This was added in 9b07a2dc02 as an ABI hack to allow older
code built with lavf 52 to register protocols even if the size
of the URLProtocol struct was increased. Later, registering
protocols from outside of lavf was removed and this workaround
isn't needed any longer since lavf 53.
This removes an unchecked malloc and a memory leak for the cases
when this workaround actually was used - which it hasn't since
lavf 53.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'feeafb4adabd5c17de1738ed9962e40892b20edb':
lavf: do not export av_register_{rtp,rdt}_dynamic_payload_handlers from shared objects
Merged-by: Michael Niedermayer <michaelni@gmx.at>
F4V is Adobe's mp4/iso media variant, with the most significant
addition/change being supporting other flash codecs than just
aac/h264.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is admittedly kind of pointless since usually -f image2pipe
can be used for the purpose, but this is more user-friendly.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* commit '2f3bada63e57345329c4f9b48e9b81b5cfc03d05':
lavf: Add a protocol for SRTP encryption/decryption
rtsp: Support decryption of SRTP signalled via RFC 4568 (SDES)
Conflicts:
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is mostly useful for encryption together with the RTP muxer,
but could also be set up as IO towards the peer with the SDP
demuxer with custom IO.
Signed-off-by: Martin Storsjö <martin@martin.st>
The function find_things() in configure is confused by component
registration calls as part of multiline macros defining combined
component registration. Coalesce those macros into one line to
work around the issue.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Signed-off-by: Martin Storsjö <martin@martin.st>
Note that the linebreaks text codec option (but not the feature) has
been removed; its main goal was to allow demuxers to configure the text
decoder (and not meant to be used by users), but the AVOption are not a
viable solution. This is solved differently in this commit.
Gif demuxer is capable of extracting multiple frames from gif file.
In conjunction with gif decoder it implements support for reading
animated gifs.
Demuxer has two options available to user: default_delay and min_delay.
These options are for protection from too rapid gif animations. In practice
it is standard approach to slow down rendering of this kind of gifs. If you try to
play gif with delay between frames of one hundredth of second (100fps) using
one of major web browsers, you get significantly slower playback,
around 10 fps. This is because browser detects that delay value is less than some
threshold (usually 2 hundredths of second) and reset it to default value (usually 10
hundredths of second, which corresponds to 10fps). Manipulating these options user
can achieve the same effect during conversion to some video format. Otherwise user
can set them to not protect from rapid animations at all.
The other case when these options necessary is for gif images encoded according to
gif87a standard since prior to gif89a there was no delay information included in file.
Bump lavf minor version.
Signed-off-by: Vitaliy E Sugrobov <vsugrob@hotmail.com>
* commit '07584eaf4a95db3f11d3bc411f9786932829e82b':
mpegts: check substreams before discarding
Add a smooth streaming segmenter muxer
file: Add an avoption for disabling truncating existing files on open
img2dec: always close AVIOContexts
rtpdec_jpeg: Error out on other unsupported type values as well
rtpdec_jpeg: Disallow using the reserved q values
rtpdec_jpeg: Fold the default qtables case into an existing if statement
rtpdec_jpeg: Store and reuse old qtables for q values 128-254
rtpdec_jpeg: Simplify the calculation of the number of qtables
rtpdec_jpeg: Add more comments about the fields in the SOF0 section
rtpdec_jpeg: Clarify where the subsampling magic numbers come from
rtpdec_jpeg: Don't use a bitstream writer for the EOI marker
rtpdec_jpeg: Don't needlessly use a bitstream writer for the header
rtpdec_jpeg: Simplify writing of the jpeg header
rtpdec_jpeg: Merge two if statements
rtpdec_jpeg: Write the DHT section properly
Conflicts:
libavformat/Makefile
libavformat/allformats.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This muxer splits the output from the ismv muxer into individual
files, in realtime.
The same can also be done by the standalone tool ismindex, but this
muxer is needed for doing it in realtime (especially for live
streams that need extra handling for updating the lookahead fields
in the fragment headers).
Using this muxer, one can deliver live smooth streaming from a
normal static file web server. (Using ismindex, one can deliver
premade smooth streaming files from a static file web server,
or prepare files for serving with IIS.)
Signed-off-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Michael Bradshaw <mbradshaw@sorensonmedia.com>
Reviewed-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
v410dec: Implement explode mode support
zerocodec: fix direct rendering.
wav: init st to NULL to avoid a false-positive warning.
wavpack: set bits_per_raw_sample for S32 samples to properly identify 24-bit
h264: refactor NAL decode loop
RTMPTE protocol support
RTMPE protocol support
rtmp: Add ff_rtmp_calc_digest_pos()
rtmp: Rename rtmp_calc_digest to ff_rtmp_calc_digest and make it global
swscale: add missing HAVE_INLINE_ASM check.
lavfi: place x86 inline assembly under HAVE_INLINE_ASM.
vc1: Add a test for interlaced field pictures
swscale: Mark all init functions as av_cold
swscale: x86: Drop pointless _mmx suffix from filenames
lavf: use conditional notation for default codec in muxer declarations.
swscale: place inline assembly bilinear scaler under HAVE_INLINE_ASM.
dsputil: ppc: cosmetics: pretty-print
dsputil: x86: add SHUFFLE_MASK_W macro
configure: respect CC_O setting in check_cc
Conflicts:
Changelog
configure
libavcodec/v410dec.c
libavcodec/zerocodec.c
libavformat/asfenc.c
libavformat/version.h
libswscale/utils.c
libswscale/x86/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This adds two protocols, but one of them is an internal implementation
detail just used as an abstraction layer/generalization in the code. The
RTMPE protocol implementation uses ffrtmpcrypt:// as an alternative to the
tcp:// protocol. This allows moving most of the lower level logic out
from the higher level generic rtmp code.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
libopenjpeg: introduce encoding support
libopenjpeg: rename decoder source file.
RTMPTS protocol support
RTMPS protocol support
avconv: print an error message when demuxing fails.
tscc2: DCT output should not be clipped
rtmp: Rename rtmphttp to ffrtmphttp
Conflicts:
Changelog
configure
doc/general.texi
libavcodec/libopenjpegenc.c
libavcodec/version.h
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This simplifies usage for segment streaming formats with no global
headers, tipically MPEG 2 transport stream "ts" files.
The seg class duplication is required in order to avoid an infinite loop
in libavformat/utils.c:format_child_next_class().
* qatar/master: (24 commits)
flvdec: remove incomplete, disabled seeking code
mem: add support for _aligned_malloc() as found on Windows
lavc: Extend the documentation for avcodec_init_packet
flvdec: remove incomplete, disabled seeking code
http: replace atoll() with strtoll()
mpegts: remove unused/incomplete/broken seeking code
af_amix: allow float planar sample format as input
af_amix: use AVFloatDSPContext.vector_fmac_scalar()
float_dsp: add x86-optimized functions for vector_fmac_scalar()
float_dsp: Move vector_fmac_scalar() from libavcodec to libavutil
lavr: Add x86-optimized function for flt to s32 conversion
lavr: Add x86-optimized function for flt to s16 conversion
lavr: Add x86-optimized functions for s32 to flt conversion
lavr: Add x86-optimized functions for s32 to s16 conversion
lavr: Add x86-optimized functions for s16 to flt conversion
lavr: Add x86-optimized function for s16 to s32 conversion
rtpenc: Support packetizing iLBC
rtpdec: Add a depacketizer for iLBC
Implement the iLBC storage file format
mov: Support muxing/demuxing iLBC
...
Conflicts:
Changelog
configure
libavcodec/avcodec.h
libavcodec/dsputil.c
libavcodec/version.h
libavformat/movenc.c
libavformat/mpegts.c
libavformat/version.h
libavutil/mem.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavr: add x86-optimized functions for mixing 1-to-2 s16p with flt coeffs
lavr: add x86-optimized functions for mixing 1-to-2 fltp with flt coeffs
Add Dolby/DPLII downmix support to libavresample
vorbisdec: replace div/mod in loop with a counter
fate: vorbis: add 5.1 surround test
rtpenc: Allow requesting H264 RTP packetization mode 0
configure: Sort the library listings in the help text alphabetically
dwt: remove variable-length arrays
RTMPT protocol support
http: Properly handle chunked transfer-encoding for replies to post data
http: Fail reading if the connection has gone away
amr: Mark an array const
amr: More space cleanup
rtpenc: Fix memory leaks in the muxer open function
Conflicts:
Changelog
configure
doc/APIchanges
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This adds two protocols, but one of them is an internal implementation
detail just used as an abstraction layer/generalization in the code. The
RTMPT protocol implementation uses rtmphttp:// as an alternative to the
tcp:// protocol. This allows moving most of the lower level logic out
from the higher level generic rtmp code.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (27 commits)
libxvid: Give more suitable names to libxvid-related files.
libxvid: Separate libxvid encoder from libxvid rate control code.
jpeglsdec: Remove write-only variable in ff_jpegls_decode_lse().
fate: cosmetics: lowercase some comments
fate: Give more consistent names to some RealVideo/RealAudio tests.
lavfi: add avfilter_get_audio_buffer_ref_from_arrays().
lavfi: add extended_data to AVFilterBuffer.
lavc: check that extended_data is properly set in avcodec_encode_audio2().
lavc: pad last audio frame with silence when needed.
samplefmt: add a function for filling a buffer with silence.
samplefmt: add a function for copying audio samples.
lavr: do not try to copy to uninitialized output audio data.
lavr: make avresample_read() with NULL output discard samples.
fate: split idroq audio and video into separate tests
fate: improve dependencies
fate: add convenient shorthands for ea-vp6, libavcodec, libavutil tests
fate: split some combined tests into separate audio and video tests
fate: fix dependencies for probe tests
mips: intreadwrite: fix inline asm for gcc 4.8
mips: intreadwrite: remove unnecessary inline asm
...
Conflicts:
cmdutils.h
configure
doc/APIchanges
doc/filters.texi
ffmpeg.c
ffplay.c
libavcodec/internal.h
libavcodec/jpeglsdec.c
libavcodec/libschroedingerdec.c
libavcodec/libxvid.c
libavcodec/libxvid_rc.c
libavcodec/utils.c
libavcodec/version.h
libavfilter/avfilter.c
libavfilter/avfilter.h
libavfilter/buffersink.h
tests/Makefile
tests/fate/aac.mak
tests/fate/audio.mak
tests/fate/demux.mak
tests/fate/ea.mak
tests/fate/image.mak
tests/fate/libavutil.mak
tests/fate/lossless-audio.mak
tests/fate/lossless-video.mak
tests/fate/microsoft.mak
tests/fate/qt.mak
tests/fate/real.mak
tests/fate/screen.mak
tests/fate/video.mak
tests/fate/voice.mak
tests/fate/vqf.mak
tests/ref/fate/ea-mad
tests/ref/fate/ea-tqi
Merged-by: Michael Niedermayer <michaelni@gmx.at>
