* commit 'ab35ec29a4071871934856c00da7d6ebcc0c095b':
vf_overlay: get rid of pointless messing with timebase.
samplefmt: make av_samples_alloc() initialize the data to silence.
libspeexdec: handle NULL return value from speex_packet_to_header()
h264probe: Don't error out on bits that no longer are reserved
mpegvideo: set extended_data in ff_update_duplicate_context()
libspeexdec: properly handle DTX for multiple frames-per-packet
libspeexdec: move the SpeexHeader from LibSpeexContext to where it is used
libspeexdec: simplify setting of frame_size
libspeexdec: set channel_layout
Conflicts:
libavfilter/vf_overlay.c
libavformat/h264dec.c
libavutil/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This makes sure any buffered data is written to the segment, for
muxers that buffer up data internally (e.g. fragmented mp4).
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes sure new inline headers are emitted when the next
packet is written. This allows segmenting mpegts without calling
write_header/write_trailer (nor freeing/reiniting the muxer)
for each segment.
Signed-off-by: Martin Storsjö <martin@martin.st>
Some segmented formats (such as fragmented mp4) are "bare", as in,
the segment files do not have the same headers/trailers as full normal
files of that format have.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes sure the muxers are set up in the way they expect
with no data left around from the previous run (which could
cause various issues including memory leaks, depending on the chaine
muxer).
This fixes memory leaks with the mpegts and flv muxers. It also
makes the usage of chained muxers correct.
Signed-off-by: Martin Storsjö <martin@martin.st>
With this change, the segmenter muxer doesn't rely on anything
not available/supported to libavformat external users, making
the segmenter muxer do things just like a normal segmenter
application using libavformat would do.
Signed-off-by: Martin Storsjö <martin@martin.st>
Before, the chained muxer reused the AVStreams array from
the outer muxer, which made it impossible to use the proper
public functions (such as av_write_frame) when calling the
chained muxer.
Signed-off-by: Martin Storsjö <martin@martin.st>
The timebases before where only guranteed to be 1/fps precisse
and could cause AV sync errors on low fps
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
otherwise a unexpected timebase could be choosen
that is one that is thousand times more precisse than requested
which can have sideeffects.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The new fields are only printed when they differ from their defaults
this way only few fate refs change
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
ARM: use numeric ID for Tag_ABI_align_preserved
segment: Pass the interrupt callback on to the chained AVFormatContext, too
ARM: bswap: drop armcc version of av_bswap16()
ARM: set Tag_ABI_align_preserved in all asm files
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This might not be needed at the moment, but it's good practice to
pass it to all chained AVFormatContexts, if it would happen to be
used there at a later point.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
ARM: fix Thumb PIC on Apple
nut: add do {} while (0) to GET_V
tiffenc: Check av_malloc() results.
tiffenc: Simplify pixel format setup using AVPixFmtDescriptor.
Use atexit() instead of defining a custom exit_program() interface.
msvc: Fix detection of VFW & Avisynth required libs
Conflicts:
ffmpeg.c
ffmpeg_opt.c
ffplay.c
ffprobe.c
ffserver.c
libavcodec/tiffenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
While a 25 fps stream can in general store frame durations in 1/25
units, this is not true for the timestamps. For example a 25fps
and a 25000/1001 fps stream when they are stored together might have
a matching 0 timestamp point but when for example a chapter from
this is cut the new start is no longer aligned. The issue gets
MUCH worse when the streams are lower fps, like 1 or 2 fps.
This commit thus makes the muxer choose a multiple of the
framerate as timebase that is at least about 20 micro seconds precise
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
With this, when we use a finer timebase than neccessary to store
durations the demuxer still knows what the original timebase was.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
It is not correct in all cases and it is less predictable than a skip of 0
for user applications.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
libopus: Remap channels using libopus' internal remapping.
Opus decoder using libopus
avcodec: document the use of AVCodecContext.delay for audio decoding
vc1dec: add flush function for WMV9 and VC-1 decoders
http: Increase buffer sizes to cope with longer URIs
nutenc: const correctness for ff_put_v_trace/put_s_trace function arguments
h264_refs: Fix debug tprintf argument types
golomb: const correctness for get_ue()/get_se() function arguments
get_bits: const correctness for get_bits_trace()/get_xbits_trace() arguments
Conflicts:
Changelog
libavcodec/Makefile
libavcodec/version.h
libavformat/http.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
dwt: Drop unused functions spatial_compose{53|97}i()
nutdec: Remove unused and broken debug function stub
avcodec: Drop long-deprecated imgconvert.h header
Add Opus support to the Ogg muxer.
Add Opus codec id and codec description.
avformat: Identify anonymous AVIO typedef structs.
Conflicts:
libavcodec/avcodec.h
libavcodec/codec_desc.c
libavcodec/imgconvert.h
libavcodec/version.h
libavformat/oggenc.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Use the MAX_URL_SIZE define where applicable. Increase buffer
sizes for all buffers that need to fit a long pathname - buffers
that need to fit only the hostname (and other short strings, but
not the pathname - such as "headers" in http_connect) are kept
at 1024 bytes for now.
Also increase the max line length in http_read_header, since it
might need to contain a full url for Location: redirects.
Signed-off-by: Martin Storsjö <martin@martin.st>
Fixes some DNXHD files generated by AVID TM, where codec UL was set to A-law
meanwhile the real audio codec was PCM S16. According to SMPTE RP 224, A-law is
the default value for sound essence parameters therefore we should handle it
specially.
Signed-off-by: Marton Balint <cus@passwd.hu>
Reviewed-by: Tomas Härdin <tomas.hardin@codemill.se>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
nutdec: const correctness for get_v_trace/get_s_trace function arguments
truemotion2: Request samples for old TM2 headers
rtpdec: Remove a useless ff_ prefix from a static symbol
rtpdec: Support depacketizing speex
rtpenc: Add support for packetizing speex
Conflicts:
libavformat/rtpdec.c
libavformat/sdp.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Anonymous typedef structs prevent forward declaration, this
change gives the AVIOContext and AVIOInterruptCB structures
a name. These structures are now in line with other common
structures such as AVFormatContext and AVCodecContext.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
At the moment, the moov header is written at the end of the file, so we
can use the current offset (which focus on the end of the mdat already
written) to guess if 64-bits offset will be required or not.
Though, the next commits will make possible the writing of this table at
the beginning, so this heuristic can't work. As a consequence, we check
all the values within the potential offset table for any value >
32-bits.
Previously we had ignored the past dts and just filled in from the
point where we have had sufficient information.
This should fix Ticket1734
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This packetization scheme simply places the full packets into the
RTP packet without any extra header bytes.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
pthread: make sure AVFrame.extended_data is set properly.
libfdk-aac: reindent after last commit
libfdk-aac: Limit to supported sample rates.
cbrt_tablegen: Include libm.h
oggparsetheora: make it more robust
ogg: prevent NULL pointer deference in theora gptopts
Conflicts:
libavformat/oggparsetheora.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'd1f05dd18375f2f8e68372edee11436927e43ba8':
ogg: calculate the start position once all the headers are parsed
Conflicts:
libavformat/oggdec.c
libavformat/oggparseskeleton.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '7751e4693dd10ec98c20fbd9887233b575034272':
ogg: check that the expected number of headers had been parsed
libx264: change default to closed gop to match x264cli
Use avcodec_free_frame() to free AVFrames.
lavf: use a malloced AVFrame in try_decode_frame().
lavc: add avcodec_free_frame().
lavc: ensure extended_data is set properly on decoding
lavc: initialize AVFrame.extended_data in avcodec_get_frame_defaults()
lavc: use av_mallocz to allocate AVFrames.
lavc: rename the argument of avcodec_alloc_frame/get_frame_defaults
Conflicts:
doc/APIchanges
doc/examples/decoding_encoding.c
libavcodec/utils.c
libavcodec/version.h
libavfilter/src_movie.c
libavformat/oggdec.c
libavformat/oggdec.h
libavformat/oggparsetheora.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Additional safety in case a special ogg stream is crafted
with the proper number of
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtp: Packetization of JPEG (RFC 2435)
smoothstreamingenc: Copy the SAR on the AVStreams as well
Conflicts:
Changelog
libavformat/rtpenc.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
RTPDynamicProtocolHandler for speex is added. Initial support for
speex depacketization from RTP stream comes with it.
Currently, only codec audio rate can be applied based on sdp:
* Narrowband ( 8K)
* Wideband (16K)
* Ultrawideband (32K)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* libspeex audio codec is no longer considered unsupported
when using rtp as output format.
* SDP rtpmap is added for speex payload, formatted according to RFC
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fix Ticket #1617, revealing a regression I introduced in 8f3eebd.
We need to make sure no stream is added in between Ogg context save and
restore operations (because it would likely lead to a mismatch between
ogg->nstreams and AVFormatContext->nb_streams after the restore op).
This is the reason the ogg->state check is added in ogg_new_stream().
Before this patch, checking for ogg->headers was preventing this:
ogg->headers is always set before any ogg save/restore (though, it was
also preventing from creating the stream when necessary).
* qatar/master:
libx264: add forgotten ;
matroskadec: fix a sanity check.
matroskadec: only return corrupt packets that actually contain data
lavf: zero data/size of the packet passed to read_packet().
ARM: use 2-operand syntax for ADD Rd, PC in Apple PIC code
ARM: align PIC offset pools to 4 bytes
ARM: swap source operands in some add instructions
configure: update tms470 detection for latest version
lavf probe: prevent codec probe with no data at all seen
motion_est: fix use of inline on extern functions
Conflicts:
libavcodec/motion_est_template.c
libavformat/matroskadec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mp3dec: read Xing frame TOC index
mp3dec: use named constants for Xing header flags
libx264: add support for nal-hrd, required for Blu-ray streams.
mov: support random access point grouping
matroskadec: properly support BlockDuration
Conflicts:
libavcodec/libx264.c
libavformat/isom.h
libavformat/matroskadec.c
libavformat/mov.c
libavformat/mp3dec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '117d8c6d1f1c187ffc6098d9618457e00534e013':
matroska: implement support for ProRes
matroska: implement support for ALAC
Conflicts:
libavformat/matroskaenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This occurs with fuzzed mpeg-ts files. set_codec_from_probe_data() is
called with a zeroed AVProbeData since no packet made through for
specific stream.
* commit '581281e242609a222233a2e5538b89dfb88fb18e':
matroskadec: check realloc in lzo encoding
matroska: honor error_recognition on unknown doctypes
tiffdec: Add support for GRAY16LE.
tiffenc: Add support for little endian RGB48 and GRAY16
mpeg4: support frame parameter changes with frame-mt
mpegvideo: check ff_find_unused_picture() return value for errors
mpegvideo: release frame buffers before freeing them
configure: msvc: default to 'lib' as 'ar' tool
build: support some non-standard ar variants
Conflicts:
libavcodec/h263dec.c
libavcodec/mpegvideo.c
libavcodec/tiff.c
libavcodec/tiffenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Frames described by this grouping are the starter of a closed or
an open GOP.
This is useful for open GOP of H.264 stream which is not described
by sync sample atom.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Support Matroska native formatting.
On demuxing prepend a Frame container atom (32bit big endian encoded
frame size and 'icpf' string).
On muxing remove it.
* commit '1b3439b3055b083df51d7f7838ecc6b3f708b15c':
mpegvideo: move frame size dependent memory management to separate functions
configure: add --toolchain option
configure: Make the smoothstreaming muxer enable the ismv muxer
smoothstreaming: Export the mp4 codec tags
mov: check for EOF in long lasting loops
avcodec: cleanup utils.c
binkaudio: remove unneeded GET_BITS_SAFE macro
binkaudio: use float sample format
binkaudio: use a different value for the coefficient scale for the DCT codec
Conflicts:
configure
libavcodec/mpegvideo.c
libavcodec/utils.c
libavformat/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes stream copy from a format that already has incompatible
codec tags set. The chained ismv muxer exports this same codec tag
list, so set it on this one as well, to allow the caller (and
lavf common code) to set them correctly.
Signed-off-by: Martin Storsjö <martin@martin.st>
A quite widespread pattern in the demuxer is read a 32bit unsigned
integer and then loop till this value is reached.
Checking for EOF prevents pathological situations.
The compiler fails to figure out that enc->codec_type can only
have 3 different values.
Thus when an if/else is encountered it triggers on the possibility
of the else case has not initialized the flags variable.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
It would have been done anyway in the av_dict_set() call.
This simplifies the code and avoid a warning because of assigning a
const string from ff_id3v1_genre_str to a non-const variable.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
avconv: flush filtered frames before reconfiguring filters
mov: stsd entries must be at least 16 byte
mov: detect EOF in mov_read_dref()
file: return proper error on seek failures
Conflicts:
libavformat/file.c
libavformat/mov.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fix near infinite loop in stsd parsing.
Bug found by: Diana Elena Muscalu
The size is unsigned according the specification.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Avoid a near infinite loop.
Issue discovered by cosminamironesei.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* commit '3f7fd59d151a2773f0e2e93e56b6b13ec6e5334b':
avformat: fix typo in avformat_close_input
mp3enc: write Xing TOC
mp3enc: support MPEG-2 and MPEG-2.5 in Xing header.
mp3enc: downgrade some errors in writing Xing frame to warnings
lavf: flush the output AVIOContext in av_write_trailer().
lavf: cosmetics, reformat av_write_trailer().
avio: flush the internal buffer in avio_close()
Enhance doc on asyncts audiofilter
cmdutils: avoid setting data pointers to invalid values in alloc_buffer()
libavcodec: remove av_destruct_packet_nofree()
Conflicts:
libavcodec/avpacket.c
libavformat/mp3enc.c
libavformat/nutenc.c
libavformat/utils.c
libavformat/version.h
tests/ref/lavf/voc
tests/ref/lavf/voc_s16
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Remove a bogus warning when using segment_list_type csv.
The LIST_TYPE_EXT constant is only used internally, so it can
be removed when the feature (segment_list_type ext) gets removed.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Should also be faster (though I doubt that hardly ever matters
for the usage here).
Also remove the pointer copy. Since we do not need to reset the
pointer to the start of the string, it is not needed anymore.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
This updated version does not deviate from previous behavior on default value of 'buffer_size'
I skipped porting 'sources', 'block' options for now as they're parsed seriously. So i added TODO remarks.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Based on the code by:
Peter Belkner <pbelkner@snafu.de>,
Michael Niedermayer <michaelni@gmx.at>,
Clément Bœsch <clement.boesch@smartjog.com>,
Reimar Döffinger <Reimar.Doeffinger@gmx.de>, and
Tobias Rapp <t.rapp@noa-audio.com>
Alex Converse <alex.converse@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This is consistent with stdio and is what we want to do in all cases.
Fixes a bug in the voc muxer which didn't flush in write_trailer()
previously. This is the cause of the change in the test results.
This is used by avidemux, and is likely usefull to others too.
Patch by: gruntster (Avidemux Rev 7990 — 2012-05-30 13:02:27)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
os_support: Choose between direct.h and io.h using a configure check
os_support: Include io.h instead of direct.h on mingw32ce
x86: ac3dsp: Only refer to the ac3_downmix_sse symbol if it has been declared
swscale: Remove two bogus asserts
ac3: move ac3_downmix() from dsputil to ac3dsp
lavr/audio_mix_matrix: acknowledge the existence of LFE2.
mlp_parser: avoid mapping multiple disctinct TrueHD channels to the same Libav channel.
lavu/audioconvert: add a second low frequency channel.
Conflicts:
doc/APIchanges
libavcodec/ac3dsp.c
libavcodec/ac3dsp.h
libavcodec/mlp_parser.c
libavutil/audioconvert.c
libavutil/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
Windows CE doesn't have neither mkdir nor _mkdir officially (only
CreateDirectoryW), but mingw32ce has compat wrappers with these names
(declared in io.h since direct.h is unavailable).
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
Everything written with this bitstream writer is 8/16 bit units
(except for a pair of 4 bit values), so using a bitstream writer
isn't necessary.
Signed-off-by: Martin Storsjö <martin@martin.st>
Generalize writing of any number of qtables. Don't manually write
16 bit values in two separate calls.
Signed-off-by: Martin Storsjö <martin@martin.st>
Currently the size header of the generated DHT section is
incorrect, making the mjpeg decoder just skip it. Since the
written huffman tables are the default ones, this failure had
gone undetected.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
rtpdec_jpeg: Add support for default quantizers
x86: dsputil: Move specific optimization settings out of global init function
avplay: get rid of ugly casts in the options table
avplay: fix prototypes for option callbacks.
flvdec: always set AVFMTCTX_NOHEADER.
file: Use a normal private context for storing the file descriptor
configure: Adjust the xgetbv instrinsic check
configure: Add --disable-inline-asm command line option
configure: Don't try to enable the log2 function on msvcrt
Conflicts:
configure
ffplay.c
libavcodec/x86/dsputil_mmx.c
libavformat/file.c
libavformat/flvdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Allow to specify options affecting the segment list generation.
In particular: add +live and +cache flags.
For a full discussion read trac ticket #1642:
http://ffmpeg.org/trac/ffmpeg/ticket/1642
Also add live M3U8 generation example.
* qatar/master:
x86: dsputil: Only compile motion_est code when encoders are enabled
mem: fix typo in check for __ICC
fate: mp3: drop redundant CMP setting
rtp: Depacketization of JPEG (RFC 2435)
Rename ff_put_string to avpriv_put_string
mjpeg: Rename some symbols to avpriv_* instead of ff_*
yadif: cosmetics
Conflicts:
Changelog
libavcodec/mjpegenc.c
libavcodec/x86/Makefile
libavfilter/vf_yadif.c
libavformat/version.h
libavutil/mem.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mov_chan: Only set the channel_layout if setting it to a nonzero value
mov_chan: Reindent an incorrectly indented line
mp2 muxer: mark as AVFMT_NOTIMESTAMPS.
x86: float_dsp: fix ff_vector_fmac_scalar_avx() on Win64
x86: more specific checks for availability of required assembly capabilities
x86: avcodec: Drop silly "_mmx" suffix from dsputil template names
fate: Drop redundant setting of FUZZ to 1
cavsdsp: set idct permutation independently of dsputil
x86: allow using add_hfyu_median_prediction_cmov on any cpu with cmov
Conflicts:
libavcodec/x86/dsputil_mmx.c
libavformat/mp3enc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If regularly parsing new chan atoms (as in rtpdec_qt), but the
chan atoms don't actually contain any channel layout, don't reset
the value that the caller has filled in (by guessing).
Signed-off-by: Martin Storsjö <martin@martin.st>
It is included for the open/read/write/close functions. On
MSVC, where this header does not exist, the same functions
are provided by io.h, which is already included.
On windows, these functions are provided by io.h. Make sure
io.h is included if it exists, regardless of the setmode
function.
Signed-off-by: Martin Storsjö <martin@martin.st>
Conflicts:
configure
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
unistd.h used to be required for gethostname. On windows, gethostname
is provided by winsock2.h. Now network.h includes both unistd.h and
winsock2.h if they exist.
Signed-off-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mov_chan: Pass a separate AVIOContext for reading
af_asyncts: check return value from lavr when flushing.
mss2: simplify loop in decode_rle()
mss12: avoid unnecessary division in arith*_get_bit()
mss2: do not try to read too many palette entries
mpegvideo: set AVFrame fields to NULL after freeing the base memory
configure: Set the right cc_e flags for msvc
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes crashes when called from rtpdec_qt, where
AVFormatContext->pb is null, a crash present since 3bab7cd128.
Signed-off-by: Martin Storsjö <martin@martin.st>
Make internal small_strptime() function public, and use it in place of
strptime().
This allows to avoid a dependency on strptime() on systems which do not
support it.
In particular, fix trac ticket #992.
* commit 'd58dd4b5b5d31cfd4092e38a5f2c894eee2ab078':
avopt: Store defaults for AV_OPT_TYPE_FLAGS in the i64 union member
Conflicts:
libavcodec/libvpxenc.c
libavcodec/options_table.h
libavfilter/vf_drawtext.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '124134e42455763b28cc346fed1d07017a76e84e':
avopt: Store defaults for AV_OPT_TYPE_CONST in the i64 union member
Conflicts:
libavcodec/aacenc.c
libavcodec/libopenjpegenc.c
libavcodec/options_table.h
libavdevice/bktr.c
libavdevice/v4l2.c
libavdevice/x11grab.c
libavfilter/af_amix.c
libavfilter/vf_drawtext.c
libavformat/movenc.c
libavformat/options_table.h
libavutil/opt.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
configure: add support for bdver1 and bdver2 CPU types.
avio: make avio_close NULL the freed buffer
pixdesc: cosmetics
proresenc: Don't free a buffer not owned by the codec
proresenc: Write the full value in one put_bits call
adpcmenc: Calculate the IMA_QT predictor without overflow
x86: Add convenience macros to check for CPU extensions and flags
x86: h264dsp: drop some unnecessary ifdefs around prototype declarations
mss12: merge decode_pixel() and decode_top_left_pixel()
mss12: reduce SliceContext size from 1067 to 164 KB
mss12: move SliceContexts out of the common context into the codec contexts
Conflicts:
libavformat/aviobuf.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes stream lookup in flv demuxer. When used with librtmp
protocol streams are sometimes added after head arrived. If the
first stream added in flv header reader is Audio stream then it
is messed with Video stream added later in the code patched.
The result is I have 2 Audio streams (first of them is said to have
a video codec like h264) instead of Audio/Video pair.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
crypto: Remove a stray double space
Rename missed cases of FF_OPT_TYPE_* to AV_OPT_TYPE_*
Conflicts:
libavfilter/vf_delogo.c
libavfilter/vf_lut.c
libavfilter/vsrc_testsrc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
After much discussion and back-and-forth, we reached the conclusion
that matroska uses convergence_duration for subtitle duration because
a 32bit value isn't large enough to store the duration if sub-micro-second
timebases are used. Matroska may not be the only one that supports these
timebases, but it's certainly the only one that ffmpeg attempts to support
in this way.
The long term solution that we seemed to reach was that if we encounter
a matroska file with a sub-micro-second timebase, we should internally
scale it up to at least micro-second, and then duration can be used
normally. This suggests that on the encode side, we should not allow
generation of files with sub-micro-second timebases, but that's a separate
issue.
That being a non-trivial change, and the subtitle interoperability breakage
being very real, I'm re-submitting this small change for consideration.
In this diff, we make sure that duration is populated by the matroska
demuxer, and that convergence_duration is respected in matroskaenc and
srtenc, but that duration is used otherwise. This ends up being a strict
improvement - pipelines that use convergence duration are unchanged, and
ones that are currently broken due to the duration mismatch will start
working - except for the ones with the extreme timebases, but those were
already broken.
Signed-off-by: Philip Langdale <philipl@overt.org>
Closed caption data is definitely not teletext.
Since it contains a EIA-608 compatibility stream,
the EIA_608 codec ID is at least not completely wrong.
Fixes subtitle playback in MPlayer with the sample in
trac ticket #1482.
To fix the ticket itself I expect FFmpeg will need
a closed-caption to SRT decoder first.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
MSS1 and MSS2: set final pixel format after common stuff has been initialised
MSS2 decoder
configure: handle --disable-asm before check_deps
x86: Split inline and external assembly #ifdefs
configure: x86: Separate inline from standalone assembler capabilities
pktdumper: Use a custom define instead of PATH_MAX for buffers
pktdumper: Use av_strlcpy instead of strncpy
pktdumper: Use sizeof(variable) instead of the direct buffer length
Conflicts:
Changelog
configure
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/codec_desc.c
libavcodec/dct-test.c
libavcodec/imgconvert.c
libavcodec/mss12.c
libavcodec/version.h
libavfilter/x86/gradfun.c
libswscale/x86/yuv2rgb.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This improves dts validity checks and consequently fps detection of files with invalid dts
Fixes Ticket1681
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'd488c3bcbaf7ddda42597e014deb661a7e9e2112':
configure: support Bitrig OS
yuv2rgb: handle line widths that are not a multiple of 4.
graph2dot: Use the fallback getopt implementation if needed
tools: Include io.h for open/read/write/close if unistd.h doesn't exist
testprogs: Remove unused includes
qt-faststart: Use other seek/tell functions on MSVC than on mingw
ismindex: Include direct.h for _mkdir on windows
sdp: Use static const char arrays instead of pointers to strings
x86: avcodec: Drop silly "_mmx" suffixes from filenames
x86: avcodec: Drop silly "_sse" suffixes from filenames
sdp: Include profile-level-id for H264
utvideoenc: use ff_huff_gen_len_table
huffman: add ff_huff_gen_len_table
cllc: simplify/fix swapped data buffer allocation.
rtpdec_h264: Don't set the pixel format
h264: Check that the codec isn't null before accessing it
audio_frame_queue: Define af_queue_log_state before using it
Conflicts:
libavcodec/audio_frame_queue.c
libavcodec/h264.c
libavcodec/huffman.h
libavcodec/huffyuv.c
libavcodec/utvideoenc.c
libavcodec/x86/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This enables replacing the -l and -L flags used to specify the
just-built libraries when linking the tools and shared libs with
non-standard syntaxes. System library flags are already handled
by the filtering mechanism in configure.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This patch accepts 'timeout' option for input mode only. As far as i know, UDP output cannot introduce delays.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If set non-zero, limits duration of retry_transfer_wrapper() loop, thus
affects ffurl_read*(), ffurl_write()
Measured in microseconds.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Implements support for tmpo atom for mp4 files, typically used to store BPM. -metadata "tmpo=127" as a command line option will record 127 as the BPM in the meta data.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
vf_hqdn3d: Don't declare the loop variable within the for loop
huffyuv: update to current coding style
huffman: update to current coding style
rtsp: Free the rtpdec context properly
build: fft: x86: Drop unused YASM-OBJS-FFT- variable
Conflicts:
libavcodec/huffman.c
libavcodec/huffyuv.c
libavcodec/x86/Makefile
libavfilter/vf_hqdn3d.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
While not explicitly stated in the specs, the original author
has stated that S_TEXT/UTF-8 is expected to be text using Subrip
markup, but without Subrip in-band timing.
So, now that we have a decoder that conforms to this expectation,
let's use it.
Note that this change will impact tools that use libavformat. If
they expect srt subtitles to have CODEC_ID_TEXT, they must be
adjusted to expect CODEC_ID_SUBRIP. The actual content is, obviously,
unchanged.
Signed-off-by: Philip Langdale <philipl@overt.org>
* qatar/master:
doc/APIchanges: add an entry for codec descriptors.
vorbisenc: set AVCodecContext.bit_rate to 0
vorbisenc: fix quality parameter
FATE: add ALAC encoding tests
lpc: fix alignment of windowed samples for odd maximum LPC order
alacenc: use s16p sample format as input
alacenc: remove unneeded sample_fmt check
alacenc: fix max_frame_size calculation for the final frame
adpcm_swf: Use correct sample offsets when using trellis.
rtmp: support strict rtmp servers
mjpegdec: support AVRn interlaced
x86: remove FASTDIV inline asm
Conflicts:
doc/APIchanges
libavcodec/mjpegdec.c
libavcodec/vorbisenc.c
libavutil/x86/intmath.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In order to send or receive a stream FCPublish, FCSubscribe and _checkbw
are completely optional and often not implemented. releaseStream over a
non-existen stream might report an error instead of being silent.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The special cases in demuxers and decoders are a mess otherwise (and more
would be needed to support it fully)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mpegvideo_enc: don't use deprecated avcodec_encode_video().
cmdutils: refactor -codecs option.
avconv: make -shortest a per-output file option.
lavc: add avcodec_descriptor_get_by_name().
lavc: add const to AVCodec* function parameters.
swf(dec): replace CODEC_ID with AV_CODEC_ID
dvenc: don't use deprecated AVCODEC_MAX_AUDIO_FRAME_SIZE
rtmpdh: Do not generate the same private key every time when using libnettle
rtp: remove ff_rtp_get_rtcp_file_handle().
rtsp.c: use ffurl_get_multi_file_handle() instead of ff_rtp_get_rtcp_file_handle()
avio: add (ff)url_get_multi_file_handle() for getting more than one fd
h264: vdpau: fix crash with unsupported colorspace
amrwbdec: Decode the fr_quality bit properly
Conflicts:
Changelog
cmdutils.c
cmdutils_common_opts.h
doc/ffmpeg.texi
ffmpeg.c
ffmpeg.h
ffmpeg_opt.c
libavcodec/h264.c
libavcodec/options.c
libavcodec/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
