An obscure Japanese lossless video codec, originally intended
for use with a remote desktop application.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fate: make compare() function compatible with POSIX bc
Update Janne's email address.
APIchanges: Replace Subversion revision numbers by Git hashes.
bytestream: Eliminate one level of pointless macro indirection.
xwd: convert to bytestream2.
vqavideo: port to bytestream2 API
Read preset files with suffix .avpreset
prores: allow user to set fixed quantiser
lavf: remove some disabled code.
lavf: only set average frame rate for video.
lavf: remove a pointless check.
avcodec: add XBM encoder
Conflicts:
Changelog
cmdutils.c
cmdutils.h
doc/APIchanges
libavcodec/Makefile
libavcodec/avcodec.h
libavcodec/version.h
libavcodec/vqavideo.c
libavformat/img2enc.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
We can't do this in general since we could be reading a file with B-frames while
lacking an index.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
In non-blocking mode, lowest-level read protocols are
supposed block only for a short amount of time to let
retry_transfer_wrapper() check for interrupts.
Also, checking the interrupt_callback in the receiving thread is
wrong, as interrupt_callback is not guaranteed to be thread-safe
and the job is already done by retry_transfer_wrapper(). The error
code was also incorrect.
Bug reported by Andrey Utkin.
* qatar/master:
dxa: remove useless code
lavf: don't select an attached picture as default stream for seeking.
avconv: remove pointless checks.
avconv: check for get_filtered_frame() failure.
avconv: remove a pointless check.
swscale: convert hscale() to use named arguments.
x86inc: add *mp named argument support to DEFINE_ARGS.
swscale: convert hscale to cpuflags().
Conflicts:
ffmpeg.c
libswscale/x86/scale.asm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This avoids problems
where avio_tell() returns 0. I've updated all the checks against
cluster_pos
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
doc/general: update supported devices table.
doc/general: add missing @tab to codecs table.
h264: Fix invalid interlaced/progressive MB combinations for direct mode prediction.
avconv: reindent
avconv: link '-passlogfile' option to libx264 'stats' AVOption.
libx264: add 'stats' private option for setting 2pass stats filename.
libx264: fix help text for slice-max-size option.
http: Clear the auth state on redirects
http: Retry auth if it failed due to being stale
rtsp: Resend new keepalive commands if they used stale auth
rtsp: Retry authentication if failed due to being stale
httpauth: Parse the stale field in digest auth
dxva2_vc1: pass the overlap flag to the decoder
dxva2_vc1: fix decoding of BI frames
FATE: add shorthand to wavpack test
dfa: convert to bytestream2 API
anm decoder: move buffer allocation from decode_init() to decode_frame()
h264: improve parsing of broken AVC SPS
Conflicts:
ffmpeg.c
libavcodec/anm.c
libavcodec/dfa.c
libavcodec/h264.c
libavcodec/h264_direct.c
libavcodec/h264_ps.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Allow up to 4 retries for normal requests, where both the
proxy and the target server might need to authenticate.
Signed-off-by: Martin Storsjö <martin@martin.st>
These commands are sent asynchronously, not waiting for the reply.
This reply is parsed later by ff_rtsp_tcp_read_packet or
udp_read_packet. If the reply indicates that we used stale
authentication and need to use a new nonce, resend a new keepalive
command immediately.
This is the only request sent asynchronously, so currently there's
no other command that needs to be resent in the same way.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
pcm-mpeg: convert to bytestream2 API
Revert "h264: clear trailing bits in partially parsed NAL units"
remove iwmmxt optimizations
mimic: do not continue if swap_buf_size is 0
mimic: convert to bytestream2 API
frwu: use MKTAG to check marker instead of AV_RL32
txd: port to bytestream2 API
c93: convert to bytestream2 API
iff: make .long_name more descriptive
FATE: add test for cdxl demuxer
rtsp: Fix a typo
Conflicts:
libavcodec/arm/dsputil_iwmmxt.c
libavcodec/arm/dsputil_iwmmxt_rnd_template.c
libavcodec/arm/mpegvideo_iwmmxt.c
libavcodec/c93.c
libavcodec/txd.c
libavutil/arm/cpu.c
tests/fate/demux.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
According to video_file_format_spec_v10_1.pdf flv stores AAC RAW
thanks to Baptiste Coudurier for pointing that out
thanks to Aℓex Converse for explaining:
This can't be at the start of a non-ADTS payload. 111 is the
EndOfFrame syntax element.
Together these proof that the check was correctly rejecting ADTS which
is not supposed to be in flv. Many players are able to play such ADTS
in flv though but its better if we conform to the spec as this should
ensure that not many but all players can play files generated by ffmpeg.
This reverts commit 3c9a86df0e.
mpjpeg video streamings would break and stop on Firefox after 1 - 30
seconds.
In order to fix this, two changes were made:
1. Replaced all occurrences of '\n' character in mjpeg metadata
with occurences of "\r\n".
2. Added "Content-length: <packet-size>" metadata entry for each
sent frame.
The change has been tested on Google Chrome 17.0.963.78 and Firefox 10.0.2
on lubuntu 11.10 and the streaming seems to work fine now.
* qatar/master:
Fix a bunch of common typos.
build: Skip compiling xvmc.h under the correct condition.
configure: darwin: Change dylib install names to include major version.
mpegts: Always honor a registration descriptor if present and there is no other codec information.
aacdec: Fix SCE parity check.
aacdec: Fix out of array writes (stack).
rtsp: Only set the ttl parameter if the server actually gave a value
udp: Set ttl for read-write streams, too, not only for write-only ones
udp: Only bind to the multicast address if in read-only mode
udp: Clarify the comment about binding the multicast address
udp: Reorder comments
Conflicts:
libavcodec/aacdec.c
tools/patcheck
Merged-by: Michael Niedermayer <michaelni@gmx.at>
With this we can always know if a timestamp is based on added durations
from an unknown origin or if it is based on a correct timestamp (and possibly
added durations)
This should fix some bugs where this distinction was mixed up.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes sending back RTCP RR packets if receiving RTP over
multicast.
If the multicast stream is sent on demand (set up and signalled
via RTSP), the sender might depend on getting RTCP RR packets
knowing that there are listeners, otherwise the stream can be
closed after a certain timeout.
This fixes receiving RTSP streams over multicast on unix, from
certain Axis cameras.
Signed-off-by: Martin Storsjö <martin@martin.st>
When this code was added in 36b532815c, the new code was added
between the existing comment and the existing line of code, making
the old comment seem to refer to the new code. This makes it read
correctly.
Signed-off-by: Martin Storsjö <martin@martin.st>
