The implicit network initialization is set to be removed in the
future, but is kept for compatibility. By not doing the implicit
initialization for non-network protocols, we avoid the warning
about avformat_network_init() not being called for these, where
it really doesn't make much sense.
Signed-off-by: Martin Storsjö <martin@martin.st>
This definition is in two files, since the definitions will move
to the private header at the next bump.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (46 commits)
mtv: Make sure audio_subsegments is not 0
v4l2: use V4L2_FMT_FLAG_EMULATED only if it is defined
avconv: add symbolic names for -vsync parameters
flvdec: Fix compiler warning for uninitialized variables
rtsp: Fix compiler warning for uninitialized variable
ulti: convert to new bytestream API.
swscale: Use standard multiple inclusion guards in ppc/ header files.
Place some START_TIMER invocations in separate blocks.
v4l2: list available formats
v4l2: set the proper codec_tag
v4l2: refactor device_open
v4l2: simplify away io_method
v4l2: cosmetics
v4l2: uniform and format options
v4l2: do not force interlaced mode
avio: exit early in fill_buffer without read_packet
vc1dec: fix invalid memory access for small video dimensions
rv34: fix invalid memory access for small video dimensions
rv34: joint coefficient decoding and dequantization
avplay: Don't call avio_set_interrupt_cb(NULL)
...
Conflicts:
Changelog
avconv.c
doc/APIchanges
doc/indevs.texi
libavcodec/adxenc.c
libavcodec/dnxhdenc.c
libavcodec/h264.c
libavdevice/v4l2.c
libavformat/flvdec.c
libavformat/mtv.c
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Author: Michael Niedermayer <michaelni@gmx.at>
Date: Thu Nov 3 22:38:10 2011 +0100
lavf: fix null pointer dereference in rdt
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is no longer needed and causes various problems with RTSP
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
a realloc()
BUG=100492
Review URL: http://codereview.chromium.org/8366004
Fixes: 1 of 2 for CVE-2011-3893
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes an invalid free() with ass in avi. The sample in bug 98 passes
parts of AVPacket.data as buffer for the AVIOContext. Since the packet
is quite large fill_buffer tries to reallocate the buffer before doing
nothing. Fixes bug 98.
* qatar/master:
fate: add dxtory test
adx_parser: rewrite.
adxdec: Validate channel count to fix a division by zero.
adxdec: Do not require extradata.
cmdutils: K&R reformatting cosmetics
alacdec: implement the 2-pass prediction type.
alacenc: implement the 2-pass prediction type.
alacenc: do not generate invalid multi-channel ALAC files
alacdec: fill in missing or guessed info about the extradata format.
utvideo: proper median prediction for interlaced videos
lavu: bump lavu minor for av_popcount64
dca: K&R formatting cosmetics
dct: K&R formatting cosmetics
lavf: flush decoders in avformat_find_stream_info().
win32: detect number of CPUs using affinity
Add av_popcount64
snow: Restore three mistakenly removed casts.
Conflicts:
cmdutils.c
doc/APIchanges
libavcodec/adx_parser.c
libavcodec/adxdec.c
libavcodec/alacenc.c
libavutil/avutil.h
tests/fate/screen.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mpegenc: use avctx->slices as number of slices
v410enc: fix undefined signed left shift caused by integer promotion
Release notes: mention cleaned up header includes
fix Changelog file
Fix a bunch of typos.
Drop some pointless void* return value casts from av_malloc() invocations.
wavpack: fix typos in previous cosmetic clean-up commit
wavpack: cosmetics: K&R pretty-printing
avconv: remove the 'codec framerate is different from stream' warning
wavpack: determine sample_fmt before requesting a buffer
bmv audio: implement new audio decoding API
mpegaudiodec: skip all channels when skipping granules
mpegenc: simplify muxrate calculation
Conflicts:
Changelog
avconv.c
doc/RELEASE_NOTES
libavcodec/h264.c
libavcodec/mpeg12.c
libavcodec/mpegaudiodec.c
libavcodec/mpegvideo.c
libavformat/mpegenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The fate-h264-bsf-mp4toannexb failures were caused by an integer
overflow of the unneeded multiplication.
Inspired by patch by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
FATE: add tests for dfa
mpegaudiodec: fix seeking.
mpegaudiodec: fix compilation when testing the unchecked bitstream reader
threads: add sysconf based number of CPUs detection
threads: always include necessary headers for number of CPUs detection
threads: default to automatic thread count detection
Changelog: restore version <next> header
cook: K&R formatting cosmetics
Conflicts:
Changelog
libavcodec/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This works around issues arising from inputs that claim to have a
filesize of 0.
Reported-by: buzz_
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fate: split off DPCM codec FATE tests into their own file
fate: split off PCM codec FATE tests into their own file
libvorbis: K&R reformatting cosmetics
libmp3lame: K&R formatting cosmetics
fate: Add a video test for xxan decoder
mpegvideo_enc: K&R cosmetics (line 1000-2000).
avconv: K&R cosmetics
qt-faststart: Fix up indentation
indeo4: remove two unused variables
doxygen: cleanup style to support older doxy
fate: add more tests for VC-1 decoder
applehttpproto: Apply the same reload interval changes as for the demuxer
applehttp: Use half the target duration as interval if the playlist didn't update
applehttp: Use the last segment duration as reload interval
lagarith: add decode support for arith rgb24 mode
Conflicts:
avconv.c
libavcodec/libmp3lame.c
libavcodec/mpegvideo_enc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
According to draft-pantos-http-live-streaming-07, 6.3.4,
the duration of the last media segment in the playlist
should be used as initial minimum reload delay.
Signed-off-by: Martin Storsjö <martin@martin.st>
With the current default PES packet size, and very small audio bitrates,
audio packet duration gets too long. For players, which wait for a whole
audio packet (or more) it takes a very long time to start playing sound.
For 24kbps audio, one PES packet is about 1 second long. On Motorola STBs,
we observe about 3 second delay before the playback starts with the
default setting.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Do not assume the audio packets being always smaller than
DEFAULT_PES_PAYLOAD_SIZE.
Signed-off-by: Jindřich Makovička <makovick@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
ID3v2.4 allows for zlib compressed tags, but libavformat skips them.
Implement code to inflate compressed tags.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
build: fix standalone compilation of OMA muxer
build: fix standalone compilation of Microsoft XMV demuxer
build: fix standalone compilation of Core Audio Format demuxer
kvmc: fix invalid reads
4xm: Add a check in decode_i_frame to prevent buffer overreads
adpcm: fix IMA SMJPEG decoding
options: set minimum for "threads" to zero
bsd: use number of logical CPUs as automatic thread count
windows: use number of CPUs as automatic thread count
linux: use number of CPUs as automatic thread count
pthreads: reset active_thread_type when slice thread_init returrns early
v410dec: include correct headers
Drop ALT_ prefix from BITSTREAM_READER_LE name.
lavfi: always build vsrc_buffer.
ra144enc: zero the reflection coeffs if the filter is unstable
sws: readd PAL8 to isPacked()
mov: Don't stick the QuickTime field ordering atom in extradata.
truespeech: fix invalid reads in truespeech_apply_twopoint_filter()
Conflicts:
configure
libavcodec/4xm.c
libavcodec/avcodec.h
libavfilter/Makefile
libavfilter/allfilters.c
libavformat/Makefile
libswscale/swscale_internal.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When no data was available both the buffer thread as well as
the main thread would block in select(), when data becomes
available both should move forward and as data is read in the
buffer thread the main thread would block in select() later
the read data was put in the fifo but the main thread still
would be blocked in select() until either the timeout or
another packet would come in.
This is solved in this commit by using a mutex and a condition
variable
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* tjoppen/fuzz_fixes:
mxfdec: Don't crash in mxf_packet_timestamps() if current_edit_unit overflows
mxfdec: Zero nb_ptses in mxf_compute_ptses_fake_index()
mxfdec: Sanity check PreviousPartition
mxfdec: Never seek back in local sets and KLVs
mxfdec: Move the current_partition check inside mxf_read_header()
mxfdec: Fix infinite loop in mxf_packet_timestamps()
mxfdec: Check url_feof() in mxf_read_local_tags()
mxfdec: Check for NULL component
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The 'fiel' atoms can be found in H.264 tracks clobbering the extradata.
MJPEG supports non field based extradata, and this data should be
preserved when copying.
* qatar/master: (27 commits)
asfdec: add side data to ASFStream packet instead of output packet.
idroqdec: set AVFMTCTX_NOHEADER and create streams as they occur.
nellymoserdec: Indicate that the decoder can handle changed parameters
libavcodec: Apply parameter change side data when decoding audio
flvdec: Add param change side data if the sample rate or channels have changed
libavformat: Add a utility function for adding parameter change side data
libavcodec: Define a side data type for parameter changes
aacdec: Handle new extradata passed as side data
flvdec: Export new AAC/H.264 extradata as side data on the next packet
libavcodec: Define a side data type for new extradata
flacdec: skip all track indices at once instead of looping.
mxf: Add PictureEssenceCoding UL for V210.
mxfdec: consider QuantizationBits between 17 and 24 to be pcm_s24*
mxfenc: Add support for MPEG-2 MP@HL-14 in mxf container.
mxf: H.264/MPEG-4 AVC Intra support
configure: Show whether the safe bitstream reader is enabled
x86: Tighten register constraints for decode_significance*_x86.
Replace Subversion revisions in comments by Git hashes.
h264_cabac: synchronize decode_significance_*_x86 conditionals
w32threads: wait for the waked thread in pthread_cond_signal.
...
Conflicts:
libavcodec/avcodec.h
libavcodec/version.h
libavformat/flvdec.c
libavformat/utils.c
tests/ref/lavfi/pixdesc
tests/ref/lavfi/pixfmts_copy
tests/ref/lavfi/pixfmts_null
tests/ref/lavfi/pixfmts_scale
tests/ref/lavfi/pixfmts_vflip
Merged-by: Michael Niedermayer <michaelni@gmx.at>
