffmpeg/trunk
This makes the code dependencies correct. Previously, the SDP demuxer
wasn't buildable on its own.
This also reverts rev 25343.
Originally committed as revision 25354 to svn://svn.ffmpeg.org/ffmpeg/trunk
They reside within a large CONFIG_RTSP_DEMUXER block and are thus pointless.
Originally committed as revision 25343 to svn://svn.ffmpeg.org/ffmpeg/trunk
It is only useful for debugging, so it doesn't have to be shown every time.
Originally committed as revision 25323 to svn://svn.ffmpeg.org/ffmpeg/trunk
Do the same change for ff_rdt_parse_packet, too, to keep the interfaces
similar.
Originally committed as revision 25289 to svn://svn.ffmpeg.org/ffmpeg/trunk
message, if available (RFC 2326, section 12.39), fixes issue 2212.
Patch by John Wimer <john at god vtic net>.
Originally committed as revision 25032 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes a bug from rev 22917. Now RTSP streams where the individual RTCP
sender reports aren't sent at the same time actually are synced properly.
Originally committed as revision 25029 to svn://svn.ffmpeg.org/ffmpeg/trunk
this prevents a time-out which closes the TCP connection and kills our
session.
see "Re: [FFmpeg-devel] [PATCH] rtsp.c: keep-alive" thread on mailinglist.
Originally committed as revision 24785 to svn://svn.ffmpeg.org/ffmpeg/trunk
That makes easier understand what went wrong.
In debug mode the whole reply gets printed.
Originally committed as revision 24212 to svn://svn.ffmpeg.org/ffmpeg/trunk
ff_url_split() is retained as an alias, as it was used by ffserver,
to avoid breaking ABI compatibility with it.
Originally committed as revision 23822 to svn://svn.ffmpeg.org/ffmpeg/trunk
Also make the RTSP protocol use url_alloc and url_connect instead of relying
on the delay open behaviour.
Originally committed as revision 23710 to svn://svn.ffmpeg.org/ffmpeg/trunk
This removes some useless copying of handles, and simplifies error handling.
Patch by Josh Allmann, joshua dot allmann at gmail
Originally committed as revision 23648 to svn://svn.ffmpeg.org/ffmpeg/trunk
Since rtsp_hd isn't assigned to rt->rtsp_hd until after the setup phase,
the initialized URLContext could be leaked on failures.
Originally committed as revision 23643 to svn://svn.ffmpeg.org/ffmpeg/trunk
Since the parsing of Vorbis/Theora fmtp headers is handled by the
parse_sdp_a_line function pointer now, the buffer in sdp_parse_fmtp
doesn't need to be this large any longer.
Patch by Josh Allmann, joshua dot allmann at gmail
Originally committed as revision 23599 to svn://svn.ffmpeg.org/ffmpeg/trunk
Done in preparation for RTSP over HTTP.
Patch by Josh Allmann, joshua dot allmann at gmail
Originally committed as revision 23494 to svn://svn.ffmpeg.org/ffmpeg/trunk
in its place.
av_metadata_set() is going to be dropped at the next major bump.
Originally committed as revision 22961 to svn://svn.ffmpeg.org/ffmpeg/trunk
If these aren't reset, the timestamps make a huge jump when the next RTCP
is received.
Originally committed as revision 22918 to svn://svn.ffmpeg.org/ffmpeg/trunk
In order to sync RTP streams that get their initial RTCP timestamp at
different times, propagate the NTP timestamp of the first RTCP packet
to all other streams.
This makes the timestamps of returned packets start at (near) zero instead
of at any random offset.
Originally committed as revision 22917 to svn://svn.ffmpeg.org/ffmpeg/trunk
This patch also changes FF_NETERROR() to be an AVERROR(), i.e. it is always
negative, whereas it was previously positive.
Originally committed as revision 22887 to svn://svn.ffmpeg.org/ffmpeg/trunk
The status_code field is read in the fail codepath, where it could be
read uninitialized earlier. Found by clang.
Originally committed as revision 22801 to svn://svn.ffmpeg.org/ffmpeg/trunk
This helps if the URL (erroneously?) contains question marks within the path.
Originally committed as revision 22643 to svn://svn.ffmpeg.org/ffmpeg/trunk
Don't modify the user-specified s->filename at all, keep all modifications
locally and in rt->control_uri.
Originally committed as revision 22642 to svn://svn.ffmpeg.org/ffmpeg/trunk
Currently, the caller doesn't get the status_code and location for rediects,
since rtsp_setup_input_streams uses a copy of RTSPMessageHeader of its own.
Originally committed as revision 22630 to svn://svn.ffmpeg.org/ffmpeg/trunk
This makes sure that the streams get correctly synchronized when viewed,
previously the streams were out of sync by as much time as it took
between the initialization of the individual muxers.
Originally committed as revision 22545 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes compilation on some OSes
Patch by Dave Yeo, daveryeo at telus dot net
Originally committed as revision 22426 to svn://svn.ffmpeg.org/ffmpeg/trunk
instead of using the original host name, since the RTP (and UDP) protocols
may choose another IP address if the host name resolves into several different
addresses.
Originally committed as revision 22398 to svn://svn.ffmpeg.org/ffmpeg/trunk
This ensures proper escaping of numerical IPv6 addresses.
The RTSP (de)muxer needs its own network initialization, since it isn't
a protocol and url_open hasn't been called yet.
Originally committed as revision 22226 to svn://svn.ffmpeg.org/ffmpeg/trunk
