Allows avoiding the buffer when using avio read, write and seek functions.
When using the ffmpeg executable -avioflags direct can be used to enable
this mode for input files, but has no effect on output files.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The reason for this is that such files have IndexTableSegments which when parsed
cover EditUnit ranges like this:
[0,1)
[249,250)
[249,377)
[0,249)
where each interval is [IndexStartPosition,IndexStartPosition+IndexDuration).
This would be reduced to a sparse index like:
[0,1), [249,250)
instead of the full range:
[0,249), [249,377)
See TimeCode_HD.mxf, UMID =
060a2b340101010101010410130000000004001aa0e59175025b2a5600da4101.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The audio codecs property is composed by all values except
SUPPORT_SND_INTEL (0x0008) and SUPPORT_SND_UNUSED (0x0010) which are
unused.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* qatar/master:
vsrc_buffer: allow buffering arbitrary number of frames.
vf_scale: avoid a pointless memcpy in no-op conversion.
avfiltergraph: try to reduce format conversions in filters.
avfiltergraph: add an AVClass to AVFilterGraph on next major bump.
id3v2: fix skipping extended header in id3v2.4
Conflicts:
libavfilter/vf_scale.c
libavfilter/vsrc_buffer.c
libavformat/id3v2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
apedec: check bits <= 32.
cavs: Remove unused code.
oggenc: fix condition when not to flush due to keyframe granule.
oggenc: add pagesize option to set preferred page size
libspeexdec: set frame size in libspeex_decode_init()
smacker audio: sign-extend the initial 16-bit predicted value
Conflicts:
libavcodec/apedec.c
libavcodec/libspeexdec.c
libavformat/oggenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes crashes when copying a data track as in trac
issue #236.
No proper timecode tracks will be written though.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This patch fixes the sample from trac issue #522.
The issue is that the mov demuxer insists on using its
calculated sample_size (which is nonsense for old-style tracks)
instead of the one encoded in the track.
The old raw audio code should be using the value in stsz, because
the size of a single sample never makes sense for the size of
a full audio packet, whereas the new code will multiply the
sample size by the chunk size, so it should use the calculated value.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This patch fixes the sample from trac issue #733.
The issue is that the size of the trak elements is coded
too large, so that the next trak element would be parsed
as part of the first and truncated incorrectly.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The previous condition of 0 page size was wrong,
that would disable the mechanism for all frames at
a start of a page, thus some keyframes still would not
get their own granule.
The real problem is that header packets must not be flushed,
but they have (and must have) 0 granule and thus would
be detected as keyframes.
Add a separate parameter to mark header packets.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
When set, if an Ogg stream buffer has enough data, a page is made
instead of filling maximum-size pages. Using smaller pages results
smaller seek intervals at the expense of higher container overhead.
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* qatar/master:
h264: drop ff_h264_ prefix from static function ff_h264_decode_rbsp_trailing()
h264: Make ff_h264_decode_end() static, it is not used externally.
output-example: K&R formatting cosmetics, comment spelling fixes
avf: make the example output the proper message
avf: fix audio writing in the output-example
mov: don't overwrite existing indexes.
lzw: fix potential integer overflow.
truemotion: forbid invalid VLC bitsizes and token values.
truemotion2: handle out-of-frame motion vectors through edge extension.
configure: Check for a different SDL function
Conflicts:
configure
doc/examples/muxing.c
libavcodec/truemotion2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mp3dec: perform I/S and M/S only when frame mode is joint stereo.
id3v2: add another mimetype for JPEG image
lzw: prevent buffer overreads.
WMAL: Remove inaccurate and unnecessary doxy
h264: fix cabac-on-stack after safe cabac reader.
truemotion2: convert packet header reading to bytestream2.
Conflicts:
libavcodec/lzw.c
libavcodec/truemotion2.c
libavformat/id3v2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
tilde expansion should/can be done by the shell
Reviewed-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This changes globbing support to only be used if the character
contains at least one glob meta character that is preceded by
an unescaped %. To escape a literal % one would use %% which is
identical to the way to match a % with image2 sequence generation
feature.
* Makes it possible to have patterns like %04d-[720p].jpg work
again with sequence number generation. Previously this would
always be detected as a glob pattern and was interpreted by
the image2 glob code instead.
* Makes it possible to use %*-[720p].jpg to match above pattern
without having to double escape it to be not interpreted by most
shells and not by the image2 glob code (previously one would
need to use \*-\\\[720p\\\].jpg to achieve the same)
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
asf: only set index_read if the index contained entries.
cabac: add overread protection to BRANCHLESS_GET_CABAC().
cabac: increment jump locations by one in callers of BRANCHLESS_GET_CABAC().
cabac: remove unused argument from BRANCHLESS_GET_CABAC_UPDATE().
cabac: use struct+offset instead of memory operand in BRANCHLESS_GET_CABAC().
h264: add overread protection to get_cabac_bypass_sign_x86().
h264: reindent get_cabac_bypass_sign_x86().
h264: use struct offsets in get_cabac_bypass_sign_x86().
h264: fix overreads in cabac reader.
wmall: fix seeking.
lagarith: fix buffer overreads.
dvdec: drop unnecessary dv_tablegen.h #include
build: fix doc generation errors in parallel builds
Replace memset(0) by zero initializations.
faandct: Remove FAAN_POSTSCALE define and related code.
dvenc: print allowed profiles if the video doesn't conform to any of them.
avcodec_encode_{audio,video}: only reallocate output packet when it has non-zero size.
FATE: add a test for vp8 with changing frame size.
fate: add kgv1 fate test.
oggdec: calculate correct timestamps in Ogg/FLAC
Conflicts:
libavcodec/4xm.c
libavcodec/cook.c
libavcodec/dvdata.c
libavcodec/dvdsubdec.c
libavcodec/lagarith.c
libavcodec/lagarithrac.c
libavcodec/utils.c
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Version from vqa header does not dictate which sound chunks may
appear in file.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* qatar/master:
make av_interleaved_write_frame() flush packets when pkt is NULL
mpegts: Fix dead error checks
vc1: Do not read from array if index is invalid.
targa: convert to bytestream2.
rv34: set mb_num_left to 0 after finishing a frame
Conflicts:
libavcodec/targa.c
libavcodec/vc1data.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This patch allows the user to force flushing of all queued packets
by calling av_interleaved_write_frame() with pkt set to NULL.
Signed-off-by: Jindrich Makovicka <jindrich.makovicka@nangu.tv>
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
build: ppc: drop stray leftover backslash
build: Only clean the architecture subdirectory we build for.
build: drop some unnecessary dependencies from the H.264 parser
build: prettyprinting cosmetics
libavutil: Remove pointless rational test program.
libavutil: Remove broken and pointless lzo test program.
lavf doxy: expand AVStream.codec doxy.
lavf doxy: improve AVStream.time_base doxy.
lavf doxy: add some basic documentation about reading from the demuxer.
lavf doxy: document passing options to demuxers.
lavf doxy: clarify that an AVPacket contains encoded data.
mpegtsenc: allow user triggered PES packet flushing
APIchanges: mark the place where 0.7 was cut.
APIchanges: mark the place where 0.8 was cut.
APIchanges: fill in missing dates and hashes.
smacker: convert palette and header reading to bytestream2.
alac: convert extradata reading to bytestream2.
Conflicts:
doc/APIchanges
libavcodec/smacker.c
libavcodec/x86/Makefile
libavfilter/Makefile
libavutil/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This depends on the proposed parser change for 0-size packets
in previous mail, otherwise video now plays far too fast.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Currently, the duration of those packets is just discarded
when enabling parsing, thus the output of the Metal Gear Solid
demuxer breaks completely when just setting AVSTREAM_PARSE_HEADERS.
The result will not be correct if a parser creates a delay even
with PARSER_FLAG_COMPLETE_FRAMES and there might be other cases
where it does not work correct, but just discarding them as it
is done currently seems worse.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
avc: Add a function for converting mp4 style extradata to annex b
pthread: free progress if buffer allocation failed.
lavc/avconv: support changing frame sizes in codecs with frame mt.
libavformat: Document who sets the AVStream.id field
utvideo: mark output picture as keyframe.
sunrast: Add support for negative linesize.
vp8: fix update_lf_deltas in libavcodec/vp8.c
ralf: read Huffman code lengths without GetBitContext
Conflicts:
ffmpeg.c
libavcodec/sunrastenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This isn't exactly equivalent with the earlier code for codecs
other than H264 and VC1, but those are two only codecs supported
by this codepath anyway, and it simplifies it a bit.
Signed-off-by: Martin Storsjö <martin@martin.st>
This prevents a null ptr dereference.
It could be checked differently but this way it should
be possible to return some data.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
xwma: Validate channels and bits_per_coded_sample.
mov: Do not read past the end of the ctts_data table.
mov: Add missing terminator to mov_ch_layout_map_1ch.
asf: reset side data elements on packet copy.
wmavoice: fix stack overread.
wmalossless: error out if a subframe is not used by any channel.
vqa: check palette chunk size before reading data.
wmalossless: reset sample pointer for each subframe.
wmalossless: error out on invalid values for order.
Conflicts:
libavcodec/vqavideo.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
FATE: Add ZeroCodec test
oggparseogm: fix order of arguments of avpriv_set_pts_info().
pngenc: better upper bound for encoded frame size.
aiffdec: set block_duration to 1 for PCM codecs that are supported in AIFF-C
aiffdec: factor out handling of integer PCM for AIFF-C and plain AIFF
aiffdec: use av_get_audio_frame_duration() to set block_duration for AIFF-C
aiffdec: do not set bit rate if block duration is unknown
wmall: output packet only if we have decoded some samples
Conflicts:
libavcodec/pngenc.c
tests/fate/lossless-video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Version from vqa header does not dictate which sound chunks may
appear in file.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (27 commits)
avconv: free packet in write_frame() when discarding due to frame number limit
FATE: use +/- flag option syntax for vp8 emu-edge tests
lavf: make av_interleave_packet_per_dts() private.
lavf: deprecate av_read_packet().
oggdec: output correct timestamps for Vorbis
avconv: pass input stream timestamps to audio encoders
lavc: shrink encoded audio packet size after encoding.
xa: set correct bit rate
xa: do not set bit_rate, block_align, or bits_per_coded_sample
xa: fix end-of-file handling
xa: fix timestamp calculation
bink: fix typo in FFALIGN() argument
bink: align plane width to 8 when calculating bundle sizes
doc: pass -Idoc texi2html and texi2pod
doc: texi2pod: add -I flag
movenc: Add a min_frag_duration option
rtsp: Set the default delay to 0.1 s for the RTSP/SDP/RTP demuxers
libavformat: Set the default for the max_delay option to -1
Generate manpages for AV{Format,Codec}Context AVOptions.
doc/avconv: remove entries for AVOptions.
...
Conflicts:
doc/Makefile
doc/ffmpeg.texi
doc/muxers.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/options.c
libavcodec/vp8.c
libavformat/options.c
tests/fate/demux.mak
tests/ref/fate/truemotion1-15
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Takes encoder delay into account by comparing first the coded page
duration with the calculated page duration. Handles last packet duration
if needed, also by comparing coded duration with calculated duration.
Also does better handling of timestamp generation for packets in the
first page for streamed ogg files where the start time is not
necessarily zero.
The other fragmentation options (frag_duration, frag_size and
frag_keyframe) are combined with OR, cutting fragments at the
first of the conditions being fulfilled.
Signed-off-by: Martin Storsjö <martin@martin.st>
Make the muxers/demuxers that use the field handle the default
-1 in the same way as 0.
This allows distinguishing an intentionally set 0 from the default
value where the user hasn't set it.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (35 commits)
fix space type in Changelog
ZeroCodec Decoder
RealAudio Lossless decoder
rtpenc: Use AVFormatContext.packet_size instead of a private option
url: Document the expected behaviour of url_read
libavformat: Use AVFormatContext.probesize in init_input
docs: Fix a stray reference to tags in the generic doxy on dicts
cosmetics: Align some AVInput/OutputFormat declarations
zmbv: check decompress result
zmbv: correct indentation
adpcm: convert adpcm_thp to bytestream2.
adpcm: convert adpcm_yamaha to bytestream2.
adpcm: convert adpcm_swf to bytestream2.
adpcm: convert adpcm_sbpro to bytestream2.
adpcm: convert adpcm_ct to bytestream2.
adpcm: convert adpcm_ima_amv/smjpeg to bytestream2.
adpcm: convert adpcm_ea_xas to bytestream2.
adpcm: convert adpcm_ea_r1/2/3 to bytestream2.
adpcm: convert ea_maxis_xa to bytestream2.
adpcm: convert adpcm_ea to bytestream2.
...
Conflicts:
Changelog
libavcodec/Makefile
libavcodec/adpcm.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/version.h
libavcodec/zerocodec.c
libavcodec/zmbv.c
libavformat/riff.c
libavformat/url.h
tests/ref/fate/truemotion1-15
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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: Kostya Shishkov <kostya.shishkov@gmail.com>
The private option has not been part of any release yet (and
it is only of use in quite rare cases), so just remove it instead
of keeping it with deprecation warnings.
Signed-off-by: Martin Storsjö <martin@martin.st>
This was forgotten in the transition from av_open_input_file to
avformat_open_input, see 603b8bc2a1.
This doesn't change anything for the default case where the
option isn't set, since PROBE_BUF_MAX is 1048576 (which was
used as max probe size earlier) while the default value for
the probesize option is 5000000, which for the probe function
is clipped to PROBE_BUF_MAX anyway.
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
