Whitespace of the patch cleaned up by Aurel
Some of the issues have been reported by Steve Manzuik / Microsoft Vulnerability Research (MSVR)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 956c901c68)
The width and height must be copied from the input before
being used.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit e9f98c9022)
cxtypes.h works with version 2.1 and older, cxcore.h works with 2.2 and older.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
(cherry picked from commit 9bc8bcddbd)
EBML_STOP leaves matroska->current_id set. Then matroska_read_seek changes
the stream position without resetting current_id. The next
matroska_parse_cluster fails due to calculation of incorrect pos. So clear
current_id when avio_seek happens in matroska_read_seek.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit cdc2c1c576)
Specifically crafted samples can reinit ogg->streams[] while
reading samples, and thus we should not cache old pointers since
these may no longer be valid.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 4cc3467e7a)
Do not use Doxygen for comments that apply to specific implementation
details; merge some duplicated Doxygen comment blocks.
(cherry picked from commit f75e3da535)
Code would allocate a new context but forget to assign it
to the pointer actually passed to avformat_open_input,
potentially causing a crash.
Even if it was initialized it would cause a memleak.
This caused crashes with e.g. mpd, see also
http://bugs.gentoo.org/show_bug.cgi?id=373423
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This is the version available in Debian stable, so it should be a reasonable
baseline that can be expected to be present on all developer machines.
Moreover, this is the version that is used by the nightly cronjob that
generates the online html version.
(cherry picked from commit 10dde477c7)
The previously suggested replacement - av_get_bits_per_sample_fmt() -
was also deprecated.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit ccfa626db8)
Before this, almost all module groups have been used for grouping functions
and fields in structures semantically. This causes them to not appear
properly in the file documentation and needlessly clutters up the "Modules"
index.
Additionally, this commit streamlines some spelling and appearances.
(cherry picked from commit 21a19b7912)
convert the comment that documents the metadata API to use
the doxygen markup
(cherry picked from commit 1a53a438dc)
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
av_open_input_stream used to allow this, even though it makes no sense.
Make it just print a warning instead of failing, thus restoring
compatibility.
Note that avformat_open_input() will still reject this combination.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
(cherry picked from commit 4f731c4429)
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
In these cases it can't be guessed from the operands (at least
not necessarily), and it seems some clang versions refuse to
compile it.
Fixes ticket #303.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
(cherry picked from commit 5c13b5bb39)
ogg_save() copies streams[], but doesn't keep track of free()'ed
struct members. Thus, if in between a call to ogg_save() and
ogg_restore(), streams[].private was free()'ed, this would result
in a double free -> crash, which happened when e.g. playing small
chained ogg fragments.
(cherry picked from commit 9ed6cbc3ee)
ogg_save() copies streams[], but doesn't keep track of free()'ed
struct members. Thus, if in between a call to ogg_save() and
ogg_restore(), streams[].private was free()'ed, this would result
in a double free -> crash, which happened when e.g. playing small
chained ogg fragments.
(cherry picked from commit 9ed6cbc3ee)
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
These blocks don't report audio stream parameters and they are not needed
for decoding.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit cb7b55b096)
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
In commit bebe72f4a0, the enum AV_PICTURE_TYPE_* was introduced. There are still places in the code where pict_type is used as an integer and there is a case where "pict_type = 0" with the explanation "let ffmpeg decide what to do". The new enum does not know a value of 0 and C++ will fail if compiling such programs anyway as it is refered as an int (and you cannot patch them properly).
(cherry picked from commit 5129336714)
2tap qpel isn't implemented yet for high bit depth, so it just breaks decoding.
(cherry picked from commit 9a0dda8b3a)
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
This silences warnings about pointer target sign mismatches as
already done for gcc with -Wno-pointer-sign.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit d0ce090ec5)
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
buffersink_filter is a strong requirement for compiling ffmpeg.
Fixes ffmpeg compilation with --disable-everything.
(cherry picked from commit e65d6e22e3)
From some tests it results that:
1. All of the AVI/MOV WRAW files need to be flipped.
2. MOV WRAW files need to use AVI color modes.
3. Assigning PAL8 mode by default to WRAW codec is not correct.
(cherry picked from commit 67e7dc5404)
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
This removes an unsightly override of the 'optimizations' setting
only to make the configure report print 'small' when --enable-small
is used.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit f082a0fb42)
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
This fourcc is used by the "mpegable AVC" codec and files encoded with
this codec decode correctly with our H.264 decoder.
(cherry picked from commit 2ea1ca1714)
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
The assert referenced a variable that no longer exists since 4:4:4 support.
(cherry picked from commit 6371ce4b0f)
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
This avoid a crash with in avfilter_merge_formats() in case one of the
filter formats list has multiple time the same entry.
Thanks to Mina Nagy Zaki for helping figuring out the issue.
Some operands need to be accessed in byte mode, which restricts the
available registers in 32-bit mode. Using the 'q' constraint selects
a suitable register.
Signed-off-by: Mans Rullgard <mans@mansr.com>
There are 2 known Bink audio codecs. Additionally they have
a different flavor if they are found inside Bink-b "BIKb" file.
In order to guess the correct flavor, the demuxer sets the audio
codec_tag to be the same as the file format tag.
This causes problem because same tag is used for both audio codecs.
The hack works in FFmpeg because audio codecs are identified by their
codec_id, but other players rely on standard behavior.
This fix removes the codec_tag hack and instead uses artificial
extradata format to signal the codec flavor. This would also
allow proper embedding of Bink audio in other containers.
Signed-off-by: Ivan Kalvachev <ikalvachev@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fix warnings of the type:
vf_drawtext.c:NNN: warning: missing braces around initializer
vf_drawtext.c:NNN: warning: (near initialization for ‘drawtext_options[X].default_val’)
Signed-off-by: Anton Khirnov <anton@khirnov.net>
On architectures such as x86 (both 32 bit and 64bit), the stack element
size is fixed, which maintains alignment. Here, this change does not
break anything. However, we also support also other architectures where
this property is not maintained and therefore, applications will crash
horribly.
This change effectively forces all applications to be recompiled against
libswscale.
* qatar/master:
Drop git- prefix from version labels
Use the contents of RELEASE as version fallback instead of 'UNKNOWN'
path64/open64: filter out unsupported flags
Add Release Notes
sipr: Drop unused DSPContext
sipr: include string.h for mem*()
sipr: Use memmove() to copy overlapped buffers.
h264: drop some ugly casts
Conflicts:
version.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The list type was changed to int64_t to be able to hold
channel layouts.
avfilter_make_format_list() still takes a int32_t array and converts
it to int64_t. A new function, avfilter_make_format64_list, that
takes int64_t arrays has been added.
Replace sizeof((*avff)->formats)
with sizeof(*(*avff)->formats)
as the size of the array element is given by the pointed element
rather than by its pointer.
In particular fix computation with the pending patch when
sizeof(int64_t) != sizeof(int64_t *).
This makes binaries produced by source tarballs identify themselves with
the version number of the corresponding release series, unless overriden
by a 'VERSION' file.
These flags are accepted without error but produce an annoying
warning. Filtering them out makes the build less noisy.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
lavf: prevent crash in av_open_input_file() if ap == NULL.
more Changelog additions
lavf: add a forgotten NULL check in convert_format_parameters().
Fix build if yasm is not available.
H.264: Add x86 assembly for 10-bit MC Chroma H.264 functions.
Conflicts:
Changelog
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavc: add opt_find to AVCodecContext class.
h264: Complexify frame num gap shortening code
intreadwrite.h: fix AV_RL32/AV_RB32 signedness.
Fix decoding of mpegts streams with h264 video that does *NOT* have b frames
Add minor bumps and APIChanges entries for lavf private options.
ffmpeg: deprecate -vc and -tvstd
ffmpeg: use new avformat_open_* API.
ffserver: use new avformat_open_* API.
ffprobe: use new avformat_open_* API.
ffplay: use new avformat_open_* API.
cmdutils: add opt_default2().
dict: add AV_DICT_APPEND flag.
lavf: add avformat_write_header() as a replacement for av_write_header().
Deprecate av_open_input_* and remove their uses.
lavf: add avformat_open_input() as a replacement for av_open_input_*
AVOptions: add av_opt_find() as a replacement for av_find_opt.
AVOptions: add av_opt_set_dict() mapping a dictionary struct to a context.
ffmpeg: don't abuse a global for passing frame size from input to output
ffmpeg: don't abuse a global for passing pixel format from input to output
ffmpeg: initialise encoders earlier.
Conflicts:
cmdutils.c
doc/APIchanges
ffmpeg.c
ffplay.c
ffprobe.c
libavcodec/h264.c
libavformat/avformat.h
libavformat/utils.c
libavformat/version.h
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fix a crash occurring when open_inputs is NULL and *open_inputs is
checked, the crash was introduced by the recent avfilter_graph_parse()
syntax change.
In particular, fix graph2dot crash.
By observation it did not seem to handle prev_frame_num > frame_num.
This does not affect any files I have.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The output type of the AV_RL32/AV_RB32 macros was signed int. The
resulting overflow broke at least some ASF streams with large
timestamps. Fix by adding a cast to uint32_t.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
One of the causes of this bug is that the h264 parser defaults low_delay
to 1, but the h264 codec defaults low_delay to 0. Really Ugly.
After many hours of looking at this, I'm still not sure how has_b_frames
is *intended* to behave, but to me the implementation appears way more
complicated than it ought to be.
My patch relies on the encoder to set an optional field in the SPS. This
works for libx264 streams, but I'm not sure that all h264 encoders will
set it.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
There is no need to write two HTTP newlines (\r\n) into "headers",
because http_connect (in http.c) already appends one HTTP newline at
the end of the given headers chunk, which would result in sending
three HTTP newlines after the headers. Most of the time it's okay
(although not RFC-conforming), but many proxy servers and the
occasional strict httpd will puke with a "400 bad request".
In this case, the string that was passed couldn't contain
user-defined data and thus there was no risk for injection
bugs, but it's safer this way, if we later change the
content of the options string.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
ac3enc: use correct alignment and length in channel coupling dsp functions.
ffmpeg: don't abuse a global for passing framerate from input to output
ffmpeg: don't abuse a global for passing channels from input to output
ffmpeg: don't abuse a global for passing samplerate from input to output
ARM: update ff_h264_idct8_add4_neon for 4:4:4 changes
swscale: use SwsContext for av_log when available
swscale: Remove HAVE_MMX from files that are only compiled with MMX enabled.
swscale: Fix compilation with --disable-mmx2.
Conflicts:
ffmpeg.c
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
swscale: remove misplaced comment.
ffmpeg: fix streaming to ffserver.
swscale: split out RGB48 output functions from yuv2packed[12X]_c().
build: move vpath directives to main Makefile
swscale: fix JPEG-range YUV scaling artifacts.
build: move ALLFFLIBS to a more logical place
ARM: factor some repetitive code into macros
Fix SVQ3 after adding 4:4:4 H.264 support
H.264: fix CODEC_FLAG_GRAY
4:4:4 H.264 decoding support
ac3enc: fix allocation of floating point samples.
Conflicts:
ffmpeg.c
libavcodec/dsputil_template.c
libavcodec/h264.c
libavcodec/mpegvideo.c
libavcodec/snow.c
libswscale/swscale.c
libswscale/swscale_internal.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is part of the Great Evil Plan to simplify swscale. Note that
you'll see some code duplication between the output functions for
different RGB variants, and even between packed-YUV and RGB
variants. This is intentional because it improves readability.
The H.264 parser that we use to detect interlacing can only handle
an Annex B stream, so we need to actually use the filter. This is
unfortunate as the crystalhd library is already doing this conversion
internally. A future change will reorganise the decode path more
completely so that we can feed the converted stream into libcrystalhd
and avoid the second conversion.
Signed-off-by: Philip Langdale <philipl@overt.org>
In preparation for using the filter on the actual bitstream, we need
to extend it's lifetime to match that of the decoder.
Signed-off-by: Philip Langdale <philipl@overt.org>
* qatar/master: (24 commits)
utils: Drop pointless '#if 1' preprocessor directive.
ac3enc: remove empty ac3_float function that is never called
ac3enc: split templated float vs. fixed functions into a separate file.
ac3enc: dynamically allocate AC3EncodeContext fields windowed_samples and mdct
ac3enc: use function pointer to choose between AC-3 and E-AC-3 header output functions.
Roll back 4:4:4 H.264 for now Needs some ARM/PPC asm modifications.
Fix SVQ3 after adding 4:4:4 H.264 support
H.264: fix CODEC_FLAG_GRAY
4:4:4 H.264 decoding support
h264_parser: Fix whitespace after previous change.
h264_parser: Fix behaviour when PARSER_FLAG_COMPLETE_FRAMES is set.
wav: remove an invalid free().
lavf: initialise reference_dts in av_estimate_timings_from_pts.
h264: don't be so picky on decoding pps in extradata.
avcodec.h: add or elaborate on some documentation comments.
h264: change a few comments into error messages
ac3dec: fix doxy-style for comment ("///>" should be "///<" instead).
img2: add .dpx to the list of supported file extensions.
ffv1: fix undefined behavior with insane widths.
ARM: jrevdct_arm: simplify stack usage
...
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Currently, the parser is buggy and only processes the stream extradata
when the flag is set. This fixes it to actually inspect the frames.
Whitespce will be fixed in a separate change.
Signed-off-by: Philip Langdale <philipl@overt.org>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The new tables is large enough to prevent this together with our image size checks.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
cmdutils: add missing NULL check in parse_options()
x11grab: remove a memory allocation and the associated memcpy.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This reverts commit a50f0bea25.
This has been implemented differently in qatar and its better they
maintain it for me instead of me having to spend an average 5sec more
per merge
Conflicts:
libavcodec/h264.c
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
By observation it did not seem to handle prev_frame_num > frame_num.
This does not affect any files I have.
(cherry picked from commit 43c0092a80f8212cbb783260bafa157f7b85126e)
* qatar/master:
bitstream: Properly promote av_reverse values before shifting.
libavutil/swscale: YUV444P10/YUV444P9 support.
H.264: Fix high bit depth explicit biweight
h264: Fix 10-bit H.264 x86 chroma v loopfilter asm.
Replace DEBUG_SEEK/DEBUG_SI + av_log combinations by av_dlog.
Update copyright year for ac3enc_opts_template.c.
adts: Adjust frame size mask to follow the specification.
movenc: Add RTP muxer/hinter options
movenc: Pass the RTP AVFormatContext to the SDP generation
rtspenc: Add RTP muxer options
rtspenc: Add an AVClass for setting muxer specific options
rtpenc_chain: Pass the rtpflags options through to the chained muxer
rtpenc: Declare the rtp flags private AVOptions in rtpenc.h
sdp: Reindent after the previous commit
rtpenc: MP4A-LATM payload support
avoptions: Add an av_opt_flag_is_set function for inspecting flag fields
sdp: Allow passing an AVFormatContext to the SDP generation
mov: Fix wrong timestamp generation for fragmented movies that have time offset caused by the first edit list entry.
mpeg12: more advanced ffmpeg mpeg2 aspect guessing code.
swscale: split YUYV output out of yuv2packed[12X]_c().
Conflicts:
doc/APIchanges
libavcodec/Makefile
libavcodec/h264dsp_template.c
libavcodec/mpeg12.c
libavformat/aacdec.c
libavformat/avidec.c
libavformat/internal.h
libavformat/movenc.c
libavformat/rtpenc.c
libavformat/rtpenc_latm.c
libavformat/sdp.c
libavformat/version.h
libavutil/avutil.h
libavutil/pixfmt.h
libswscale/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is enabled with an AVOption on the RTP muxer. The SDP
generator looks for a latm flag in the rtpflags field.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
lavf: make compute_pkt_fields2() return meaningful error values
matroskadec: set timestamps for RealAudio packets.
intelh263dec: aspect ratio processing fix.
intelh263dec: fix "Strict H.263 compliance" file playback
oss,sndio: simplify by using FFMIN.
swscale: extract monowhite/black output from yuv2packed[12X]_c().
swscale: de-macro'ify RGB15/16/32 input functions.
swscale: rearrange code.
movdec: Add support for the 'wfex' atom.
ffmpeg.c: Add a necessary const qualifier
riff: Fix potential memleak.
swscale: change 48bit RGB input macros to inline functions.
swscale: change 9/10bit YUV input macros to inline functions.
swscale: extract gray16 output functions from yuv2packed[12X]().
swscale: use standard clipping functions.
swscale: merge macros that are used only once.
swscale: fix function declarations in swscale.c.
swscale: fix function declaration keywords in x86/swscale_template.c.
Conflicts:
ffmpeg.c
libavcodec/intelh263dec.c
libswscale/swscale.c
libswscale/x86/swscale_template.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Inline functions are easier to read, maintain, modify and test,
which justifies the slightly increased source size. This patch
also adds support for non-native endianness RGB15/16 and fixes
isSupportedOutput() to no longer claim that we support writing
non-native RGB565/555/444.
Remove inline keyword from functions that are never inlined.
Use av_always_inline for functions that should be force-inlined
for performance reasons. Use av_cold for init functions.
Remove inline keyword for functions that are only called through
their function pointers (and thus cannot be inlined); add av_cold
keyword to init function, and use av_always_inline instead of
inline for functions that must be inlined for performance reasons.
* qatar/master:
crypto: Use av_freep instead of av_free
lavf: don't try to free private options if priv_data is NULL.
swscale: fix types of assembly arguments.
swscale: move two macros that are only used once into caller.
swscale: remove unused function.
options: Add missing braces around struct initializer.
mov: Remove leftover crufty debug statement with references to a local file.
dvbsubdec: Fix compilation of debug code.
Remove all uses of now deprecated metadata functions.
Move metadata API from lavf to lavu.
Conflicts:
doc/APIchanges
libavformat/aiffdec.c
libavformat/asfdec.c
libavformat/avformat.h
libavformat/avidec.c
libavformat/cafdec.c
libavformat/matroskaenc.c
libavformat/mov.c
libavformat/mp3enc.c
libavformat/wtv.c
libavutil/avutil.h
libavutil/internal.h
libswscale/swscale.c
libswscale/x86/swscale_template.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This prevents the following compiler warnings: "warning:
initialization from incompatible pointer type". Since the
variables are only ever used in inline assembly, their type
is actually irrelevant (so the part where it was wrong did
not invoke any buggy behaviour).
read_sequence_header can change width/height; therefore, re-initialize
all tables if width/height changed
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Deprecate av_get_bits_per_sample_fmt(), which was a misnamed function.
For the moment we don't have sample formats with a non-integer number
of bytes, in that case we may need to create a new
av_get_bits_per_sample() function. In the meanwhile we prefer to adopt
this variant, since avoids divisions by 8 all over the place.
They are hacks added to reuse the same scaling function for
different formats and they may cause problems when SIMD
implementation of the same functions are used along with pure
C functions.
This target was added to prevent some files being deleted
by make when using chains of implicit rules. This is no
longer required.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Current GCC versions know how to generate these instructions
properly and avoiding inline asm gives better code. The MULH
function for ARMv5 uses the same instruction and is also not
needed any more.
The MLS64 macro remains since negating an input would normally
not be allowed as it would fail for INT_MIN. In our uses, the
inputs never have this value and thus negating is safe.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Remove AVFilterBufferRefAudioProps.size, and use nb_samples in
avfilter_get_audio_buffer() and avfilter_default_get_audio_buffer() in
place of size.
This is required as the size in the audio buffer may be aligned, so it
may not contain a well defined number of samples.
The new layout is consistent with that of the av_image_() API, and
simplifies understanding and copy operations, it also preserves
alignment information which was lost with the previous layout.
This breaks API/ABI, but since the function was never referenced in
the code (and it isn't unlikely already used by someone) then this
should not be a problem.
* qatar/master:
Remove some unused scripts from tools/.
Add x86 assembly for some 10-bit H.264 intra predict functions.
v4l2: do not force NTSC as standard
Skip tableprint.h during 'make checkheaders'.
Remove unnecessary LIBAVFORMAT_BUILD #ifdef.
Drop explicit filenames from @file Doxygen tags.
Skip generated table headers during 'make checkheaders'.
lavf,lavc: free avoptions in a generic way.
AVOptions: add av_opt_free convenience function.
tableprint: Restore mistakenly deleted common.h #include for FF_ARRAY_ELEMS.
tiff: print log in case of unknown / unsupported tag.
tiff: fix linesize for mono-white/black formats.
Fix build of eval-test program
configure: Document --enable-vaapi
ac3enc: extract all exponents for the frame at once
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The new flags parameter allows to specify if the video ref to add
should overwrite the cache, if the flag is not set vsrc_buffer will
complain and abort; otherwise it will clean the already cached video
ref before to overwrite it, thus avoiding a leak.
Parts are inspired from the 8-bit H.264 predict code in Libav.
Other parts ported from x264 with relicensing permission from author.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Avoid to add frames to the vsrc_buffer in the case ist->pts <
start_time, as these frames are unused (and never released). In
particular this condition is verified with commands of the kind:
ffmpeg -i INPUT -ss TIME OUTPUT
Also allow a minor simplification.
The -i INPUT option can be implemented more cleanly by using a
function option, which can easily be done now that the
parse_arg_function passed to parse_options has a standard signature.
VAAPI is disabled by default so it should have a --enable-vaapi option
documented, not a --disable-vaapi.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
ARM: remove MULL inline asm
mathops: use MUL64 macro where it forms part of other ops
tty: factorise returning error codes.
rawdec: add framerate private option.
x11grab: add framerate private option.
fbdev,v4l2: remove some forgotten uses of AVFormatParameters.time_base.
bktr: don't error when AVFormatParameters.time_base isn't set.
cmdutils: add missing const qualifier
Skip headers not designed to work standalone during 'make checkheaders'.
Add missing #includes to make headers self-contained.
musepack: remove unnecessary #include from mpcdata.h
musepack: remove extraneous mpcdata.h inclusions
Fix error check in av_file_map()
Conflicts:
cmdutils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (21 commits)
build: simplify commands for clean target
swscale: split swscale.c in unscaled and generic conversion routines.
swscale: cosmetics.
swscale: integrate (literally) swscale_template.c in swscale.c.
swscale: split out x86/swscale_template.c from swscale.c.
swscale: enable hScale_altivec_real.
swscale: split out ppc _template.c files from main swscale.c.
swscale: remove indirections in ppc/swscale_template.c.
swscale: split out unscaled altivec YUV converters in their own file.
mpegvideoenc: fix multislice fate tests with threading disabled.
mpegts: Wrap #ifdef DEBUG and av_hex_dump_log() combination in a macro.
build: Simplify texi2html invocation through the --output option.
Mark some variables with av_unused
Replace avcodec_get_pix_fmt_name() by av_get_pix_fmt_name().
svq3: Check negative mb_type to fix potential crash.
svq3: Move svq3-specific fields to their own context.
rawdec: initialize return value to 0.
Remove unused get_psnr() prototype
rawdec: don't leak option strings.
bktr: get default framerate from video standard.
...
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Remove duplicate "inC" and "_c" functions that do the same thing;
give each function that handles data and acts as a function pointer
a "_c" suffix; remove "_c" suffix from functions that are inherently
not optimizable. Remove inline keyword from functions that are only
used through function pointers.
On Windows/MinGW the SDL cflags re-define the main() function, which
results in a linking error if the define is not undeffed.
Since the addition of the SDL output device, SDL cflags are used also
for compiling ffmpeg and ffprobe, so we need to move this trick from
ffplay.c to a common header.
Fix trac issue #256.
Most of these variables are only used in av_dlog statements, some
are required but not used by other macros.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Reintroduce the internal symbol which was removed in:
commit e1197b9e17
Author: Stefano Sabatini <stefano.sabatini-lala@poste.it>
Date: Sun May 29 17:57:40 2011 +0200
swscale: remove sws_format_name()
Use av_get_pix_fmt_name() instead.
The symbol is used by some external libs (hi libx264!), this gives
time to them to use the recently added av_get_pix_fmt_name() rather
than an internal symbol.
Since this is only a compilation check (the actual function used is
selected at runtime) and HAVE_SSE indicates that we can also compile
SSE2 code, this is correct.
Some parameters passed to the av_dlog can be either float or int, depending on
the mode the file is being compiled as. Cast those parameters to float and use
appropriate conversion specifiers.
Fixes regression in 836f47d34b in ICC-10.x,
since ICC<=11.0 doesn't align stack upon function calls.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master:
vf_drawtext: Replace FFmpeg by Libav in license boilerplate.
mpegaudiodec: remove unusued code and variables
improved 'edts' atom writing support
mpegaudio: clean up compute_antialias() definition
vp8: fix segmentation race during frame-threading.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Ports the majority of IDCT functions for 10-bit H.264.
Parts are inspired from 8-bit IDCT code in Libav; other parts ported from x264 with relicensing permission from author.
Signed-off-by: Ronald S. Bultje <rbultje@google.com>
Prior to ARMv6, the destination registers of the SMULL instruction
must be distinct from the first source register. Marking the
output early-clobber ensures it is allocated unique registers.
This restriction is dropped in ARMv6 and later, so allowing overlap
between input and output registers there might give better code.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The 'edts' write function can now generate an initial empty edit resulting in a track-specific presentation delay.
This is automatically calculated and inserted for any track where the initial DTS != 0.
Added support for long (version==1) timecodes.
This merges the float and fixed-point versions of the compute_antialias
function, fixes invalid array indexing, and eliminates a dead copy of
csa_table.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
ARM: enable UAL syntax in asm.S
v4l2: don't leak video standard string on error.
swscale: Remove disabled code.
avfilter: Surround function only used in debug mode by appropriate #ifdef.
vf_crop: Replace #ifdef DEBUG + av_log() by av_dlog().
build: remove BUILD_ROOT variable
vp8: use av_clip_uintp2() where possible
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This enables UAL syntax for all asm files instead of only those
which happen to be incompatible with the old, deprecated syntax.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master: (27 commits)
ac3enc: fix LOCAL_ALIGNED usage in count_mantissa_bits()
ac3dsp: do not use the ff_* prefix when referencing ff_ac3_bap_bits.
ac3dsp: fix loop condition in ac3_update_bap_counts_c()
ARM: unbreak build
ac3enc: modify mantissa bit counting to keep bap counts for all values of bap instead of just 0 to 4.
ac3enc: split mantissa bit counting into a separate function.
ac3enc: store per-block/channel bap pointers by reference block in a 2D array rather than in the AC3Block struct.
get_bits: add av_unused tag to cache variable
sws: replace all long with int.
ARM: aacdec: fix constraints on inline asm
ARM: remove unnecessary volatile from inline asm
ARM: add "cc" clobbers to inline asm where needed
ARM: improve FASTDIV asm
ac3enc: use LOCAL_ALIGNED macro
APIchanges: fill in git hash for av_get_pix_fmt_name (0420bd7).
lavu: add av_get_pix_fmt_name() convenience function
cmdutils: remove OPT_FUNC2
swscale: fix crash in bilinear scaling.
vpxenc: add VP8E_SET_STATIC_THRESHOLD mapping
webm: support stereo videos in matroska/webm muxer
...
Conflicts:
Changelog
cmdutils.c
cmdutils.h
doc/APIchanges
doc/muxers.texi
ffmpeg.c
ffplay.c
libavcodec/ac3enc.c
libavcodec/ac3enc_float.c
libavcodec/avcodec.h
libavcodec/get_bits.h
libavcodec/libvpxenc.c
libavcodec/version.h
libavdevice/libdc1394.c
libavformat/matroskaenc.c
libavutil/avutil.h
libswscale/rgb2rgb.c
libswscale/swscale.c
libswscale/swscale_template.c
libswscale/x86/swscale_template.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
commit 93681fbd50
Author: Ronald S. Bultje <rsbultje@gmail.com>
Date: Thu May 26 11:32:32 2011 -0400
swscale: fix compile on ppc.
commit e758573a88
Author: Ronald S. Bultje <rsbultje@gmail.com>
Date: Thu May 26 10:36:47 2011 -0400
swscale: fix compile on x86-32.
commit 0f4eb8b043
Author: Ronald S. Bultje <rsbultje@gmail.com>
Date: Thu May 26 09:17:52 2011 -0400
swscale: remove VOF/VOFW.
commit b4a224c5e4
Author: Ronald S. Bultje <rsbultje@gmail.com>
Date: Wed May 25 14:30:09 2011 -0400
swscale: split chroma buffers into separate U/V planes.
Preparatory step to implement support for sizes > VOFW.
Make ff* tools only accept opt_* functions taking two arguments.
The distinction between functions with one and two arguments is quite
pointless. Simplify parse_options() code.
Aligned local variables must use the LOCAL_ALIGNED macro to ensure
correct alignment on all systems. The unusual size argument to
memset is required since the implementation of LOCAL_ALIGNED may
use a pointer to an array.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Make ff* tools only accept opt_* functions taking two arguments.
The distinction between functions with one and two arguments is quite
pointless. Simplify parse_options() code.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Many functions have such a prefix, but do not actually use any
instructions or features from that set, thus giving the false
impression that swscale is highly optimized for a particular
system, whereas in reality it is not.
Interleave macros and code so that it's easier to find the
actual code that belongs to a function. Also reindent where
appropriate and remove dead code.
This function uses old-style vector operations deprecated in VFPv3.
Some implementations, e.g. Cortex-A9, support them only through
slow software emulation. Cortex-A8 does have this functionality
in hardware, but as it also has NEON, this function is not used
there regardless.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The header is always required for files which directly deal with
devices, since libavdevice uses the AVFormat* structures defined in
avformat.h.
Avoid the need to explicitely add libavformat/avformat.h.
Remove reference to the mplayerhq.hu rsync command, and replace it
with a reference to the make fate-rsync command.
rsync needs still to be enabled on mplayerhq.hu, and it is currently
not working. Also the fate-rsync Makefile target can be easily updated
without the need to keep the docs in synch.
Not needed as overwritten later on (look for "// update AVCodecContext
with x264 parameters"), and not accessed inbetween.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (31 commits)
ARM: add ARMv6 optimised av_clip_uintp2
ARM: remove volatile from asm statements in libavutil/intmath
ARM: fix av_clipl_int32_arm()
v4l: include avdevice.h
ffserver: move close_connection() call to avoid a temporary string and copy.
lavf: initialize demuxer private options.
AVOptions: set string default values.
lavdevice: mark v4l for removal on next major bump.
swscale: fix compile on ppc.
swscale: fix compile on x86-32.
build: Remove generated .version file on distclean.
configure: Add -D_GNU_SOURCE to CPPFLAGS on OS/2.
doc: Drop hint at --enable-memalign-hack for MinGW, it is now autodetected.
ffplay: Remove disabled code.
Mark parameterless function declarations as 'void'.
swscale: use av_clip_uint8() in yuv2yuv1_c().
swscale: remove VOF/VOFW.
swscale: split chroma buffers into separate U/V planes.
swscale: replace formatConvBuffer[VOF] by allocated array.
rgb2rgb: remove duplicate mmx/mmx2/3dnow/sse2 functions.
...
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The volatile qualifiers are not needed on these statements as
their effects are fully specified by constraints.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Many functions have such a prefix, but do not actually use any
instructions or features from that set, thus giving the false
impression that swscale is highly optimized for a particular
system, whereas in reality it is not.
Interleave macros and code so that it's easier to find the
actual code that belongs to a function. Also reindent where
appropriate and remove dead code.
Fixed another dereference in the RTSP code.
Removed a useless variable.
Changed an unnecessary looping assignment to a simple assignment suggested by
Maksym.
Added fixes and tweaks suggested by Maksym Veremeyenko [verem@m1stereo.tv] and
Clément B.
* qatar/master: (32 commits)
doc: create separate section for audio encoders
swscale: Remove orphaned, commented-out function declaration.
swscale: Eliminate rgb24toyv12_c() duplication.
Remove h263_msmpeg4 from MpegEncContext.
APIchanges: Fill in git hash for fps_probe_size (30315a8)
avformat: Add fpsprobesize as an AVOption.
avoptions: Return explicitly NAN or {0,0} if the option isn't found
rtmp: Reindent
rtmp: Don't try to do av_malloc(0)
tty: replace AVFormatParameters.sample_rate abuse with a private option.
Fix end time of last chapter in compute_chapters_end
ffmpeg: get rid of useless AVInputStream.nb_streams.
ffmpeg: simplify managing input files and streams
ffmpeg: purge redundant AVInputStream.index.
lavf: deprecate AVFormatParameters.channel.
libdc1394: add a private option for channel.
dv1394: add a private option for channel.
v4l2: reindent.
v4l2: add a private option for channel.
lavf: deprecate AVFormatParameters.standard.
...
Conflicts:
doc/APIchanges
doc/encoders.texi
ffmpeg.c
libavdevice/alsa-audio.h
libavformat/version.h
libavutil/opt.c
libswscale/rgb2rgb.h
libswscale/rgb2rgb_template.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Uninitialized coefficients were being used to generate exponents, some
of which actually ended up in the final stream. Even though, they were
just extra exponents that are not used by any decoder, it is still
better to have consistent output for testing. This also fixes valgrind
errors.
Having a separate section for audio encoders simplifies navigation
and is slightly more consistent with the rest of the manual.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
The attached patch fixes the crash which happens when user passes lowres value lower than 0 to FFplay.
ffplay -lowres -1 test.mpg
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This actually matches what av_get_double did earlier, the
0.0/0.0 division was intentional, for producing NAN.
Still keeping the check for the return value from
av_get_number, for clarity.
Signed-off-by: Martin Storsjö <martin@martin.st>
Some received packets can have size 0. The return value from
av_malloc(0) may be NULL, which is ok if the size was 0. On
OS X, however, the returned pointer is non-null but leads to
crashes when trying to free it.
Signed-off-by: Martin Storsjö <martin@martin.st>
Grow the file and stream list in opt_input_file() instead of creating it
all at once in transcode(). This is simpler and will be useful for
following commits.
* qatar/master: (22 commits)
configure: enable memalign_hack automatically when needed
swscale: unbreak the build on non-x86 systems.
swscale: remove if(bitexact) branch from functions.
swscale: remove if(canMMX2BeUsed) conditional.
swscale: remove swScale_{c,MMX,MMX2} duplication.
swscale: use emms_c().
Move emms_c() from libavcodec to libavutil.
tiff: set palette in the context when specified in TIFF_PAL tag
rtsp: use strtoul to parse rtptime and seq values.
pgssubdec: fix incorrect colors.
dvdsubdec: fix incorrect colors.
ape: Allow demuxing of files with metadata tags.
swscale: remove dead macro WRITEBGR24OLD.
swscale: remove AMD3DNOW "optimizations".
swscale: remove duplicate code in ppc/ subdirectory.
swscale: remove duplicated x86/ functions.
swscale: force --enable-runtime-cpudetect and remove SWS_CPU_CAPS_*.
vsrc_buffer.h: add file doxy
vsrc_buffer: tweak error message in init()
msmpeg4: reindent.
...
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This will be cleaned up in the next merge
Authorship / merged commits:
commit f668afd489
Author: Janne Grunau <janne-libav@jannau.net>
Date: Fri Apr 15 09:12:34 2011 +0200
swscale: fix "ISO C90 forbids mixed declarations and code" warning
only hit with --enable-runtime-cpudetect
commit 7f2ae5c7af
Author: Janne Grunau <janne-libav@jannau.net>
Date: Fri Apr 15 02:09:44 2011 +0200
swscale: fix compilation with --enable-runtime-cpudetect
commit b6cad3df82
Author: Janne Grunau <janne-libav@jannau.net>
Date: Fri Apr 15 00:31:04 2011 +0200
swscale: correct include path to fix ppc altivec build
commit 6216fc70b7
Author: Luca Barbato <lu_zero@gentoo.org>
Date: Thu Apr 14 22:03:45 2011 +0200
swscale: simplify rgb2rgb templating
MMX is always built. Drop the ifdefs
commit 33a0421bba
Author: Josh Allmann <joshua.allmann@gmail.com>
Date: Wed Apr 13 20:57:32 2011 +0200
swscale: simplify initialization code
Simplify the fallthrough case when no accelerated functions
can be initialized.
commit 735bf19511
Author: Josh Allmann <joshua.allmann@gmail.com>
Date: Wed Apr 13 20:57:31 2011 +0200
swscale: further cleanup swscale.c
Move x86-specific constants out of swscale.c
commit 86330b4c92
Author: Luca Barbato <lu_zero@gentoo.org>
Date: Wed Apr 13 20:57:30 2011 +0200
swscale: partially move the arch specific code left
PPC and x86 code is split off from swscale_template.c. Lots of code is
still duplicated and should be removed later.
Again uniformize the init system to be more similar to the dsputil one.
Unset h*scale_fast in the x86 init in order to make the output
consistent with the previous status. Thanks to Josh for spotting it.
commit c003832883
Author: Luca Barbato <lu_zero@gentoo.org>
Date: Wed Apr 13 20:57:29 2011 +0200
swscale: move away x86 specific code from rgb2rgb
Keep only the plain C code in the main rgb2rgb.c and move the x86
specific optimizations to x86/rgb2rgb.c
Change the initialization pattern a little so some of it can be
factorized to behave more like dsputils.
Conflicts:
libswscale/rgb2rgb.c
libswscale/swscale_template.c
Fix decoding of QT WRAW files.
From some tests it results that:
1. all of the AVI/MOV WRAW files (at least from the link posted
to the trac ticket #108) need to be flipped
2. mov WRAW files need to use AVI color modes
3. assigning PAL8 mode by default to WRAW codec is not correct
Fix decoding of file CarltonMovie2.mov, fix trac issue #108.
Instead, only set the function pointers if bitexact flag is
not set during initialization. Since a change in flags triggers
a re-init anyway, this doesn't situations where flag values
change during runtime.
Since image initialization was moved after tag parsing, the
palette needs to be specified in the context and then copied
to the allocated image in init_image().
Fixes a regression with TIFF images that have palette data,
trac issue #230, file Test_Flate_8bpp.tif.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
strtol could return negative values, leading to various error messages,
mainly "non-monotonically increasing dts".
Signed-off-by: Anton Khirnov <anton@khirnov.net>
On Blu-ray colors are stored in the order YCrCb (and not YCbCr) as mentioned in the specifications:
see System Description Blu-ray Disc Read-Only Format, 9.14.4.2.2.1 Palette Definition Segment
When decoding a Blu-ray subtitle, the colors were incorrectly set.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
On DVD and HD-DVD colors are stored in the order YCrCb (and not YCbCr) as mentioned in the specifications:
see DVD Specifications for Read-Only Disc / Part 3, 4.3 Program Chain Information (7) PGC_SP_PLT
see DVD Specifications for High Definition Disc, 5.2 Navigation for Standard Content (11) PGC_SDSP_PLT
see DVD Specifications for High Definition Disc, 5.2 Navigation for Standard Content (12) PGC_HDSP_PLT
see DVD Specifications for High Definition Disc, 5.5 Presentation Data (4) SET_COLOR2
When decoding a DVD or HD-DVD subtitle, the colors were incorrectly set.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The functions are identical to their MMX counterparts. Thus,
pretending that swscale is highly optimized for AMD3DNOW
extensions is a poorly executed practical joke at best.
Change:
Expected 7 arguments, but only %d found in '%s'\n
to:
Expected 7 arguments, but %d found in '%s'\n
as the user may provide more than 7 arguments, in that case the error
is not misleading.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Channel coupling is an optional AC-3 feature that increases quality by
combining high frequency information from multiple channels into a
single channel. The per-channel high frequency information is sent with
less accuracy in both the frequency and time domains. This allows more
bits to be used for lower frequencies while preserving enough
information to reconstruct the high frequencies.
* qatar/master:
configure: Add -U__STRICT_ANSI__ to CPPFLAGS on Cygwin and DOS.
aacdec: fix typo in scalefactor clipping check
fate: fix fate-h264-conformance-frext-pph10i4-panasonic-a crcs.
fate: update 9/10bit refs.
h264: Properly set coded_{width, height} when parsing H.264.
x86 asm: Add SECTION_TEXT to dct32_sse.asm.
Fix 9/10 bit in swscale.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In -std=c99 mode GCC defines __STRICT_ANSI__ to hide non-ANSI interfaces.
This causes declarations for some POSIX functions to be omitted from system
headers, which causes compilation failures.
* qatar/master:
ffmpeg: Don't trigger url_interrupt_cb on the first signal
avoptions: Check the return value from av_get_number
dct32_sse: eliminate some spills
Fix dct32() compilation with --disable-yasm
Conflicts:
ffmpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Currently, the url_interrupt_cb callback will abort all IO
after the first received signal. This makes the output files
from e.g. the mov muxer to be unreadable if the transcode is
aborted with ctrl+c.
After this patch, the first signal cleanly breaks out of
the transcoding loop, but won't forcibly abort all IO.
After the second signal is received, the url_interrupt_cb
callback will abort all IO.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids doing a division by zero if the option wasn't found,
or wasn't an option of an appropriate type.
Signed-off-by: Martin Storsjö <martin@martin.st>
The new function accepts a slightly more intuitive order of paramters,
and returns an error code, thus allowing applications to report a
meaningful error message.
* qatar/master:
ffmpeg: get rid of the -vglobal option.
dct32: Add AVX implementation of 32-point DCT
dct32: Change pass 6 permutation to allow for AVX implementation
dct32: port SSE 32-point DCT to YASM
multiple inclusion guard cleanup
avio: document buffer must created with av_malloc() and friends
avio: check AVIOContext malloc failure
swscale: point out an alternative to sws_getContext
svq3: Do initialization after parsing the extradata
add changelog entries for 0.7_beta2
mp3lame: add #include required for AV_RB32 macro.
Conflicts:
Changelog
libavcodec/svq3.c
libavcodec/x86/dct32_sse.c
libavfilter/vsrc_buffer.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If done before, some parameters aren't known yet.
With svq3/rtp, initializing before some parameters are known
can lead to calling av_malloc(0), which on OS X currently returns
broken pointers.
* qatar/master:
configure: make executable again
LATM/AAC: Free previously initialized context on reinit.
configure: Do not unconditionally add -Wall to host CFLAGS.
configure: Set OS/2 objformat to a.out.
Add support for a.out object format to assembler macros.
fate: disable threading for encoding
fate: add comment field
fate: allow overriding default build and install dirs
mpegtsenc: Add an AVClass pointer to the private data
mpegaudio: clean up #includes
mpegaudio: move all header parsing to mpegaudiodecheader.[ch]
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* ffmpeg-mt/master:
Update todo.
h264: add an assert that copied pictures are valid picture pointers
valgrind-check: run with 1 and 3 threads
h264: When decoding a packet with multiple PPS/SPS, don't start the next thread until all of them have been read
Allow some pictures to be released earlier after 51ead6d2c40c5defdd211f435aec49b19f5f6a18
h264: fix slice threading MC reading uninitialized frame edges.
Please see ffmpeg-mt for a list of authors of these changes.
Conflicts:
libavcodec/h264.c
mt-work/valgrind-check.sh
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes memory leaks which are the result of overwriting already-initialized
MDCT contexts during context reinitialization, e.g. in valgrind
fate-aac-latm_000000001180bc60.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
This explicitly disables threading for encoding as slices are otherwise
automatically activated. This should be dropped once option resetting
between files is fully implemented.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This adds a comment field to the report header, suitable for
extra information not covered by the automatic fields.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Since a private class is set for this muxer, the callers will
assume that the private data starts with an AVClass pointer.
If no such member exists, the first few bytes of the struct
will be overwritten, and the class pointer may be broken at
any later time.
Signed-off-by: Martin Storsjö <martin@martin.st>
Require the presence of opencv/cxcore.h in place of opencv/cxtypes.h,
which has been removed. Fix compilation with libopencv > 2.1.0.
Fix trac issue #221.
Fix warnings:
cmdutils.c: In function ‘opt_default’:
cmdutils.c:304: warning: initialization discards qualifiers from pointer target type
cmdutils.c: In function ‘set_context_opts’:
cmdutils.c:431: warning: passing argument 2 of ‘alloc_priv_context’ discards qualifiers from pointer target type
cmdutils.c:414: note: expected ‘struct AVClass *’ but argument is of type ‘const struct AVClass *’
Make Audio Encoders a separate chapter, and decrese the contained
sections level by one.
Improve rendering, and improve consistency with the other parts of the
manual.
* qatar/master:
qdm2: Use floating point synthesis filter.
h264: correct border check.
h264: fix loopfilter with threading at slice boundaries.
Fix ff_mpa_synth_filter_fixed() prototype
Rename costablegen.c ---> cos_tablegen.c.
Collapse tableprint.c into tableprint.h.
Simplify trig table rules
Remove potentially unstable filenames from comments in generated files.
Ignore generated tables and generated table generator programs.
Simplify CLEANFILES make variable by using wildcards.
Remove silly insults from avformat_version() Doxygen documentation.
mpegaudiodsp: fix x86 and ppc makefiles
configure: Adjust AVX assembler check.
mpegaudio: remove unused version of SAME_HEADER_MASK
mpegaudio: remove useless #undef at end of file
asfdec: add missing #include for av_bswap32()
mpegaudio: merge two #if CONFIG_FLOAT blocks
mpegaudio: move some struct definitions from mpegaudio.h
Move some mpegaudio functions to new mpegaudiodsp subsystem
Conflicts:
libavcodec/h264.c
libavcodec/x86/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When backing up the top-left border, check that the top-left
(rather than left) MB indeed does belong to our slice. If it
doesn't, backing up has no positive effect but may accidentally
interfere with other threads writing in the same space.
Fixes occasional one-off effects when enabling slice-MT.
The prototype should use the same typedefs as the definition, or it
will fail where int32_t is not int (DOS apparently).
Signed-off-by: Mans Rullgard <mans@mansr.com>
The new function is a wrapper around
av_vsrc_buffer_add_video_buffer_ref(), and allows to simplify the act
of pushing AVFrame data to the source buffer.
Rename av_vsrc_buffer_add_frame to
av_vsrc_buffer_add_video_buffer_ref(), and change its inteface to make
it accept in input an AVFilterBufferRef rather than an AVFrame.
This way the interface can be used without requiring the
inclusion/installation of libavcodec headers.
This is consistent with how all other table generation programs are named.
Moreover this ensures that the cos table generation program is correctly
deleted when cleaning the tree.
This collapses the make rules for the trig tables into a pattern
rule. Based on a patch by Diego, modified to avoid using fragile
make constructs and allow future addition of fixed-point sin tables.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Older nasm versions have trouble assembling certain AVX instructions, but the
current AVX check did not detect this. Update the check to use an instruction
that triggers the nasm problem.
These structs are only used in mpegaudiodec.c, so move them there
and remove no longer needed #include lines from mpegaudio.h.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This separation allows these functions to be used in a cleaner
fashion from other codecs (e.g. qdm2) and simplifies creating
optimised versions of them.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This early morning merge should fix --disable-yasm
* qatar/master:
Clean up #includes in cmdutils.h.
g729: Merge g729.h into g729dec.c.
10l: wrap float_interleave functions in HAVE_YASM.
Conflicts:
libavcodec/g729.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
inttypes.h is not necessary, just stdint.h is enough.
Unconditionally #include avfilter.h in cmdutils.h. It is an installed
header with no non-standard external dependencies, so it is safe.
* qatar/master:
APIchanges: fill in date and commit for request_sample_fmt
Add floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders.
Add support for request_sample_format in ffmpeg and ffplay.
Add APIchanges entry for request_sample_fmt.
Add request_sample_fmt field to AVCodecContext.
Add float_interleave() to FmtConvertContext with x86-optimized versions.
Remove unused make variable SEEK_REFFILE
fate: remove redundant aref and vref references
fate: remove do_ffmpeg_nocheck function
fate: do not collect -benchmark output
mpegaudiodec: remove decode_end() function
fate: run aref and vref as regular tests
mpegaudio: sanitise compute_antialias_* names
mpeg12: add slice-threading checks to slice-threading initializers.
h264: copy pixel_shift between slice threading contexts.
mdec: enable frame-level multithreading.
mdec.c: fix overread.
Conflicts:
libavcodec/aacdec.c
libavcodec/ac3dec.c
libavcodec/avcodec.h
libavcodec/dca.c
libavcodec/h264.c
libavcodec/mdec.c
libavcodec/mpeg12.c
libavcodec/options.c
libavcodec/version.h
libavcodec/vorbisdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The ref targets are included in the FATE_[AV]CODEC lists created
by configure so they do not need to be listed separately in the
makefile. Filter them out when setting dependencies to avoid make
warnings about circular deps.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Idea taken from mencoder. This should fix conflicts with presets as long as
the new system is used.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This function is essentially an alias for run_ffmpeg and is only
used in one place. This patch removes the function and replaces
the call with the equivalent (simpler) run_ffmpeg call.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The old regtest scripts pass -benchmark and collect the utime values.
As these values are never used, this machinery can be removed.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This function is not needed since 721d6f2dc5 removed the DCT table
allocations for the configuration used here.
Signed-off-by: Mans Rullgard <mans@mansr.com>
These tests create reference files used for psnr calculation in
the other codec tests. Treating them as (mostly) regular tests
simplifies the makefile and makes them visible in the fate reports.
The latter makes errors in these runs easier to identify.
Signed-off-by: Mans Rullgard <mans@mansr.com>
In ff_id3v2_parse(), prevent unsigned integer overflow if data length
indicator is skipped and tlen is < 4.
Fix crash decoding file Allaby_cut.mp3, fix trac issue #182.
* qatar/master:
Fix compilation of iirfilter-test.
libx264: handle closed GOP codec flag
lavf: remove duplicate assignment in avformat_alloc_context.
lavf: use designated initializers for AVClasses.
flvdec: clenup debug code
asfdec: fix possible overread on broken files.
asfdec: do not fall back to binary/generic search
asfdec: reindent after previous commit c7bd5ed
asfdec: fallback to binary search internally
mpegaudio: add _fixed suffix to some names
Modify x86util.asm to ease transitioning to 10-bit H.264 assembly.
dct: build dct32 as separate object files
qdm2: include correct header for rdft
Conflicts:
ffpresets/libx264-fast.ffpreset
ffpresets/libx264-fast_firstpass.ffpreset
ffpresets/libx264-faster.ffpreset
ffpresets/libx264-faster_firstpass.ffpreset
ffpresets/libx264-medium.ffpreset
ffpresets/libx264-medium_firstpass.ffpreset
ffpresets/libx264-placebo.ffpreset
ffpresets/libx264-placebo_firstpass.ffpreset
ffpresets/libx264-slow.ffpreset
ffpresets/libx264-slow_firstpass.ffpreset
ffpresets/libx264-slower.ffpreset
ffpresets/libx264-slower_firstpass.ffpreset
ffpresets/libx264-superfast.ffpreset
ffpresets/libx264-superfast_firstpass.ffpreset
ffpresets/libx264-ultrafast.ffpreset
ffpresets/libx264-ultrafast_firstpass.ffpreset
ffpresets/libx264-veryfast.ffpreset
ffpresets/libx264-veryfast_firstpass.ffpreset
ffpresets/libx264-veryslow.ffpreset
ffpresets/libx264-veryslow_firstpass.ffpreset
libavformat/flvdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Also update libx264 presets to keep closed gop as default.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Moving the search and parsing of the 'fmt ' info the main loop of wav_read_header() allows tags that precede it to be parsed.
Creating wav_parse_fmt_tag() makes wav_read_header() easier to read.
asf_read_seek() inside the asf demuxer already does the
right thing, it tries the index and if that fails it uses
binary search. If binary search is called from outside of asfdec.c
it will fail because the asf code cannot clean up after itself.
Therefore introduce AVFMT_NOBINSEARCH that prevents the seek
code to fallback to binary search and AVFMT_NOGENSEARCH that
prevents the seek code to fallback to generic search.
This adds a _fixed suffix to the fixed-point versions of things
with both float and fixed-point variants. This makes it more
consistent with other dual-implementation things, e.g. fft.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Arguments for variable size instructions are added to many macros, along
with other various changes. The x86util.asm code was ported from x264.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
This builds the float and fixed-point versions of dct32 separately
instead of #including the file in dct.c and mpegaudiodec.c.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Make the iff demuxer send the whole audio chunk to the decoder as a
single packet, move stereo interleaving from the iff demuxer to the
decoder, and introduce an 8svx_raw decoder which performs
stereo interleaving.
This is required for handling stereo data correctly, indeed samples
are stored like:
LLLLLL....RRRRRR
that is all left samples are at the beginning of the chunk, all right
samples at the end, so it is necessary to store and process the whole
buffer in order to decode each frame. Thus the decoder needs all the
audio chunk before it can return interleaved data.
Fix decoding of files 8svx_exp.iff and 8svx_fib.iff, fix trac issue #169.
Create separate fields 8svx_compression (for audio compression), and
bitmap_compression (for video compression), and perform minor related
logging tweaks.
Improve clarity, also simplify the case when both types of compression
are employed in the same file.
Fix warnings of the type:
vf_drawtext.c:NNN: warning: missing braces around initializer
vf_drawtext.c:NNN: warning: (near initialization for ‘drawtext_options[X].default_val’)
Redesign the way -aspect option is handled. This is done by making
ffmpeg read the sample aspect ratio set in the corresponding input
stream by default, and overriding it using the value specified by
-aspect.
If the output display aspect ratio is specified with -aspect, it is
set at the end of the filterchain, thus overriding the value set by
filters in the filterchain.
This implementation is more robust, since does not modify the
filterchain description (which was creating potential syntax errors).
(Cherry-pick abf8342aa9)
Another aspect ratio fix try. This leaves the setdar addition at the end
(preferred by people).
(Cherry-pick e7c7b0d000)
This fixes several bugs like multiple outputs and -aspect mixed with -vf
(cherry picked from commit 1762d9ced7)
(cherry picked from commit 5c20c81bfa)
(cherry picked from commit a7844c580d)
Iam not sure this is the best way to implement it, but its the simplest
and keeps the code seperate from the application. Keeping ffmpeg.c
simple and not requireing user apps to duplicate this code.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
These macros are no longer needed after the s32 output was removed.
Change the relevant code to use av_clip_int16() instead of using
explicit limits.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The size-32 DCT_II has a special implementation which doesn't use
the normal tables. Skipping allocation of these in this case saves
some memory.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Ignore color_type information for setting the monoblack pixel format,
only rely on bit_depth.
In particular: fix Test_1bpp_grayscale.png from issue #172.
This may work fine for some broken encoder, and not for others.
* qatar/master:
aacdec: Use float instead of int16_t for ltp_state to avoid needless rounding.
acelp: Remove unused gray_decode table.
dfa: Remove unused variable.
configure: Include AVX availability in summary output.
configure: use same CPPFLAGS in kFreeBSD as Linux
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Make it a subcase of the general algorithm used for the non-bitstream
case. Simplify, and make av_image_get_linesize() and
av_image_fill_linesizes() correctly return the right value when plane
!= 0.
In particular fix a crash occurring with:
-vf format=monow,showinfo,format=monow.
046f081b46 reorganized the CPPFLAGS to no
longer add -D_POSIX_C_SOURCE unconditionally, but only on systems (e.g.,
glibc based ones) that require it. As kFreeBSD uses glibc, it needs to
be treated similar.
Additionally, _BSD_SOURCE is turned on to enable some additional types
such as caddr_t, which are normally enabled on BSD but not with glibc.
Also remove code that overwrites the C versions of functions in
sws_init_swScale_altivec(), so that it uses the C functions of files
if no altivec-optimized version exists.
Surround memset and ff_vp8_dct_cat_prob by X() in order to fix iOS build
Includes patch by Luca Barbato <lu_zero@gentoo.org>.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* qatar/master:
swscale: properly inline bits/endianness in yuv2yuvX16inC().
(We didnt pull the bug) swscale: fix clipping of 9/10bit YUV420P.
Add av_clip_uintp2() function
(our patch / duplicate) dfa: fix buffer overflow checks to avoid integer overflows.
(our patch / duplicate) movenc: always write esds descriptor length using 4 bytes.
(our patch / duplicate) ffmpeg: use parse_number_and_die() when it makes sense
(No thanks) ffmpeg: get rid of the 'q' key schizofrenia
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Prefer parse_number_or_die() over atoi()/atol() parsing for the options:
-pass, -top, -vc, and -qscale.
Improve input validation.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master: (33 commits)
rtpdec_qdm2: Don't try to parse data packet if no configuration is received
ac3enc: put the counting of stereo rematrixing bits in the same place to make the code easier to understand.
ac3enc: clean up count_frame_bits() and count_frame_bits_fixed()
mpegvideo: make FF_DEBUG_DCT_COEFF output coeffs via av_log() instead of just via AVFrame.
srtdec: make sure we don't write past the end of buffer
wmaenc: improve channel count and bitrate error handling in encode_init()
matroskaenc: make sure we don't produce invalid file with no codec ID
matroskadec: check that pointers were initialized before accessing them
lavf: fix function name in compute_pkt_fields2 av_dlog message
lavf: fix av_find_best_stream when providing a wanted stream.
lavf: fix av_find_best_stream when decoder_ret is given and using a related stream.
ffmpeg: factorize quality calculation
tiff: add support for SamplesPerPixel tag in tiff_decode_tag()
tiff: Prefer enum TiffCompr over int for TiffContext.compr.
mov: Support edit list atom version 1.
configure: Enable libpostproc automatically if GPL code is enabled.
Cosmetics: fix prototypes in oggdec
oggdec: fix memleak with continuous streams.
matroskaenc: add missing new line in av_log() call
dnxhdenc: add AVClass in private context.
...
swscale changes largely rewritten by me or replaced by baptsites due to lots of bugs in ronalds code.
Above code is also just in case its not obvios to a large extended duplicates that where cherry picked
from ffmpeg.
Conflicts:
configure
ffmpeg.c
libavformat/matroskaenc.c
libavutil/pixfmt.h
libswscale/ppc/swscale_template.c
libswscale/swscale.c
libswscale/swscale_template.c
libswscale/utils.c
libswscale/x86/swscale_template.c
tests/fate/h264.mak
tests/ref/lavfi/pixdesc_le
tests/ref/lavfi/pixfmts_copy_le
tests/ref/lavfi/pixfmts_null_le
tests/ref/lavfi/pixfmts_scale_le
tests/ref/lavfi/pixfmts_vflip_le
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The later parsing of payload data depends on the configuration
being present. If it hasn't been configured properly yet,
parsing a data packet may lead to a crash.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows the values to be used without changing C code and is closer to how
the other DEBUG flags work.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
In the main loop, stream_number is incremented after checking the stream type,
so the search usually will not find the wanted stream.
This patch eliminates the useless stream_number variable and introduces a new
one, called real_stream_index to store the real stream index of the current
stream, no matter if we are looping through all the streams or only the streams
of a program.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Yet another fix for the code originally designed for use without related_stream.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Calculate quality value once per stream in print_report().
Also fix segfault, as coded_frame can be NULL.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Format detection and internal frame initialization is moved to a
separate init_image() function, which is called when all the tags have
been read, and so both BitsPerSample and SamplesPerPixel information
has been collected.
This fixes decoding of the file 11.tiff from roundup issue #1925.
Based on a patch by Kostya Shishkov <kostya.shishkov@gmail.com>.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
To use posix_memalign() "#define _XOPEN_SOURCE 600" have to be defined.
FFmpeg configure defines it from the command line through config.mak,
but when libavutil is used outside of FFmpeg the define is missing.
Signed-off-by: Ivan Kalvachev <ikalvachev@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Add an extra size validity check in asf_read_frame_header(). Without
this asf->packet_size_left may become negative, which triggers an
assertion failure later.
This avoids the creation of a new AVStream instead of replacing it when
a stream reset occurs (track change with some webradios for example).
Signed-off-by: Diego Biurrun <diego@biurrun.de>
int/unsigned is the natural memory access type for CPUs, using sized types
for temporary variables, counters and similar just increases code size and
can possibly cause a slowdown.
Adding _POSIX_C_SOURCE to CPPFLAGS globally produces all sorts of problems
since it causes certain system functions to be hidden on some (BSD) systems.
The solution is to only add the flag on systems that really require it, i.e.
glibc-based ones.
This change makes BSD systems compile out-of-the-box without the need for
adding specific flags manually. It also allows dropping a number of flags
set manually on a file-per-file basis, but were only present to work around
breakage introduced by the presence of _POSIX_C_SOURCE.
Also add _XOPEN_SOURCE to CPPFLAGS for glibc systems. We use XSI extensions
in several places already, so it is preferable to define it globally instead
of littering source files with individual #defines only needed for glibc.
If an MP3 file contains the string NSVs, the NSV probe will confuse it for an
NSV file. Check for 0xBEEF after a Video/Audio chunk to achieve more accuracy.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
* qatar/master: (32 commits)
10-bit H.264 x86 chroma v loopfilter asm
Port SMPTE S302M audio decoder from FFmbc 0.3. [Copyright headers corrected]
Fix crash of interlaced MPEG2 decoding
h264pred: fix one more aliasing violation.
doc/APIchanges: fill in missing hashes and dates.
flacenc: use proper initializers for AVOption default values.
lavc: deprecate named constants for deprecated antialias_algo.
aac: workaround for compilation on cygwin
swscale: extend YUV422p support to 10bits depth
tiff: add support for inverted FillOrder for uncompressed data
Remove unused softfloat implementation.
h264pred: fix aliasing violations.
rotozoom: Eliminate French variable name.
rotozoom: Check return value of fread().
rotozoom: Return an error value instead of calling exit().
rotozoom: Make init_demo() return int and check for errors on invocation.
rotozoom: Drop silly UINT8 typedef.
rotozoom: Drop some unnecessary parentheses.
rotozoom: K&R coding style cosmetics
rtsp: Only do keepalive using GET_PARAMETER if the server supports it
...
Conflicts:
Changelog
cmdutils.c
doc/APIchanges
doc/general.texi
ffmpeg.c
ffplay.c
libavcodec/h264pred_template.c
libavcodec/resample.c
libavutil/pixfmt.h
libavutil/softfloat.c
libavutil/softfloat.h
tests/rotozoom.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It seems sws-PPC did hardcode 2048 at various places instead of using VOFW.
This also means that all past VOFW benchmarks on PPC are meaningless
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fix handling of input if not in native endianness, and add support for
9/10-bit output. This allows us to force endianness of YUV420P 9/10bit
in the H264/10bit fate tests, which should fix them on big-endian
systems.
The QuickTime RLE encoder only supports 16, 24, and 32-bit color. This patch adds support
for 8-bit grayscale.
Signed-off-by: John Horigan <john@glyphic.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This FourCC is used by "mpegable AVC" codec and the file encoded with this codec decodes correctly with FFmpeg's H264 decoder.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The file does not decode correctly yet the checksums match this wrongly
decoded file. Thus the checksums must be wrong.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
framecrc returns different values when one swiches endianness,
this apparently has been missed by "the fork" who added the 10bit fate
tests. Sorry for missing this during the merge.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
On cygwin, math.h needs to be included before float.h because of a bug
in the system headers. Including libavutil/libm.h first works around
this issue.
Longer discussion of the topic:
http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/128582
The softfloat functionality is unused, not installed and incomplete.
On platforms without floating point units, the compiler provides a softfloat
implementation so there is no point in carrying this code around locally.
This fixes the warnings:
tests/rotozoom.c:252: warning: ignoring return value of ‘fread’, declared with attribute warn_unused_result
tests/rotozoom.c:254: warning: ignoring return value of ‘fread’, declared with attribute warn_unused_result
This is more like what VLC does. If the server doesn't mention
supporting GET_PARAMETER in response to an OPTIONS request,
VLC doesn't send any keepalive requests at all. After this patch,
libavformat will still send OPTIONS keepalives if GET_PARAMETER
isn't explicitly said to be supported.
Some RTSP cameras don't support GET_PARAMETER, and will
close the connection if this is sent as keepalive request
(but support OPTIONS just fine, but probably don't need any
keepalive at all). Some other cameras don't support using
OPTIONS as keepalive, but require GET_PARAMETER instead.
Signed-off-by: Martin Storsjö <martin@martin.st>
In output_packet(), move the pre_process_video_frame() call inside the
if (ist->decoding_needed) { } block. This way
pre_process_video_frame() is not called when stream-copy has been
selected.
Also simplify.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Use av_log() rather than fprintf(stderr, ...), and show information
related to the previous size/pixel format configuration.
Consistent with the corresponding message issued in case of audio
configuration change.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master: (30 commits)
AVOptions: make default_val a union, as proposed in AVOption2.
arm/h264pred: add missing argument type.
h264dsp_mmx: place bracket outside #if/#endif block.
lavf/utils: fix ff_interleave_compare_dts corner case.
fate: add 10-bit H264 tests.
h264: do not print "too many references" warning for intra-only.
Enable decoding of high bit depth h264.
Adds 8-, 9- and 10-bit versions of some of the functions used by the h264 decoder.
Add support for higher QP values in h264.
Add the notion of pixel size in h264 related functions.
Make the h264 loop filter bit depth aware.
Template dsputil_template.c with respect to pixel size, etc.
Template h264idct_template.c with respect to pixel size, etc.
Preparatory patch for high bit depth h264 decoding support.
Move some functions in dsputil.c into a new file dsputil_template.c.
Move the functions in h264idct into a new file h264idct_template.c.
Move the functions in h264pred.c into a new file h264pred_template.c.
Preparatory patch for high bit depth h264 decoding support.
Add pixel formats for 9- and 10-bit yuv420p.
Choose h264 chroma dc dequant function dynamically.
...
Conflicts:
doc/APIchanges
ffmpeg.c
ffplay.c
libavcodec/alpha/dsputil_alpha.c
libavcodec/arm/dsputil_init_arm.c
libavcodec/arm/dsputil_init_armv6.c
libavcodec/arm/dsputil_init_neon.c
libavcodec/arm/dsputil_iwmmxt.c
libavcodec/arm/h264pred_init_arm.c
libavcodec/bfin/dsputil_bfin.c
libavcodec/dsputil.c
libavcodec/h264.c
libavcodec/h264.h
libavcodec/h264_cabac.c
libavcodec/h264_cavlc.c
libavcodec/h264_loopfilter.c
libavcodec/h264_ps.c
libavcodec/h264_refs.c
libavcodec/h264dsp.c
libavcodec/h264idct.c
libavcodec/h264pred.c
libavcodec/mlib/dsputil_mlib.c
libavcodec/options.c
libavcodec/ppc/dsputil_altivec.c
libavcodec/ppc/dsputil_ppc.c
libavcodec/ppc/h264_altivec.c
libavcodec/ps2/dsputil_mmi.c
libavcodec/sh4/dsputil_align.c
libavcodec/sh4/dsputil_sh4.c
libavcodec/sparc/dsputil_vis.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/x86/dsputil_mmx.c
libavformat/options.c
libavformat/utils.c
libavutil/pixfmt.h
libswscale/swscale.c
libswscale/swscale_internal.h
libswscale/swscale_template.c
tests/ref/seek/lavf_avi
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Looks better for some cases, worse for others, overall not much difference.
Its more correct though.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Just some cosmetics & comments, the only functional change
"error_resilience: use s->last_picture for accessing last MVs."
Is so buggy that it needs a full rewrite, guess -1.0 PSNR loss wasnt
enough for ronald to realize there was a problem.
This fixes some overflow in bright areas and ensures that the maximum brightness level is
mapped to the maximum without cliping and without showing dither patterens in flat max
brightness areas.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This should fix behavior introduced by commit
96573c0d76. Av_rescale_rnd() is not
lossless so if two timestamps are equal after being rescaled they are
not always actually identical. This patch use av_compare_ts() to get
always a correct result.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This patch lets e.g. dsputil_init chose dsp functions with respect to
the bit depth to decode. The naming scheme of bit depth dependent
functions is <base name>_<bit depth>[_<prefix>] (i.e. the old
clear_blocks_c is now named clear_blocks_8_c).
Note: Some of the functions for high bit depth is not dependent on the
bit depth, but only on the pixel size. This leaves some room for
optimizing binary size.
Preparatory patch for high bit depth h264 decoding support.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
In high bit depth, the QP values may now be up to (51 + 6*(bit_depth-8)).
Preparatory patch for high bit depth h264 decoding support.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
In high bit depth the pixels will not be stored in uint8_t like in the
normal case, but in uint16_t. The pixel size is thus 1 in normal bit
depth and 2 in high bit depth.
Preparatory patch for high bit depth h264 decoding support.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
The functions moved are used when decoding h264.
Preparatory patch for high bit depth h264 decoding support.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Mostly useful for the audio-filters branch, to make more apparent the
distinction between configure_audio_filters() and
configure_video_filters().
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The function was only used in opt_sample_fmt() for listing the sample
formats. Move list_fmts() functionality directly into
opt_sample_fmt().
Also fix the warning:
ffmpeg.c: In function ‘opt_audio_sample_fmt’:
ffmpeg.c:2877: warning: passing argument 1 of ‘list_fmts’ from incompatible pointer type
cmdutils.h:163: note: expected ‘void (*)(char *, int, int)’ but argument is of type ‘char * (*)(char *, int, enum AVSampleFormat)’
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
mpegaudiodec: group #includes more sanely
mpegaudio: remove #if 0 blocks
ffmpeg.c: reset avoptions after each input/output file.
ffmpeg.c: store per-output stream sws flags.
mpegaudio: remove CONFIG_MPEGAUDIO_HP option
mpegtsenc: Clear st->priv_data when freeing it
udp: Fix receiving RTP data over multicast
rtpproto: Remove an unused variable
regtest: fix wma tests
NOT pulled: mpegaudio: remove CONFIG_AUDIO_NONSHORT
regtest: separate flags for encoding and decoding
Conflicts:
ffmpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
With this change libavutil uses av_freep() everywhere where it makes sense.
Remaining av_free() use in it has the used pointer become inaccessible quickly
soo zeroing makes no sense.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The low quality mode is off by default and never tested. The high
quality mode is also plenty fast enough.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Earlier, AVIO_RDWR was handled as READ, and all the checks
for the AVIO_WRONLY flag explicitly meant the write-only case.
When changed from old AVIO/URL contants to AVIO_FLAG in
59d96941f0, these comparisons were updated incorrectly, by
mapping checks for AVIO_WRONLY to checks for AVIO_FLAG_WRITE.
When receiving RTP over UDP, the urlcontext is opened with
READ_WRITE flags.
This patch updates the flag comparisons to check for the same
conditions as the code did prior to 59d96941f0.
Signed-off-by: Martin Storsjö <martin@martin.st>
This adds $DEC_OPTS to the wma decode commands, making tests pass
on systems where the bitexact flag is needed.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This fake option cannot be enabled by normal means, and the
floating-point decoder should provide the higher quality this
option presumably was intended for.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This separates encoding and decoding flags, and passes them together
with the related file argument instead of all at the start of the
command line.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
log: Fix an oob array read.
cosmetics: trim trailing whitespace in postproc
Ban strncpy() it's too easy to misuse.
psymodel: Remove wrapper functions.
aacenc: Replace loop counters in aac_encode_frame() with more descriptive 'ch' and 'w'.
regtest: remove redundant flags in jpg test
regtest: use run_ffmpeg in do_image_formats
regtest: simplify encoding functions
ffmpeg.c: check for interlaced flag in the correct place.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When the file to decode contains a sequence of binary values like
"1101110...", decode_frame() was reading the sequence of digits like a
unique integer value, which was resulting in integer overflows.
The change add support for parsing non-space-separated pixel digits
for mono formats, in particular fix decoding of file battrace.pbm, and
fix trac issue #154.
The video encoding options were needlessly split in two parameters
which are merged. The do_audio_encoding function did not use its
second argument, so this can be removed.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
drawtext: add documentation for the shadow parameters
drawtext: add shadow support.
drawtext: factor draw_glyphs.
drawtext: fix and optimize yuv blend.
drawtext: get rid of divisions in the inner loop.
drawtext: simplify chroma blend and fix chroma alpha.
lavfi: Port drawtext filter by Hemanth from the libavfilter soc repo
Conflicts:
configure
libavfilter/avfilter.h
libavfilter/vf_drawtext.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
All decode_channel_map calls together can easily read
more data than the amount of padding available.
Thus below patch adds an input length check before reading them.
Fixes some invalid reads with sample from
http://bugzilla.mplayerhq.hu/show_bug.cgi?id=1138
This is required specifically for setting frame->format to -1,
otherwise it will be set to 0 = PIX_FMT_YUV420P and code reading
the format from the output decoded frame will get misled.
In particular fix regressions occurring with the pending vsrc_buffer
patch.
The check and the corresponding message in do_video_out() are
unnecessary in the libavfilter path, as a similar check and message
are performed within the buffer source.
Implement lazy initialization of the image resample context, so that
the initialization code does not need to be duplicated.
Also manage the case in which resample/size change mid-stream. For
each packet the code computes if resampling is needed.
Previously the resample check was done only at the initialization, in
case size/format changed and no resample was detected at the init
stage ffmpeg was silently exiting.
avfilter_copy_frame_props() avoids code duplication and increases
robustness.
The added files libavfilter/avcodec.[ch] are used for containing
utilities useful for gluing togheter libavfilter and libavcodec.
The format is a per-frame property, having it in AVFrame simplify the
operation of extraction of that information, since avoids the need to
access the codec/stream context.
width and height are per-frame properties, setting these values in
AVFrame simplify the operation of extraction of that information,
since avoids the need to check the codec/stream context.
With the following additions:
* support to anti-aliased glyph rendering
* support to UTF-8 text and Unicode chars rendering
* support for RGB packed formats
* fix minor errors and typos in the filter description
* extend/clarify examples in the filter description
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The patch below provides exactly that to the MP3 muxer. A XING header
containing
* the numer of frames,
* the size, and
* a TOC
is generated.
It's based on an idea by Anton Khirnov (restricted to the number of
frames) found at
http://patches.ffmpeg.org/patch/1891/
The TOC is generated as found in lame's "VbrTag.c".
According to my tests the following reproduces the number of frames, the
size and the TOC in "c.mp3" from "b.mp3" (except a shift due to shorter
XING header generated by FFmpeg):
lame -V2 a.wav b.mp3
ffmpeg -i b.mp3 -acodec copy -y c.mp3
This fixes, for instance, the case that there is a track that has some samples with composition time offset and has a track run without sample-composition-time-offsets-present.
In do_lavfi_pixfmts(), provide the filter arguments to showfiltfmts,
since some filter may require non-null or non-empty argument string
for working properly.
* qatar/master: (23 commits)
doc: Check standalone compilation before submitting new components.
Fix standalone compilation of pipe protocol.
Fix standalone compilation of ac3_fixed encoder.
Fix standalone compilation of binkaudio_dct / binkaudio_rdft decoders.
Fix standalone compilation of IMC decoder.
Fix standalone compilation of WTV demuxer.
Fix standalone compilation of MXPEG decoder.
flashsv: K&R cosmetics
matroskaenc: fix memory leak
vc1: make overlap filter for I-frames bit-exact.
vc1dec: use s->start/end_mb_y instead of passing them as function args.
Revert "VC1: merge idct8x8, coeff adjustments and put_pixels."
Replace strncpy() with av_strlcpy().
indeo3: Eliminate use of long.
get_bits: make cache unsigned to eliminate undefined signed overflow.
asfdec: fix assert failure on invalid files
avfilter: check malloc return values.
Not pulled as reason for reindent is not pulled: mpegvideo: reindent.
nutenc: check malloc return values.
Not pulled due to much simpler solution in ffmpeg *: don't av_malloc(0).
...
Conflicts:
doc/developer.texi
libavcodec/Makefile
libavcodec/get_bits.h
libavcodec/mpegvideo.c
libavformat/Makefile
libavutil/log.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This reverts commit f8bed30d8b. The reason
for this is that the overlap filter, which runs after IDCT, should run
on unclamped values, and thus IDCT and put_pixels() cannot be merged if
we want to attempt to be bitexact.
Add an extra size validity check in asf_read_frame_header(). Without
this asf->packet_size_left may become negative, which triggers an
assertion failure later.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master:
Implement fate-rsync target
Make ffmpeg support generic data stream
Conflicts:
cmdutils.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
lowres option crashes with wmv2 files which I have for test,
so I think it's better to remove the support to the time when
lowres will be really supported by this codec
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
vp8: frame-multithreading.
Duplicate Replace deprecated FF_*_TYPE symbols with AV_PICTURE_TYPE_*.
Duplicate Replace deprecated av_get_pict_type_char() with av_get_picture_type_char().
Bug spoted&removed in last merge: dpx: Do not use DPX encoder for decoding.
Conflicts:
ffmpeg.c
ffplay.c
libavcodec/h264.c
libavcodec/mpeg12.c
libavcodec/mpeg4videodec.c
libavcodec/mpegvideo.c
libavcodec/mpegvideo_enc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Do not use rlelen for buffer size in init_get_bits, it is
only the size of the data for the first field.
Since it is not reliable, just use the size of the whole buffer.
Additional comments add removal of unused rlelen variable by
Reimar Döffinger.
This means it uses full brightness range and brightness
increasing instead of decreasing with index of non-opaque
color.
Based on patch by Alexandre Colucci [alexandre elgato com]
* qatar/master:
ALPHA: Replace sized int_fast integer types with plain int/unsigned.
Duplicate DPX image encoder
Duplicate DPX decoder: read sample aspect ratio
Duplciate DPX decoder: add buffer size checks.
ac3enc: clip large coefficient values and negative exponents rather than using av_assert2().
ac3enc: do not store a bandwidth code for each channel.
ac3enc: remove bandwidth reduction as fallback for bit allocation failure.
ac3enc: merge compute_exp_strategy_ch() into compute_exp_strategy()
ac3enc: return error if frame+exponent bits are too large instead of using av_assert2().
ac3enc: differentiate between current block and reference block in bit_alloc()
ac3enc: simplify exponent_init() by calculating exponent_group_tab[] based on exponent group sizes.
ac3enc: simplify stereo rematrixing decision options
Include both URLs: Update URL to fate samples
Conflicts:
Changelog
doc/fate.txt
libavcodec/ac3enc.c
libavcodec/dpxenc.c
libavcodec/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
int/unsigned is the natural memory access type for CPUs, using sized types
for temporary variables, counters and similar just increases code size and
can possibly cause a slowdown.
They use now code identical to the AAC decoder.
The AC3 decoder previously did not check the data_size and
the dca decoder checked against and set wrong values for float.
Improve consistency with libavcodec.
This breaks libavfilter API/ABI.
The non-sequential 2.1.0 -> 2.4.0 bump is due to the mess previously
done with the lavfi minor number.
The sample aspect ratio is a per-frame property, so it makes sense to
define it in AVFrame rather than in the codec/stream context.
Simplify application-level sample aspect ratio information extraction,
and allow further simplifications.
This is similar to what was done with pkt_pts. This simplifies the
operation of extracting the pos information from the AVPacket, and
allows further simplifications.
One of the causes of this bug is that the h264 parser defaults low_delay
to 1, but the h264 codec defaults low_delay to 0. Really Ugly.
After many hours of looking at this, I'm still not sure how has_b_frames
is *intended* to behave, but to me the implementation appears way more
complicated than it ought to be.
My patch relies on the encoder to set an optional field in the SPS. This
works for libx264 streams, but I'm not sure that all h264 encoders will
set it.
* commit '85770f2a2651497861ed938efcd0df3696ff5e45':
AVOptions: make default_val a union, as proposed in AVOption2.
Move ff_dynarray_add to lavu and make it public.
lavf: remove duplicate assignment in avformat_alloc_context.
lavf: use designated initializers for AVClasses.
options: simplify av_find_opt by using av_next_option.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
I still don't fully understand the cause but the difference between
the samples that trigger the bug and the samples that don't is
that the former uses delay frames and the later uses drop frames
as placeholders for the packed frame. So, if we see the one type
of frame, we can assume the bug will or won't be present.
Right now, I'm detecting the frame types by size, which may not be
safe in general, but given the specific codec and file type, I
expect any scenario where we encounter these frames where they
aren't being used for b-frame packing won't care one way or
another whether the work around is in effect or not.
Signed-off-by: Philip Langdale <philipl@overt.org>
* qatar/master:
Duplicate AMV: disable DR1 and don't override EMU_EDGE
Duplicate lavf: inspect more frames for fps when container time base is coarse
Wrong and we have correct fix: Fix races in default av_log handler
vorbis: Replace sized int_fast integer types with plain int/unsigned.
Remove disabled non-optimized code variants.
NO bswap.h: Remove disabled code.
Remove some disabled printf debug cruft.
Replace more disabled printf() calls by av_dlog().
NO tests: Remove disabled code.
NO Replace some commented-out debug printf() / av_log() messages with av_dlog().
vorbisdec: Replace some sizeof(type) by sizeof(*variable).
NO vf_fieldorder: Replace FFmpeg by Libav in license boilerplate.
Conflicts:
libavcodec/h264.c
libavcodec/vorbisdec.c
libavutil/log.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This works around a possibly exploitable crash.
Appearently, vlc can be exploited with a malicous file. This should get
reverted as soon as a proper fix is found.
Reported-at: Thu, 21 Apr 2011 14:38:25 +0000
Reported-by: Dominic Chell <Dominic.Chell@ngssecure.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 89f903b3d5)
(cherry picked from commit 9b919571e5)
As per issue2629, most 23.976fps matroska H.264 files are incorrectly
detected as 24fps, as the matroska timestamps usually have only
millisecond precision.
Fix that by doubling the amount of timestamps inspected for frame rate
for streams that have coarse time base. This also fixes 29.970 detection
in matroska.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 78431098f9)
Tested with mplayer based on this report
http://thread.gmane.org/gmane.comp.video.mplayer.user/66043/focus=66063
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
int/unsigned is the natural memory access type for CPUs, using sized types
for temporary variables, counters and similar just increases code size and
can possibly cause a slowdown.
No longer needed after martins change to the values.
Though this would have been nicer ABI wise but iam too lazy to maintain this difference
This reverts commit 183401b924.
* qatar/master:
LOTS of duplicates ...
avio: Fix the deprecated fallback URL-prefixed open flags
avoid duplicate -lm in .pc files
mpeg12: terminate mpeg2_video_profiles arrays
network: Check POLLERR and POLLHUP in ff_network_wait_fd
mpegtsenc: make PMT PID really start on pmt_start_pid
nut format: support PIX_FMT_BGR48LE and PIX_FMT_BGR48BE
rawvideo codec: support PIX_FMT_BGR48LE and PIX_FMT_BGR48BE
hflip: make the filter accept PIX_FMT_BGR48LE and PIX_FMT_BGR48BE pixel formats
crop: make the filter accept PIX_FMT_BGR48LE and PIX_FMT_BGR48BE pixel formats
libswcale: PIX_FMT_BGR48LE and PIX_FMT_BGR48BE scaler implementation
Conflicts:
configure
libswscale/swscale.h
tests/ref/lavf/ts
Merged-by: Michael Niedermayer <michaelni@gmx.at>
While deprecated, they're totally useless as long as their values
are different from the AVIO_FLAG values that are used internally.
Currently, this leads to old libav applications still compiling
correctly (since we haven't removed the fallback wrappers), but
failing since the functions internally compare to the new AVIO_FLAG
values.
These should be removed at some point, but they aren't removed yet.
The intent is to be able to recompile an old application against
the new ABI without modifying the code, and this doesn't work
currently.
Signed-off-by: Martin Storsjö <martin@martin.st>
Make av_get_profile_name() return NULL if no profile is detected.
Fix trac issue #130, fix crash reading file tek3.m2v.
(cherry picked from commit e5d80c7b2d)
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Previously, the function would lead to an infinite wait (by
returning AVERROR(EAGAIN)) on sockets indicating an error
via either of these poll flags.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
vorbisdec: Rename silly "class_" variable to plain "class".
simple_idct_alpha: Drop some useless casts.
Simplify av_log_missing_feature().
ac3enc: remove check for mismatching channels and channel_layout
If AVCodecContext.channels is 0 and AVCodecContext.channel_layout is non-zero, set channels based on channel_layout.
If AVCodecContext.channel_layout and AVCodecContext.channels are both non-zero, check to make sure they do not contradict eachother.
cosmetics: indentation
Check AVCodec.supported_samplerates and AVCodec.channel_layouts in avcodec_open().
aacdec: remove sf_scale and sf_offset.
aacdec: use a scale of 2 in the LTP MDCT rather than doubling the coefficient table values from the spec.
Define POW_SF2_ZERO in aac.h and use for ff_aac_pow2sf_tabp[] offsets instead of hardcoding 200 everywhere.
Large intensity stereo and PNS indices are legal. Clip them instead of erroring out. A magnitude of 100 corresponds to 2^25 so the will most likely result in clipped output anyway.
qpeg: use reget_buffer() in decode_frame()
ultimotion: use reget_buffer() in ulti_decode_frame()
smacker: remove unnecessary call to avctx->release_buffer in decode_frame()
avparser: don't av_malloc(0).
Merged-by: Michael Niedermayer <michaelni@gmx.at>
avcodec_open().
If the encoder has a channel_layouts list and AVCodecContext.channel_layout
is 0, then only print a warning and let the encoder decide how to handle it.
Instead, scalefactors are adjusted by the offset amount, removing the need
for sf_scale, and the MDCT scales are adjusted to compensate for the higher
scalefactors. Floating-point output will be handled by modifying the MDCT
scales.
erroring out. A magnitude of 100 corresponds to 2^25 so the will most
likely result in clipped output anyway.
None of the conformance streams fall in the range that need to be clipped.
Decoder relies on previous frame data, so use reget_buffer().
This also set frame->reference to 3, as the frame will be requested
unmodified later so it shouldn't be modified by the application.
Fix playback of file Clock.avi.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
The release_buffer was cleaning the provided frame, thus causing the
successive call to avctx->reget_buffer() to allocate a new frame. In
case the returned frame was not the same one previously returned but a
new one with different data, it resulted in artifacts.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Also remove unnecessary call to avctx->release_buffer(). reget_buffer
is required since apparently the codec needs to be feeded with the
previous frame data.
Releasing the frame and using get_buffer was working only in the case
get_buffer() was returning the old frame data, and resulting in
playback artifacts otherwise.
Fix trac issue #85.
Store palette data in the codec context, do not assume that the frame
allocated the first time by avctx->get_buffer will be returned again.
This requires to store the palette which comes with the first frame in
the context, and copy it to the following frames.
Fix trac issue #84.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
Also deprecate av_get_pict_type_char() in favor of
av_get_picture_type_char().
The new enum and av_get_picture_type_char() are defined in libavutil.
This allows the use in libavfilter without the need to link against
libavcodec.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Also set movie->picref to NULL, in order to avoid a crash in uninit()
when movie->picref is unreffed again and it was already freed.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
graphparser: add a NULL check on the argument passed to strstr
setdar: prefer "sar" over "par" in log info message
fade: fix draw_slice() check on fade->factor value
fade: make draw_slice() chroma check against planes 1 and 2
win32: include the correct header in cmdutils.c
ac3: fix memleak in fixed-point encoder
flashsv: Return more meaningful error values.
flashsv: Employ explicit AVCodec struct initializers.
read AVI palette from the end of extradata
cosmetics: K&R coding style and more whitespace for Flash Screen Video
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is required for setting up the filterchain, as it requires the
pixel format to be set, which is usually done when probing the file
with av_find_stream_info().
Fix trac issue #83.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
draw_slice() checks that the fade factor is < 65536 and only
calculates the fade if so. But the fade factor is clipped in
end_frame() by av_clip_uint16() to 65535, so the fade is calculated
for every frame. This patch alters the check so that it compares with
< 65535 (UINT16_MAX).
Signed-off-by: Anton Khirnov <anton@khirnov.net>
draw_slice() checks that planes 0 and 1 of AVFilterBufferRef's data
are not NULL before manipulating planes 1 and 2. This patch makes the
check against planes 1 and 2. More senseful and possibly more robust.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
In the loop:
for (i=0; i<dstH; i++) {
int chrI= i*c->chrDstH / dstH;
when i*c->chrDstH > INT_MAX this leads to an integer overflow, which
results in a negative value for chrI and in out-of-buffer reads. The
overflow is avoided by forcing int64_t arithmetic by casting i to
int64_t.
Fix crash, and trac issue #72.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
Also automatically select the show mode only if not specified by the
user.
Fix trac issue #109.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
It does not make sense (DV is interleaved by design) and
it avoids a crash when the non-interleaved code tries to
use the priv_data of streams created by the DV demuxer.
The crash could be avoided differently, but then that stream
would still lack an index and would not play correctly in
non-interleaved mode.
Fixes e.g. samples/ffmpeg-bugs/roundup/issue1514/Dennis0002_video1.avi
Official AVI specification says that stream header in case of video contains
BITMAPINFO, which is equal to BITMAPINFOHEADER and optional palette. Currently
lavf AVI demuxer thinks otherwise which produces garbage on codecs that have
both palette and extradata (luckily, there are not so many such codecs).
An example of such file is:
http://samples.multimedia.cx/V-codecs/KMVC/baseball1.avi
(IIRC, MSS1 or MSS2 also had such situation but they are still not supported
by lavc).
As a side note, passing palette in extradata as it's been done previously is
not quite correct since proper _extra_ data is surplus bytes in
BITMAPINFOHEADER, not including palette.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master:
vorbisdec: Employ proper printf format specifiers for uint_fast32_t.
applehttp: Don't export variant_bitrate if it isn't known
crypto: Use av_freep instead of av_free
configure: Add missing libm library dependencies to .pc files.
oggdec: reindent after 8f3eebd6
configure: Add missing libavutil inter-library dependencies to .pc files.
Conflicts:
configure
Merged-by: Michael Niedermayer <michaelni@gmx.at>
libavcodec/vorbisdec.c:543: warning: format ‘%d’ expects type ‘int’, but argument 4 has type ‘uint_fast32_t’
libavcodec/vorbisdec.c:543: warning: format ‘%d’ expects type ‘int’, but argument 5 has type ‘uint_fast32_t’
In video_image_display(), fix exception occurring when the size of the
rectangle passed to SDL_DisplayYUVOverlay() is 0x0, which happens when
interactively resizing the SDL window.
This is done by forcing the minimum size to 1x1.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
This enables non interleaved AVI mode for example.
Players that are picky on strict interleaving can set this.
Patches to only switch to non intereaved AVI mode when the index is not strictly
correctly interleaved are welcome.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If there are no variants, the total bitrate of the single
stream isn't known, and exporting variant_bitrate = 0 does
look weird, since there really aren't any variants.
Signed-off-by: Martin Storsjö <martin@martin.st>
The CrystalHD hardware can do scaling, which is particularly
desirable when dealing with some high resolution clips that take
so long to decode and copy out that they end up playing back
slower than realtime. By using scaling, we can make the output
frames smaller and reduce the copy out time.
This option takes the desired horizontal width in pixels, and
the hardware will do an aspect-scale. Upscaling is not supported
and the hardware will simply ignore any request to do so.
Signed-off-by: Philip Langdale <philipl@overt.org>
This unbreaks static compilation using pkg-config on systems in need of -lm.
Based on an mplayer2 patch by Uoti Urpala <uau@mplayer2.org>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
This unbreaks static compilation when using pkg-config.
Based on an mplayer2 patch by Uoti Urpala <uau@mplayer2.org>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
* qatar/master:
Handle unicode file names on windows
rtp: Rename the open/close functions to alloc/free
Lowercase all ff* program names.
Refer to ff* tools by their lowercase names.
NOT Pulled Replace more FFmpeg instances by Libav or ffmpeg.
Replace `` by $() syntax in shell scripts.
patcheck: Allow overiding grep program(s) through environment variables.
NOT Pulled Remove stray libavcore and _g binary references.
vorbis: Rename decoder/encoder files to follow general file naming scheme.
aacenc: Fix whitespace after last commit.
cook: Fix small typo in av_log_ask_for_sample message.
aacenc: Finish 3GPP psymodel analysis for non mid/side cases.
Remove RDFT dependency from AAC decoder.
Add some debug log messages to AAC extradata
Fix mov debug (u)int64_t format strings.
bswap: use native types for av_bwap16().
doc: FLV muxing is supported.
applehttp: Handle AES-128 encrypted streams
Add a protocol handler for AES CBC decryption with PKCS7 padding
doc: Mention that DragonFly BSD requires __BSD_VISIBLE set
Conflicts:
ffplay.c
ffprobe.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
All file names should be in UTF-8 within libavformat.
This is handled by mapping the open() function to an internal one
in os_support.h for windows.
fopen() could be overridden in the same way, but if that would be
used from ffmpeg.c, it would add a dependency on an ff prefixed
internal lavf function.
Signed-off-by: Martin Storsjö <martin@martin.st>
patcheck hardcodes the binary names for grep/egrep. This makes overriding the
binary names a pain, e.g. when calling a GNU version of grep on BSD systems.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
The new name is less misleading, since the function will resume the
stream if it is currently paused.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
Add a check in flic_decode_frame_8BPP(), in case chunk_size is >
frame_size issue a warning and resize chunk_size to frame_size, in
order to avoid out-of-buffer reads.
Fix roundup issue #2520, trac issue #69.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
There is still are still a few sections missing relating to TNS (not present)
and mid/side (contains other bugs).
Overall this improves quality, and vastly improves rate-control.
Signed-off-by: Martin Storsjö <martin@martin.st>
$subj
>From 557176d961c70604c2a96d81aff4bd6faa670d8a Mon Sep 17 00:00:00 2001
From: Alex Converse <aconverse@google.com>
Date: Thu, 21 Apr 2011 12:11:42 -0700
Subject: [PATCH] Remove RDFT dependency from AAC decoder.
MIME-Version: 1.0
Content-Type: multipart/mixed; boundary="------------1"
This is a multi-part message in MIME format.
--------------1
Content-Type: text/plain; charset=UTF-8; format=fixed
Content-Transfer-Encoding: 8bit
It was used for an old implementation of the SBR filterbank.
./configure --disable-everything --disable-ffplay --enable-decoder=aac works.
This prevents a call to bytestream_get_be16() using a movzwl both before
and after the ror instruction, which is obviously inefficient. Arm uses
the same trick also.
Sintel decoding goes from (avg+SD) 9.856 +/- 0.003 to 9.797 +/- 0.003 sec.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master:
Use av_log_ask_for_sample() to request samples from users.
Make av_log_ask_for_sample() accept a variable number of arguments.
vqavideo: We no longer need to ask for version 1 samples.
aacdec: indentation cosmetics
Conflicts:
libavcodec/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This should hopefully fix roundup issue 2586.
This commit only implements it in the demuxer, not in the
protocol handler. If desired, some of the code could be
refactored to be shared by both implementations.
Signed-off-by: Martin Storsjö <martin@martin.st>
This can later be extended to support other AES bit sizes,
encryption, other crypto algorithms, reading the key from a URL, etc.
In order to use it, the key and initialization vector has to be
passed via AVOptions. Since such options can't be passed to
protocols from the command line, the protocol is currently
only for libavformat internal use.
Signed-off-by: Martin Storsjö <martin@martin.st>
Setting this should fix the FATE build failure on DragonFly BSD,
since inet_aton only is visible if __BSD_VISIBLE is set.
Alternatively, a line defining __BSD_VISIBLE=1 could be
added at the top of os_support.c. For FreeBSD, similar lines
are required in libavdevice/bktr.c and libavdevice/oss_audio.c, too.
Signed-off-by: Martin Storsjö <martin@martin.st>
The removal was not applied when merging the Libav patches, because
there was still code referencing the symbols.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
AVERROR_NOTSUPP has been dropped with the major bump. Since the
signalled error denotes a configuration problem, AVERROR(EINVAL) looks
an adequate replacement.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
* qatar/master:
APIChanges: document git revision for CODEC_CAP_SLICE_THREADS addition.
Introduce slice threads flag.
FATE: allow forcing thread-type when doing threaded fate runs.
Use av_log_ask_for_sample() where appropriate.
error: sort, pack, and align error code and string definitions
The stabilization period after version bumps should be one month, not one week.
applehttp: Expose the stream bitrate via metadata
doc: Add some initial docs on the applehttp demuxer
Provide a fallback version of the libm function trunc
libavdevice: Define _XOPEN_SOURCE for usleep
lavc: provide deprecated avcodec_thread_init until next major version
lavc: provide the opt.h header until the next bump
error: change AVERROR_EOF value
error: remove AVERROR_NUMEXPECTED
error: add error code AVERROR_OPTION_NOT_FOUND, and use it in opt.c
Conflicts:
libavcodec/h264.c
libavutil/error.c
libavutil/error.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The old API was deprecated and will be possibly removed, fix warnings
in libavfilter/libmpcodecs/vf_geq.c and libavfilter/libmpcodecs/vf_qp.c.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
In write_packet(), return AVERROR(EINVAL) rather than AVERROR(EIO) if
the provided output filename pattern is not valid. Indeed this has to
be considered a configuration error rather than an I/O error.
Allow a less confusing error report.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
In output_packet(), move the pre_process_video_frame() call inside the
if (ist->decoding_needed) { } block. This way
pre_process_video_frame() is not called when stream-copy has been
selected.
Also simplify.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
The current value is masking the POSIX error code EPIPE, which has a
different semantics.
This breaks API.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
AVERROR_NUMEXPECTED is used only in the image muxer and demuxer, and
has a too much specific meaning, which is better explained through a
log message. Thus it can be replaced by AVERROR(EINVAL).
This breaks API.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The new error code is better than AVERROR(ENOENT), which has a
completely different semantics ("No such file or directory").
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
Makefile: Include dependencies for test tools, too
Remove a version check in av_log made unnecessary by the big bump.
update last major version increase dates in APIchanges
Reduce picture size for yadif.
oggdec: use av_freep() instead of av_free()
avio: Fix sanity checks in ffurl_read*
libavformat: Free AVFormatContext->streams
libavformat: Make protocols pass URLContext as log context where available
asf: remove commented out code in asf_read_seek
not pulled: libpostproc: Remove crufty code disabled by the big bump.
Reflect 0.7_beta1 release in the Changelog
not pulled: sws: remove disabled cruft.
lavu: remove disabled ff_random_get_seed cruft.
lavu: remove disabled sha1 cruft.
Conflicts:
Changelog
libavutil/sha1.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This makes seek_test to be rebuilt when its dependencies
has changed. The changes to the dependencies didn't usually matter
in practice, but the introduction of side data in AVPacket required
a recompilation.
Signed-off-by: Martin Storsjö <martin@martin.st>
After switching this from a statically allocated array to a
dynamically allocated one in the major bump, this needs explicit
freeing.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (37 commits)
In avcodec_open(), set return code to an error value only when an error occurs instead of unconditionally at the start of the function.
lavc: remove reference to opt.h from Makefile.
prefer avio_check() over url_exist()
avio: remove AVIO_* access symbols in favor of new AVIO_FLAG_* symbols
lavu: remove misc disabled cruft
lavu: remove FF_API_OLD_IMAGE_NAMES cruft
NOT PULLED lavu: remove FF_API_OLD_EVAL_NAMES cruft
lavc: remove misc disabled cruft.
lavc: remove the FF_API_INOFFICIAL cruft.
lavc: remove the FF_API_SET_STRING_OLD cruft.
lavc: remove the FF_API_USE_LPC cruft.
lavc: remove the FF_API_SUBTITLE_OLD cruft.
lavc: remove the FF_API_VIDEO_OLD cruft.
lavc: remove the FF_API_AUDIO_OLD cruft.
lavc: remove the FF_API_OPT_SHOW cruft.
lavc: remove the FF_API_MM_FLAGS cruft.
lavf: remove misc disabled cruft.
lavf: remove FF_API_INDEX_BUILT cruft
lavf: remove FF_API_URL_CLASS cruft.
lavf: remove FF_API_SYMVER cruft
...
Merged-by: Michael Niedermayer <michaelni@gmx.at>
instead of unconditionally at the start of the function.
This fixes a bug where a successful call to ff_thread_init() masks errors that
occur after that point in the function. It also makes future bugs like this
less likely since the error code is now set near to the point in the code
where the error is found.
The problem with url_exist() is that it tries to open a resource in
RDONLY mode. If the file is a FIFO and there is already a reading
client, the open() call will hang.
By using avio_check() with access mode of 0, the second reading
process will check if the file exists without attempting to open it,
thus avoiding the lock.
Fix issue #1663.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Make AVIO_FLAG_ access constants work as flags, and in particular fix
the behavior of functions (such as avio_check()) which expect them to
be flags rather than modes.
This breaks API.
Use av_log() rather than fprintf(stderr, ...), and show information
related to the previous size/pixel format configuration.
Consistent with the corresponding message issued in case of audio
configuration change.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
* commit 'f0a41afd8a37ebe972436fabfa3d289178bbd83b':
Not pulled: Replace mplayerhq.hu URLs by libav.org.
asfdec: Remove dead code from asf_read_close().
ptx: Use av_log_ask_for_sample() where appropriate.
Conflicts:
libavcodec/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Also set movie->picref to NULL, in order to avoid a crash in uninit()
when movie->picref is unreffed again and it was already freed.
Fix trac issue #66.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
* ffmpeg-mt/master:
Release unused pictures even when not calling ff_h264_frame_start()
h264: Fix decoding race condition with PAFF
h264: cosmetic whitespace change
Duplicate Fix REBASE_PICTURE with h.264
Not pulled Update test scripts to use ffmpeg instead of ffmpeg_g
Duplicate Fix ffmpeg-mt fixme in h264
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
ac3dec: fix processing of delta bit allocation information.
vc1: fix fate-vc1 after previous commit.
wmv3dec: fix playback of complex WMV3 files using simple_idct.
make av_dup_packet() more cautious on allocation failures
make containers pass palette change in AVPacket
introduce side information for AVPacket
Politic commits that have not been pulled:
Update regtest checksums after revision 6001dad.
Replace more FFmpeg references by Libav.
Replace references to ffmpeg-devel with libav-devel; fix roundup URL.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The constant values were never set, thus the evaluation of expressions
referencing those constants were resulting in an error.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
PROFILE_ADVANCED doesn't set res_fasttx, so make that a special case
in the condition that decides which IDCT to use (and whether to read
coefficients transposed or not).
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
Redesign the way -aspect option is handled. This is done by making
ffmpeg read the sample aspect ratio set in the corresponding input
stream by default, and overriding it using the value specified by
-aspect.
If the output display aspect ratio is specified with -aspect, it is
set at the end of the filterchain, thus overriding the value set by
filters in the filterchain.
This implementation is more robust, since does not modify the
filterchain description (which was creating potential syntax errors).
There is no need to have 2 encoders, the input sample format can,does and should choose which is used
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Prefer parse_number_or_die() over atoi()/atol() parsing for the options:
-pass, -top, -vc, and -qscale.
Improve input validation.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
IDCT coefficients are read transposed, but simple_idct does not expect
this. Therefore, only do tranposed coefficient reading if we're not
using simple_idct.
Fixes http://forum.videolan.org/viewtopic.php?f=14&t=89651
I was using the wrong value to track the position of the parser in the
stream. For an error-free stream, the size of the frame and number of
bytes consumed will be the same, but in an error situation they can
diverge.
Signed-off-by: Philip Langdale <philipl@overt.org>
* commit 'f668afd4896ee65683619b6f165dda4cdd46766f':
swscale: fix "ISO C90 forbids mixed declarations and code" warning
swscale: fix compilation with --enable-runtime-cpudetect
swscale: correct include path to fix ppc altivec build
swscale: simplify rgb2rgb templating
swscale: simplify initialization code
swscale: further cleanup swscale.c
swscale: partially move the arch specific code left
swscale: move away x86 specific code from rgb2rgb
None of above is merged, the changes are not yet ready they add over 2000 lines of duplicated code and a few bugs
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Also add inttypes.h #include for PRId64 macros.
This fixes the following warnings:
libavformat/xwma.c:147: warning: too many arguments for format
libavformat/xwma.c:151: warning: too many arguments for format
The warning is printed with loglevel error because the output
stream is known to be broken.
Also warn if no channel layout was provided for multichannel input.
* commit 'a3a0af4fb1237bed0af75868073f9a63db8b1864':
avutil: make aes and sha api public
handle malloc failures in ff_get_wav_header
add xWMA demuxer
mpegts: force the default timebase
libavcodec: Bump micro after adding the dtx AVOptions
Merged-by: Michael Niedermayer <michaelni@gmx.at>
PPC and x86 code is split off from swscale_template.c. Lots of code is
still duplicated and should be removed later.
Again uniformize the init system to be more similar to the dsputil one.
Unset h*scale_fast in the x86 init in order to make the output
consistent with the previous status. Thanks to Josh for spotting it.
Keep only the plain C code in the main rgb2rgb.c and move the x86
specific optimizations to x86/rgb2rgb.c
Change the initialization pattern a little so some of it can be
factorized to behave more like dsputils.
The functions are already av_ prefixed and sha1 header is already provided.
Install libavutil/{aes,sha}.h
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
ff_get_wav_header is reading data from a WAVE file and then uses it
(without validation) to malloc a buffer. It then proceeded to read
data into the buffer, without verifying that the allocation succeeded.
To address this, change ff_get_wav_header to return an error if
allocation failed, and adapted all calling code to handle that error.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The mpegts muxer does not set the stream time base using
av_set_pts_info, but expects it to have the default value of 1/90000.
If the calling code changes stream pts before writing the header,
other muxers override the time base at that point (like mpegenc.c).
* qatar/master: (32 commits)
libopencore-amr, libvo-amrwbenc: Allow enabling DTX via private AVOptions
libopencore-amr, libvo-amrwbenc: Only check the bitrate when changed
libopencore-amr, libvo-amrwbenc: Find the closest matching bitrate
libvo-*: Fix up the long codec names
libavcodec: Mark AVCodec->priv_class const
swscale: Factorize FAST_BGR2YV12 definition.
libvo-aacenc: Only produce extradata if the global header flag is set
lavf: postpone removal of public metadata conversion API
lavc: postpone removal of request_channels
lavc: postpone removal of audioconvert and sample_fmt wrappers
lavf: postpone removal of deprecated avio functions
libopencore-amr: Cosmetics: Rewrap and align
libopencore-amr, libvo-amrbwenc: Rename variables and functions
libopencore-amr: Convert commented out debug logging into av_dlog
libopencore-amr: Remove an unused state variable
libvo-amrwbenc: Don't explicitly store bitrate modes in the bitrate table
libopencore-amr: Remove a useless local variable
libopencore-amr, libvo-amrwbenc: Make the bitrate/mode mapping array static const
libopencore-amr, libvo-amrwbenc: Return proper error codes in most places
libopencore-amr: Don't print carriage returns in log messages
...
Conflicts:
doc/developer.texi
libavcodec/avcodec.h
libavcodec/libvo-aacenc.c
libavcodec/libvo-amrwbenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
DTX, discontinuous transmission, allows emitting frames with
comfort noise when no voice is detected in the input audio.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also rename the incorrectly named enc_bitrate to enc_mode, use the
enc_bitrate variable for storing the last chosen bitrate.
This avoids continuous warning log messages if not using an
exactly matching bitrate, while still allowing changing bitrate
at any point.
Signed-off-by: Martin Storsjö <martin@martin.st>
Dynamically print the supported bitrates from the local table,
instead of using a hardcoded log message.
Signed-off-by: Martin Storsjö <martin@martin.st>
Include VisualOn to clarify the codec name, but remove the
actual library name from the user-friendly description.
Also mention Android, to clarify which VisualOn implementation
this refers to, since they do sell other variants of the same
code, too.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids warnings of this kind, everywhere priv_class
is initialized:
warning: initialization discards qualifiers from pointer target type
Signed-off-by: Martin Storsjö <martin@martin.st>
This code could be executed if the demuxer reads more than one
segment before returning from av_open_input_stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
When CONFIG_AVFILTER, use a AVFrame -> AVPicture cast rather than
explicitely copy the image information from AVFrame to a new
AVPicture.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
The new function is more flexible than url_exist(), as it allows to
specify which access flags to check, and does not require an explicit
open of the checked resource.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
Partially merged:flvdec: Allow parsing keyframes metadata without seeking in most cases
Error out if vaapi is not found
avio: undeprecate av_url_read_fseek/fpause under nicer names
libvo-*: Don't use deprecated sample format names and enum names
DUPLICATE flvdec: Fix support for flvtool2 "keyframes based" generated index
DUPLICATE libavcodec: Use "const enum AVSampleFormat[]" in AVCodec initialization
Fix the conversion of AV_SAMPLE_FMT_FLT and _DBL to AV_SAMPLE_FMT_S32.
Convert some undefined 1<<31 shifts into 1U<<31.
Conflicts:
configure
libavcodec/libvo-aacenc.c
libavcodec/libvo-amrwbenc.c
libavformat/flvdec.c
Marged-by: Michael Niedermayer <michaelni@gmx.at>
Stop the avio input at a point where amf_parse_object can
continue parsing the end of the object seamlessly, when all
data is available.
If unsupported data is encountered within the keyframes object,
try seeking to the start of the keyframes object - if the seek
back was successful, the caller can continue parsing the rest
of the AMF data.
Signed-off-by: Martin Storsjö <martin@martin.st>
Current keyframes data parser unconditionally rewind metadata to
the end at the end of function. As result ALL metadata located
after keyframes index not parsed, and as metadata object can have
ANY placement inside metadata it can lead to unpredictable result
(bitrate can not be found, etc.). As result FLV movie will not
play at all in such situation.
Signed-off-by: Martin Storsjö <martin@martin.st>
According to ISO 9899:1999 S 6.5.7/4:
The result of E1 << E2 is E1 left-shifted E2 bit positions; vacated bits
are filled with zeros. If E1 has an unsigned type, the value of the
result is E1× 2^E2, reduced modulo one more than the maximum value
representable in the result type. If E1 has a signed type and
nonnegative value, and E1× 2^E2 is representable in the result type, then
that is the resulting value; otherwise, the behavior is undefined.
via the equivalent AVCodecContext::crf modifying the range allowed by
options.c (0-51 -> 0-63)
update configure check to require libvpx 0.9.6 (Bali)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Not pulled due to wrong author, according to carl: Add unconditional return statement to yuva420_rgb32_MMX() / yuva420_bgr32_MMX().
Fix compilation of timefilter test program.
Add support for AMR-WB encoding via libvo-amrwbenc
Add an AAC encoder by using the libvo-aacenc library
g722: Return 8 bits per sample for g722 in av_get_bits_per_sample
Conflicts:
Changelog
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Call this new function before decode() to replace the custom and
inconsistant initialization in various decoders.
This function is equivalent to avcodec_get_frame_defaults() for AVFrame.
Signed-off-by: Aurelien Jacobs <aurel@gnuage.org>
When HAVE_7REGS was not defined these functions had an empty body
causing the following warnings during compilation.
In file included from libswscale/x86/yuv2rgb_mmx.c:58:
libswscale/x86/yuv2rgb_template.c: In function ‘yuva420_rgb32_MMX’:
libswscale/x86/yuv2rgb_template.c:412: warning: no return statement in function returning non-void
libswscale/x86/yuv2rgb_template.c: In function ‘yuva420_bgr32_MMX’:
libswscale/x86/yuv2rgb_template.c:457: warning: no return statement in function returning non-void
Signed-off-by: Diego Biurrun <diego@biurrun.de>
The printf function is forbidden in library code, but not in test programs,
so #undefine the printf macro that disables the system printf function.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Change:
Expected 7 arguments, but only %d found in '%s'\n
to:
Expected 7 arguments, but %d found in '%s'\n
as the user may provide more than 7 arguments, in that case the error
is not misleading.
This fixes a failing assert in ff_raw_read_header (in
fate-g722dec-1), where bits_per_coded_sample is set using this
function and is required to have a positive value.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
avio: add more documentation for AVIOContext.
Parse sprite data for WMVP and WVP2, and decode sprites for the latter
Replace outdated info on the FAQ
Redefine sameq
pad: fix example explanation
gradfun: add notice from the MPlayer manual
eval: add support for trunc, ceil, and floor functions
documentation: add setdar and setsar description to filters.texi
avio: document some members of AVIOContext.
avio: document avio_close().
avio: cosmetics, vertically align comments.
avio: cosmetics, group the reading functions.
avio: cosmetics, merge all the FF_API_OLD_AVIO blocks.
avio: cosmetics, move AVIOContext to start of the file.
avio: update file header.
os: fix OpenBSD/PowerPC compilation
pixfmt: add PIX_FMT_BGR48LE and PIX_FMT_BGR48BE
oggdec: fix demuxing chained audio streams
fix typo
Conflicts:
doc/filters.texi
libavformat/avio.h
libavutil/pixfmt.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Mostly useful for the audio-filters branch, to make more apparent the
distinction between configure_audio_filters() and
configure_video_filters().
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
This patch lets e.g. dsputil_init chose dsp functions with respect to
the bit depth to decode. The naming scheme of bit depth dependent
functions is <base name>_<bit depth>[_<prefix>] (i.e. the old
clear_blocks_c is now named clear_blocks_8_c).
Note: Some of the functions for high bit depth is not dependent on the
bit depth, but only on the pixel size. This leaves some room for
optimizing binary size.
Preparatory patch for high bit depth h264 decoding support.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
In high bit depth, the QP values may now be up to (51 + 6*(bit_depth-8)).
Preparatory patch for high bit depth h264 decoding support.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
In high bit depth the pixels will not be stored in uint8_t like in the
normal case, but in uint16_t. The pixel size is thus 1 in normal bit
depth and 2 in high bit depth.
Preparatory patch for high bit depth h264 decoding support.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The functions moved are used when decoding h264.
Preparatory patch for high bit depth h264 decoding support.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Unfortunately the output buffer size check assumes that the
input buffer is never over-consumed, thus this actually
also allowed to write outside the output buffer if "lucky".
FAQ could use some work. This updates a few little details.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
I think the documentation of sameq is causing confusion and misuse of
this option.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
When attempting to re-enable the AltiVec support it was noticed
that we need to undefine _POSIX_C_SOURCE to appease the headers
for ff_get_cpu_flags_ppc() to be able to compile.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
AS libavcodec/arm/ac3dsp_armv6.o
ffmpeg-src/libavcodec/arm/ac3dsp_armv6.S: Assembler messages:
ffmpeg-src/libavcodec/arm/ac3dsp_armv6.S:40: Error: selected processor
does not support `movw r8,#0x1fe0'
make[1]: *** [libavcodec/arm/ac3dsp_armv6.o] Error 1
MOVW is ARMv7 way to load constant:
* movw, or move wide, will move a 16-bit constant into a register,
implicitly zeroing the top 16 bits of the target register.
* movt, or move top, will move a 16-bit constant into the top half
of a given register without altering the bottom 16 bits
To load 32 bit constant, movw lower16; movt upper16; is better than
ldr if available, because:
While this approach takes two instructions, it does not require any
extra space to store the constant so both the movw/movt method and the
ldr method will end up using the same amount of memory. Memory
bandwidth is precious in and the movw/movt approach avoids an extra
read on the data side, not to mention the read could have missed the
cache.
But here it is armv6 optimization, so that we have to use ldr.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
daud: Don't write packets that are too large to have their size expressed in the bytestream.
DUPLICATE pthread: validate_thread_parameters() ignored slice-threading being intentionally off
DUPLICATE Use consistent condition for whether to run slice-threading execute function.
avformat.h: fix a misplaced #endif
DUPLCIATE vaapi: cope with VA-API 0.32, i.e. fix VC-1 decoding on Sandy Bridge.
DUPLICATE vaapi: filter out DELAYED_PIC_REF flag to determine field.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The thread_type API allows you to request only FF_THREAD_FRAME (instead of
FRAME | SLICE), but it was being ignored.
We don't implement both of them at the same time, so there isn't an effect
on current codecs, except that you can request no kinds of threading now
(a bit useless).
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
As previously discussed, the CrystalHD hardware returns exceptionally
useless information about interlaced h.264 content - to the extent
that it's not possible to distinguish most MBAFF and PAFF content until
it's too late.
In an attempt to compensate for this, I'm introducing two mechanisms:
1) Peeking at the picture number of the next picture
The hardware provides a capability to peek the next picture number. If
it is the same as the current picture number, then we are clearly dealing
with two fields and not a frame or fieldpair.
If this always worked, it would be all we need, but it's not guaranteed
to work. Sometimes, the next picture may not be decoded sufficiently
for the number to be known; alternately, a corruption in the stream may
cause the hardware to refuse to return the number even if the next
intact frame is decoded. In either case, the query will return 0.
If we are unable to peek the next picture number, we assume that the
picture is a frame/fieldpair and return it accordingly. If that turns
out to be incorrect, we discard the second field, and the user has
to live with the glitch. In testing, false detection can occur for
the first couple of seconds, and then the pipeline stabalizes and
we get correct detection.
2) Use the h264_parser to detect when individual input fields have
been combined into an output fieldpair.
I have multiple PAFF samples where this behaviour is detected. The
peeking mechanism described above will correctly detect that the
output is a fieldpair, but we need to know what the input type was
to ensure pipeline stability (only return one output frame per input
frame).
If we find ourselves with an output fieldpair, yet the input picture
type was a field, as reported by the parser, then we are dealing with
this case, and can make sure not to return anything on the next
decode() call.
Taken together, these allow us to remove the hard-coded hacks for
different h.264 types, and we can clearly describe the conditions
under which we can trust the hardware's claim that content is
interlaced.
Signed-off-by: Philip Langdale <philipl@overt.org>
Now that we know the type of the input picture, we have to bring
that information to the output picture to help identify its type.
We do this by adding a field to the opaque_list node.
Signed-off-by: Philip Langdale <philipl@overt.org>
As the hardware is unreliable, we will have to use the h.264 parser
to identify whether an input picture is a field or a frame. This
change loads the parser and extracts the picture type.
Signed-off-by: Philip Langdale <philipl@overt.org>
In preparation for adding additional fields to the node, return
the node instead of the pts value. This requires the caller to
free the node.
Signed-off-by: Philip Langdale <philipl@overt.org>
I found another MBAFF sample where the input:output pattern is
the same as mpeg2 and vc1 (fieldpair input, individual field output).
While I'm not sure how you can output individual fields from MBAFF,
if I apply the mpeg2/vc1 handling to this file, it plays correctly.
So, this changes the detection algorithm to handle the known cases.
Whitespace will be fixed in a separate change.
Signed-off-by: Philip Langdale <philipl@overt.org>
Current keyframes data parser unconditionally rewind metadata to the end
at the end of function. As result ALL metadata located after keyframes
index not parsed,
and as metadata object can have ANY placement inside metadata it can
lead to unpredictable result
(bitrate can not be found, etc.). As result FLV movie will not play at
all in such situation.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavf: bump minor and add an APIChanges entry for avformat cleanup
lavf: get rid of ffm-specific stuff in avformat.h
Not pulled: avio: deprecate av_protocol_next().
avio: add a function for iterating though protocol names.
lavf: rename a parameter of av_sdp_create from buff->buf
lavf: rename avf_sdp_create to av_sdp_create.
lavf: make av_guess_image2_codec internal
avio: make URLProtocol internal.
avio: make URLContext internal.
lavf: mark av_pkt_dump(_log) for remove on $next+1 bump.
lavf: use designated initializers for all protocols
applehttp: don't use deprecated url_ functions.
avio: move two ff_udp_* functions from avio_internal to url.h
asfdec: remove a forgotten declaration of nonexistent function
avio: deprecate the typedef for URLInterruptCB
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* ffmpeg-mt/master:
DUPLICATE mingw32 compilation after 'unbreak avcodec_thread_init'
pthread: validate_thread_parameters() ignored slice-threading being intentionally off
DUPLICATE Remove unnecessary parameter from ff_thread_init() and fix behavior
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
proto: include os_support.h in network.h
matroskaenc: don't write an empty Cues element.
lavc: add a FF_API_REQUEST_CHANNELS deprecation macro
avio: move extern url_interrupt_cb declaration from avio.h to url.h
avio: make av_register_protocol2 internal.
avio: avio_ prefix for url_set_interrupt_cb.
avio: AVIO_ prefixes for URL_ open flags.
proto: introduce listen option in tcp
doc: clarify configure features
proto: factor ff_network_wait_fd and use it on udp
Conflicts:
ffmpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Fix parser: mark av_parser_parse() for removal on next major bump
swscale: postpone sws_getContext removal until next major bump.
fate: add AAC LATM test
mmst: get rid of deprecated AVERRORs
lxfdec: use AVERROR(ENOMEM) instead of deprecated AVERROR_NOMEM.
Reemove remaining uses of deprecated AVERROR_NOTSUPP.
REIMPLEMENTED in 2 lines of code: lavf: if id3v2 tag is present and all else fails, guess by file extension
Conflicts:
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Support the URL_FLAG_NONBLOCK semantic and uniform the protocol.
The quick retry loop is already part of retry_transfer_wrapper.
The polling routine is common to the network protocols:
udp, tcp and, once merged, sctp.
Not yet complete, for demuxing AAC the AAC header must be generated
manually.
Possibly the decoder could accept the header as extradata to simplify
this.
* qatar/master:
psymodel: extend API to include PE and bit allocation.
avio: always compile dyn_buf functions
Remove unnecessary parameter from ff_thread_init() and fix behavior
Revert "aac_latm_dec: use aac context and aac m4ac"
configure: tell user if libva is enabled like the rest of external libs.
Add silence support for AV_SAMPLE_FMT_U8.
avio: make URL_PROTOCOL_FLAG_NESTED_SCHEME internal
avio: deprecate av_url_read_seek
avio: deprecate av_url_read_pause
ac3enc: NEON optimised extract_exponents
Conflicts:
libavcodec/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
CONFIG_VDPAU is the condition on which ff_vdpau_mpeg_picture_complete
is compiled in, so it is more appropriate, particularly since the
separate VDPAU decoder should be removed in the longer term.
thread_count passed to ff_thread_init() is only used to set AVCodecContext.
thread_count, and can be removed. Instead move it to the legacy implementation
of avcodec_thread_init().
This also fixes the problem that calling avcodec_thread_init() with pthreads
enabled did not set it since ff1efc524c.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
This is done by adding a setsar filter at the beginning of the
configured filterchain.
This implementation is more robust, since does not modify the
filterchain description (which was creating potential syntax errors),
but directly modifies the filterchain structure.
This also changes the ffmpeg -aspect behavior, as following filters in
the filterchain can change the DAR/SAR set by the inserted setsar
filter.
Signed-off-by: Stefano Sabatini <stefano.sabatini-lala@poste.it>
Currently, the parser is buggy and only processes the stream extradata
when the flag is set. This fixes it to actually inspect the frames.
Whitespce will be fixed in a separate change.
Signed-off-by: Philip Langdale <philipl@overt.org>
swscale doesnt ever actually do any runtime detection at all when
runtime cpu detection is enabled, it simply passes whatever is passed
to -sws_flags, which could be nothing at all making swscale default to
the C implementation.
* qatar/master: (22 commits)
ac3enc: move extract_exponents inner loop to ac3dsp
avio: deprecate url_get_filename().
avio: deprecate url_max_packet_size().
avio: make url_get_file_handle() internal.
avio: make url_filesize() internal.
avio: make url_close() internal.
avio: make url_seek() internal.
avio: cosmetics, move AVSEEK_SIZE/FORCE declarations together
avio: make url_write() internal.
avio: make url_read_complete() internal.
avio: make url_read() internal.
avio: make url_open() internal.
avio: make url_connect internal.
avio: make url_alloc internal.
applehttp: Merge two for loops
applehttp: Restructure the demuxer to use a custom AVIOContext
applehttp: Move finished and target_duration to the variant struct
aacenc: reduce the number of loop index variables
avio: deprecate url_open_protocol
avio: deprecate url_poll and URLPollEntry
...
Conflicts:
libavformat/applehttp.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The pulldown flags should be communicated to the client of the libavcodec library. Not doing so causes jerky playback with pulldown content. Note that this change requires the patch previously provided here: http://ffmpeg.org/pipermail/ffmpeg-devel/2011-April/110314.html
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The attached patch fixes the jerky playback of VC-1 content with pulldown. The pulldown flags were incorrectly set. They must be correct in order to display the frames with the correct timing as mentioned in the specifications: "SMPTE 421M: VC-1 Compressed Video Bitstream Format and Decoding Process". More precisely the following tables:
Table 20: Progressive P picture layer bitstream for Advanced Profile
Table 22: Progressive B picture layer bitstream for Advanced Profile
Table 23: Progressive Skipped picture layer bitstream for Advanced Profile
Table 82: Interlaced Frame I and BI picture layer bitstream for Advanced Profile
Table 83: Interlaced Frame P picture layer bitstream for Advanced Profile
Table 84: Interlaced Frame B picture layer bitstream for Advanced Profile
Table 85: Picture Layer bitstream for Field 1 of Interlace Field Picture for Advanced Profile
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This avoids issues where EOF at the end of the segment is given
the variant demuxer. Now the demuxers only see one single data
stream (as when using the applehttp protocol handler).
* qatar/master:
fate: fix partial run when no samples path is specified
ARM: NEON fixed-point forward MDCT
ARM: NEON fixed-point FFT
lavf: bump minor version and add an APIChanges entry for avio changes
avio: simplify url_open_dyn_buf_internal by using avio_alloc_context()
avio: make url_fdopen internal.
avio: make url_open_dyn_packet_buf internal.
avio: avio_ prefix for url_close_dyn_buf
avio: avio_ prefix for url_open_dyn_buf
avio: introduce an AVIOContext.seekable field
ac3enc: use generic fixed-point mdct
lavfi: add fade filter
Change yadif to not use out of picture lines.
lavc: deprecate AVCodecContext.antialias_algo
lavc: mark mb_qmin/mb_qmax for removal on next major bump.
Conflicts:
doc/filters.texi
libavcodec/ac3enc_fixed.h
libavcodec/ac3enc_float.h
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/vf_fade.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Currently bits_per_raw_sample is exposed as an AVCodecContext option.
The option is not very useful, because ffmpeg 1) overwrites it with
a value from the upstream codec, or 2) it resets the value whenever
the video is resampled.
This patch adds the -bits_per_raw sample option to FFmpeg, and
caches the value like is done for the -pix_fmt option.
Example usage:
ffmpeg v210.avi -pix_fmt rgb48 -bits_per_raw_sample 10 out%d.dpx
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This restores the behaviour of running only the builtin
regression tests when no path to external samples is given.
aa3805a inadvertently broke this by always setting SAMPLES
to something. Using := makes the ifdef test work as expected
when the FATE_SAMPLES environment variable is empty.
Signed-off-by: Mans Rullgard <mans@mansr.com>
It doesn't look fit to be a part of the public API.
Adding a temporary hack to ffserver to be able to use it, should be
cleaned up when somebody is up for it.
This makes the AC3 encoder use the shared fixed-point MDCT rather
than its own implementation. The checksum changes are due to
different rounding in the MDCT.
Signed-off-by: Mans Rullgard <mans@mansr.com>
commit c0ec9918b0
Author: Måns Rullgård <mans@mansr.com>
Date: Tue Aug 24 17:47:05 2010 +0000
Remove global mm_flags variable
Originally committed as revision 24909 to svn://svn.ffmpeg.org/ffmpeg/trunk
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Port fade filter from libavfilter soc repo, with minor fixes by
Stefano.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
As per issue2629, most 23.976fps matroska H.264 files are incorrectly
detected as 24fps, as the matroska timestamps usually have only
millisecond precision.
Fix that by doubling the amount of timestamps inspected for frame rate
for streams that have coarse time base. This also fixes 29.970 detection
in matroska.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Fixed-point MDCT with 32-bit unscaled output
lavc: deprecate rate_emu
lavc: mark hurry_up for removal on next major bump
parser: mark av_parser_parse() for removal on next major bump
lavc: add missing audioconvert includes
jvdec: don't use deprecated CODEC_TYPE_*/PKT_FLAG_KEY
Conflicts:
libavcodec/h264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Reason, is this was requested by the windows experts / seems dlltool causes alot of problems
This reverts commit ec10a9ab46.
Conflicts:
configure
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
ac3enc: ARM optimised ac3_compute_matissa_size
ac3: armv6 optimised bit_alloc_calc_bap
fate: simplify fft test rules
avio: document avio_alloc_context.
lavf: make compute_chapters_end less picky.
sierravmd: fix Indeo3 videos
FFT: simplify fft8()
fate: add fixed-point fft/mdct tests
Fixed-point support in fft-test
ape: check that number of seektable entries is equal to number of frames
Merged-by: Michael Niedermayer <michaelni@gmx.at>
TrueHD supports more channels than FFmpeg, so a valid sample
could set the channel layout to a value that represents less
channels than the sample actually consists of.
In particular, now it assumes that
a) chapters are chronologically ordered
b) chapters have the same timebases
c) duration of the stream is known
and asserts if any of these is not met.
Make it properly deal with harsher conditions.
fixes issue2320
* newdev/master:
mpegts: propagate avio EOF in read_packet()
configure: Initial support for --target-os=symbian
Fixed-point FFT and MDCT
Include dependencies for test programs
ac3enc: simplify sym_quant()
flvdec: read index stored in the 'keyframes' tag.
mov: Add support for zero-sized stsc runs.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Allows distinguishing between EOF and IO error in read_packet return code.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
These expressions are equivalent since levels is always odd, and
overflow is impossible due to the constraints set by the assert().
Signed-off-by: Mans Rullgard <mans@mansr.com>
* newdev/master:
rtsp: Use GET_PARAMETER for keep-alive for generic RTSP servers
mlp_parse.c: set AVCodecContext channel_layout
APIChanges: mark the place where 0.6 was branched.
avio: make get_checksum() internal.
avio: move ff_crc04C11DB7_update() from avio.h -> avio_internal.h
avio: make init_checksum() internal.
NOT MERGED Add MxPEG decoder
NOT MERGED Add support for picture_ptr field in MJpegDecodeContext
NOT MERGED Move MJPEG's input buffer preprocessing in separate public function
NOT MERGED Support reference picture defined by bitmask in MJPEG's SOS decoder
sndio bug fix
Merged-by: Michael Niedermayer <michaelni@gmx.at>
'keyframes' metatag is not part of the standard, it is just
convention to use such kind of metatag information for indexing.
Structure is following, it allows to have it inconsistent:
keyframes:
times (array):
time0 (num)
time1 (num)
time2 (num)
filepositions (array)
position0 (num)
position1 (num)
Signed-off-by: Anton Khirnov <anton@khirnov.net>
A zero sized stsc run doesn't make a lot of sense but the spec does not
prohibit them and MPlayer VLC demuxers support them.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This reverts commit cb48e245e6.
Reason being we like pressing "q" to quit ffmpeg.
Conflicts:
ffmpeg.c
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
According to the RFC, GET_PARAMETER should be used for
this, and according to a report from Tim Ouellette,
OPTIONS doesn't work for keeping the connection alive for some
servers. Also, live555 uses GET_PARAMETER for this purpose.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Heres a benchmark
VOFW 5120 VOFW 21504 Note
10438.8 10344.2 timex ffmpeg -y -i tulip2.mp4 tulip2.yuv
25611.9 24256.9 timex ffmpeg -y -itulip2.mp4 -s 1272x724 tulip2.yuv
24485.7 26006.6
28573.1 24291.3
24069.1 26995.9
25684.95 25387.675 average
This is on a Nehalem i7. Despite being a 500 frame, 720p video, the numbers
fluctuate, so I took an average of 4 runs.
Another reason I'd like this change is youtube accepts videos with higher
resolutions than 5120.
16384 is vp8's maximum resolution, so that should keep us amused for a
little longer.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Since the code already supports both little- and big-endian
audio for recording, do not fail just because the endianness is not
what we expect.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* newdev/master:
ac3enc: move compute_mantissa_size() to ac3dsp
ac3enc: move mant*_cnt and qmant*_ptr out of AC3EncodeContext
Remove support for stripping executables
ac3enc: NEON optimised float_to_fixed24
ac3: move ff_ac3_bit_alloc_calc_bap to ac3dsp
dfa: protect pointer range checks against overflows.
Duplicate: mimic: implement multithreading.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* ffmpeg-mt/master:
Update todo. More items appeared...
Fix mdec
Duplicate: id3v1: change filesize to int64_t.
Duplicate: id3v1: Seek back to old position after reading.
Conflicts:
libavcodec/mpegvideo.c
libavcodec/snow.c
libavformat/id3v1.c
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
These fields are only used in quantize_mantissas() and reset
on each call, no need to store them in the main context.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Stripping is generally best left to package management tools, and
since unstripped copies are kept in the build tree, any arguments
about saving space (no matter how insignificant) are void.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Since the code already supports both little- and big-endian
audio for recording, do not fail just because the endianness is not
what we expect.
It is possible that 8-bit recording would not have worked at all on
some systems without that change.
FFmpeg did not seek back to the original position, but to "0", making
reading a VBR tag impossible.
(issue 2645)
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* newdev/master:
ac3enc: avoid memcpy() of exponents and baps in EXP_REUSE case by using exponent reference blocks.
Chronomaster DFA decoder
DUPLICATE: framebuffer device demuxer
NOT MERGED: cosmetics: fix dashed line length after 070c5d0
http: header field names are case insensitive
Conflicts:
LICENSE
README
doc/indevs.texi
libavdevice/fbdev.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Make the mp wrapper accept the syntax mp=filter=params as alternative
to mp=filter:params. The alternative syntax is sligthly more readable
and should simplify copy&paste of MPlayer filter strings to the mp
filter.
* newdev/master:
ac3enc: Add codec-specific options for writing AC-3 metadata.
NOT MERGED: Remove arrozcru URL from documentation
sndio support for playback and record
Conflicts:
doc/faq.texi
doc/general.texi
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fixes issue2632 if interl=1 is used or the automatic interlace detection is enabled
and works. This has the advantage compared to the patch in issue2632 that it causes
no speed loss and it also works when scaling is used. The disadvantage is that
interlacing autodetection does not yet work very well it seems.
This is the same method mplayer uses
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* newdev/master:
dsputil: allow to skip drawing of top/bottom edges.
Split fate-psx-str-v3 into a video-only and audio-only test.
Conflicts:
libavcodec/dsputil.c
libavcodec/mpegvideo.c
libavcodec/snow.c
libavcodec/x86/dsputil_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
As previously discussed, the CrystalHD hardware treats some PAFF
clips different from others; even when input fields are always in
separate packets, the hardware might return a single fieldpair for
one clip and individual fields for another.
Given the bogus flags set by the hardware, it is impossible to
distinguish these two cases without knowing about the current
picture and the next one. The hardware can usually provide the
picture number of the next picture and when that is available,
we can detect the two cases.
When it is not available, we have to guess - and find out later
if we were right or wrong.
With this change, clips will play correctly unless they are PAFF
where individual fields are returned *and* no next picture number
is available. Generally speaking, the incorrect cases arise in
the first couple of seconds of a clip as the delay calibration takes
place. Once that's set, things work fine.
As previously discussed, the CrystalHD hardware returns exceptionally
useless information about interlaced h.264 content - to the extent
that it's not possible to distinguish MBAFF and PAFF content until
it's too late.
This change introduces use of the h264_parser to help bridge the
gap; it can indicate if the input data is PAFF fields or not.
With this clarity, some of heuristics can be removed from the code,
making this less convoluted.
Finally, I found an MBAFF clip that acts like non h.264 content so
I had to make allowances for that.
Note that I still cannot distinguish between two forms of PAFF,
where the hardware either returns individual fields or a field-pair.
It's not clear that there's even a spec relevant difference between
the two forms, as opposed to hardware ideosyncracies.
Currently, only S16 quad, 5.1 and 7.1 are implemented.
Implementing support for other formats/layouts and capture should be
straightforward.
7.1 support by Carl Eugen Hoyos.
Provide a non-NULL AVFormatParameters structure to
av_open_input_file() in open_input_file().
This is required because otherwise av_open_input_file() will allocate
and use a new format context, discarding the options set in the
provided format context.
The function was only used in opt_sample_fmt() for listing the sample
formats. Move list_fmts() functionality directly into
opt_sample_fmt().
Als fix the warning:
ffmpeg.c: In function ‘opt_audio_sample_fmt’:
ffmpeg.c:2877: warning: passing argument 1 of ‘list_fmts’ from incompatible pointer type
cmdutils.h:163: note: expected ‘void (*)(char *, int, int)’ but argument is of type ‘char * (*)(char *, int, enum AVSampleFormat)’
* newdev/master:
mov: set audio service type for AC-3 from bitstream mode in the 'dac3' atom.
Get audio_service_type for AC-3 based on bitstream mode in the AC-3 parser and decoder, and vice-versa for the AC-3 encoder.
Use audio_service_type to set stream disposition.
Add APIchanges entry for audio_service_type.
Add audio_service_type field to AVCodecContext for encoding and reporting of the service type in the audio bitstream.
configure: in check_ld, place new -l flags before existing ones
support @heading, @subheading, @subsubheading, and @subsubsection in texi2pod.pl
doc: update build system documentation
aacenc: indentation
aacenc: fix the side calculation in search_for_ms
vp8.c: rename EDGE_* to VP8_EDGE_*.
Conflicts:
doc/APIchanges
libavcodec/avcodec.h
libavcodec/version.h
libavcodec/vp8.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
On Blu-ray colors are stored in the order YCrCb (and not YCbCr) as mentioned in the specifications:
see System Description Blu-ray Disc Read-Only Format, 9.14.4.2.2.1 Palette Definition Segment
When decoding a Blu-ray subtitle, the colors were incorrectly set.
On DVD and HD-DVD colors are stored in the order YCrCb (and not YCbCr) as mentioned in the specifications:
see DVD Specifications for Read-Only Disc / Part 3, 4.3 Program Chain Information (7) PGC_SP_PLT
see DVD Specifications for High Definition Disc, 5.2 Navigation for Standard Content (11) PGC_SDSP_PLT
see DVD Specifications for High Definition Disc, 5.2 Navigation for Standard Content (12) PGC_HDSP_PLT
see DVD Specifications for High Definition Disc, 5.5 Presentation Data (4) SET_COLOR2
When decoding a DVD or HD-DVD subtitle, the colors were incorrectly set.
Split normal build system documentation and the fate one.
Document THREADS make variable and provide an introduction to fate.
Document DESTDIR make variable and add some generic targets.
* newdev/master:
matroskadec: set default duration for simple block
When building for MinGW32 disable strict ANSI compliancy.
ARM: fix ff_apply_window_int16_neon() prototype
configure: check for --as-needed support early
ARM: NEON optimised apply_window_int16()
ac3enc: NEON optimised shift functions
ac3enc: NEON optimised ac3_max_msb_abs_int16 and ac3_exponent_min
mpeg12.c: fix slice threading for mpeg2 field picture mode.
ffmetadec.c: fix compiler warnings.
configure: Don't explicitly disable ffplay or in/outdevices on dos
configure: Remove the explicit disabling of ffserver
configure: Add fork as a dependency to ffserver
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When a normal Block is parsed, duration is initialized to
AV_NOPTS_VALUE. If it is not changed, then the track's default
duration is used. But for SimpleBlock, duration is initialized to
0 instead of AV_NOPTS_VALUE. This is due to the difference in how
EBML_NEST vs EBML_PASS are processed. Setting duration to 0 leads
eventually to wrongly estimate the frame duration in util.c
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
When -std=c99 is used, GCC defines __STRICT_ANSI__ to hide non-ANSI
interfaces; on MinGW32, when this macro is defined, some functions that are
part of POSIX but not of Windows APIs are disabled, including strcasecmp().
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This moves the check for --as-needed linker support before all
library tests, ensuring consistent behaviour between the checks
and the actual link step.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* newdev/master:
avio: make udp_set_remote_url/get_local_port internal.
asfdec: also subtract preroll when reading simple index object
matroskaenc: remove a variable that's unused after bc17bd9.
avio: cosmetics - nicer vertical alignment.
Remove unnecessary icc version checks
Disable 'attribute "foo" ignored' warnings from icc
rtsp: Don't use a locale dependent format string
Add xd55 codec tag for XDCAM HD422 720p25 CBR files.
configure: get libavcodec version from new version.h header
lavc: move the version macros to a new installed header.
matroskaenc: simplify get_aac_sample_rates by using ff_mpeg4audio_get_config
Do not use format string "%0.3f" for RTSP Range field.
Add apply_window_int16() to DSPContext with x86-optimized versions and use it in the ac3_fixed encoder.
Document usage of import libraries created by dlltool
configure: Set the correct lib target for arm/wince dlltool
fate: simplify regression-funcs.sh
fate: add support for multithread testing
Conflicts:
libavformat/rtspdec.c
libavutil/attributes.h
libavutil/internal.h
libavutil/mem.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
With unknown attribute warnings disabled, these checks are no
longer needed. Removing them improves readability while having
no effect on generated code.
Signed-off-by: Mans Rullgard <mans@mansr.com>
ICC lies about the version of gcc it emulates, which results
in unsupported attributes sometimes being used. The warning
is an annoyance and should be disabled.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* newdev/master: (33 commits)
Fix an infinite loop when RoQ encoded generated a frame with a size greater than the maximum valid size.
Add kbdwin.o to AC3 decoder
Detect byte-swapped AC-3 and support decoding it directly.
cosmetics: indentation
Always copy input data for AC3 decoder.
ac3enc: make sym_quant() branch-free
cosmetics: indentation
Add a CPU flag for the Atom processor.
id3v2: skip broken tags with invalid size
id3v2: don't explicitly skip padding
Make sure kbhit() is in conio.h
fate: update wmv8-drm reference
vc1: make P-frame deblock filter bit-exact.
configure: Add the -D parameter to the dlltool command
amr: Set the AVFMT_GENERIC_INDEX flag
amr: Set the pkt->pos field properly to the start of the packet
amr: Set the codec->bit_rate field based on the last packet
rtsp: Specify unicast for TCP interleaved streams, too
Set the correct target for mingw64 dlltool
applehttp: Change the variable for stream position in seconds into int64_t
...
Conflicts:
ffmpeg.c
ffplay.c
libavcodec/ac3dec.c
libavformat/avio.h
libavformat/id3v2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows the AC-3 decoder to be used directly with RealMedia
decoders that unlike the libavformat one do not byte-swap automatically.
Since the new code is only used in case we would fail directly otherwise
there should be no risk for regressions.
Depending on error_recognition is not correct, low values do
certainly not mean it is ok to crash.
Based on a patch by Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The Atom has SSSE3 support, which is useful in many cases, but sometimes the
SSSE3 version is slower than the SSE2 equivalent on the Atom, but is generally
faster on other processors supporting SSSE3. This flag allows for selectively
disabling certain SSSE3 functions on the Atom.
Conio.h is a non-standard header and may not have kbhit()
prototyped. This fixes compile on OS/2 where the EMX version (we're
using a fork) of conio.h only has getch() and getche().
Signed-off-by: Mans Rullgard <mans@mansr.com>
This allows libavformat to guess an estimated duration for
amr files.
For streams with varying bit rates (or with silence descriptors
or "no frame" blocks) the guess is, of course, inaccurate.
According to the RFC, the default is multicast if nothing is
specified, which doesn't make sense for TCP.
According to a bug report, some Axis camera models give a
"400 Bad Request" error if this is omitted.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
A similar variable for the total stream duration was changed to
int64_t in b79c3df088, due to overflows in some odd
streams.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This fixes crashes on exit when plaing some RealRTSP streams
(among other), e.g.
rtsp://dl.lib.brown.edu:554/areserves/1093545294660883.mp3.
These crashes have been present since 3e68b3ba7b.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
It's an evil hack that assumes an AVIOContext is always based on top of
an URLContext.
It's also not used anywhere.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Remove usage of tcgetattr and tcsetattr to modify terminal
parameters, and rely on ctrl-c to stop instead of pressing 'q'.
On systems with conio.h, keep the old behavior.
Changing the terminal settings causes problems if multiple instances
are running asynchronously on the same terminal, such as during a
parallel FATE run, or if the process crashes before restoring the
terminal. In both cases, the terminal state is messed up requiring
a manual reset.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Change "file:stream[:syncfile:syncstream]" to
"file.stream[:syncfile.syncstream]", the latter syntax is more
consistent with the stream identifier form printed by
av_dump_format(), and more readable. Both syntaxes are currently
accepted by the opt_map() code.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
These windows do not really belong in fft/mdct files and were
easily confused with the similarly named tables used by rdft.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This function is not tightly coupled to mdct, and it's in the way
of making a fixed-point mdct implementation.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This removes the rather pointless wrappers (one not even inline)
for calling the fft_calc and related function pointers.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This makes existing pkg-config uses as well as the libsdl checks
use the new pkg-config helper functions, which should be more
robust against broken systems.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This adds helper functions for checking packages with pkg-config
and managing the associated flags.
Note that pkg-config use is still discouraged due to widespread
poor practices resulting in broken flags in many situations. A
few badly designed packages require flags only obtainable using
pkg-config, and these functions are intended for those cases.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 3f8040db3e)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This makes it possible to pass a space-separated list of functions
to check_func_headers and check_lib2. If any function is missing,
none are enabled as available, so this should only be used for
all-or-nothing sets, i.e. groups in which none will be used if any
one is missing.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit edaf1ae276)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Not Pulled:
commit 42cfb3835b
Author: Mans Rullgard <mans@mansr.com>
Date: Mon Feb 28 18:06:58 2011 +0000
Remove Sonic experimental audio codec
commit 2912e87a6c
Author: Mans Rullgard <mans@mansr.com>
Date: Fri Mar 18 17:35:10 2011 +0000
Replace FFmpeg with Libav in licence headers
Fix warning:
ffplay.c: At top level:
ffplay.c:3027: warning: missing braces around initializer
ffplay.c:3027: warning: (near initialization for ‘options[53].u’)
Since initially committed in 2004, this codec has only been touched
for maintenanance. Functionally, it contains no novel ideas and
its intended audience is better served by existing mature codecs.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This allows the AC-3 decoder to be used directly with RealMedia
decoders that unlike the libavformat one do not byte-swap automatically.
Since the new code is only used in case we would fail directly otherwise
there should be no risk for regressions.
The "buf" pointer needs to be overwritten since otherwise the CRC check fails.
draw_slice() checks that the fade factor is < 65536 and only
calculates the fade if so. But the fade factor is clipped in
end_frame() by av_clip_uint16() to 65535, so the fade is calculated
for every frame. This patch alters the check so that it compares with
< 65535 (UINT16_MAX).
draw_slice() checks that planes 0 and 1 of AVFilterBufferRef's data
are not NULL before manipulating planes 1 and 2. This patch makes the
check against planes 1 and 2. More senseful and possibly more robust.
This fixes ugliness when configure outputs its (empty) results on HPUX
(maybe some other UNIXes too):
Enabled indevs:
pr: -- empty file
Signed-off-by: Mans Rullgard <mans@mansr.com>
This updates the seek test reference to match de11ee9. Before this
change, most of the seeks requested positions before the supposed
start of the file (the preroll time), resulting in the first packet
being returned. With the preroll subtracted, some of these seeks
will land within the file and some beyond the end, thus returning
a different set of packets.
Signed-off-by: Mans Rullgard <mans@mansr.com>
right now swab is defined in libavfilter/libmpcodecs/mpbswap.h
this breaks mingw32 cross compiles. since it its not used, it should be
removed. alternatively swab has to be added to HAVE_LIST in configure.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Add an APIchanges entry for the av_pkt_dump2 and av_pkt_dump_log2
functions, and a changelog entry for the apple http live streaming
protocol handler.
Since neither of them got a minor bump at commit time, but were
applied before the jv demuxer, they all can be considered added
in this minor version.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Turns a comment into an av_dlog() instruction, also add a commented
issues.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 77f21ce464)
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
3GPP:
Remove ffac from and move min_snr out of AacPsyBand.
Rearrange AacPsyCoeffs to make it easier to implement energy spreading.
Rename the band[] array to bands[]
Copy energies and thresholds at the end of analysis.
LAME:
Use a loop instead of an if chain in LAME windowing.
There are several places where a buffer is byte-swapped in 16-bit units.
This allows them to share code which can be optimised for various
architectures.
Signed-off-by: Mans Rullgard <mans@mansr.com>
If the function is not inlined, an immmediate cannot be used for the
shift parameter, so the %cl register must be used instead in that case.
This fixes compilation for x86-32 using gcc with --disable-optimizations.
This fixes unexpected name collisions that were occurring with variables
declared within the macros.
It also fixes the fate-acodec-ac3_fixed regression test on x86-32.
This reverts commit cc4d3dd3e2.
revert at authors request due to better impementation being available
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Functions interrupted by url_interrupt_cb should not be restarted.
Therefore using AVERROR(EINTR) was wrong, as it did not allow to distinguish
when the underlying system call was interrupted and actually needed to be
restarted.
This fixes roundup issues 2657 and 2659 (ffplay not exiting for streamed
content).
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This is different from AVERROR(EINTR) because calls that fail with EINTR
should usually be restarted.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This increases the accuracy of coefficients, leading to improved quality.
Rescaling of the coefficients to full 25-bit accuracy is done rather than
offsetting the exponent values. This requires coefficient scaling to be done
before determining the rematrixing strategy. Also, the rematrixing strategy
calculation must use 64-bit math to prevent overflow due to the higher
precision coefficients.
In the event of overflow, the JV_PADDING state will avio_skip over
any overflow bytes (using JVFrame.total_size).
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
In fbdev_read_packet(): nanosleep returns -1 in case of errors,
the EINTR check has to be done on errno.
Spotted by Nicolas.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This makes existing pkg-config uses as well as the libsdl checks
use the new pkg-config helper functions, which should be more
robust against broken systems.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This adds helper functions for checking packages with pkg-config
and managing the associated flags.
Note that pkg-config use is still discouraged due to widespread
poor practices resulting in broken flags in many situations. A
few badly designed packages require flags only obtainable using
pkg-config, and these functions are intended for those cases.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This makes it possible to pass a space-separated list of functions
to check_func_headers and check_lib2. If any function is missing,
none are enabled as available, so this should only be used for
all-or-nothing sets, i.e. groups in which none will be used if any
one is missing.
Signed-off-by: Mans Rullgard <mans@mansr.com>
In the main loop, stream_number is incremented after checking the stream type,
so the search usually will not find the wanted stream.
This patch eliminates the useless stream_number variable and introduces a new
one, called real_stream_index to store the real stream index of the current
stream, no matter if we are looping through all the streams or only the streams
of a program.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The Broadcom CrystalHD decoder chips provide hardware video
decoding for a number of video formats. It does so using a
memory:memory interface where a compressed bitstream is fed
in and decompressed pictures are copied out. As such, it works
independent of any graphics hardware in the system.
Features supported in this initial version:
* Support for Linux (using current drivers/library from git.wilsonet.com)
* Support for 70015 hardware
* Formats: MPEG2, MPEG4 Part 2, H.264, VC1 and DivX 3.11 (untested)
* Progressive content
* Non-H.264 Interlaced content
* H.264 MBAFF content
Features missing in this initial version:
* Support for OSX (might work - untested)
* Support for Windows
* Support for 70012 hardware
* H.264 PAFF content
Signed-off-by: Philip Langdale <philipl@overt.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This reverts the removal of scoefs from AACEncContext.
It resulted in scoefs being a NULL pointer when
search_for_quantizers() is called.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This makes channel coupling more accurate, increasing quality for stereo
content. It also simplifies exponent extraction and mantissa quantization
by no longer needing to apply an offset to the exponents.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This reverts the removal of scoefs from AACEncContext.
It resulted in scoefs being a NULL pointer when
search_for_quantizers() is called.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is used at least on some older DVB broadcasts for dual-mono audio
tracks.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 6a7e074eb9)
when writing and pressing q during encoding. Instead, check url_interrupt_cb
at the end.
Note that when a protocol is interrupted by url_interrupt_cb, some data may
be silently discarded: the protocol context is not suitable for anything
anymore.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 688c22e033)
AVIOContext.max_packet_size should be used directly instead.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit e8bb2e2439)
When decoding latm config, use the corresponding aac context and its
m4ac instead of using NULL and a local variable. This fixes decoding of
audio in MPEG TS from SBTVD (the Brazillian Digital TV Sytem), when
there is no extradata. This is the case when using the decoder with
gst-ffmpeg and a GStreamer mpegts demuxer.
Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 36864ac354)
It's not used anywhere and doesn't look ver useful to be public.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 655e45e7df)
Should an AVC-1 in MP4 stream not contain SPS or PPS NAL units,
this BSF is then unable to allocate an output buffer for the
modified stream. Warn that the resulting stream may be unplayable.
Fix roundup issue #2386.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 02dd3666c2)
If this flag is set, the protocol can handle URLs where the
scheme is a nested scheme such as applehttp+file: - the protocol
can handle any URL where the first segment of the nested scheme
belongs to this protocol.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
(cherry picked from commit 8f73c06077)
This fixes compilation with preprocessors which do not add whitespace
around replaced tokens, resulting in invalid expressions like 1--1.
Signed-off-by: Mans Rullgard <mans@mansr.com>
when writing and pressing q during encoding. Instead, check url_interrupt_cb
at the end.
Note that when a protocol is interrupted by url_interrupt_cb, some data may
be silently discarded: the protocol context is not suitable for anything
anymore.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
When decoding latm config, use the corresponding aac context and its
m4ac instead of using NULL and a local variable. This fixes decoding of
audio in MPEG TS from SBTVD (the Brazillian Digital TV Sytem), when
there is no extradata. This is the case when using the decoder with
gst-ffmpeg and a GStreamer mpegts demuxer.
Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This is a substitute for the url_fskip function that was deprecated by
commit 0300db8ad7. avio_fskip is provided to
improve demuxer code readability. It distinguishes the act of skipping over
unknown or irrelevant bytes from the standard avio_seek operation.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Should an AVC-1 in MP4 stream not contain SPS or PPS NAL units,
this BSF is then unable to allocate an output buffer for the
modified stream. Warn that the resulting stream may be unplayable.
Fix roundup issue #2386.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
If this flag is set, the protocol can handle URLs where the
scheme is a nested scheme such as applehttp+file: - the protocol
can handle any URL where the first segment of the nested scheme
belongs to this protocol.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This fixes compilation with preprocessors which do not add whitespace
around replaced tokens, resulting in invalid expressions like 1--1.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Currently it is always 4, but this change will allow it to be adjusted when
bandwidth-related features are added such as channel coupling, enhanced
channel coupling, and spectral extension.
(cherry picked from commit 53e35fd340)
Currently it is always 4, but this change will allow it to be adjusted when
bandwidth-related features are added such as channel coupling, enhanced
channel coupling, and spectral extension.
This moves setting the thread count to a minimum of 1 to
frame_thread_init(), allowing a value of zero to propagate
through to the codec if frame threading is not used. This
makes auto-threads work in libx264.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit ff1efc524c)
This makes dumped packet timestamps proper for streams with
timebases other than AV_TIME_BASE.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
(cherry picked from commit 5e33e7bdac)
This removes a fixme issue, by allowing the av_pkt_dump functions
to use the correct time base.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
(cherry picked from commit 863c471638)
In most cases, s->buf_ptr will be equal to s->buf_end when
fill_buffer is called, but this may not always be the case, if
we're seeking forward by reading (permitted by the short seek
threshold).
If fill_buffer is writing to s->buf_ptr instead of s->buf_end (when
they aren't equal and s->buf_ptr is ahead of s->buffer), the data
between s->buf_ptr and s->buf_end is overwritten, leading to
inconsistent buffer content. This could return incorrect data if
later seeking back into the area before the current s->buf_ptr.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
(cherry picked from commit e360ada2d1)
For intra codecs, ff_thread_finish_setup() is called before decoding starts
automatically. However, get_buffer can only be used before it's called, so
adding this requirement broke frame threading for them. Fixed by moving the
call until after get_buffer is finished.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit ad9791e12b)
This moves setting the thread count to a minimum of 1 to
frame_thread_init(), allowing a value of zero to propagate
through to the codec if frame threading is not used. This
makes auto-threads work in libx264.
Signed-off-by: Mans Rullgard <mans@mansr.com>
In most cases, s->buf_ptr will be equal to s->buf_end when
fill_buffer is called, but this may not always be the case, if
we're seeking forward by reading (permitted by the short seek
threshold).
If fill_buffer is writing to s->buf_ptr instead of s->buf_end (when
they aren't equal and s->buf_ptr is ahead of s->buffer), the data
between s->buf_ptr and s->buf_end is overwritten, leading to
inconsistent buffer content. This could return incorrect data if
later seeking back into the area before the current s->buf_ptr.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
For intra codecs, ff_thread_finish_setup() is called before decoding starts
automatically. However, get_buffer can only be used before it's called, so
adding this requirement broke frame threading for them. Fixed by moving the
call until after get_buffer is finished.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Although not documented, clang does support the --sysroot flag, and it
does the right thing. Use this flag intead of -isysroot which only
applies to header file searches, not the linker.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 00ba041cb3)
Although not documented, clang does support the --sysroot flag, and it
does the right thing. Use this flag intead of -isysroot which only
applies to header file searches, not the linker.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The assembler emits literal pools too far from the load instructions,
so we must do it explicitly at a suitable location.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 8b454c352f)
Use the FATE_SAMPLES environment variable if samples location
is not set with the --samples configure option or on the make
command line.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit aa3805a486)
The assembler emits literal pools too far from the load instructions,
so we must do it explicitly at a suitable location.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Use the FATE_SAMPLES environment variable if samples location
is not set with the --samples configure option or on the make
command line.
Signed-off-by: Mans Rullgard <mans@mansr.com>
decode_init sets bands[0] == 2, so this loop always sets the band table
index (k) to zero.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit a304def1dc)
This causes lxlite to use stdout instead of vioXXX
functions. This improves fate and build logs readability.
Affects OS/2 only.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit cc4e9d2a24)
This avoids the core substream extensions scan when the EXT_AUDIO_ID
field indicates no extensions or only unsupported extensions. The scan
is done only if the value of EXT_AUDIO_ID is unknown or indicates a
present XCh extension which we can decode.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 7e06e0ede3)
There is no need to expand to 16-bits. Just use memcpy() to copy the raw data.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 1108f8998c)
This also adds output buffer size checks for AUDIO and SILENCE block types.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 1574eff3d2)
The size should depend on the output sample size, not the internal bit depth.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit a58bcb40b1)
This causes lxlite to use stdout instead of vioXXX
functions. This improves fate and build logs readability.
Affects OS/2 only.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This avoids the core substream extensions scan when the EXT_AUDIO_ID
field indicates no extensions or only unsupported extensions. The scan
is done only if the value of EXT_AUDIO_ID is unknown or indicates a
present XCh extension which we can decode.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This allows the values to be used without changing C code and is closer to how
the other DEBUG flags work.
If this causes a problem for any user of this flag, please tell me and
ill split the flag in 2.
Map EAGAIN and EINTR from ff_neterrno to the normal AVERROR()
error codes. Provide fallback definitions of other errno.h network
errors, mapping them to the corresponding winsock errors.
This eases catching these error codes in common code, without having
to distinguish between FF_NETERRNO(EAGAIN) and AVERROR(EAGAIN).
This fixes roundup issue 2614, unbreaking blocking network IO on
windows.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 28c4741a66)
Also change the comments a bit since the FOURCCs aren't specific to Flip4Mac
and different ones are used for 720 versus 1080 lines.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 8f935b9271)
Map EAGAIN and EINTR from ff_neterrno to the normal AVERROR()
error codes. Provide fallback definitions of other errno.h network
errors, mapping them to the corresponding winsock errors.
This eases catching these error codes in common code, without having
to distinguish between FF_NETERRNO(EAGAIN) and AVERROR(EAGAIN).
This fixes roundup issue 2614, unbreaking blocking network IO on
windows.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Also change the comments a bit since the FOURCCs aren't specific to Flip4Mac
and different ones are used for 720 versus 1080 lines.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This adds the AAC codec to the list of audio codecs that results
in a PES stream_id of 0xc0 (audio stream).
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 6c065f0acd)
The 4-tap filters should only access one row/column before the
reference block.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit e0e46cae37)
GCC 4.3 and later are more particular about signedness matching
in vector operations. The operations under if(rangered) were
missing assignments and thus had no effect.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 381efba0ec)
In the name of consistency:
put_byte -> avio_w8
put_<type> -> avio_w<type>
put_buffer -> avio_write
put_nbyte will be made private
put_tag will be merged with avio_put_str
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 77eb5504d3)
In the name of consistency:
get_byte -> avio_r8
get_<type> -> avio_r<type>
get_buffer -> avio_read
get_partial_buffer will be made private later
get_strz is left out becase I want to change it later to return
something useful.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit b7effd4e83)
Merging these functions allows merging some loops, which makes the
results (particularly after SIMD optimizations) much faster.
(cherry picked from commit f8bed30d8b)
Advantage is that it allows us to combine several loops into a single
one, and these can eventually be merged into the IDCT itself. Also, it
allows us to remove vc1_put_block(), and makes CODEC_FLAG_GRAY faster.
(cherry picked from commit bbfd2e7ab4)
This ensures the tools are rebuilt when necessary. Specifically,
lavfi-showfiltfmts was sometimes not rebuilt causing spurious test
failures.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 08df7f8666)
This allows the CPU detection to work with assemblers not supporting
the xgetbv mnemonic. These include clang and some BSD versions.
All AVX code will be written for yasm, where the main assembler
is not involved.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit ef66953875)
the following additions:
* support to anti-aliased glyph rendering
* support to UTF-8 text and Unicode chars rendering
* support for RGB packed formats
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This adds the AAC codec to the list of audio codecs that results
in a PES stream_id of 0xc0 (audio stream).
Signed-off-by: Mans Rullgard <mans@mansr.com>
GCC 4.3 and later are more particular about signedness matching
in vector operations. The operations under if(rangered) were
missing assignments and thus had no effect.
Signed-off-by: Mans Rullgard <mans@mansr.com>
In the name of consistency:
put_byte -> avio_w8
put_<type> -> avio_w<type>
put_buffer -> avio_write
put_nbyte will be made private
put_tag will be merged with avio_put_str
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
In the name of consistency:
get_byte -> avio_r8
get_<type> -> avio_r<type>
get_buffer -> avio_read
get_partial_buffer will be made private later
get_strz is left out becase I want to change it later to return
something useful.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Advantage is that it allows us to combine several loops into a single
one, and these can eventually be merged into the IDCT itself. Also, it
allows us to remove vc1_put_block(), and makes CODEC_FLAG_GRAY faster.
Add drawutils.h and drawutils.c, and use them in the pad filter.
The new functions are going to be shared by other filters.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Add drawutils.h and drawutils.c, and use them in the pad filter.
The new functions are going to be shared by other filters.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Update libavformat/version.h and doc/APIChanges after renaming
init_put_byte() and ByteIOContext to ffio_init_context() (private)
and AVIOContext, (public), and deprecating the originals.
(cherry picked from commit d2bbf82e65)
init_put_byte should never be used outside of lavf, since
sizeof(AVIOContext) isn't part of public ABI.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit e731b8d872)
Advanced profile never uses "range reduction", so vc1_put_block() quite
literally just calls put_pixels_clamped() from vc1_decode_i_blocks_adv().
By inlining the function, we can prevent calling IDCT8x8 if
CODEC_FLAG_GRAY is set, and we don't have to scale the coeffs in the
[0,256] range, but can instead use put_signed_pixels_clamped().
(cherry picked from commit 70aa916e46)
This adds configure and runtime checks for AVX support on x86 CPUs.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 87f1355f9b)
Allows playback of nonprimary audio streams in multiple bitrate sources,
such as mmsh://wmscr1.dr.dk/e02ch03m
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 74d6871d62)
With negative stride, the start of the edge_emu buffer should be pointing to
the last line, not the end of the buffer.
With positive stride, pointing to the end of the buffer was completely wrong.
(cherry picked from commit a89f4ca005)
This ensures the tools are rebuilt when necessary. Specifically,
lavfi-showfiltfmts was sometimes not rebuilt causing spurious test
failures.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This allows the CPU detection to work with assemblers not supporting
the xgetbv mnemonic. These include clang and some BSD versions.
All AVX code will be written for yasm, where the main assembler
is not involved.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Update libavformat/version.h and doc/APIChanges after renaming
init_put_byte() and ByteIOContext to ffio_init_context() (private)
and AVIOContext, (public), and deprecating the originals.
init_put_byte should never be used outside of lavf, since
sizeof(AVIOContext) isn't part of public ABI.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Advanced profile never uses "range reduction", so vc1_put_block() quite
literally just calls put_pixels_clamped() from vc1_decode_i_blocks_adv().
By inlining the function, we can prevent calling IDCT8x8 if
CODEC_FLAG_GRAY is set, and we don't have to scale the coeffs in the
[0,256] range, but can instead use put_signed_pixels_clamped().
Allows playback of nonprimary audio streams in multiple bitrate sources,
such as mmsh://wmscr1.dr.dk/e02ch03m
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
With negative stride, the start of the edge_emu buffer should be pointing to
the last line, not the end of the buffer.
With positive stride, pointing to the end of the buffer was completely wrong.
On some versions of gcc, these weren't always getting inlined due to hitting
the inline cap limit in some files. This is generally bad, as most of these
functions are smaller inlined than not.
(cherry picked from commit eb3755a5aa)
If the client sends PLAY/PAUSE requests with the same url as
specified in Content-Base, these requests may have urls with
trailing slashes.
(cherry picked from commit c2ca851b23)
The rematrixing strategy reuse flags are not reset between frames, so they
need to be initialized for all blocks, not just block 0.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 5b54d4b376)
The new av_parse_time() is created in libavutil/parseutils.h, all the
internal functions used by parse_date are moved to
libavutil/parseutils.c and made static.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit f6c7375a17)
If udp_read_packet returns 0, rtsp_st isn't set and we shouldn't
treat it as a successfully received packet (which is counted and
possibly triggers a RTCP receiver report).
This fixes issue 2612.
(cherry picked from commit 2c35a6bde9)
Hi.
It seems that ffserver sets sample_aspect_ratio to an invalid value and lavf
rejects it.
I am not sure what I am doing here, but the attached patch actually solves
something: using the following config:
CustomLog -
NoDaemon
RTSPPort 5454
<Stream test1-rtsp.mpg>
Format rtp
File "/tmp/test1-rtsp.mpg"
</Stream>
it allows a somewhat old ffplay (unaffected by the content-base issue I
spoke of in another thread) to play the stream.
Without it, ffserver logs this and closes the stream:
Wed Feb 16 14:52:14 2011 [rtp @ 0x1399de0]Aspect ratio mismatch between encoder and muxer layer
Regards,
--
Nicolas George
From 1b89c3c2164335060e87567b27deb0d354e0a814 Mon Sep 17 00:00:00 2001
From: Nicolas George <nicolas.george@normalesup.org>
Date: Wed, 16 Feb 2011 14:44:31 +0100
Subject: [PATCH] ffserver: set the sample aspect ratio.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
(cherry picked from commit 6741f7c9be)
On some versions of gcc, these weren't always getting inlined due to hitting
the inline cap limit in some files. This is generally bad, as most of these
functions are smaller inlined than not.
The rematrixing strategy reuse flags are not reset between frames, so they
need to be initialized for all blocks, not just block 0.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The new av_parse_time() is created in libavutil/parseutils.h, all the
internal functions used by parse_date are moved to
libavutil/parseutils.c and made static.
Signed-off-by: Mans Rullgard <mans@mansr.com>
If udp_read_packet returns 0, rtsp_st isn't set and we shouldn't
treat it as a successfully received packet (which is counted and
possibly triggers a RTCP receiver report).
This fixes issue 2612.
Hi.
It seems that ffserver sets sample_aspect_ratio to an invalid value and lavf
rejects it.
I am not sure what I am doing here, but the attached patch actually solves
something: using the following config:
CustomLog -
NoDaemon
RTSPPort 5454
<Stream test1-rtsp.mpg>
Format rtp
File "/tmp/test1-rtsp.mpg"
</Stream>
it allows a somewhat old ffplay (unaffected by the content-base issue I
spoke of in another thread) to play the stream.
Without it, ffserver logs this and closes the stream:
Wed Feb 16 14:52:14 2011 [rtp @ 0x1399de0]Aspect ratio mismatch between encoder and muxer layer
Regards,
--
Nicolas George
From 1b89c3c2164335060e87567b27deb0d354e0a814 Mon Sep 17 00:00:00 2001
From: Nicolas George <nicolas.george@normalesup.org>
Date: Wed, 16 Feb 2011 14:44:31 +0100
Subject: [PATCH] ffserver: set the sample aspect ratio.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
The current implementation has a bug, it is returning the stream index
in the found program, and not the stream index in the list of all
streams. The attached patch fixes this issue.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 22ec6b738f)
This is to match the value in every (E-)AC-3 file from commercial sources.
It has a negligible effect on audio quality.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 50d7140441)
Our poll implementation does not iterate over the pollfd array properly
while setting the revents.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
(cherry picked from commit 9ac2085dbf)
The current implementation has a bug, it is returning the stream index
in the found program, and not the stream index in the list of all
streams. The attached patch fixes this issue.
Signed-off-by: Mans Rullgard <mans@mansr.com>
VBV delay is useful for T-STD compliance in some TS muxers. It is
certainly possible to retrieve it by parsing the output of FFmpeg, but
getting it from the context makes it simpler and less error-prone.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is to match the value in every (E-)AC-3 file from commercial sources.
It has a negligible effect on audio quality.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Perform validity check on AVFormatContext.channels instead of
uninitialised field.
This fixes issue 2001.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 9806fbd535)
It is pretty hopeless that other considerable projects will adopt
libavutil alone in other projects. Projects that need small footprint
are better off with more specialized libraries such as gnulib or rather
just copy the necessary parts that they need. With this in mind, nobody
is helped by having libavutil and libavcore split. In order to ease
maintenance inside and around FFmpeg and to reduce confusion where to
put common code, avcore's functionality is merged (back) to avutil.
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
VBV delay is useful for T-STD compliance in some TS muxers. It is
certainly possible to retrieve it by parsing the output of FFmpeg, but
getting it from the context makes it simpler and less error-prone.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Perform validity check on AVFormatContext.channels instead of
uninitialised field.
This fixes issue 2001.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Fixes errors after a few minutes (first ping) when playing back
mmst://wm.bbc.co.uk/wms/bbc7coyopa/bbc7_-_friday_0430.wma
(cherry picked from commit 275189a2bd)
The code was setting aspect ratio from the DTG active format, and was
present and disabled since the creation of ffplay.
See thread:
Subject: [FFmpeg-devel] [PATCH] Cosmetics: adopt compact notation in
disabled code.
Date: Sat, 12 Feb 2011 18:46:19 +0100
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit b7ec5cc401)
AC3DSPContext.ac3_max_msb_abs_int16() finds the maximum MSB of the absolute
value of each element in an array of int16_t.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit fbb6b49dab)
The colon operator of expr always anchors the pattern at the start
of the string. An explicit ^ in the pattern has unspecified
behaviour, so remove it.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 3ad464bfc7)
This makes the cc_ident value, which is used in FATE reports, include
all interesting parts of the gcc version string.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 5d57846bba)
This function removes leading and trailing spaces and collapses
multiple spaces into one.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit bb7a22b584)
There is a check for HAVE_BIGENDIAN when outputting the IEC 61937
stream. On big-endian systems the payload data is not byteswapped,
causing in effect the outputted payload data to be in a different byte
order on big-endian than on little-endian systems.
However, the IEC 61937 preamble (and the final odd byte if present) is
always outputted in the same byte order. This means that on big-endian
systems the headers have a different byte order than the payload,
preventing useful use of the output.
Fix that by outputting the data in a format suitable for sending to an
audio device in S16LE format by default. Output as big-endian (S16BE)
is added as an AVOption. This makes the muxer output the same on all
archs by default.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 57f2c9aed9)
This fixes visual glitches in Bink version 'b' files, as the quantization
tables were not being permuted.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 2315392174)
When built with gcc 4.6, the MMX rgb24 to yuv conversion gives
wrong output. The compiler produces this warning:
libswscale/swscale_template.c:1885:5: warning: use of memory input without lvalue in asm operand 4 is deprecated
Changing the memory operand to a register makes it work.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit f344903ca5)
The code was setting aspect ratio from the DTG active format, and was
present and disabled since the creation of ffplay.
See thread:
Subject: [FFmpeg-devel] [PATCH] Cosmetics: adopt compact notation in
disabled code.
Date: Sat, 12 Feb 2011 18:46:19 +0100
Signed-off-by: Mans Rullgard <mans@mansr.com>
AC3DSPContext.ac3_max_msb_abs_int16() finds the maximum MSB of the absolute
value of each element in an array of int16_t.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
The colon operator of expr always anchors the pattern at the start
of the string. An explicit ^ in the pattern has unspecified
behaviour, so remove it.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Some MPEG4 cameras produce files with empty GOP headers.
This patch makes the decoder ignore such broken headers and proceed
with the following I-frame. Without this change, the following
start code is missed resulting in the entire I-frame being skipped.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This makes the cc_ident value, which is used in FATE reports, include
all interesting parts of the gcc version string.
Signed-off-by: Mans Rullgard <mans@mansr.com>
There is a check for HAVE_BIGENDIAN when outputting the IEC 61937
stream. On big-endian systems the payload data is not byteswapped,
causing in effect the outputted payload data to be in a different byte
order on big-endian than on little-endian systems.
However, the IEC 61937 preamble (and the final odd byte if present) is
always outputted in the same byte order. This means that on big-endian
systems the headers have a different byte order than the payload,
preventing useful use of the output.
Fix that by outputting the data in a format suitable for sending to an
audio device in S16LE format by default. Output as big-endian (S16BE)
is added as an AVOption. This makes the muxer output the same on all
archs by default.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
This fixes visual glitches in Bink version 'b' files, as the quantization
tables were not being permuted.
Signed-off-by: Mans Rullgard <mans@mansr.com>
When built with gcc 4.6, the MMX rgb24 to yuv conversion gives
wrong output. The compiler produces this warning:
libswscale/swscale_template.c:1885:5: warning: use of memory input without lvalue in asm operand 4 is deprecated
Changing the memory operand to a register makes it work.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Using doubles make the double -> int cast well defined for all the values
used, with the exception of when s[i]==1.0, which is special-cased.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 47d62c965b)
In the FAQ section "How do I encode single pictures into movies?", use
-s for generating symbolic links with the ln command.
The script was generating hard links, which is not likely what it was
supposed to do.
Fix issue 2488.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit e063f5886b)
Allow make clean to remove the corresponding binaries.
Fix issue 2162.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit cd6a5a57b8)
This is used for mapping AVStreams back to their corresponding
RTSPStream. Since d9c0510, the RTSPStream pointer isn't stored in
AVStream->priv_data any longer, breaking this mapping from AVStreams
to RTSPStreams.
Also, we don't need to clear the priv_data in rdt cleanup any longer,
since it isn't set to duplicate pointers.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit b2dd842d21)
The early disabling of irrelevant arch extensions is no longer
required, and removing it makes dependencies involving these
work as expected.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 4b884207eb)
it's not touched anywhere in ffmpeg, the code setting it was removed
over two years ago (e9b78eeba2).
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit b0294c80d3)
Using doubles make the double -> int cast well defined for all the values
used, with the exception of when s[i]==1.0, which is special-cased.
Signed-off-by: Mans Rullgard <mans@mansr.com>
In the FAQ section "How do I encode single pictures into movies?", use
-s for generating symbolic links with the ln command.
The script was generating hard links, which is not likely what it was
supposed to do.
Fix issue 2488.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This is used for mapping AVStreams back to their corresponding
RTSPStream. Since d9c0510, the RTSPStream pointer isn't stored in
AVStream->priv_data any longer, breaking this mapping from AVStreams
to RTSPStreams.
Also, we don't need to clear the priv_data in rdt cleanup any longer,
since it isn't set to duplicate pointers.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
The early disabling of irrelevant arch extensions is no longer
required, and removing it makes dependencies involving these
work as expected.
Signed-off-by: Mans Rullgard <mans@mansr.com>
it's not touched anywhere in ffmpeg, the code setting it was removed
over two years ago (e9b78eeba2).
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
s->windowed_samples will always have a range of [-32767,32767] due to the
window function, so the return value from log2_tab() will always be in the
range [0,14].
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 626264b11b)
This code will be later split out into a function which takes a 'size'
argument, so I'm keeping the name 'sizeX' here.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 0b1d291a71)
Only trivial splits are done here -- i.e. copy/paste + reindent +
missing variable declarations.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit c1fea23070)
Instead of returning an error when bytes are left over, just return
the number of actually used bytes as other decoders do.
Instead add a special case so an error will be returned when none
of the data looks valid to avoid making debugging a pain.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 4a72765a1c)
The function return type is void, so a return statement with an
expression is forbidden (and pointless).
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit b4668274b9)
Keep the original corner case behaviour, where reuse is enabled
for the case where no argument is given to the reuse url option.
Signed-off-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 00952be424)
The avcodec_thread_free() compatibility wrapper calls ff_thread_free(),
which is not defined when threading is disabled. Make this call
conditional.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 9a77a92c2b)
check AVCodecContext->sample_fmt against AVCodec->sample_fmts[] to ensure
that the encoder supports the specified sample format. Error out if it doesn't.
Previously, it would continue and output garbage. Fixes issue 2587.
(cherry picked from commit 2cfa2d9258)
Decode times for big_buck_bunny_720p_stereo:
1 thread:
real 1m14.227s
user 1m13.104s
sys 0m1.108s
2 threads: (33% faster)
real 0m49.329s
user 1m33.735s
sys 0m1.834s
3 threads: (44% faster)
real 0m41.593s
user 1m44.884s
sys 0m1.967s
(cherry picked from commit d23845f311)
As a side effect of the last commit, avcodec_open() now calls it automatically,
so there is no longer any need for clients to call it.
Instead they should set AVCodecContext.thread_count.
avcodec_thread_free() is deprecated, and will be removed from avcodec.h at the
next MAJOR libavcodec bump.
Rename the functions to ff_thread_init/free, since they are now internal.
Wrappers are provided to maintain API compatibility.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit c0b102ca03)
s->windowed_samples will always have a range of [-32767,32767] due to the
window function, so the return value from log2_tab() will always be in the
range [0,14].
Signed-off-by: Mans Rullgard <mans@mansr.com>
This code will be later split out into a function which takes a 'size'
argument, so I'm keeping the name 'sizeX' here.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Instead of returning an error when bytes are left over, just return
the number of actually used bytes as other decoders do.
Instead add a special case so an error will be returned when none
of the data looks valid to avoid making debugging a pain.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
Keep the original corner case behaviour, where reuse is enabled
for the case where no argument is given to the reuse url option.
Signed-off-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
The avcodec_thread_free() compatibility wrapper calls ff_thread_free(),
which is not defined when threading is disabled. Make this call
conditional.
Signed-off-by: Mans Rullgard <mans@mansr.com>
check AVCodecContext->sample_fmt against AVCodec->sample_fmts[] to ensure
that the encoder supports the specified sample format. Error out if it doesn't.
Previously, it would continue and output garbage. Fixes issue 2587.
Decode times for big_buck_bunny_720p_stereo:
1 thread:
real 1m14.227s
user 1m13.104s
sys 0m1.108s
2 threads: (33% faster)
real 0m49.329s
user 1m33.735s
sys 0m1.834s
3 threads: (44% faster)
real 0m41.593s
user 1m44.884s
sys 0m1.967s
As a side effect of the last commit, avcodec_open() now calls it automatically,
so there is no longer any need for clients to call it.
Instead they should set AVCodecContext.thread_count.
avcodec_thread_free() is deprecated, and will be removed from avcodec.h at the
next MAJOR libavcodec bump.
Rename the functions to ff_thread_init/free, since they are now internal.
Wrappers are provided to maintain API compatibility.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Also allow qmin/qmax to go up to 69 (the current max value for libx264). This
will have to increase when we add 9/10-bit support.
(cherry picked from commit c7ac200d15)
Due to being pants-on-head retarded, libavcodec defaults this to zero, which
results in broken output. This didn't affect ffmpeg.c, which sets it itself,
but caused problems for other calling apps using VBV.
(cherry picked from commit f7f8120fb9)
Fix emu_edge_v_extend_15 to be <128 bytes on Win64, by being more strict
on the size of registers and which registers are being used for operations
where multiple are available. This fixes segfaults in emulated_edge()
function calls on Win64.
(cherry picked from commit 17cf7c68ed)
It is useful for applications that hand input data directly to lavf via
a ByteIOContext.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 3940caad02)
There is no need to pass the ByteIOContext via a pointer to a pointer
anymore.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit aad216fd7e)
Currently (since the data_offset fix) the ogg demuxer assumes that
after the first non-header packets in any stream no more header packets
will follow.
This is not guaranteed, so change the code back again to wait until it
has finished the headers for all streams before returning from ogg_get_headers.
This fixes issue 2428.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 6bd69e6ada)
Haiku does not have a separate libm, so do not try to link to it.
Signed-off-by: François Revol <revol@free.fr>
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit f59c4bd625)
In all 3 cases, the decoding continues and thus a warning would be sufficient.
Helps application that catch them with own log handers to handle them
accordingly.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit ae2104791f)
This adds NEON optimised versions of all functions in VP8DSPContext.
Based on initial work by Rob Clark.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit a1c1d3c003)
This validate the length of a mkv element directly after reading
it.
This has the advantage that it is easy to add new limits and makes
it less likely to forget to add checks and also avoids issues like
bits of the length value above the first 32 being ignored because
the parsing functions only takes an int.
Previously discussed in the "mkv 0-byte integer parsing" thread.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 95ec3d4cac)
Change int64_t into a int, which caused this compiler warning:
libavformat/oggparseskeleton.c:64: warning: passing argument 2 of ‘av_reduce’ from incompatible pointer type
(cherry picked from commit 69ff149204)
Use avio functions instead of bytestream ones (also drops dependency on
lavc and removes a bunch of warnings).
Drop custom version of avio_get_str16 and use that instead.
Tested on mewmew-ssa.avi sample.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 47fdf00a77)
Makes playing QDMC files in MPlayer work when using the libavformat demuxer.
Problem was that the extradata was not passed from demuxer to decoder.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit ed19fafd48)
NASM versions older than 2.08 fail to build ffmpeg with several
"error: operation size not specified" errors but this is not caught in
configure.
Fix that by checking if "pextrd [eax], xmm0, 1" works in configure.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 48545a8f72)
Due to being pants-on-head retarded, libavcodec defaults this to zero, which
results in broken output. This didn't affect ffmpeg.c, which sets it itself,
but caused problems for other calling apps using VBV.
Fix emu_edge_v_extend_15 to be <128 bytes on Win64, by being more strict
on the size of registers and which registers are being used for operations
where multiple are available. This fixes segfaults in emulated_edge()
function calls on Win64.
Currently (since the data_offset fix) the ogg demuxer assumes that
after the first non-header packets in any stream no more header packets
will follow.
This is not guaranteed, so change the code back again to wait until it
has finished the headers for all streams before returning from ogg_get_headers.
This fixes issue 2428.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Haiku does not have a separate libm, so do not try to link to it.
Signed-off-by: François Revol <revol@free.fr>
Signed-off-by: Mans Rullgard <mans@mansr.com>
In all 3 cases, the decoding continues and thus a warning would be sufficient.
Helps application that catch them with own log handers to handle them
accordingly.
Signed-off-by: Mans Rullgard <mans@mansr.com>
In some places, dvbsubdec passes improper input buffer size to
bitstream reading functions, not accounting for reading pointer
updates.
Fixed by using buffer_end - buffer pointer instead of fixed buffer length.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
In some places, dvbsubdec passes improper input buffer size to
bitstream reading functions, not accounting for reading pointer
updates.
Fixed by using buffer_end - buffer pointer instead of fixed buffer length.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Mans Rullgard <mans@mansr.com>
This validate the length of a mkv element directly after reading
it.
This has the advantage that it is easy to add new limits and makes
it less likely to forget to add checks and also avoids issues like
bits of the length value above the first 32 being ignored because
the parsing functions only takes an int.
Previously discussed in the "mkv 0-byte integer parsing" thread.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Change int64_t into a int, which caused this compiler warning:
libavformat/oggparseskeleton.c:64: warning: passing argument 2 of ‘av_reduce’ from incompatible pointer type
Use avio functions instead of bytestream ones (also drops dependency on
lavc and removes a bunch of warnings).
Drop custom version of avio_get_str16 and use that instead.
Tested on mewmew-ssa.avi sample.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Makes playing QDMC files in MPlayer work when using the libavformat demuxer.
Problem was that the extradata was not passed from demuxer to decoder.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Using a preset file caused the address of a stack variable to be stored
in opt_names/values. This change causes the strings to be dup'd then
freed in uninit_opts.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 60ff9de6ff)
No behavior change; this makes DTS reliable with the next patch.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 6b47495397)
If required, the caller can do this itself. ff_write_chained rescales
timestamps as necessary, and all current callers of rtpenc_chain
use ff_write_chained, making this timebase copy unnecessary.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 397ffde115)
This function is useful for freeing data structures allocated by
muxers, which currently have to be freed manually by the caller.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit f124b087ee)
This avoids having the chained AVStream->codec point to the same
AVCodecContext owned by the outer AVStream. The downside is that
changes to the AVCodecContext made after calling av_write_header
cannot be detected automatically within the chained muxer.
This avoids having to manually unlink the chained AVStream->codec
by setting it to null before freeing the chained muxer via generic
freeing functions.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 1338dc0823)
Make avfilter_graph_free() free not only the internal structures, but
also the allocated graph, and set the graph pointer to NULL for
increased safety.
Simplify usage.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 4359288c56)
Make avfilter_graph_parse() only release the internal structures
allocated during the parsing, and leave to free the graph itself to
the calling code.
This approach looks cleaner, as the graph is not allocated by the
function.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit e8e5dde779)
NASM versions older than 2.08 fail to build ffmpeg with several
"error: operation size not specified" errors but this is not caught in
configure.
Fix that by checking if "pextrd [eax], xmm0, 1" works in configure.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Using a preset file caused the address of a stack variable to be stored
in opt_names/values. This change causes the strings to be dup'd then
freed in uninit_opts.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
If required, the caller can do this itself. ff_write_chained rescales
timestamps as necessary, and all current callers of rtpenc_chain
use ff_write_chained, making this timebase copy unnecessary.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This function is useful for freeing data structures allocated by
muxers, which currently have to be freed manually by the caller.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This avoids having the chained AVStream->codec point to the same
AVCodecContext owned by the outer AVStream. The downside is that
changes to the AVCodecContext made after calling av_write_header
cannot be detected automatically within the chained muxer.
This avoids having to manually unlink the chained AVStream->codec
by setting it to null before freeing the chained muxer via generic
freeing functions.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Make avfilter_graph_free() free not only the internal structures, but
also the allocated graph, and set the graph pointer to NULL for
increased safety.
Simplify usage.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Make avfilter_graph_parse() only release the internal structures
allocated during the parsing, and leave to free the graph itself to
the calling code.
This approach looks cleaner, as the graph is not allocated by the
function.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Now the first argument is URLContext *h. However, the function logs to
LOG_CONTEXT, which is #defined as 's' for new lavf major versions.
Therefore, rename h -> s.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 9ad4c65f6f)
This fixes memory leaks in the RTSP muxer and RTP hinting in the
mov muxer present since SVN rev 25418.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
(cherry picked from commit ce41c51b0c)
For mpegts in RTP, there isn't a direct mapping between RTSPStreams
and AVStreams, and the RTSPStream isn't ever stored in
AVStream->priv_data, which was earlier leaked. The fix for this
leak, in ea7f080749, lead to
double frees for other, normal RTP streams.
This patch avoids storing RTSPStreams in AVStream->priv_data, thus
avoiding the double free. The RTSPStreams are always available via
RTSPState->rtsp_streams anyway.
Tested with MS-RTSP, RealRTSP, DSS and mpegts/RTP.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
(cherry picked from commit d9c0510e22)
It's consistent with the -metadata option and easier to write.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 87e4d9b252)
The bumps are for adding version.h and avio_{get/put}_str functions in
lavf and making av_dlog public in lavu.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
This patch changes the exponent difference threshold in the exponent
strategy decision function of the AC-3 encoder. I tested lowering in
increments of 100. From 1000 down to 500 generally increased in quality
with each step, but 400 was generally much worse.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit c3beafa0f1)
This will be beneficial for use with the audio conversion API without
requiring it to depend on all of dsputil.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit c73d99e672)
Gcc 4.6 only preserves the first value when using an array with an "m"
constraint.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 770c410fbb)
Now the first argument is URLContext *h. However, the function logs to
LOG_CONTEXT, which is #defined as 's' for new lavf major versions.
Therefore, rename h -> s.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This fixes memory leaks in the RTSP muxer and RTP hinting in the
mov muxer present since SVN rev 25418.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
For mpegts in RTP, there isn't a direct mapping between RTSPStreams
and AVStreams, and the RTSPStream isn't ever stored in
AVStream->priv_data, which was earlier leaked. The fix for this
leak, in ea7f080749, lead to
double frees for other, normal RTP streams.
This patch avoids storing RTSPStreams in AVStream->priv_data, thus
avoiding the double free. The RTSPStreams are always available via
RTSPState->rtsp_streams anyway.
Tested with MS-RTSP, RealRTSP, DSS and mpegts/RTP.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The bumps are for adding version.h and avio_{get/put}_str functions in
lavf and making av_dlog public in lavu.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
This patch changes the exponent difference threshold in the exponent
strategy decision function of the AC-3 encoder. I tested lowering in
increments of 100. From 1000 down to 500 generally increased in quality
with each step, but 400 was generally much worse.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This will be beneficial for use with the audio conversion API without
requiring it to depend on all of dsputil.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This makes "make documentation" build the man/html pages only for
the tools enabled in the build. It also fixes the dependency
tracking for the built man pages.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 7f939f55bb)
Adds some duplicated code, but avoids duplicate edge checks and similar.
~0.5% faster overall on Parkjoy test sample.
(cherry picked from commit 64233e702a)
The first part of the metadata, the "vendor" string, is required by
libvorbis, it will refuse to play when it is not available.
Also we do not currently parse that part into metadata so it would also
be lost if we removed it as well.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 8cb3c557a9)
This makes "make documentation" build the man/html pages only for
the tools enabled in the build. It also fixes the dependency
tracking for the built man pages.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The first part of the metadata, the "vendor" string, is required by
libvorbis, it will refuse to play when it is not available.
Also we do not currently parse that part into metadata so it would also
be lost if we removed it as well.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Around 01/28/11 18:56, Ronald S. Bultje scribbled:
> That patch is now merged, can you submit the update to muxers.texi?
> Then we'll apply the whole thing.
See attached. I hope the documentation is enough.
--
Georgi Chorbadzhiyski
http://georgi.unixsol.org/
From c236024b8254f5c2c45934c30fff390cb0e55a5e Mon Sep 17 00:00:00 2001
From: Georgi Chorbadzhiyski <gf@unixsol.org>
Date: Tue, 25 Jan 2011 13:09:17 +0200
Subject: [PATCH] mpegts: Replace defines in with AVOptions
This patch adds support for setting transport_stream_id,
original_network_id, service_id, pmt_start_pid and start_pid
in mpegts muxer.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 445996aa51)
AVCodecContext.reordered_opaque is deprecated for this specific use.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 2855080447)
Disable logging of rescaled timestamps if DEBUG is not enabled.
Avoid debug log spamming with -loglevel debug.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 13156f40e1)
Around 01/28/11 18:56, Ronald S. Bultje scribbled:
> That patch is now merged, can you submit the update to muxers.texi?
> Then we'll apply the whole thing.
See attached. I hope the documentation is enough.
--
Georgi Chorbadzhiyski
http://georgi.unixsol.org/
From c236024b8254f5c2c45934c30fff390cb0e55a5e Mon Sep 17 00:00:00 2001
From: Georgi Chorbadzhiyski <gf@unixsol.org>
Date: Tue, 25 Jan 2011 13:09:17 +0200
Subject: [PATCH] mpegts: Replace defines in with AVOptions
This patch adds support for setting transport_stream_id,
original_network_id, service_id, pmt_start_pid and start_pid
in mpegts muxer.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Disable logging of rescaled timestamps if DEBUG is not enabled.
Avoid debug log spamming with -loglevel debug.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This moves the fields needed by asm near the top, before any
structs or other members which complicate the offset calculation.
Modifying other structs will no longer require updating the offsets,
and the asm code is slightly simpler due to the smaller offsets.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit d461a47317)
The generated HTML files are similar to the ones generated with
texi2html 1.56k used on the website.
Tested with texi2html 1.78 and 5.0. 1.78 is the minimal recommended
version.
The removed @sp from the titlepage section were ignored until
texi2html 5.0. If not removed the pages generated by 5.0 will have ugly
empty space around the title.
(cherry picked from commit a8f0814a74)
From ~780 cycles to 551 cycles, mostly just by using libc memcpy()
instead of manually shuffling individual bytes around.
(cherry picked from commit e5262ec44a)
$subject. Have used this for loopback testing with mpegts.c.
-- Peter
(A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B)
[2. text/x-diff; 0001-mpegtsenc-support-CODEC_ID_AAC_LATM.patch]
From 0f7f9db4b7da1793996af6dda84298507703759a Mon Sep 17 00:00:00 2001
From: Peter Ross <pross@xvid.org>
Date: Sun, 9 Jan 2011 09:45:50 +1100
Subject: [PATCH] mpegtsenc: support CODEC_ID_AAC_LATM
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 4d54df8e07)
poll() is only used by networking code, so the fallback should
only be built if networking is enabled. Also remove CONFIG_FFSERVER
condition from the declarations.
This should fix building on systems without poll(), broken
by a8475bbdb6.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 362d8f7d9e)
Add an error message in case the user requests to write more than one file
and the path does not contain a "%d" or "%0Nd" pattern.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 4fc9ff0ad6)
According to EN 300 468 section 3.1 (Definitions):
Unless otherwise specified within the present document all
"reserved_future_use" bits is set to "1".
This was not the case for SDT generation so this patch fixes it.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit a7827a17c6)
That's required because -1 is evaluated as NAN, which converted back
to int looks like a random number, this is especially annoying when
debugging sources with undefined pos (as the video4linux2 device).
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 3e5bc7ff6a)
This significantly reduces the size of the symbol table in the generated ELF
shared object (as well as the other linked tables).
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit ac28ce5fac)
This moves the fields needed by asm near the top, before any
structs or other members which complicate the offset calculation.
Modifying other structs will no longer require updating the offsets,
and the asm code is slightly simpler due to the smaller offsets.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The generated HTML files are similar to the ones generated with
texi2html 1.56k used on the website.
Tested with texi2html 1.78 and 5.0. 1.78 is the minimal recommended
version.
The removed @sp from the titlepage section were ignored until
texi2html 5.0. If not removed the pages generated by 5.0 will have ugly
empty space around the title.
VC1 test container always uses time-base 1 ms, so we must convert
from whatever time-base the application gave us to that, otherwise
the video will play at ridiculous speeds.
It would be possible to signal that a container supports only one
time-base and have code in a layer above do the conversion, but
for a single format this seems over-engineered.
$subject. Have used this for loopback testing with mpegts.c.
-- Peter
(A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B)
[2. text/x-diff; 0001-mpegtsenc-support-CODEC_ID_AAC_LATM.patch]
From 0f7f9db4b7da1793996af6dda84298507703759a Mon Sep 17 00:00:00 2001
From: Peter Ross <pross@xvid.org>
Date: Sun, 9 Jan 2011 09:45:50 +1100
Subject: [PATCH] mpegtsenc: support CODEC_ID_AAC_LATM
Signed-off-by: Mans Rullgard <mans@mansr.com>
poll() is only used by networking code, so the fallback should
only be built if networking is enabled. Also remove CONFIG_FFSERVER
condition from the declarations.
This should fix building on systems without poll(), broken
by a8475bbdb6.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Add an error message in case the user requests to write more than one file
and the path does not contain a "%d" or "%0Nd" pattern.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
According to EN 300 468 section 3.1 (Definitions):
Unless otherwise specified within the present document all
"reserved_future_use" bits is set to "1".
This was not the case for SDT generation so this patch fixes it.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
That's required because -1 is evaluated as NAN, which converted back
to int looks like a random number, this is especially annoying when
debugging sources with undefined pos (as the video4linux2 device).
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
This also lists the objects from those two libraries as internal (by adding
the ff_ prefix) so that they can then be hidden via linker scripts.
(cherry picked from commit c6610a216e)
Fix C VP8 H+V MC functions which do two-dimensional 4/6-tap filters to
not overread beyond their edges if the second filter is 4-tap, since
the outer pixels aren't there anymore since
44002d8323.
(cherry picked from commit 22893e10ae)
None of these symbols should be accessed directly, so declare them as
hidden.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit d36beb3f69)
The iff.h header only declared one function that is now static, the
libavformat/iff.c source file wasn't using it before. Drop the file
entirely.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 2d162e3825)
It is only used to generate band_start_tab, which about the same size, at
runtime, so it's simpler just to always hardcode band_start_tab.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 24e3ad3031)
This removes ff_get_ref_perms_string, ff_dprintf_ref and ff_dprintf_link
fro the interface of libavfilter.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit eb7ccf8f33)
The reserved bits between PCR base and extension fields must be
set to 1.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 535638b55f)
This significantly reduces the size of the symbol table in the generated ELF
shared object (as well as the other linked tables).
Signed-off-by: Mans Rullgard <mans@mansr.com>
Fix C VP8 H+V MC functions which do two-dimensional 4/6-tap filters to
not overread beyond their edges if the second filter is 4-tap, since
the outer pixels aren't there anymore since
44002d8323.
The iff.h header only declared one function that is now static, the
libavformat/iff.c source file wasn't using it before. Drop the file
entirely.
Signed-off-by: Mans Rullgard <mans@mansr.com>
It is only used to generate band_start_tab, which about the same size, at
runtime, so it's simpler just to always hardcode band_start_tab.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This removes ff_get_ref_perms_string, ff_dprintf_ref and ff_dprintf_link
fro the interface of libavfilter.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The header is empty after making the function static, so delete it and
drop its usage.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 13eb6b9097)
Both functions seem to be commanded by the ff_spatial_idwt function
instead.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit ebb06d96ed)
Also make it an anonymous structure as never it is accessed by name.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit a3dffc0627)
It will be useful in mp3 demuxer and hopeful some other places.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 93b78d1210)
This allows passing armcc-specific flags with --extra-cflags without
choking the assembler.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit d0f0f6287c)
This adds a function to filter out words matching a pattern
from a list.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 9d201b2606)
Do not emulate larger edges than we will actually use for this round of
MC. Decoding goes from avg+SE 29.972+/-0.023sec to 29.856+/-0.023, i.e.
0.12sec or ~0.4% faster.
(cherry picked from commit 44002d8323)
This allows this file to build on systems lacking NAN or INFINITY
in math.h.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit e781c4e6ff)
This symbol is only ever used to calculate the non-hardcoded tables, so
only enable it in that case, and static to the source unit that uses it.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 6ed3b504f9)
The variable TESTPROGS is reset by the library makefiles,
use another name.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 3d157bf31f)
This moves network_extralibs setup before use so that the link tests
for network functions work correctly.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 54fe299b88)
Remove two variables that were not used and caused the following
warnings:
CC libavformat/mpegtsenc.o
libavformat/mpegtsenc.c: In function 'mpegts_write_section':
libavformat/mpegtsenc.c:72:18: warning: unused variable 'ts'
libavformat/mpegtsenc.c: In function 'mpegts_insert_null_packet':
libavformat/mpegtsenc.c:586:18: warning: unused variable 'ts'
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 6e78c8ee94)
The dprintf macro is a no-op if DEBUG is not defined, so there
is no need to guard it here.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 6081f8c4e2)
1d4da6a460 added static to the
prototypes for these fuctions. Adding it to the definitions
as well.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit aa61e39eac)
The PCM_DVD encoder would be left unused, as allcodecs.c properly declared
it as being decoder-only, but it would still be built into the object file.
Since there is no block of code to properly encode this PCM format, it's
not a full codec.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 5b5083b5fe)
The dprintf macro is no-op when DEBUG is unset, so there is no need to
put it conditional to DEBUG.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 73a0b19ba3)
Also drop some CONFIG_DV_MUXER #ifdefs probably vestigial from before the
split of dvenc and dvdec.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit f0a8676958)
This ensures a locally-unique name as well as marks the symbol as
FFmpeg-private at least by declaration.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 82e1f217f2)
The key string is supposed to contain the equals character,
too. Since the checked string was wrong, and the return value
check was wrong too, it incorrectly seemed to work right before.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 2b0decf60b)
If audio_sync_method is >1 the resampler is used for audio drift
compensation, and do_audio_out() was causing an assert failure because
audio_resample was not set.
Fix issue 2516, which was introduced by SVN r25939.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 07b48f8c7a)
The PathScale compiler miscompiles wrapping arithmetic without
these flags.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 5f3b8314a4)
Dragonfly, NetBSD, and OpenBSD do not support symbol versioning
although our link test passes. Disable it explicitly for these
systems.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit a1e4b3f6d3)
Do not emulate larger edges than we will actually use for this round of
MC. Decoding goes from avg+SE 29.972+/-0.023sec to 29.856+/-0.023, i.e.
0.12sec or ~0.4% faster.
This symbol is only ever used to calculate the non-hardcoded tables, so
only enable it in that case, and static to the source unit that uses it.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This moves network_extralibs setup before use so that the link tests
for network functions work correctly.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Remove two variables that were not used and caused the following
warnings:
CC libavformat/mpegtsenc.o
libavformat/mpegtsenc.c: In function 'mpegts_write_section':
libavformat/mpegtsenc.c:72:18: warning: unused variable 'ts'
libavformat/mpegtsenc.c: In function 'mpegts_insert_null_packet':
libavformat/mpegtsenc.c:586:18: warning: unused variable 'ts'
Signed-off-by: Mans Rullgard <mans@mansr.com>
The PCM_DVD encoder would be left unused, as allcodecs.c properly declared
it as being decoder-only, but it would still be built into the object file.
Since there is no block of code to properly encode this PCM format, it's
not a full codec.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This ensures a locally-unique name as well as marks the symbol as
FFmpeg-private at least by declaration.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The key string is supposed to contain the equals character,
too. Since the checked string was wrong, and the return value
check was wrong too, it incorrectly seemed to work right before.
Signed-off-by: Mans Rullgard <mans@mansr.com>
If audio_sync_method is >1 the resampler is used for audio drift
compensation, and do_audio_out() was causing an assert failure because
audio_resample was not set.
Fix issue 2516, which was introduced by SVN r25939.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Dragonfly, NetBSD, and OpenBSD do not support symbol versioning
although our link test passes. Disable it explicitly for these
systems.
Signed-off-by: Mans Rullgard <mans@mansr.com>
With the removal of the libmpeg2 bitstream reader, MIN_CACHE_BITS
is always >= 25, so tests against smaller values can be removed.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit f162e988aa)
These whitespace changes improve the readability of the get_bits
macros.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit fb5c841d5f)
Some of the macros in get_bits.h include a final semicolon,
some do not. This removes these or adds do {} while(0) around
the macros as appropriate and adds semicolons where needed in
calling code.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit bf5f9b528b)
Using the libmpeg2 reader causes errors in a multitude of places,
including MPEG and H264 codecs. As the advantage of this reader
is questionable, removing it seems the sensible course of action,
especially considering the simplifications this allows elsewhere
with the bit cache size increasing from 17 to 25 bits as minimum.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 938f72e199)
Don't free RBSP tables (containing decoded NAL units) on resolution
change, because we actually need this data to decode the frame after
reiniting (with new resolution). Fixed issue 2393.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 9107892624)
C99 variadic macros require more arguments than there are named
parameters in the definition. This means we must use an extra
indirection to avoid having two different macros for arrays with
one resp more than one dimension.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 96aad41e81)
It's incomplete, no one is working on it, and when someone asks about
working on it we advise them not to.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit ff3d43104f)
Use backwards compatible explicit signalling to denote the absence of
SBR.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 8ae0fa243e)
I did not notice that the filter implementation uses a reversed history state.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 98cfadd648)
fixes issue2465.
The problem is that the ffmpeg (the app) -streamid option did not work
with -newaudio/-newvideo/-newsubtitle.
The cause was a conflict between the feature where streamid values were
reset to default for each output filename, and the implementation of
-new*, which requires that the -new* option be specified after the
target filename.
My patch changes the ffmpeg behavior so that user-specified streamid
values apply to all the following output files on the command line
(rather than just the next output filename.)
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 20ac9de3df)
make the initialization of put clearer
this are the differences between
[FFmpeg-devel] [PATCH 1/3] mp3enc: add support for writing UTF-16 tags
and the already applied 187e23478b
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit 8c3caf7fb1)
Instead of saving huge raw files, use the md5: output pseudo-protocol
to calculate the checksum of the file directly. This is especially
useful when testing on remote targets as it avoids transferring 3.6GB
over the network.
(cherry picked from commit f4b1e21a63)
DTS-HD HRA streams do not always have an XBR extension in the extension
substream. Instead they can have only XXCh and X96 extensions in
there and still be considered DTS-HD HRA.
This is also confirmed with Onkyo TX-SR607 receiver which recognizes
such a stream as HiRes Audio.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 8f4a5d225c)
Some variants of mktemp require a template, so provide one when
checking for the command. We already supply a template in the
subsequent uses of mktemp.
Thanks to Michael Kostylev.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit a210bce298)
This does the same thing and also fixes the trapping in
some (possibly broken) shells.
Suggested by Michael Kostylev.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 93bb9ff08e)
With the removal of the libmpeg2 bitstream reader, MIN_CACHE_BITS
is always >= 25, so tests against smaller values can be removed.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Some of the macros in get_bits.h include a final semicolon,
some do not. This removes these or adds do {} while(0) around
the macros as appropriate and adds semicolons where needed in
calling code.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Using the libmpeg2 reader causes errors in a multitude of places,
including MPEG and H264 codecs. As the advantage of this reader
is questionable, removing it seems the sensible course of action,
especially considering the simplifications this allows elsewhere
with the bit cache size increasing from 17 to 25 bits as minimum.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Don't free RBSP tables (containing decoded NAL units) on resolution
change, because we actually need this data to decode the frame after
reiniting (with new resolution). Fixed issue 2393.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
C99 variadic macros require more arguments than there are named
parameters in the definition. This means we must use an extra
indirection to avoid having two different macros for arrays with
one resp more than one dimension.
Signed-off-by: Mans Rullgard <mans@mansr.com>
It's incomplete, no one is working on it, and when someone asks about
working on it we advise them not to.
Signed-off-by: Mans Rullgard <mans@mansr.com>
fixes issue2465.
The problem is that the ffmpeg (the app) -streamid option did not work
with -newaudio/-newvideo/-newsubtitle.
The cause was a conflict between the feature where streamid values were
reset to default for each output filename, and the implementation of
-new*, which requires that the -new* option be specified after the
target filename.
My patch changes the ffmpeg behavior so that user-specified streamid
values apply to all the following output files on the command line
(rather than just the next output filename.)
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
make the initialization of put clearer
this are the differences between
[FFmpeg-devel] [PATCH 1/3] mp3enc: add support for writing UTF-16 tags
and the already applied 187e23478b
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
Instead of saving huge raw files, use the md5: output pseudo-protocol
to calculate the checksum of the file directly. This is especially
useful when testing on remote targets as it avoids transferring 3.6GB
over the network.
DTS-HD HRA streams do not always have an XBR extension in the extension
substream. Instead they can have only XXCh and X96 extensions in
there and still be considered DTS-HD HRA.
This is also confirmed with Onkyo TX-SR607 receiver which recognizes
such a stream as HiRes Audio.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Some variants of mktemp require a template, so provide one when
checking for the command. We already supply a template in the
subsequent uses of mktemp.
Thanks to Michael Kostylev.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This does the same thing and also fixes the trapping in
some (possibly broken) shells.
Suggested by Michael Kostylev.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This makes configure --disable-everything --enable-demuxer=foo
work as expected.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit c2dd0e9eba)
Simplifies error handling and makes it easier to add additional filter types.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 0361d13cf3)
title metadata is only used as fallback if no service_name is available.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
(cherry picked from commit e5fe65512b)
Set service_provider and service_name in mpegts demuxer, previously
name and provider_name were set but since the muxer uses service_provider
and service_name use them.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit e8315e7881)
Apparently some broken taggers prepend a new ID3v2 tag leaving the
existing one intact. Our parser currently reads all tags and overwrites
existing values with supposedly outdated ones.
fixes issue2419
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 75aded8328)
This patch adds support in mpegts muxer for using service_provider and
service_name metadata to set service_provider_name and service_name
fields in SDT.
Example usage:
ffmpeg -i file.ts -f mpegts -re -acodec copy -vcodec copy -f mpegts \
-metadata service_provider="Some provider" \
-metadata service_name="Some Channel" \
"udp://239.0.70.2:5000?pkt_size=1316&ttl=1"
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 69688e71bc)
The original functions did not work correctly for edge pixels, e.g.
when CODEC_FLAG_EMU_EDGE is set, leading to corrupt output in e.g. VLC.
Based on a patch by Daniel Kang <daniel d kang gmail com>.
Signed-off-by: Ronald S. Bultje <rsbultje gmail com>
(cherry picked from commit b9c7f66e6d)
Set service_provider and service_name in mpegts demuxer, previously
name and provider_name were set but since the muxer uses service_provider
and service_name use them.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Apparently some broken taggers prepend a new ID3v2 tag leaving the
existing one intact. Our parser currently reads all tags and overwrites
existing values with supposedly outdated ones.
fixes issue2419
Signed-off-by: Mans Rullgard <mans@mansr.com>
This patch adds support in mpegts muxer for using service_provider and
service_name metadata to set service_provider_name and service_name
fields in SDT.
Example usage:
ffmpeg -i file.ts -f mpegts -re -acodec copy -vcodec copy -f mpegts \
-metadata service_provider="Some provider" \
-metadata service_name="Some Channel" \
"udp://239.0.70.2:5000?pkt_size=1316&ttl=1"
Signed-off-by: Mans Rullgard <mans@mansr.com>
The original functions did not work correctly for edge pixels, e.g.
when CODEC_FLAG_EMU_EDGE is set, leading to corrupt output in e.g. VLC.
Based on a patch by Daniel Kang <daniel d kang gmail com>.
Signed-off-by: Ronald S. Bultje <rsbultje gmail com>
Recent gcc versions define __ARM_PCS or __ARM_PCS_VFP to indicate the
VFP ABI in use, and ARM RVCT defines __SOFTFP__ when using this ABI.
If none of these are defined, check $cross_prefix and $cc for the
substring "hardfloat", and finally fall back to a linker test. This
gives the correct result in most configurations.
Some linkers fail if this flag is used when creating an executable file.
Since the flag is meaningful only for shared libraries, dropping it from
the main link flags is correct.
libavfilter/defaults.c: In function 'avfilter_default_get_audio_buffer':
libavfilter/defaults.c:129: warning: 'buf' is used uninitialized in this function
Originally committed as revision 26382 to svn://svn.ffmpeg.org/ffmpeg/trunk
Improves CABAC performance about ~1.2%.
Trick originates from x264 and has also been used in ffvp8. It's useful because
coded block flags are usually zero, so it helps to have the early termination
inlined into the main function.
Originally committed as revision 26375 to svn://svn.ffmpeg.org/ffmpeg/trunk
More consistent with the rest of FFmpeg and sounds more natural to
English speaking people.
Originally committed as revision 26374 to svn://svn.ffmpeg.org/ffmpeg/trunk
The hunk is not fully understood but it just makes a check tighter so its
safer for us to apply until it is fully understood.
Might fix issue 2550 (and Chrome issue 68115 and unknown CERT issues).
Our bugtracker issue though should stay open until this has been fully
investiagted
Patch by Frank Barchard, fbarchard at google
Originally committed as revision 26368 to svn://svn.ffmpeg.org/ffmpeg/trunk
This should improve duration accuracy slightly and avoids a warning about its
inaccuracy when accurate values are available. Idea by Frank Barchard
Originally committed as revision 26366 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fixes issue 2548 (and Chrome issue 68115 and unknown CERT issues).
Patch by Frank Barchard, fbarchard at google
Originally committed as revision 26365 to svn://svn.ffmpeg.org/ffmpeg/trunk
instead of real width, this fixes decoding of some Bink files with odd width.
Originally committed as revision 26364 to svn://svn.ffmpeg.org/ffmpeg/trunk
color value instead of always taking 0 (resulting in green frames).
Fixes issue issue2531.
Originally committed as revision 26363 to svn://svn.ffmpeg.org/ffmpeg/trunk
exponent strategies for a single channel to compute_exp_strategy_ch().
This allows for removal of the temporary pointer arrays.
Originally committed as revision 26356 to svn://svn.ffmpeg.org/ffmpeg/trunk
No speed improvement, but necessary for some future stuff.
Also opens up the possibility of asm chroma dc idct/dequant.
Originally committed as revision 26349 to svn://svn.ffmpeg.org/ffmpeg/trunk
Doesn't help speed as there isn't an asm implementation yet, but consistency
is a good thing.
Originally committed as revision 26348 to svn://svn.ffmpeg.org/ffmpeg/trunk
Since we no longer have non-transposed scantables, the problem it warns about
no longer exists.
Originally committed as revision 26339 to svn://svn.ffmpeg.org/ffmpeg/trunk
Useful so that we don't have to run the hierarchical DC iDCT if there aren't
any coefficients. Opens up some future opportunities for optimization as well.
Originally committed as revision 26337 to svn://svn.ffmpeg.org/ffmpeg/trunk
About 2.5x the speed.
NOTE: the way that the asm code handles large qmuls is a bit suboptimal.
If x264-style dequant was used (separate shift and qmul values), it might
be possible to get some extra speed.
Originally committed as revision 26336 to svn://svn.ffmpeg.org/ffmpeg/trunk
It was an ugly hack to begin with and didn't give any performance.
NOTE: this patch opens up some future simplifications to be made (such as
removing some of the scantables from H264Context) but doesn't take advantage
of them yet.
Originally committed as revision 26329 to svn://svn.ffmpeg.org/ffmpeg/trunk
Previously the pad filter just drawed borders in the surrounding of the input
without checking if this area was allocated or writeable. Now we check and
allocate a new buffer if the input is unsuitable.
Originally committed as revision 26315 to svn://svn.ffmpeg.org/ffmpeg/trunk
These fields are needed to be able to know which area of memory is allocated and
which is off limits.
This reverts 24291 and parts of r24424.
Originally committed as revision 26314 to svn://svn.ffmpeg.org/ffmpeg/trunk
ffmpeg.c: In function ‘new_video_stream’:
ffmpeg.c:3701:18: warning: ‘codec_id’ may be used uninitialized in this function
ffmpeg.c: In function ‘new_audio_stream’:
ffmpeg.c:3848:18: warning: ‘codec_id’ may be used uninitialized in this function
ffmpeg.c: In function ‘new_subtitle_stream’:
ffmpeg.c:3926:18: warning: ‘codec_id’ may be used uninitialized in this function
Originally committed as revision 26313 to svn://svn.ffmpeg.org/ffmpeg/trunk
ffmpeg.c: In function ‘parse_meta_type’:
ffmpeg.c:3323:13: warning: assignment discards qualifiers from pointer target type
Originally committed as revision 26312 to svn://svn.ffmpeg.org/ffmpeg/trunk
This improves the audio quality significantly for stereo source with both the
fixed-point and floating-point AC-3 encoders.
Update acodec-ac3_fixed and seek-ac3_rm test references.
Originally committed as revision 26271 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fixes issue 2072 and issue 2421.
Patch by Anatoly Nenashev, anatoly d nenashev a ovsoft d ru
Originally committed as revision 26252 to svn://svn.ffmpeg.org/ffmpeg/trunk
for invalid header up before reading data.
Fixes issue 2500.
Patch by Daniel Kang, daniel.d.kang at gmail
Originally committed as revision 26248 to svn://svn.ffmpeg.org/ffmpeg/trunk
If filtered, only packets from the right source address and port
are received.
To test, play back e.g. some mpeg4 video RTSP stream (where the
video stream is the first stream in the presentation) over UDP.
While receiving this stream, send another stream to the same port:
ffmpeg -re -i <whatever> -vcodec mpeg4 -an -f rtp
rtp://127.0.0.1:5000?localport=1234
Normally, the RTSP playback reports lots of errors at this point.
If the RTSP stream has the ?filter_src option enabled, these
interferring packets are ignored.
Originally committed as revision 26246 to svn://svn.ffmpeg.org/ffmpeg/trunk
If the remote address is updated later with this function, the caller
shouldn't set the connect option until in this call.
Originally committed as revision 26245 to svn://svn.ffmpeg.org/ffmpeg/trunk
By calling connect on the UDP socket, only packets from the chosen
peer address and port are received on the socket. This is one
solution to issue 1688.
Originally committed as revision 26244 to svn://svn.ffmpeg.org/ffmpeg/trunk
For MS-RTSP, we don't always get RTCP packets (never?), so the earlier
timestamping code never wrote anything into pkt->pts. The rtpdec_asf
depacketizer just sets the dts of the packet, so if the generic RTP
timestamping is used, too, we get inconsistent timestamps.
Therefore, skip the generic RTP timestamp algorithm if the depacketizer
already has set something.
This fixes "Invalid timestamps" warnings, present since SVN rev 26187.
Originally committed as revision 26241 to svn://svn.ffmpeg.org/ffmpeg/trunk
close the stream.
This way the flushed packets can still reference the still unclosed
format context.
In particular this fixes a spurious error issued when closing the
video4linux2 buffer in mmap_release_buffer(), which tries to access
the file descriptor of an already closed file.
Originally committed as revision 26237 to svn://svn.ffmpeg.org/ffmpeg/trunk
instead of doing it separately in 2 different functions.
This makes float AC-3 encoding approx. 3-7% faster overall.
Also, the coefficient conversion can now be easily SIMD-optimized.
Originally committed as revision 26232 to svn://svn.ffmpeg.org/ffmpeg/trunk
accessing of structs and arrays inside the loop.
Approx. 30% faster in function extract_exponents().
Originally committed as revision 26226 to svn://svn.ffmpeg.org/ffmpeg/trunk
timebase value already set in the driver, and set it back in the codec
stream, rather than leaving the invalid value of 0/0.
In particular, fix ffmpeg grabbing timestamps when the timebase value
is not set through the CLI.
Originally committed as revision 26224 to svn://svn.ffmpeg.org/ffmpeg/trunk
cmdutils.c:init_opts(), in the case libswscale compilation is not
enabled.
Fix ffprobe and ffserver compilation with --disable-swscale.
Originally committed as revision 26212 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fixed-point AC-3 encoder renamed to ac3_fixed.
Regression test acodec-ac3 renamed to acodec-ac3_fixed.
Regression test lavf-rm changed to use ac3_fixed encoder.
Originally committed as revision 26209 to svn://svn.ffmpeg.org/ffmpeg/trunk
This avoids having a large temporary buffer in the struct used for
storing the rtsp reply headers.
Originally committed as revision 26192 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allows ff_rtsp_parse_line to do more changes directly in RTSPState
when parsing the reply, instead of having to store large amounts of
temporary data in RTSPMessageHeader.
Originally committed as revision 26190 to svn://svn.ffmpeg.org/ffmpeg/trunk
Emitted timestamps in each stream start from 0, for the first received
RTP packet. Once an RTCP packet is received, that one is used for
sync, emitting timestamps that fit seamlessly into the earlier ones.
Originally committed as revision 26187 to svn://svn.ffmpeg.org/ffmpeg/trunk
maximum value of 1023.
This speeds up overall encoding depending on the content and bitrate.
The most improvement is with high bitrates and/or low complexity content.
Originally committed as revision 26181 to svn://svn.ffmpeg.org/ffmpeg/trunk
instead of 64. This will change output in some cases, but it happens to not
affect the AC-3 regression tests.
Originally committed as revision 26180 to svn://svn.ffmpeg.org/ffmpeg/trunk
Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot d dot kang
at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26162 to svn://svn.ffmpeg.org/ffmpeg/trunk
wav. In that case, DTS can be transmitted through S/PDIF without
the IEC 61937 headers.
Patch by Anssi Hannula, anssi d hannula a iki d fi
Originally committed as revision 26160 to svn://svn.ffmpeg.org/ffmpeg/trunk
Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot d dot kang
at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26159 to svn://svn.ffmpeg.org/ffmpeg/trunk
Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot d dot kang
at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26158 to svn://svn.ffmpeg.org/ffmpeg/trunk
(authors:Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot
d dot kang at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26157 to svn://svn.ffmpeg.org/ffmpeg/trunk
Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot d dot kang
at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26156 to svn://svn.ffmpeg.org/ffmpeg/trunk
Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot d dot kang
at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26155 to svn://svn.ffmpeg.org/ffmpeg/trunk
(authors: Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot
d dot kang at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26151 to svn://svn.ffmpeg.org/ffmpeg/trunk
(authors: Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot
d dot kang at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26150 to svn://svn.ffmpeg.org/ffmpeg/trunk
(authors: Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot
d dot kang at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26149 to svn://svn.ffmpeg.org/ffmpeg/trunk
(authors: Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot
d dot kang at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26148 to svn://svn.ffmpeg.org/ffmpeg/trunk
(authors: Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot
d dot kang at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26147 to svn://svn.ffmpeg.org/ffmpeg/trunk
(authors: Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot
d dot kang at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26146 to svn://svn.ffmpeg.org/ffmpeg/trunk
(authors: Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot
d dot kang at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26145 to svn://svn.ffmpeg.org/ffmpeg/trunk
This gives slightly better quality in PEAQ tests.
Code 3 gives a dBpb value of 2816 = -132dB (128 psd units = -6dB), which
corresponds to 22 bits. Since the exponents have an offset applied, the
16-bit source looks like 24-bit source to the bit allocation routine.
So using dBpb code=3 is a closer match to the exponent range.
Regression test refs updated for acodec-ac3, lavf-rm, and seek-ac3_rm.
Originally committed as revision 26144 to svn://svn.ffmpeg.org/ffmpeg/trunk
Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot d dot kang
at gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26143 to svn://svn.ffmpeg.org/ffmpeg/trunk
Jason, Loren, Holger) to FFmpeg. Patch by Daniel Kang <daniel dot d dot kang at
gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26142 to svn://svn.ffmpeg.org/ffmpeg/trunk
FFmpeg. Original authors: Holger Lubitz <holger lubitz org>, Jason Garrett-
Glaser <darkshikari gmail com> (approves LGPL relicensing for this code) and
Loren Merritt <lorenm at u dot washington dot edu> (approves LGPL relicensing
for this code). Patch by Daniel Kang <daniel dot d dot kang at gmail com>, as
part of Google's GCI 2010.
Originally committed as revision 26140 to svn://svn.ffmpeg.org/ffmpeg/trunk
FFmpeg. Original authors: Holger Lubitz <holger lubitz org>, Jason Garrett-
Glaser <darkshikari gmail com> (approves LGPL relicensing for this code) and
Loren Merritt <lorenm at u dot washington dot edu> (approves LGPL relicensing
for this code). Patch by Daniel Kang <daniel dot d dot kang at gmail com>, as
part of Google's GCI 2010.
Originally committed as revision 26139 to svn://svn.ffmpeg.org/ffmpeg/trunk
Original authors: Holger Lubitz <holger lubitz org>, Jason Garrett-Glaser
<darkshikari gmail com> (approves LGPL relicensing for this code) and Loren
Merritt <lorenm at u dot washington dot edu> (approves LGPL relicensing for
this code). Patch by Daniel Kang <daniel dot d dot kang at gmail com>, as
part of Google's GCI 2010.
Originally committed as revision 26138 to svn://svn.ffmpeg.org/ffmpeg/trunk
Original authors: Holger Lubitz <holger lubitz org>, Jason Garrett-Glaser
<darkshikari gmail com> (approves LGPL relicensing for this code) and Loren
Merritt <lorenm at u dot washington dot edu> (approves LGPL relicensing for
this code). Patch by Daniel Kang <daniel dot d dot kang at gmail com>, as
part of Google's GCI 2010.
Originally committed as revision 26137 to svn://svn.ffmpeg.org/ffmpeg/trunk
authors: Holger Lubitz <holger lubitz org>, Jason Garrett-Glaser <darkshikari
gmail com> (approves LGPL relicensing for this code) and Loren Merritt <lorenm
at u dot washington dot edu> (approves LGPL relicensing for this code). Patch
by Daniel Kang <daniel dot d dot kang at gmail com>, as part of Google's GCI
2010.
Originally committed as revision 26135 to svn://svn.ffmpeg.org/ffmpeg/trunk
Original authors: Holger Lubitz <holger lubitz org>, Jason Garrett-Glaser
<darkshikari gmail com> (approves LGPL relicensing for this code) and Loren
Merritt <lorenm at u dot washington dot edu> (approves LGPL relicensing for
this code). Patch by Daniel Kang <daniel dot d dot kang at gmail com>, as
part of Google's GCI 2010.
Originally committed as revision 26132 to svn://svn.ffmpeg.org/ffmpeg/trunk
initially said he'd be OK with relicensing, he also said he wanted to have
another look at the patch, and then he went on vacation, so let's play it
safe for now. We can consider removing this again later.
Originally committed as revision 26131 to svn://svn.ffmpeg.org/ffmpeg/trunk
Noticed by CrystalP from XBMC.
Patch by Anssi Hannula, anssi d hannula a iki d fi
Originally committed as revision 26130 to svn://svn.ffmpeg.org/ffmpeg/trunk
evaluated crop area coordinates.
Prevent debug log spamming, define the DEBUG symbol for enabling them.
Originally committed as revision 26110 to svn://svn.ffmpeg.org/ffmpeg/trunk
For example MS-RTSP doesn't have RTPDemuxContexts for all streams.
This fixes issue 2448.
Originally committed as revision 26107 to svn://svn.ffmpeg.org/ffmpeg/trunk
explicit reference to the ffmpeg tool command for keeping the docs
more generic.
Originally committed as revision 26098 to svn://svn.ffmpeg.org/ffmpeg/trunk
LGPL relicensing approved by original authors: Holger Lubitz <holger lubitz
org>, Jason Garrett-Glaser <darkshikari gmail com> and Loren Merritt <lorenm
at u dot washington dot edu>. Patch by Daniel Kang <daniel dot d dot kang at
gmail com>, as part of Google's GCI 2010.
Originally committed as revision 26087 to svn://svn.ffmpeg.org/ffmpeg/trunk
This is optional for encoders, but it's a good idea and has minimal impact
on performance.
This will change the output for some files, but it happens not to affect the
regression tests.
Originally committed as revision 26083 to svn://svn.ffmpeg.org/ffmpeg/trunk
FILTERNAME=ARGS and FILTERNAME:ARGS syntax.
The same filter class will be used for managing all the libopencv
filtering functions.
Originally committed as revision 26079 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes a crash if we requested TCP interleaved transport, but the
server replies with transport data for UDP. According to the RFC, the
server isn't allowed to respond with another transport type than the
one requested.
Originally committed as revision 26077 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes compilation with --disable-everything --enable-<component>,
for all encoders, decoders, muxers, demuxers, parsers, protocols, bsfs,
indevs, outdevs and filters at the moment. (All those that work without
any external dependencies at least.)
Originally committed as revision 26076 to svn://svn.ffmpeg.org/ffmpeg/trunk
This avoids a 16-bit overflow in mdct512() due to a -32768 value in costab.
References updated for acodec-ac3, lavf-rm, and seek-ac3_rm tests.
Thanks to Måns Rullgård for finding the bug.
Originally committed as revision 26071 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fixes AC-3 encoding on OpenBSD 4.8 x86_32 and hopefully other similar
configurations.
Originally committed as revision 26070 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes building with --disable-everything --enable-muxer=matroska and/or
--enable-muxer=webm
Originally committed as revision 26067 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes one of the issues found if building with
--disable-everything --enable-muxer=webm
Originally committed as revision 26066 to svn://svn.ffmpeg.org/ffmpeg/trunk
Since SVN rev 25866, this table is used by the trellis encoder, too,
not only by the decoder.
Originally committed as revision 26065 to svn://svn.ffmpeg.org/ffmpeg/trunk
While not mentioned in RFC 4629, this is required for H.263 in
3GPP TS 26.234. It is in practice required for playback with
Android stagefright and on Samsung bada phones.
Originally committed as revision 26062 to svn://svn.ffmpeg.org/ffmpeg/trunk
commit ab8610aa176c13b5e25a2d8770df28a14665e169
Author: koorogi <koorogi@9553f0bf-9b14-0410-a0b8-cfaf0461ba5b>
Date: Thu Mar 18 23:13:04 2010 +0000
The vflip filter does not need to keep ownership of a picture reference.
Avoid creating and releasing useless extra picture references in the
start_frame and end_frame callbacks.
Originally committed as revision 22596 to svn://svn.ffmpeg.org/ffmpeg/trunk
The new reference creation is required when the linesizes are used by the
previous filters in the filterchain, for example as in "pad=W:H:X:Y,vflip".
Originally committed as revision 26059 to svn://svn.ffmpeg.org/ffmpeg/trunk
Galvão Póvoa <marspeoplester gmail com>, mentored by Robert Swain <robert
dot swain gmail com>.
Originally committed as revision 26051 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allows encoding with lower bitrates by decreasing exponent bits first,
then decreasing bandwidth if the user did not specify a specific cutoff
frequency.
Originally committed as revision 26050 to svn://svn.ffmpeg.org/ffmpeg/trunk
(largest size according to spec: 64k). Fixes playback of
mmsh://a1635.v24937.c2493.g.vm.akamaistream.net/7/1635/2493/v0001/premrad.download.akamai.com/2493/premiere_rock_report/Country_Report.wma
Patch by Zhentan Feng <spyfeng gmail com>.
Originally committed as revision 26047 to svn://svn.ffmpeg.org/ffmpeg/trunk
We can do this because exponents are the only bit allocation parameters which
change from block-to-block currently.
Approx. 57% faster in function bit_alloc().
Approx. 25% faster overall encoding.
Originally committed as revision 26040 to svn://svn.ffmpeg.org/ffmpeg/trunk
allocation for each block.
24% faster in function bit_alloc(). Approx. 10% faster overall encoding.
Originally committed as revision 26039 to svn://svn.ffmpeg.org/ffmpeg/trunk
in encode_exponents_blk_ch() by removing the inner loops. This is about 30-40%
faster for the modified sections.
Originally committed as revision 26036 to svn://svn.ffmpeg.org/ffmpeg/trunk
longer required. This gets rid of the temp buffer as well as encoded_exp in
AC3EncodeContext. It also allows for skipping the exponent grouping for
EXP_D15. 56% faster in encode_exponents_blk_ch().
Originally committed as revision 26034 to svn://svn.ffmpeg.org/ffmpeg/trunk
This macro is disabled if --enable-small or --disable-optimizations are set.
Currently, this leads to warnings about functions being defined but not used,
for functions in header files.
By defining av_always_inline as inline or as av_unused, we avoid these
warnings.
This doesn't make a normal build with --enable-small any larger,
since the compiler probably chooses not to inline these functions even
if they're marked as inline.
Originally committed as revision 26032 to svn://svn.ffmpeg.org/ffmpeg/trunk
This reduces the memory footprint when using less than 6 channels.
Modify bit allocation to swap the 2 buffers instead of using memcpy() and use
per-block pointers for bap. This is slightly faster (0.3%) in function
cbr_bit_allocation().
Originally committed as revision 26023 to svn://svn.ffmpeg.org/ffmpeg/trunk
This also reverts SVN rev 26016, which incorrectly overwrote the time base
with 90 kHz for all streams, regardless of what was set by the SDP parsing.
The stream that triggered the fix in 26016 still works after this commit.
Originally committed as revision 26022 to svn://svn.ffmpeg.org/ffmpeg/trunk
Avoids memcpy that was used to store last samples for next frame.
Approx. 3% faster in function deinterleave_input_samples() and reduces memory
usage by 3kB.
Originally committed as revision 26021 to svn://svn.ffmpeg.org/ffmpeg/trunk
The generic default is 0/0 and that obviously triggers once the value is used.
Originally committed as revision 26016 to svn://svn.ffmpeg.org/ffmpeg/trunk
svq3 still doesn't support multithreading, but it's simpler for clients if
they can enable threading for all codecs by default.
Originally committed as revision 26015 to svn://svn.ffmpeg.org/ffmpeg/trunk
This makes it possible to abort a blocking connect call.
Patch by Thomas Guillem, thomas dot guillem at gmail
Originally committed as revision 26014 to svn://svn.ffmpeg.org/ffmpeg/trunk
Th new function only needs to be called at initialization because bit
allocation parameters currently do not change during encoding.
Originally committed as revision 26003 to svn://svn.ffmpeg.org/ffmpeg/trunk
per-channel exponent strategy decision. This will also make it easier to
plug-in other exponent strategy algorithms.
Originally committed as revision 25995 to svn://svn.ffmpeg.org/ffmpeg/trunk
This reduces the amount of memcpy() by using pointers to overlap samples for
successive blocks rather than copying.
Originally committed as revision 25986 to svn://svn.ffmpeg.org/ffmpeg/trunk
Return AVERROR(EINVAL) instead of -1. Propogate errors from called functions.
Add some error log printouts.
Originally committed as revision 25982 to svn://svn.ffmpeg.org/ffmpeg/trunk
This is an av_cold function, and we don't need to duplicate variables just to
save a few characters.
Originally committed as revision 25979 to svn://svn.ffmpeg.org/ffmpeg/trunk
Turn it into 2 macros, and use av_clip_int16() and lrintf().
This matches the int16 to float sample conversion in audioconvert.c.
The regression test output is different due to lrintf() rounding.
Originally committed as revision 25956 to svn://svn.ffmpeg.org/ffmpeg/trunk
Patch by Nolan L nol888 <=> gmail >=< com.
See thread:
Subject: [FFmpeg-devel] [PATCH] Port gradfun to libavfilter (GCI)
Date: Mon, 29 Nov 2010 07:18:14 -0500
Originally committed as revision 25942 to svn://svn.ffmpeg.org/ffmpeg/trunk
transcoding.
Fix issue #2292.
Patch sponsored by KIM Keep In Mind GmbH, srl.
Originally committed as revision 25939 to svn://svn.ffmpeg.org/ffmpeg/trunk
(without buffering extra input).
Patch by Michael Chinen [mchinen at gmail]
Originally committed as revision 25932 to svn://svn.ffmpeg.org/ffmpeg/trunk
of returning packets with uninitialized data.
Returning partial packets as for other demuxers is problematice due to
packet scrambling and thus is not done.
Originally committed as revision 25931 to svn://svn.ffmpeg.org/ffmpeg/trunk
ID3v2 tags which no longer works since ID3v2 handling was moved to
generic code.
In addition, in caused false-positives for all files starting with
one or more 0-bytes.
Originally committed as revision 25929 to svn://svn.ffmpeg.org/ffmpeg/trunk
fftools-common-opts.texi.
Fix pod files updating when fftools-common-opts.texi is changed.
Originally committed as revision 25928 to svn://svn.ffmpeg.org/ffmpeg/trunk
libavcodec to libavcore.
Remove another compile-time dependancy of libavfilter on libavcodec.
Originally committed as revision 25923 to svn://svn.ffmpeg.org/ffmpeg/trunk
Test programs may be named either %.c or %-test.c. In some cases (ie. fft),
both files exist and make has to choose which rule to use. Up to gnu make
3.81, rules were applied in definition order, in this case favouring the
generic %.c rule from common.mak. gnu make 3.82 uses a shortest stem first
order, applying definition order for same length stems. In this case,
%-test.c must come first in order for make to pick the proper source file.
Originally committed as revision 25918 to svn://svn.ffmpeg.org/ffmpeg/trunk
rgb32tobgr32() has been removed in favour of shuffle_bytes_2103() in r32190
Originally committed as revision 32676 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
data thanks to the recently added FLAC parser.
Patch by Michael Chinen [mchinen at gmail]
Originally committed as revision 25915 to svn://svn.ffmpeg.org/ffmpeg/trunk
Seek test reference updated because FLAC seeking now works properly.
Fixes roundup issue 1150.
Patch by Michael Chinen [mchinen at gmail]
Originally committed as revision 25914 to svn://svn.ffmpeg.org/ffmpeg/trunk
to optionally silence the error messages.
Patch by Michael Chinen [mchinen at gmail]
Originally committed as revision 25912 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes cases where the RTP time base and the sample rate of the stream
differ. Previously, the AVStream time_base was unconditionally set to
the sample rate (which initially was set to one value when parsing the
rtpmap field in the SDP, but later overridden by an a=SampleRate field).
Additionally, this makes the code actually use the stream time base set
in rtpmap for video codecs, instead of hardcoding it to always be 90 kHz.
Originally committed as revision 25908 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes playback of a RealRTSP/MP3 URL from the RTSP samples on
MultimediaWiki.
Originally committed as revision 25906 to svn://svn.ffmpeg.org/ffmpeg/trunk
than just per-block. Patch by Sprezz [sprezzatura gmx com]. Fixes Issue 2387.
Originally committed as revision 25898 to svn://svn.ffmpeg.org/ffmpeg/trunk
According to the spec, this type shouldn't ever be used. Nevertheless,
passing it through enables decoding streams which otherwise aren't
decodeable.
Originally committed as revision 25897 to svn://svn.ffmpeg.org/ffmpeg/trunk
The current implementation is incompatible with the latest spec drafts,
this should be communicated clearly to the user.
Originally committed as revision 25887 to svn://svn.ffmpeg.org/ffmpeg/trunk
I dont know if this is the best way to handle it. But it fixes http://kuwatan.jp/temp/n-02b.3gp
Fixes issue 2373.
Originally committed as revision 25875 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes PCR drift due to accumulating TS_PACKET_SIZE*8*90000LL/ts->mux_rate each packet, due to rounding errors when mux_rate does not evenly divide 135360000.
This patch also increases the PCR precision to 27 MHz from 90 kHz and takes the location of the PCR data into account (+11 bytes according to the spec).
Originally committed as revision 25864 to svn://svn.ffmpeg.org/ffmpeg/trunk
Wraparound in ssd is mainly avoided by subtracting the ssd of the
best node from all the others once it has grown large enough.
If using very large trellis sizes (e.g. -trellis 15), the frontier
is so large that the difference between the best and the worst is
large enough to cause wraparound, even if the ssd of the best one
is subtracted regularly.
When using -trellis 10 on a 30 second sample, this causes only a slight
slowdown, from 61 to 64 seconds.
Originally committed as revision 25858 to svn://svn.ffmpeg.org/ffmpeg/trunk
Make it accept the frei0r=filtername=params as alternative to
frei0r=filtername:params.
Originally committed as revision 25856 to svn://svn.ffmpeg.org/ffmpeg/trunk
data packets before the first complete one.
Patch by Aaron Colwell [acolwell chromium org].
Originally committed as revision 25846 to svn://svn.ffmpeg.org/ffmpeg/trunk
This reverts snprintf->av_d2str part of r20840.
With it, track number is exported as a float, which is not
desirable.
Originally committed as revision 25845 to svn://svn.ffmpeg.org/ffmpeg/trunk
This works at least for some people testing it.
Patch by Anssi Hannula, anssi d hannula a iki fi
Originally committed as revision 25834 to svn://svn.ffmpeg.org/ffmpeg/trunk
Compute the outpicref PTS in start_frame(), since it is required for
comparing that value with the value of the overpicref.
Originally committed as revision 25828 to svn://svn.ffmpeg.org/ffmpeg/trunk
yet have it, this can happen if the pes stream contain data but no pes header
has been encoutered yet.
Patch by Stephen D'Angelo, SDAngelo at evertz dot com.
Originally committed as revision 25823 to svn://svn.ffmpeg.org/ffmpeg/trunk
Title is already written into special fields designated for it --
per-segment Title, per-chapter ChapString and per-stream Name (yay for
consistent naming). Therefore, don't duplicate it in Tags.
Originally committed as revision 25813 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fix issue #2269.
Interpolate pts for the second frame when using field output mode.
Originally committed as revision 25798 to svn://svn.ffmpeg.org/ffmpeg/trunk
Patch by Mark Goodman [mark goodman gmail com] with some modifications by me.
Originally committed as revision 25796 to svn://svn.ffmpeg.org/ffmpeg/trunk
The new implementation is more compact, more correct and doesn't hurt
the eyes.
Originally committed as revision 25792 to svn://svn.ffmpeg.org/ffmpeg/trunk
so extend decoder to output only one channel for it.
This fixes issue 2368.
Originally committed as revision 25790 to svn://svn.ffmpeg.org/ffmpeg/trunk
This also allows to remove a linking dependency of libavfilter on
libavcodec.
Originally committed as revision 25789 to svn://svn.ffmpeg.org/ffmpeg/trunk
const char *avcodec_get_channel_name(int channel_id)
which was never implemented.
Originally committed as revision 25788 to svn://svn.ffmpeg.org/ffmpeg/trunk
PaletteControl.
This also fixes playback of some files with ffplay (images were
corrupted for a short time after a palette change).
Originally committed as revision 25778 to svn://svn.ffmpeg.org/ffmpeg/trunk
This makes the wording consistent with how people usually talk about heaps.
Originally committed as revision 25775 to svn://svn.ffmpeg.org/ffmpeg/trunk
This increases the PSNR slightly (about 0.1 dB) for trellis sizes
below 8, and gives equal PSNR for sizes above that.
Originally committed as revision 25769 to svn://svn.ffmpeg.org/ffmpeg/trunk
This lowers the run time from 158 to 21 seconds, for -trellis 8
with a 30 second sample on my machine.
This requires 64 KB additional memory.
Originally committed as revision 25768 to svn://svn.ffmpeg.org/ffmpeg/trunk
Only works via HDMI.
Patch by Anssi Hannula (anssi d hannula a iki d fi), based on some work
by myself.
Originally committed as revision 25760 to svn://svn.ffmpeg.org/ffmpeg/trunk
beginning of the frame, so make it use skip_bits_long() instead of
skip_bits() for that.
Originally committed as revision 25754 to svn://svn.ffmpeg.org/ffmpeg/trunk
AV_PERM_REUSE and AV_PERM_REUSE2 flags, avoid conflict with the char
'r' used for AV_PERM_READ.
Originally committed as revision 25739 to svn://svn.ffmpeg.org/ffmpeg/trunk
ff_avfilter_graph_check_validity()
ff_avfilter_graph_config_links()
ff_avfilter_graph_config_formats()
and move their declaration to internal.h. These functions are never
used in application code, so it is better to consider them internal
functions, this can be changed later if necessary. Simplify API.
Originally committed as revision 25737 to svn://svn.ffmpeg.org/ffmpeg/trunk
By not looking for the exactly largest node, we avoid an O(n) seek through
the leaf nodes. Just pick one (not the same one every time) and try replacing
that node with the new one.
For -trellis 8, this lowers the run time from 190 to 158 seconds,
for a 30 second 44 kHz mono sample, on my machine.
Originally committed as revision 25733 to svn://svn.ffmpeg.org/ffmpeg/trunk
This avoids having to memmove the large parts of the array when inserting into
it.
For -trellis 8, this lowers the run time from 245 seconds to 190 seconds,
for a 30 second 44 kHz mono sample, on my machine.
Originally committed as revision 25731 to svn://svn.ffmpeg.org/ffmpeg/trunk
An intermediate value in the floor 1 linear interpolation was
overflowing
resulting in obvious artifacts on some files.
e.g.
http://upload.wikimedia.org/wikipedia/commons/7/79/Big_Buck_Bunny_small.ogv
Prior to this fix 87 out of 128 64kbit/s mono files decoded with ffmpeg
have
lower PEAQ ODG values than the same files decoded with libvorbis. With
this
fix none of that set have significantly worse ODG values than libvorbis.
Fixes issue 2352
Patch by Gregory Maxwell <greg@xiph.org>
Originally committed as revision 25724 to svn://svn.ffmpeg.org/ffmpeg/trunk
Nobody ever uses it correctly, and ffmpeg sets it incorrectly, so we'll just
leave it out.
Originally committed as revision 25720 to svn://svn.ffmpeg.org/ffmpeg/trunk
descriptors for printing the number of channels/components.
Also replace the term "nb_channels" with "nb_components" which is more
consistent with the FFmpeg internal terminology, and is somehow
different with respect to the current definition of nb_channels in
PixFmtInfo.
See thread:
Subject: [FFmpeg-devel] [PATCH 6/8] Make avcodec_pix_fmt_string() use the
information in the pixel format descriptors for printing the
number of planes. Also replace the term "nb_channels" with
"nb_planes" which is more correct.
Date: Fri, 5 Nov 2010 12:01:38 +0100
Originally committed as revision 25717 to svn://svn.ffmpeg.org/ffmpeg/trunk
files spdif.h and spdif.c.
Patch by Anssi Hannula, anssi d hannula a iki d fi
Originally committed as revision 25715 to svn://svn.ffmpeg.org/ffmpeg/trunk
Based on a patch by Nicolas George
<@var{name}.@var{surname}@@normalesup.@code{org}>.
Originally committed as revision 25709 to svn://svn.ffmpeg.org/ffmpeg/trunk
eval API.
More grep-friendly and more consistent with the rest of the FFmpeg
API.
Originally committed as revision 25708 to svn://svn.ffmpeg.org/ffmpeg/trunk
The new name is shorter and more consistent with the FFmpeg API, and
sounds less evil.
Originally committed as revision 25707 to svn://svn.ffmpeg.org/ffmpeg/trunk
avfilter_graph_parse() declaration to libavfilter/avfiltergraph.h.
Simplify, and less confusing for the user.
Originally committed as revision 25705 to svn://svn.ffmpeg.org/ffmpeg/trunk
This new version:
1. Works on 24-bit and 32-bit input, not just 32-bit.
2. Doesn't try to run on 16-bit or 8-bit, instead of outright crashing.
3. Does proper alpha-blending, so cursor shadows look correct.
4. Doesn't swap R and B.
Mostly fixes issue 1997.
Fixes issue 2056.
Originally committed as revision 25690 to svn://svn.ffmpeg.org/ffmpeg/trunk
Muxlength does not include the 3 bytes of AudioSyncStream() before the
AudioMuxElement(). If these three bytes are not accounted for the last three
bytes of the LATM packet are sent back to the decoder again.
Fixes issue244/mux2.share.ts
Originally committed as revision 25685 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes a regression since svn rev 24752, where the real_setup pointer
was set incorrectly. The arithmetic with the real_setup_cache pointer
is in units of enum AVDiscard, so the sizeof multiplication should be removed.
Patch by Yuriy Kaminskiy, yumkam at mail dot ru.
Originally committed as revision 25684 to svn://svn.ffmpeg.org/ffmpeg/trunk
the object number is, it determines whether we should continue
parsing the presentation description and whether we should
clear the subtitles on the next display command.
Based on patch by Mark Goodman [mark goodman gmail com]
Originally committed as revision 25682 to svn://svn.ffmpeg.org/ffmpeg/trunk
Previously it was releasing the buffer which was returned to the user,
which was resulting in a crash in case of direct rendering.
Originally committed as revision 25678 to svn://svn.ffmpeg.org/ffmpeg/trunk
Spotted by Alex after Carl Eugen found errors some samples. There no errors or
noticeable artifacts in the samples I used during development.
Originally committed as revision 25676 to svn://svn.ffmpeg.org/ffmpeg/trunk
The new name is shorter and more consistent with the rest of the API.
This change breaks libavfilter API/ABI.
Originally committed as revision 25674 to svn://svn.ffmpeg.org/ffmpeg/trunk
Contrary to progressive, just being able to crop up to 14/15 pixels
is not enough to encode all supported resolutions, and the new
behaviour is also consistent with e.g. MPEG-2 etc.
Originally committed as revision 25669 to svn://svn.ffmpeg.org/ffmpeg/trunk
av_get_sample_fmt_name()
av_get_sample_fmt()
av_get_sample_fmt_string()
in libavcore, and deprecate the corresponding libavcodec/audioconvert.h functions:
avcodec_get_sample_fmt_name()
avcodec_get_sample_fmt()
avcodec_sample_fmt_string()
Originally committed as revision 25653 to svn://svn.ffmpeg.org/ffmpeg/trunk
The timestamps for all the buffered frames are scaled against
AV_TIME_BASE_Q, and need to be scaled back to AV_TIME_BASE_Q when they
are extracted from the filterchain.
Originally committed as revision 25646 to svn://svn.ffmpeg.org/ffmpeg/trunk
The decoder is just a wrapper around the AAC decoder.
based on patch by Paul Kendall { paul <ät> kcbbs gen nz }
Originally committed as revision 25642 to svn://svn.ffmpeg.org/ffmpeg/trunk
Use avctx in all called functions. This allows passing a NULL AACContext
for LATM since the AACContext is only used in output_configure() which
is skipped for LATM parsing.
Originally committed as revision 25641 to svn://svn.ffmpeg.org/ffmpeg/trunk
aac_decode_frame() remains as AVPacket handling a wrapper. The actual
decoding function takes a GetBitContext as input and will be used be the
AAC LATM decoder to avoid copying the unaligned AAC bitstream.
Originally committed as revision 25640 to svn://svn.ffmpeg.org/ffmpeg/trunk
This will be used by the latm decoder to avoid AACContext changes during
audio specific config parsing.
Originally committed as revision 25638 to svn://svn.ffmpeg.org/ffmpeg/trunk
ages.
The user is requested to specify "-f image2" in place of "-f pgmyuv"
for reading/writing PGMYUV files, as for the other image formats.
Originally committed as revision 25629 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes compilation with the latest clang trunk version.
Patch by İsmail Dönmez, ismail at namtrac dot org
Originally committed as revision 25628 to svn://svn.ffmpeg.org/ffmpeg/trunk
prefix does not contain other characters which may belong to an
identifier.
This allows to distinguish for example to have different constants
with the same prefix (e.g. "foo" and "foobar").
Originally committed as revision 25626 to svn://svn.ffmpeg.org/ffmpeg/trunk
The 3GPP spec uses the following calculation for high spreading:
thr'_spr = max(thr_scaled, s_h(n) * thr_scaled(n-1))
where, n is defined as the current band, and s_h() is defined as "[...] the
distance of adjacent bands in Bark and a constant slope that is 15 dB/Bark
[...]". This is a little ambiguous as you would assume you want the Bark
width of the previous band for this calculation. However, this assumption
appears to be incorrect, and you really want the Bark width of the current
band. Coincidentally this is exactly what the spec calls for! =P
This noticeably improves Tom's Diner at low bitrates (I tested at 64kbps,
with mid/side disabled).
Patch by: Nathan Caldwell <saintdev@gmail.com>
Originally committed as revision 25622 to svn://svn.ffmpeg.org/ffmpeg/trunk
These blocks depended on the compiler keeping xmm registers untouched between
them.
Originally committed as revision 25619 to svn://svn.ffmpeg.org/ffmpeg/trunk
suncc does not like the leading commas inside the macro, but it has no problem
with trailing commas.
Originally committed as revision 25615 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fixes issue 2317.
Patch by Rocky Cardwell, rocky d cardwell a lifespringschool d org
Originally committed as revision 25603 to svn://svn.ffmpeg.org/ffmpeg/trunk
handle_jpeg may update the src/dstFormat variables, this makes sure the
updated version is stored in the context.
This fixes roundup issue 2302.
Patch by Troot, all_crap_goes_here at hotmail
Originally committed as revision 32562 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
This greatly improves bitrate handling. You will now get within a few
kbps of your requested bitrate instead of 20-40kbps higher.
There is absolutely no analog to this line in the 3GPP spec, that I
can find.
patch by Nathan Caldwell saintdev (at) gmail
Originally committed as revision 25589 to svn://svn.ffmpeg.org/ffmpeg/trunk
Removing the modification vastly improves quality (at a slight bitrate
cost) for some samples. castanets.wav is a good example. The closest
equivalent I see to the modification in the 3GPP spec is a similar
modification (over a specific frequency range) when TNS is used.
This also changes the threshold-in-quiet calculation to match the
3GPP spec.
patch by Nathan Caldwell saintdev (at) gmail
Originally committed as revision 25588 to svn://svn.ffmpeg.org/ffmpeg/trunk
According to the 3GPP spec:
"Thus the pre-echo control is inactive for the first short window (but
not all short windows in a short frame) after a start block and for
all frames with a stop window sequence."
Currently, pre-echo control is only run when the current frame is not
a short frame, and the previous frame is not a short frame.
patch by Nathan Caldwell saintdev (at) gmail
Originally committed as revision 25587 to svn://svn.ffmpeg.org/ffmpeg/trunk
This may be needed to avoid calls to implicitly defined functions
(that will be removed by dead code elimination later anyway).
Originally committed as revision 25585 to svn://svn.ffmpeg.org/ffmpeg/trunk
Reading 7 bits as an unsigned int can't result in a value exceeding 127.
Accordingly, remove error message (as it'll never be reached).
Originally committed as revision 25575 to svn://svn.ffmpeg.org/ffmpeg/trunk
Bug caused by the fact that get_bits(gb, 0) is undefined.
Doesn't affect any streams generated by the official Theora encoder, but such
streams are nevertheless valid.
Fixes decoding of CELT-933dd833-nmr-bandt.ogv.
Originally committed as revision 25573 to svn://svn.ffmpeg.org/ffmpeg/trunk
Do decode init in the init function instead of at the first frame.
Fix some possible crash cases.
Originally committed as revision 25572 to svn://svn.ffmpeg.org/ffmpeg/trunk
Some code was initializing some xmm registers in one asm block and using them
in the following block, assuming they wouldn't be changed in between blocks.
Originally committed as revision 25568 to svn://svn.ffmpeg.org/ffmpeg/trunk
thus making forced key frames work.
Patch by Nicolas George, nicolas d george a normalesup d org
Originally committed as revision 25567 to svn://svn.ffmpeg.org/ffmpeg/trunk
I used the same loop counter for the inner and outer initalization loops.
This caused initalization to only run for the first channel. This in turn lead
to any channel other than the first using only short blocks.
Patch by Nathan Caldwell, saintdev at gmail
Originally committed as revision 25566 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes a possibly exploitable buffer overflow and it will likely also be needed for future overreading fixes.
Originally committed as revision 25546 to svn://svn.ffmpeg.org/ffmpeg/trunk
The 8 bits offset (nal unit type) should not be added, as the spec says:
"This bit offset is the offset within the RBSP data for the slice, relative
to the starting position of the slice_header() in the RBSP"
This fixes DXVA2 support for intel GPU.
Patch by Rafaël Carré (funman _AT_ videolan _DOT_ org).
Originally committed as revision 25538 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allows compilation of one of them without requiring the others'
dependencies to be present.
Originally committed as revision 25535 to svn://svn.ffmpeg.org/ffmpeg/trunk
The demuxer inspects the payload type of a received RTP packet and
handles the cases where the content is fully described by the payload type.
Originally committed as revision 25527 to svn://svn.ffmpeg.org/ffmpeg/trunk
The option is useful to ensure that there is a seek point exactly at a
place the user will probably want to jump precisely sometime, the
major example would be the end of an opening and the beginning of a
chapter. The scene change detection system will often make it happen,
but not always for example if there is a fade-in.
See the thread:
Subject: [FFmpeg-devel] [PATCH] -force_key_frames option
Date: Tue, 12 Oct 2010 15:16:26 +0200
Patch by Nicolas George -mail nicolas,george,normalesup,org.
Originally committed as revision 25526 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fixes a scr issue reported with dvdauthor ([FFmpeg-user] FFMPEG encoded MPEG-2 video causes error in DVDAuthor)
Originally committed as revision 25512 to svn://svn.ffmpeg.org/ffmpeg/trunk
deprecated field AVStream.language.
Patch by Anton Khirnov %name%@%surname% dot net.
Originally committed as revision 25507 to svn://svn.ffmpeg.org/ffmpeg/trunk
greatly simplifying its handling and reducing code duplication.
Patch by Nicolas George -bsf nicolas*george|normalesup+org.
Originally committed as revision 25501 to svn://svn.ffmpeg.org/ffmpeg/trunk
code flow, in the new_video_stream() / new_audio_stream() /
new_subtitle_stream() functions.
Patch by Nicolas George <$name.$surname@normalesup.org>.
Originally committed as revision 25500 to svn://svn.ffmpeg.org/ffmpeg/trunk
This indicates that there was no error that needs to be reported to the
caller, so we can move on to parse the next packet immediately, if
available. The only error code that ff_mpegts_parse_packet can return
indicates that there was no packet to return from the provided data, for
which it returns -1.
Originally committed as revision 25496 to svn://svn.ffmpeg.org/ffmpeg/trunk
It may have returned a negative number for an error (e.g. AVERROR(EAGAIN),
if more data is required for it to be able to return a complete packet).
Originally committed as revision 25458 to svn://svn.ffmpeg.org/ffmpeg/trunk
of a filter for setting the output timebase, if there is such a link,
rather than always use AV_TIME_BASE_Q.
This fixes configuration for all the filters which do not use the
default config_props for the output link, and do not set explicitely
the timebase.
Originally committed as revision 25456 to svn://svn.ffmpeg.org/ffmpeg/trunk
filterchain.
Set timebase in the ffplay input, and make get_filtered_video_frame()
rescale the output frames PTSes according to their timebase.
Originally committed as revision 25450 to svn://svn.ffmpeg.org/ffmpeg/trunk
This is required for allowing a filter to use a time base different
from AV_TIME_BASE_Q, as it was previously assumed.
Originally committed as revision 25441 to svn://svn.ffmpeg.org/ffmpeg/trunk
The rm demuxer has timestamp bugs, so this test is sensitive to changes in
timestamp correction. The previous commit did not make output any better or worse
on this test, just different.
See https://roundup.ffmpeg.org/issue2288 for details.
Originally committed as revision 25432 to svn://svn.ffmpeg.org/ffmpeg/trunk
Functions from these files are referenced from utils.c now. This fixes
building with --disable-everything.
Originally committed as revision 25425 to svn://svn.ffmpeg.org/ffmpeg/trunk
This makes the URL handling consistent - the URL hostname is used as
destination for the RTP packets, so the URL port should also belong to that.
Now the options announce_addr= and announce_port specify where the
announcements are sent.
Originally committed as revision 25424 to svn://svn.ffmpeg.org/ffmpeg/trunk
The SAP muxer uses network functions directly, uses the rtp protocol and the
udp protocol. The network functions and the udp protocol are both transitively
enabled via the rtp protocol, so that's the only dependency that needs to
be stated.
Originally committed as revision 25423 to svn://svn.ffmpeg.org/ffmpeg/trunk
The new object file is added to the SDP demuxer in the makefile, since it
is needed in both the RTSP muxer and demuxer and in the SDP demuxer, due
to the current code coupling.
Originally committed as revision 25410 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allows us to find out the local sending address via getsockname,
otherwise we just get INADDR_ANY (or similar for v6).
This also makes writes return ECONNREFUSED if nobody is listening on the
receiving port.
Originally committed as revision 25405 to svn://svn.ffmpeg.org/ffmpeg/trunk
There was an off-by-one when addressing bitstream_filters[] in this situation.
Originally committed as revision 25398 to svn://svn.ffmpeg.org/ffmpeg/trunk
hip_decode_init() isn't used by the wrapper, and can be absent from the
library if lame was configured with --disable-decoder (like it is done
for windows builds of VLC)
Unfortunately this also removes the way to detect a libmp3lame more recent
than the buggy 3.98.2
Patch by Rafaël Carré, rafael carre at gmail
Originally committed as revision 25391 to svn://svn.ffmpeg.org/ffmpeg/trunk
Not all users of this function will have a full struct sockaddr_storage
available, and casting other sockaddrs to sockaddr_storage is wrong,
while any sockaddr can be cast to a base sockaddr.
Originally committed as revision 25388 to svn://svn.ffmpeg.org/ffmpeg/trunk
The header probably isn't the right permanent place for this function, but
it is quite small, and consensus seems to be that it can stay in the
header for now, instead of creating a new file network.c just for this one.
Originally committed as revision 25387 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes roundup issue 2270.
Patch by Robert Schlabbach, robert_s at gmx dot net
Originally committed as revision 25372 to svn://svn.ffmpeg.org/ffmpeg/trunk
On OS X, this struct is only available if _DARWIN_C_SOURCE is defined,
when we have _POSIX_C_SOURCE defined. This allows the struct to be found,
enabling proper IPv6 multicast functionality on OS X.
The define is already set within the file that uses the struct. Setting it
only for this test in configure avoids having to keep it defined for the
whole build.
Originally committed as revision 25370 to svn://svn.ffmpeg.org/ffmpeg/trunk
prediction (plus some with different rounding for svq3/rv40). Speedup (for
SSSE3) about ~6-fold, 3.6% faster overall with cathedral sample.
Originally committed as revision 25361 to svn://svn.ffmpeg.org/ffmpeg/trunk
If multiple stsd atoms are parsed for the same stream, the old
extradata would be leaked.
Originally committed as revision 25360 to svn://svn.ffmpeg.org/ffmpeg/trunk
This makes e.g. --disable-everything --enable-demuxer=rtsp actually
enable the rtsp demuxer and everything else that is needed.
Originally committed as revision 25356 to svn://svn.ffmpeg.org/ffmpeg/trunk
This makes the code dependencies correct. Previously, the SDP demuxer
wasn't buildable on its own.
This also reverts rev 25343.
Originally committed as revision 25354 to svn://svn.ffmpeg.org/ffmpeg/trunk
prevents a read-after-close-induced segfault later. Fixes issue 2266.
Patch by qrtt1 <chingyichan dot tw gmail com>.
Originally committed as revision 25349 to svn://svn.ffmpeg.org/ffmpeg/trunk
They reside within a large CONFIG_RTSP_DEMUXER block and are thus pointless.
Originally committed as revision 25343 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fixes compilation with clang's builtin assembler
Patch by İsmail Dönmez, ismail at namtrac dot org
Originally committed as revision 25331 to svn://svn.ffmpeg.org/ffmpeg/trunk
It is only useful for debugging, so it doesn't have to be shown every time.
Originally committed as revision 25323 to svn://svn.ffmpeg.org/ffmpeg/trunk
but worse it did not set up destination dimensions, thus every user
of it would necessarily fail.
Originally committed as revision 32424 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
that flag to the dca codec. This capability when set
will make sure the codec will initialize the channel
configuration instead of trusting the container. This
fixes issue 2137 and issue 850.
Originally committed as revision 25320 to svn://svn.ffmpeg.org/ffmpeg/trunk
and add a test for regular GSM as fate-gsm.
Fixes a 8kHz sample from issue 113.
Originally committed as revision 25313 to svn://svn.ffmpeg.org/ffmpeg/trunk
bitstream endianness, so they cannot reuse the same code
without a lot of function call overhead.
Thus use templating to compile two different binaries.
Originally committed as revision 25312 to svn://svn.ffmpeg.org/ffmpeg/trunk
between ffmpeg and ffplay and avoids a valgrind error by freeing
avformat_opts->key.
Originally committed as revision 25309 to svn://svn.ffmpeg.org/ffmpeg/trunk
FRAPS appears to output PC range streams. I can't guarantee this is true of
all versions.
Originally committed as revision 25308 to svn://svn.ffmpeg.org/ffmpeg/trunk
Do the same change for ff_rdt_parse_packet, too, to keep the interfaces
similar.
Originally committed as revision 25289 to svn://svn.ffmpeg.org/ffmpeg/trunk
With this the developer can now choose if he wants an assert always enabled or at which
compile time assert level. This can thus replace the #define NDEBUG hacks
Originally committed as revision 25278 to svn://svn.ffmpeg.org/ffmpeg/trunk
(576i50 25Mbps 4:1:1 special case was wrong).
Fixes issue2211
Patch by Niobos, niobos dest-unreach be
Originally committed as revision 25269 to svn://svn.ffmpeg.org/ffmpeg/trunk
rather than av_register_protocol(), which is deprecated.
Fix the GCC warning:
avio.c: In function ‘register_protocol’:
avio.c:93: warning: ‘av_register_protocol’ is deprecated (declared at avio.c:86)
Originally committed as revision 25267 to svn://svn.ffmpeg.org/ffmpeg/trunk
scale context. Prevent pointless warnings when using
av_opt_set_defaults() for setting the default values, as in a pending
patch.
Originally committed as revision 32413 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
inline asm works for gcc-3.x also (hopefully). Should fix gcc-3.x FATE
breakage after r25254.
Originally committed as revision 25262 to svn://svn.ffmpeg.org/ffmpeg/trunk
current code wouldn't always error out on errors.
Based on patch by Stephen d'Angelo <sdangelo evertz com>.
Originally committed as revision 25260 to svn://svn.ffmpeg.org/ffmpeg/trunk
increase to e.g. vc1, snow and mpeg decoding.
Patch by Eli Friedman <eli dot friedman gmail com>.
Originally committed as revision 25259 to svn://svn.ffmpeg.org/ffmpeg/trunk
That way the user app can set codec specific parameters in the private context
before opening it.
Originally committed as revision 25257 to svn://svn.ffmpeg.org/ffmpeg/trunk
from memory locations/offsets depending on b_idx plus constants, rather than
having gcc do this. This saves several lea calls and together saves about
10 cycles in h264_loop_filter_strength_mmx2().
Originally committed as revision 25256 to svn://svn.ffmpeg.org/ffmpeg/trunk
a pxor, or remove the instruction alltogether. Altogether, this saves 1
instruction.
Originally committed as revision 25255 to svn://svn.ffmpeg.org/ffmpeg/trunk
This has no measurable speed effect because the surrounding code doesn't
take advantage of this yet.
Originally committed as revision 25254 to svn://svn.ffmpeg.org/ffmpeg/trunk
of the d_idx variable and therefore allows for future optimizations. No speed
difference by this commit itself.
Originally committed as revision 25253 to svn://svn.ffmpeg.org/ffmpeg/trunk
inlining various constants within the loop code. 20 cycles faster on
cathedral sample.
Originally committed as revision 25252 to svn://svn.ffmpeg.org/ffmpeg/trunk
yadif.c:226: error: can't find a register in class 'GENERAL_REGS' while reloading 'asm'
yadif.c:220: error: 'asm' operand has impossible constraints
Patch by Alexander Strange <astrange ithinksw com>.
Originally committed as revision 25251 to svn://svn.ffmpeg.org/ffmpeg/trunk
r25218 made assumptions about the existence of past reference frames that
weren't necessarily true.
Originally committed as revision 25243 to svn://svn.ffmpeg.org/ffmpeg/trunk
Because DTS values aren't passed through decoders, they tend to be
inaccurate if decoder delay doesn't match what was expected by the encoder.
In particular this improves timestamps for H.264 without num_reorder_frames
set and with -strict 1, which causes DTS to be up to 16 frames ahead of the
picture.
Note that this doesn't really improve any file with very broken PTS/DTS,
since PTS isn't much more accurate in these.
Originally committed as revision 25242 to svn://svn.ffmpeg.org/ffmpeg/trunk
It was being set wrong for files with data_offset > 0
Patch by Michael Chinen, mchinen gmail
Originally committed as revision 25239 to svn://svn.ffmpeg.org/ffmpeg/trunk
Options are not yet set for video encoders and for codec specific
defaults it is desireable to set the codec options only after the codec
is known.
Originally committed as revision 25230 to svn://svn.ffmpeg.org/ffmpeg/trunk
See thread:
Subject: [FFmpeg-devel] [PATCH] Port MPlayer blackframe filter.
Date: Sun, 26 Sep 2010 01:10:40 +0200
Originally committed as revision 25214 to svn://svn.ffmpeg.org/ffmpeg/trunk
This prevents gcc inserting useless UXTH instructions, at least
in some cases.
Originally committed as revision 25212 to svn://svn.ffmpeg.org/ffmpeg/trunk
instead of requireing being passed through function parameters. This also
makes sws work with AVOptions.
Originally committed as revision 32368 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
r25185 | stefano | 2010-09-25 03:18:43 +0200 (Sat, 25 Sep 2010) | 1 line
Make the crop filters accept parametric expressions.
Originally committed as revision 25193 to svn://svn.ffmpeg.org/ffmpeg/trunk
Slightly more intuitive and required by a pending changes for making
the filter parametric.
Originally committed as revision 25184 to svn://svn.ffmpeg.org/ffmpeg/trunk
This can currently also be set via -muxdelay in ffmpeg for muxers,
but not for demuxers (nor for demuxers in ffplay) - this patch
allows it to be set in all those cases.
Originally committed as revision 25180 to svn://svn.ffmpeg.org/ffmpeg/trunk
inlines scan8[] and removes loop setup. 15% faster, 0.4% overall.
See "[PATCH] unroll loop in h264_idct_add8_sse2()" thread on ML.
Originally committed as revision 25172 to svn://svn.ffmpeg.org/ffmpeg/trunk
code directly also and remove loop setup. 20% faster in function, 0.8% overall.
See "[PATCH] unroll loop in h264_idct_add8_sse2()" thread on ML.
Originally committed as revision 25171 to svn://svn.ffmpeg.org/ffmpeg/trunk
- mention clobber-marking of xmm registers,
- some notes on external vs. inline asm, including tips on which to use for
what situation and to not rewrite+improve in the same patch (as with C code)
- some more best-practice guidelines
See "[PATCH] update doc/optimization.txt" thread on ML.
Originally committed as revision 25170 to svn://svn.ffmpeg.org/ffmpeg/trunk
See thread:
Subject: [FFmpeg-devel] [POC] frei0r wrapper
Date: Tue, 24 Aug 2010 21:37:32 +0200
Originally committed as revision 25165 to svn://svn.ffmpeg.org/ffmpeg/trunk
ff_asf_extended_content_header.
Fixes issue 690.
Patch by Richard Buteau, rbuteau rgbnetworks com
Originally committed as revision 25158 to svn://svn.ffmpeg.org/ffmpeg/trunk
but it used to do it only for h264 codec.
Allow it for other codecs, as mpeg2 and mpeg4 also set this flag.
Originally committed as revision 25156 to svn://svn.ffmpeg.org/ffmpeg/trunk
On windows, a connection failure doesn't trigger wfds as it does on unix.
This fixes issue 2237, based on code by yeyingxian.
Originally committed as revision 25154 to svn://svn.ffmpeg.org/ffmpeg/trunk
This is set to 1 if the stack is guaranteed to be suitably aligned
for the strictest access mode of the machine.
Originally committed as revision 25151 to svn://svn.ffmpeg.org/ffmpeg/trunk
Apparently Apple platforms do not handle movw/movt relocations
properly, leading to runtime crashes in code using them.
Originally committed as revision 25150 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes building on FreeBSD in some configurations, if the IPv6 multicast
structs are available, but IPPROTO_IPV6 isn't defined.
Originally committed as revision 25147 to svn://svn.ffmpeg.org/ffmpeg/trunk
Since we now have the official x264 presets in ffmpeg, there's no reason to
keep around the old ones.
Patch by Lou Logan <lou AT fakeoutdoorsman DOT com>.
Originally committed as revision 25142 to svn://svn.ffmpeg.org/ffmpeg/trunk
which will hopefully solve the Win64/FATE failures caused by these functions.
Originally committed as revision 25137 to svn://svn.ffmpeg.org/ffmpeg/trunk
have been accepted before).
Also do not fail if they are invalid but instead override them to 0.
This allows decoding e.g. MPEG video when only the container values are corrupted.
For encoding a value of 0,0 of course makes no sense, but was allowed
through before and will be caught by an extra check in the encode function.
Originally committed as revision 25124 to svn://svn.ffmpeg.org/ffmpeg/trunk
h264dsp_mmx.c to h264_idct.asm (as yasm code). Because the loops are now
coded in asm instead of C, this is (depending on the function) up to 50%
faster for cases where gcc didn't do a great job at looping.
Since h264_idct_add8() is now faster than the manual loop setup in h264.c,
in-asm idct calling can now be enabled for chroma as well (see r16207). For
MMX, this is 5% faster. For SSE2 (which isn't done for chroma if h264.c does
the looping), this makes it up to 50% faster. Speed gain overall is ~0.5-1.0%.
Originally committed as revision 25119 to svn://svn.ffmpeg.org/ffmpeg/trunk
Original patch by Zhou Zongyi, zhouzy A os pku edu cn, resubmitted by
James Darnley, james.darnley gmail, changes by me.
Originally committed as revision 25115 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes out of bounds accesses for big endian formats and should be
a little faster.
Originally committed as revision 25110 to svn://svn.ffmpeg.org/ffmpeg/trunk
If the CRC from the src->dst conversion matches a reference, it is not
necessary to perform a dst->yuva420p conversion and check the SSD.
Originally committed as revision 32213 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
The source format parameters are kept in static variables and conversion from
ref to source is only made when any parameter changes.
Originally committed as revision 32211 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
yvu9ToYv12Wrapper() used to support yv12 with the chroma planes either in the
uv order or the vu order. FFmpeg no longer has a pixel format in vu order.
Originally committed as revision 32156 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
For a PCE based configuration map the channels solely based on tags.
For an indexed configuration map the channels solely based on position.
This works with all known exotic samples including al17, elem_id0, bad_concat,
and lfe_is_sce.
Originally committed as revision 25098 to svn://svn.ffmpeg.org/ffmpeg/trunk
case the stride must be aligned to a multiple of 4.
The original CSCD encoder just compresses bitmaps it gets via Windows
API functions as-is, thus it uses exactly those alignment rules.
Originally committed as revision 25096 to svn://svn.ffmpeg.org/ffmpeg/trunk
Instead of defining functions in per-arch header files included
by the main cpu.c, define them normally and call them from the
generic one.
Originally committed as revision 25084 to svn://svn.ffmpeg.org/ffmpeg/trunk
mm_support instead.
Fix compilation if altivec is present and libxvidff.c is compiled.
Originally committed as revision 25075 to svn://svn.ffmpeg.org/ffmpeg/trunk
mm_support() instead.
Reduce complexity and simplify pending move to libavutil.
Originally committed as revision 25074 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fixes a black line in non-swapped, non-mod-16-height Theora videos
when vp3_draw_horiz_band is used.
Originally committed as revision 25073 to svn://svn.ffmpeg.org/ffmpeg/trunk
av_image_copy(), more consistent with the rest of the API and more
semantically correct.
Originally committed as revision 25070 to svn://svn.ffmpeg.org/ffmpeg/trunk
Using floating-point here can cause erroneous rejection of
parameters due to rounding errors leading to a slightly too
large result.
This fixes the mxf regression test with gcc 4.5 on x86_32.
Originally committed as revision 25060 to svn://svn.ffmpeg.org/ffmpeg/trunk
grepping and somewhat more consistent with the scheme adopted by other
FFmpeg modules API.
Originally committed as revision 25057 to svn://svn.ffmpeg.org/ffmpeg/trunk
that result in overdrawing areas again and again if s->flipped_image
is false.
Originally committed as revision 25051 to svn://svn.ffmpeg.org/ffmpeg/trunk
The #ifndef is required to allow for example some automated regression
tests by simply configuring with: --extra-cflags="-DFF_API_FOO=0".
Originally committed as revision 25043 to svn://svn.ffmpeg.org/ffmpeg/trunk
This increases compatibilty with nasm and is also more consistent,
e.g. with h264_intrapred.asm and h264_chromamc.asm that already
do it that way.
Originally committed as revision 25042 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes building with --disable-everything --enable-muxer=rtp, closing
issue 2159.
Originally committed as revision 25036 to svn://svn.ffmpeg.org/ffmpeg/trunk
for broken OSes. This is included in rtsp.h, as opposed to os_support.h.
Should fix OS/2 broken build on fate.
Originally committed as revision 25035 to svn://svn.ffmpeg.org/ffmpeg/trunk
message, if available (RFC 2326, section 12.39), fixes issue 2212.
Patch by John Wimer <john at god vtic net>.
Originally committed as revision 25032 to svn://svn.ffmpeg.org/ffmpeg/trunk
format), LGPL'ed with permission from Jason and Loren. This includes mmx2
code, so remove inline asm from h264dsp_mmx.c accordingly.
Originally committed as revision 25031 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes a bug from rev 22917. Now RTSP streams where the individual RTCP
sender reports aren't sent at the same time actually are synced properly.
Originally committed as revision 25029 to svn://svn.ffmpeg.org/ffmpeg/trunk
biweight code to sse2/ssse3; add sse2 weight code; and use that same code to
create mmx2 functions also, so that the inline asm in h264dsp_mmx.c can be
removed. OK'ed by Jason on IRC.
Originally committed as revision 25019 to svn://svn.ffmpeg.org/ffmpeg/trunk
still #included in dsputil_mmx.c and is part of DSPContext, and h264dsp_mmx.c,
which represents H264DSPContext and is now compiled on its own.
Originally committed as revision 25018 to svn://svn.ffmpeg.org/ffmpeg/trunk
uint16_fast_t is unsigned int (or long) on Linux, which when compared
with int results in an unsigned compare.
Originally committed as revision 24994 to svn://svn.ffmpeg.org/ffmpeg/trunk
into its own file, it doesn't belong in h264dsp_mmx.c (much less so in
dsputil_mmx.c).
Originally committed as revision 24990 to svn://svn.ffmpeg.org/ffmpeg/trunk
Therefore, init it with NULL to prevent a crash on invalid streams.
Patch by Zhentan Feng <spyfeng gmail com>.
Originally committed as revision 24960 to svn://svn.ffmpeg.org/ffmpeg/trunk
To avoid being burned at the stake by an angry mob, I am forced to
revert this commit.
Originally committed as revision 24949 to svn://svn.ffmpeg.org/ffmpeg/trunk
Release 4.1 changed the flag used to pass options through the gcc
translation layer. Detect which version is used, and choose the
appropriate flag.
Originally committed as revision 24945 to svn://svn.ffmpeg.org/ffmpeg/trunk
For this reason, a MuxerContext and write_trailer()-function was added,
to track the previous packet and flush the last packet at the end.
Originally committed as revision 24942 to svn://svn.ffmpeg.org/ffmpeg/trunk
This removes duplicated definitions of 8x8 and 16x16 fullpel MC
functions with various names reducing dsputil.o by 8k on x86_64.
Originally committed as revision 24933 to svn://svn.ffmpeg.org/ffmpeg/trunk
The stride can be negative and must be sign extended before being
used in pointer arithmetic.
Originally committed as revision 24926 to svn://svn.ffmpeg.org/ffmpeg/trunk
Because the order of evaluation of subexpressions is undefined, two
get_bits() calls may not be part of the same expression.
See also r24902.
Originally committed as revision 24906 to svn://svn.ffmpeg.org/ffmpeg/trunk
Because the order of evaluation of subexpressions is undefined, two
get_bits() calls may not be part of the same expression. In this
specific case, using get_bits_long() is simpler.
This fixes msmpeg4v1 decoding with armcc.
Originally committed as revision 24902 to svn://svn.ffmpeg.org/ffmpeg/trunk
Compilers may assume a pointer has natural alignment, even if it was
assigned from a pointer type with weaker alignment requirements. It
is thus not safe to assign a possibly unaligned value to a pointer,
regardless of how it is subsequently dereferenced.
Originally committed as revision 24897 to svn://svn.ffmpeg.org/ffmpeg/trunk
link rather than avfilter_default_get_video_buffer().
This is required as the buffer requested may be greater than the
buffer allocated locally by avfilter_default_get_video_buffer(), for
example if in filterchain there is a pad filter (like in "fifo,pad").
In that case the pad filter will try to write beyond the data of the
allocated buffer, resulting in a crash.
Originally committed as revision 24895 to svn://svn.ffmpeg.org/ffmpeg/trunk
This performs quite a bit better than the current 3GPP-inspired window decision
on all the samples I have tested. On the castanets.wav sample it performs very
similar to iTunes window selection, and seems to perform better than Nero.
On fatboy.wav, it seems to perform at least as good as iTunes, if not better.
Nero performs horribly on this sample.
Patch by: Nathan Caldwell <saintdev@gmail.com>
Originally committed as revision 24892 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allows cleaner implementation of other psymodels using the existing
structs. It also will make it easier to interchange individual parts of
the psymodel to create hybrid models.
Patch by: Nathan Caldwell <saintdev@gmail.com>
Originally committed as revision 24890 to svn://svn.ffmpeg.org/ffmpeg/trunk
instead of current "make output file of size less than ss".
Also use it to make MP3 tests more readable (using -fs xxx where xxx is
the requested output size, not something slightly lower).
Originally committed as revision 24884 to svn://svn.ffmpeg.org/ffmpeg/trunk
This is to avoid split asm sections that attempt to preserve some
registers between sections.
Originally committed as revision 24869 to svn://svn.ffmpeg.org/ffmpeg/trunk
since all of it ends up in a single 32-bit pixel.
This seems likely to be wrong though, since it is different
from the 15 and 16 bit modes and might explain the half-width
issue for 24 bit truemotion.
Originally committed as revision 24864 to svn://svn.ffmpeg.org/ffmpeg/trunk
BGR32 and not BGR24, do not swap red and blue on big-endian
for this format as well.
Originally committed as revision 24863 to svn://svn.ffmpeg.org/ffmpeg/trunk
Zhentan Feng <spyfeng gmail com> as part of Google's Summer of Code program.
Originally committed as revision 24861 to svn://svn.ffmpeg.org/ffmpeg/trunk
The relevent extract of the iso 13818-2 about the value of the syntaxical
element top_field_first of the Picture Coding Extension is:
"top_field_first -- The meaning of this element depends upon picture_structure,
progressive_sequence and repeat_first_field.
[...]
In a field picture top_field_first shall have the value '0', and the only field
output by the decoding process is the decoded field picture."
Originally committed as revision 24853 to svn://svn.ffmpeg.org/ffmpeg/trunk
The plural form is preferred as it is more consistent with the other functions:
av_fill_image_linesizes()
av_fill_image_pointers()
and looks semantically more correct as it fills an array of elements.
Originally committed as revision 24851 to svn://svn.ffmpeg.org/ffmpeg/trunk
To display the other planes is usually not useful and add noise to the
output.
Originally committed as revision 24843 to svn://svn.ffmpeg.org/ffmpeg/trunk
mmap() with MAP_ANONYMOUS requires the file descriptor to be -1 in NetBSD.
Linux just ignores this parameter.
Patch by Grant Carver <grantc at cat dot co dot za>
Originally committed as revision 31984 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
Conversion of an out of range float to int is undefined. Clipping to
the final range first avoids such problems. This fixes decoding on
MIPS, which handles these conversions differently from many other CPUs.
Originally committed as revision 24838 to svn://svn.ffmpeg.org/ffmpeg/trunk
We no longer create dependency files directly, so the rules are now pointless.
Originally committed as revision 24807 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes playback of at least one MMST stream.
Patch by Zhentan Feng <spyfeng gmail com>.
Originally committed as revision 24793 to svn://svn.ffmpeg.org/ffmpeg/trunk
ff_get_plane_bytewidth().
The new implementation is more generic, more compact and more correct.
Originally committed as revision 24786 to svn://svn.ffmpeg.org/ffmpeg/trunk
this prevents a time-out which closes the TCP connection and kills our
session.
see "Re: [FFmpeg-devel] [PATCH] rtsp.c: keep-alive" thread on mailinglist.
Originally committed as revision 24785 to svn://svn.ffmpeg.org/ffmpeg/trunk
Grab from the bitstream in 16-bit chunks instead of 8-bit chunks.
TODO: grab in 32-bit chunks on 64-bit systems.
Originally committed as revision 24783 to svn://svn.ffmpeg.org/ffmpeg/trunk
MMSContext. The other MMST-specific members go into MMSTContext.
Patch by Zhentan Feng <spyfeng gmail com>.
Originally committed as revision 24778 to svn://svn.ffmpeg.org/ffmpeg/trunk
This is required to make AVFilterBufferRef able to contain also audio
data, required by audio filtering integration.
Patch by S.N. Hemanth Meenakshisundaram smeen?ks@ucsd.ed?.
Originally committed as revision 24773 to svn://svn.ffmpeg.org/ffmpeg/trunk
wrapper of it.
The new function is more generic, and does not depend on the
definition of the AVPicture struct.
Patch by S.N. Hemanth Meenakshisundaram s + "meenakshisundaram".substr(0, 7) + "@ucsd.edu".
Originally committed as revision 24768 to svn://svn.ffmpeg.org/ffmpeg/trunk
AVFilterContext *avfilter_open(AVFilter *filter, const char *inst_name);
to:
int avfilter_open(AVFilterContext **filter_ctx, AVFilter *filter, const char *inst_name);
This way it is possible to propagate an error code telling the reason
of the failure.
Originally committed as revision 24765 to svn://svn.ffmpeg.org/ffmpeg/trunk
Define a new struct AVFilterBufferRefVideoProps and add a type field
to AVFilterBufferRef.
Video specific properties in AVFilterBufferRefVideoProps are now
referred to by *video pointer in AVFilterBufferRef.
Patch by S.N. Hemanth Meenakshisundaram smeenaks->ucsd.edu.
Originally committed as revision 24763 to svn://svn.ffmpeg.org/ffmpeg/trunk
correct mb_xy to use mb_width.
tighten allocations.
reduce the amount of zeroing.
Originally committed as revision 24760 to svn://svn.ffmpeg.org/ffmpeg/trunk
video4linux2 devices. This is used by Sensoray Model 2253 cards.
Patch by Pete Eberlein (pete AT sensoray DOT com)
Originally committed as revision 24759 to svn://svn.ffmpeg.org/ffmpeg/trunk
two padding zeroes before it. Should fix fate failures on openBSD and crashes
on MacOSX (that I cannot reproduce).
Originally committed as revision 24750 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allows using a libvpx that has been configured as decoder/encoder only.
patch by Frank Barchard, fbarchard google com
Originally committed as revision 24742 to svn://svn.ffmpeg.org/ffmpeg/trunk
Many H.264 derivatives, like RV40 and VP8, use the H.264 prediction functions
but not the weight/loopfilter functions.
This should reduce the size of builds with one of these derivatives but without
H.264 decoding itself.
Originally committed as revision 24741 to svn://svn.ffmpeg.org/ffmpeg/trunk
including libavcore/imgutils.h, which was required since the recent
avcodec_check_dimensions() -> av_check_image_size() transition.
Originally committed as revision 24734 to svn://svn.ffmpeg.org/ffmpeg/trunk
AVFilterLink.srcpic -> AVFilterLink.src_buf
AVFilterLink.cur_pic -> AVFilterLink.cur_buf
AVFilterLink.outpic -> AVFilterLink.out_buf
The new names are more generic and more consistent, since the struct
they contain, which was named AVFilterPicRef, has been renamed to
AVFilterBufferRef.
Patch by S.N. Hemanth Meenakshisundaram %smeenaks%ucsd%edu%.
Originally committed as revision 24732 to svn://svn.ffmpeg.org/ffmpeg/trunk
avfilter_(un)ref_pic -> avfilter_(un)ref_buffer
avfilter_copy_picref_props -> avfilter_copy_buffer_ref_props
AVFilterBufferRef.pic -> AVFilterBufferRef.buffer
They have been renamed to allow sharing with audio.
Patch by S.N. Hemanth Meenakshisundaram $smeenaks$ucsd$edu$.
Originally committed as revision 24731 to svn://svn.ffmpeg.org/ffmpeg/trunk
The struct is going to be used for storing audio buffer references as
well, and the new name is more generic.
Patch by S.N. Hemanth Meenakshisundaram @smeenaks@ucsd@edu@.
Originally committed as revision 24730 to svn://svn.ffmpeg.org/ffmpeg/trunk
AVFilterBuffereRef in avfilter_default_get_video_buffer().
The error was being caused by the previous patch which resized
AVFilterBuffer's data and linesize arrays to 8.
Patch by S.N. Hemanth Meenakshisundaram" &smeenaks&ucsd&edu&.
Originally committed as revision 24727 to svn://svn.ffmpeg.org/ffmpeg/trunk
When sizeof(uint_fast8_t) >= sizeof(int) there are unintended size effects.
Originally committed as revision 24716 to svn://svn.ffmpeg.org/ffmpeg/trunk
The vorbis decoder doesn't handle more than one audio frame packed into
the same AVPacket, so they need to be split in the depacketizer.
Originally committed as revision 24704 to svn://svn.ffmpeg.org/ffmpeg/trunk
Create a custom table for VP5/6/8's renorm to avoid depending on H.264's.
Saves one instruction in the arithmetic decoder as well.
Originally committed as revision 24701 to svn://svn.ffmpeg.org/ffmpeg/trunk
This moves some groups of tests for single codecs to separate files,
and adds shorthands for running all tests in a group.
Originally committed as revision 24697 to svn://svn.ffmpeg.org/ffmpeg/trunk
This reverts rev 24674 - the VP8 decoder actually depends on cabac.o.
vp8.c includes vp56.h, which includes cabac.h, which has inline functions
that reference tables from cabac.c.
This fixes compilation with --disable-everything --enable-decoder=vp8.
Originally committed as revision 24692 to svn://svn.ffmpeg.org/ffmpeg/trunk
Always inline the arithmetic coder, except in the case of header-parsing stuff,
in which case don't inline it at all to save code size.
Originally committed as revision 24677 to svn://svn.ffmpeg.org/ffmpeg/trunk
If sizeof uint_fast8_t > 1 and sizeof size_t <= 4, the expression that mallocs
classifs is susceptible to integer overflow.
Originally committed as revision 24675 to svn://svn.ffmpeg.org/ffmpeg/trunk
A lot of the time the DC block is empty: don't do the WHT in this case.
A lot of the rest of the time, there's only one coefficient: make a special
DC-only transform for that case.
When the block is empty, don't incorrectly mark luma DCT blocks as having DC
coefficients.
Originally committed as revision 24670 to svn://svn.ffmpeg.org/ffmpeg/trunk
Lets us do the zeroing in asm instead of C.
Also makes it consistent with the way the regular iDCT code does it.
Originally committed as revision 24668 to svn://svn.ffmpeg.org/ffmpeg/trunk
Without this flag -Wall generates warnings that nobody wants to see fixed or
worked around. Disabling those warnings is thus the best solution.
Originally committed as revision 24666 to svn://svn.ffmpeg.org/ffmpeg/trunk
Increase readability and robustness, as the test result is not going
to differ if the order of the pixfmts codes changes.
Originally committed as revision 24665 to svn://svn.ffmpeg.org/ffmpeg/trunk
The corresponding lavfi-pixfmts BE tests are not yet added, as there
are some bugs in the scaler (scaling rgba, argb, bgra, abgr, yuva420p)
which result in differences with the LE reference, and I cannot
visually check the generated files on BE.
Originally committed as revision 24657 to svn://svn.ffmpeg.org/ffmpeg/trunk
Introduce the function do_lavfi_pixfmts(), and use it for generating a
pixfmts test for each different filter.
Originally committed as revision 24654 to svn://svn.ffmpeg.org/ffmpeg/trunk
The 'tar' variable should be set to a command writing a tar archive
of the named files to stdout, typically "tar c" or "tar cf -"
Originally committed as revision 24649 to svn://svn.ffmpeg.org/ffmpeg/trunk
Add the lavfi-showfiltfmts dependency in the Makefile, and correctly
use the $target_exec and $target_path variables for invoking the
lavfi-showfiltfmts tool.
Originally committed as revision 24645 to svn://svn.ffmpeg.org/ffmpeg/trunk
otherwise the test will be running whatever ffmpeg is installed on the
host system.
Originally committed as revision 24644 to svn://svn.ffmpeg.org/ffmpeg/trunk
test output files, and add a prefix with the name of the test.
Make per-filter grouping of the generated output files easier, which
is more useful than per-pixel-format grouping.
Originally committed as revision 24643 to svn://svn.ffmpeg.org/ffmpeg/trunk
Consistent with the lavfi pixdesc test code, and slightly improve
readability.
Originally committed as revision 24642 to svn://svn.ffmpeg.org/ffmpeg/trunk
unchanged bytes) in the horizontal simple loopfilter. This makes the filter
quite a bit faster in itself (~30 cycles less on Core1), probably mostly
because we don't need a complex 4x4 transpose, but only a simple byte
interleave. Also allows using pextrw on SSE4, which speeds up even more
(e.g. 25% faster on Core i7).
Originally committed as revision 24638 to svn://svn.ffmpeg.org/ffmpeg/trunk
can be known exactly, so larger frame sizes can be safely encoded without buffer
overwrite.
Originally committed as revision 24630 to svn://svn.ffmpeg.org/ffmpeg/trunk
This is required for making AVFilterBuffer useful for storing audio
data.
Patch by S.N. Hemanth Meenakshisundaram revert <ude.dscu@skaneems>.
Originally committed as revision 24592 to svn://svn.ffmpeg.org/ffmpeg/trunk
the pixdescs.
The new implementation is more compact, and does not need to be
updated at each pixel format addition.
Originally committed as revision 24590 to svn://svn.ffmpeg.org/ffmpeg/trunk
max_plane_step_comp by removing the "plane_" word, and add a comment
for explaining what they represent.
Increase readability.
Originally committed as revision 24588 to svn://svn.ffmpeg.org/ffmpeg/trunk
av_fill_image_pointers() rather than their wrappers ff_fill_linesize()
and ff_fill_pointer().
Improve performance.
Originally committed as revision 24587 to svn://svn.ffmpeg.org/ffmpeg/trunk
libavcore/imgutils.c rather than ff_fill_linesize() and
ff_fill_pointer().
Also remove a dependency on libavcodec.
Originally committed as revision 24586 to svn://svn.ffmpeg.org/ffmpeg/trunk
without value when the target library is disabled.
If it does not explicitly return 0, when the last library is disabled
(swscale), the final "configure" exit value is 1, even if the
configure script is successfully executed. So it breaks scripts that
invoke configure and rely on 0 for success and 1 for failure.
Patch by Jean-Daniel Dupas reverse(<org.shadowlab@devlists>).
Originally committed as revision 24585 to svn://svn.ffmpeg.org/ffmpeg/trunk
libavcodec/imgconvert.c and make them public in libavcore/imgutils.h,
with the names av_fill_image_linesizes() and av_fill_image_pointers().
Originally committed as revision 24583 to svn://svn.ffmpeg.org/ffmpeg/trunk
r31772 | stefano | 2010-07-23 01:01:31 +0200 (Fri, 23 Jul 2010) | 2 lines
Prefer impersonal form over third person, for consistency with the
rest of FFmpeg.
The change was not approved by the maintainer.
Originally committed as revision 31847 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
Although with several limitations, lavfi can be compiled and used
without the scale filter.
Originally committed as revision 24570 to svn://svn.ffmpeg.org/ffmpeg/trunk
The extensions field was initialised first to "cavs", then to "avs".
The name "cavs" is kept since this is used for the format elsewhere
and "avs" is already used for avisynth files.
Originally committed as revision 24538 to svn://svn.ffmpeg.org/ffmpeg/trunk
This can avoid segfaults in some cases.
Patch by Josh Allmann, joshua dot allmann at gmail
Originally committed as revision 24537 to svn://svn.ffmpeg.org/ffmpeg/trunk
in favor of the newly added corresponding functions
av_parse_video_size() and av_parse_video_rate() defined in
libavcore/parseutils.h.
This change also adds a linking-time dependency of libavcodec and of
libavfilter on libavcore.
Originally committed as revision 24518 to svn://svn.ffmpeg.org/ffmpeg/trunk
should help in making mmst a little more userfriendly, or at least debuggable.
Also use helpful error return values instead of -1.
Originally committed as revision 24515 to svn://svn.ffmpeg.org/ffmpeg/trunk
5-10% faster or more on Phenom, Athlon 64, and some others.
Helps some on pre-SSSE3 Intel chips as well, but not as much.
Originally committed as revision 24513 to svn://svn.ffmpeg.org/ffmpeg/trunk
fill_image_data_ptr(). ff_fill_linesize() and ff_fill_pointer() now wrap
these functions.
The new functions are more generic, and are going to be exported in a
future patch.
Patch by S.N. Hemanth Meenakshisundaram smeenaks # ucsd § edu.
Originally committed as revision 24512 to svn://svn.ffmpeg.org/ffmpeg/trunk
mbedge loopfilter functions, by re-using space that holds a variable
that we no longer need.
Originally committed as revision 24510 to svn://svn.ffmpeg.org/ffmpeg/trunk
future new optimizations (imagine a sse5) much easier. Also fix a bug where
we used the direction (%2) rather than optimization (%1) to enable this, which
means it wasn't ever actually used...
Originally committed as revision 24507 to svn://svn.ffmpeg.org/ffmpeg/trunk
av_ prefix is reserved for libav* libraries functions, it is confusing
to use it for an application function.
Originally committed as revision 24505 to svn://svn.ffmpeg.org/ffmpeg/trunk
av_ is a prefix reserved for av* libraries functions, using that for an
application function is confusing.
Originally committed as revision 24497 to svn://svn.ffmpeg.org/ffmpeg/trunk
handled by upper layers.
Fixes double-frees (issue 2003).
Instead call ffm_close to ensure rc_eqs are freed also in the error case.
Originally committed as revision 24490 to svn://svn.ffmpeg.org/ffmpeg/trunk
splits it into small optimization-specific macros which are selected for each
DSP function. The advantage of this approach is that the sse4 functions now
use the ssse3 codepath also without needing an explicit sse4 codepath.
Originally committed as revision 24487 to svn://svn.ffmpeg.org/ffmpeg/trunk
Reduce scalefactors in non-zero bands that underflow by twice as much as those
in bands that just fail to hit psy targets.
Originally committed as revision 24482 to svn://svn.ffmpeg.org/ffmpeg/trunk
If there is little unread data in the ByteIOContext buffer, this may lead
to reducing the size of the ByteIOContext buffer to little more the probe
data size. This can lead to suboptimal aviobuf behaviour, e.g. making some
demuxers fail to do short seeks backwards (if the input isn't seekable).
Originally committed as revision 24478 to svn://svn.ffmpeg.org/ffmpeg/trunk
This is a lot more reliable to get cmov rather than trying to trick gcc into
generating it, useful since it's 2% faster overall.
Patch by Eli Friedman <eli.friedman at gmail>
Originally committed as revision 24471 to svn://svn.ffmpeg.org/ffmpeg/trunk
on the huffman tree, instead of traversing the tree in a while loop.
Based on the similar optimization in libvpx's detokenize.c
10% faster at normal bitrates, and 30% faster for high-bitrate intra-only
Originally committed as revision 24468 to svn://svn.ffmpeg.org/ffmpeg/trunk
No difference at the moment, but allows a future branchy variant
of vp56_rac_get_prob to be significantly faster
Originally committed as revision 24467 to svn://svn.ffmpeg.org/ffmpeg/trunk
sequential geometries instead of running all algorithms sequentially for each
geometry.
Originally committed as revision 31775 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
Apparently the official conformance test vectors don't test this feature,
even though libvpx uses it.
Originally committed as revision 24456 to svn://svn.ffmpeg.org/ffmpeg/trunk
Take shortcuts based on statistically common situations.
Add 4-at-a-time idct_dc function (mmx and sse2) since rows of 4 DC-only DCT
blocks are common.
TODO: tie this more directly into the MB mode, since the DC-level transform is
only used for non-splitmv blocks?
Originally committed as revision 24452 to svn://svn.ffmpeg.org/ffmpeg/trunk
Don't prefetch reference frames that were used less than 1/32th of the time so
far in the frame.
This helps speed up to ~2% on videos that, in many frames, make near-zero
(but not entirely zero) use of golden and/or alt-refs.
This is a very common property of videos encoded by libvpx.
Originally committed as revision 24451 to svn://svn.ffmpeg.org/ffmpeg/trunk
Prefetch all refs (including altref), but only if they've been used so far this
frame.
~2.5% faster overall.
TODO: Do something even smarter, like using how often each ref has been used
so far, so that a couple blocks of a rarely-used ref don't force us to prefetch
it.
Originally committed as revision 24444 to svn://svn.ffmpeg.org/ffmpeg/trunk
Making automatic reconfigure work properly in all cases proved more
difficult than it seemed, and people complained about it. Instead,
print a warning when the configuration appears to be out of date.
Originally committed as revision 24436 to svn://svn.ffmpeg.org/ffmpeg/trunk
Some converters (ie. unscaled rgb24 -> argb) may write some bytes out of
bounds. Ideally the converters should be fixed, but in the meantime we allocate
more memory to prevent heap corruption.
Originally committed as revision 31768 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
PRINT_LIB_VERSION() -> PRINT_LIB_INFO()
print_all_lib_versions() -> print_all_libs_info()
Originally committed as revision 24433 to svn://svn.ffmpeg.org/ffmpeg/trunk
Uses a slightly nonintuitive ring buffer size of (width+height*2) to simplify
addressing logic.
Also split out the segmentation map to a separate structure, necessary to
implement the ring buffer.
Originally committed as revision 24426 to svn://svn.ffmpeg.org/ffmpeg/trunk
This is needed to make the libavfilter framework work with audio
filters.
In particular add a type field to AVFilterLink, change the field types:
enum PixelFormat format -> int format in AVFilterBuffer
enum PixelFormat *formats -> int *formats in AVFilterFormats
enum PixelFormat *format -> int format in AVFilterLink
and change the function signatures:
AVFilterFormats *avfilter_make_format_list(const enum PixelFormat *pix_fmts); ->
AVFilterFormats *avfilter_make_format_list(const int *fmts);
int avfilter_add_colorspace(AVFilterFormats **avff, enum PixelFormat pix_fmt); ->
int avfilter_add_format(AVFilterFormats **avff, int fmt);
AVFilterFormats *avfilter_all_colorspaces(void); ->
AVFilterFormats *avfilter_all_formats(enum AVMediaType type);
This change breaks libavfilter API/ABI.
Patch by S.N. Hemanth Meenakshisundaram |smeenaks|ucsd|edu|.
Originally committed as revision 24424 to svn://svn.ffmpeg.org/ffmpeg/trunk
Gives better cache locality, since the VP8Macroblock structs are still in cache.
Inspired by the way x264 does it.
Originally committed as revision 24417 to svn://svn.ffmpeg.org/ffmpeg/trunk
As in the previous commit, they aren't used for context selection, so it saves
memory this way.
Originally committed as revision 24416 to svn://svn.ffmpeg.org/ffmpeg/trunk
Saves nothing except a bit of memory/cache now, but will allow future
optimizations.
Originally committed as revision 24411 to svn://svn.ffmpeg.org/ffmpeg/trunk
SSSE3 versions, improve SSE2 versions a bit.
SSE2/SSSE3 mbedge h functions are currently broken, so explicitly disable them.
Originally committed as revision 24403 to svn://svn.ffmpeg.org/ffmpeg/trunk
Avoid pextrw, since it's slow on many older CPUs.
Now it doesn't require mmxext either.
Originally committed as revision 24397 to svn://svn.ffmpeg.org/ffmpeg/trunk
The new library is meant to contain the core multimedia utilities for
FFmpeg, to make them shareable between more libav* libraries.
See thread:
Subject: [FFmpeg-devel] [RFC] New library for shared non-generic libav* utils
Date: Fri, 9 Jul 2010 01:07:40 +0200
Originally committed as revision 24393 to svn://svn.ffmpeg.org/ffmpeg/trunk
Make the buffer source accept a string for specifying the input pixel
format.
Originally committed as revision 24387 to svn://svn.ffmpeg.org/ffmpeg/trunk
darwin requires _DARWIN_C_SOURCE to be defined for MAP_ANON, which is used by
swscale to determine whether to use malloc() or mmap(). 64-bit darwin does not
have an executable heap, so mmap() must be used instead of malloc(), and
therefore _DARWIN_C_SOURCE must be defined.
Originally committed as revision 31760 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
The ff_inverse table is used by FASTDIV macro, defined in libavutil, but up
to now the table was defined only in libavcodec.
After this change, the main copy of ff_inverse is part of libavutil (just
like FASTDIV), but if CONFIG_SMALL is unset, then a different copy is made
available to libavcodec, to avoid the performance penalty of using an
external look up table.
Dynamic linking works, because the libraries are linked with -Bsymbolic, so
the local copy of the symbol has priority over the external; static linking
works because the table is on a standalone object file in both libraries,
so the linker is able to discard one of the two.
Tested on Linux/x86-64 and Mac OS X/x86-64.
Originally committed as revision 24383 to svn://svn.ffmpeg.org/ffmpeg/trunk
Should fix compilation with icc and should help prevent any future duplicates
Originally committed as revision 24380 to svn://svn.ffmpeg.org/ffmpeg/trunk
This is the correct solution to the warning "fixed" in the previous
commit.
Originally committed as revision 24367 to svn://svn.ffmpeg.org/ffmpeg/trunk
libavcodec/vp8.c:892: warning: suggest explicit braces to avoid ambiguous `else'
Originally committed as revision 24366 to svn://svn.ffmpeg.org/ffmpeg/trunk
servers happen to do. For this, we also move several header-size-related
variables to the MMSTContext.
Patch by Zhentan Feng <spyfeng gmail com>.
Originally committed as revision 24363 to svn://svn.ffmpeg.org/ffmpeg/trunk
part of the spec and causes problems otherwise.
Patch by Zhentan Feng <spyfeng gmail com>.
Originally committed as revision 24362 to svn://svn.ffmpeg.org/ffmpeg/trunk
most servers. Also remove a case where we manually aligned to 8 bytes,
since this is now no longer needed.
Patch by Zhentan Feng <spyfeng gmail com>.
Originally committed as revision 24360 to svn://svn.ffmpeg.org/ffmpeg/trunk
format ‘%d’ expects type ‘int’, but argument 3 has type ‘uint64_t’
Patch by Zhentan Feng <spyfeng gmail com>.
Originally committed as revision 24359 to svn://svn.ffmpeg.org/ffmpeg/trunk
The diff may provide useful information even if the command was
unsuccessful. The test is still treated as failed in this case.
Originally committed as revision 24353 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes a leak introduced in rev 23942, since we write padding to the
buffer unconditionally.
Originally committed as revision 24342 to svn://svn.ffmpeg.org/ffmpeg/trunk
regular MMX code. Examples of this are the Core1 CPU. Instead, set a new flag,
FF_MM_SSE2/3SLOW, which can be checked for particular SSE2/3 functions that
have been checked specifically on such CPUs and are actually faster than
their MMX counterparts.
In addition, use this flag to enable particular VP8 and LPC SSE2 functions
that are faster than their MMX counterparts.
Based on a patch by Loren Merritt <lorenm AT u washington edu>.
Originally committed as revision 24340 to svn://svn.ffmpeg.org/ffmpeg/trunk
so that it does both U and V planes at the same time. This will have speed
advantages when using SSE2 (or higher) optimizations, since we can do both
the U and V rows together in a single xmm register.
This also renames filter16 to filter16y and filter8 to filter8uv so that it's
more obvious what each function is used for.
Originally committed as revision 24337 to svn://svn.ffmpeg.org/ffmpeg/trunk
This test verifies the pixdesc code by comparing the output with and
without a filter which should have no effect on the image. Since the
available pixel formats depend on the byte order of the machine, a
simple reference checksum is not possible.
The test originally tried to solve this by generating a reference file
on the fly. The problem with this is that the test framework expects
the reference file in the source tree, and writing to the source tree
is not allowed.
To avoid complicating the test framework, we instead provide two
reference files and select which to use based on the byte order.
Originally committed as revision 24330 to svn://svn.ffmpeg.org/ffmpeg/trunk
Don't change paramater passing, but instead use casts.
Shouldn't affect asm output on anything other than win64.
libswscale should work on win64 now.
The rest of ffmpeg still isn't win64 compatible due to the issue of xmm
clobbers, but swscale doesn't use any SSE.
Patch by Anton Mitrofanov <BugMaster AT narod DOT ru>.
Originally committed as revision 31751 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
All tests use the provided helper functions so prepending $target_exec
and using eval is no longer required.
Originally committed as revision 24317 to svn://svn.ffmpeg.org/ffmpeg/trunk
This is a write-only protocol which computes the md5sum of data written,
and on close writes this to the designated output or stdout if none
is specified. It can be used to test muxers without writing an actual
file.
Originally committed as revision 24309 to svn://svn.ffmpeg.org/ffmpeg/trunk
The URL specification allows letters, numbers, plus, hyphen, and period
in the scheme part. The isalpha() test would allow additional characters
depending on locale settings while rejecting numbers and punctuation.
Originally committed as revision 24306 to svn://svn.ffmpeg.org/ffmpeg/trunk
definition.
This way it is easier to reference them in other structures, for
example in the pending AVFilterSamplesRef struct.
Patch by S.N. Hemanth Meenakshisundaram smeenaks AT ucsd DOT edu.
Originally committed as revision 24294 to svn://svn.ffmpeg.org/ffmpeg/trunk
These fields are never used, and they do not seem to belong to
AVFilterBuffer anymore, now that it is now a media-independent
structure and these fields are video-related.
Patch by S.N. Hemanth Meenakshisundaram smeenaks ! ucsd ! edu.
Originally committed as revision 24291 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allows CMP=stddev in test rules. The test passes if the reported
stddev is <= the FUZZ value (default 1).
Originally committed as revision 24289 to svn://svn.ffmpeg.org/ffmpeg/trunk
The struct is going to be used for audio data as well, so the new name
is less misleading.
Patch by S.N. Hemanth Meenakshisundaram smeenaks AT ucsd DOT edu.
Originally committed as revision 24284 to svn://svn.ffmpeg.org/ffmpeg/trunk
inner loopfilter, and it also allows us to save one register on x86-64/sse2.
Originally committed as revision 24269 to svn://svn.ffmpeg.org/ffmpeg/trunk
If the destination planes are offset within the destination buffer,
writing the extra bytes at the end may write outside of the destination
buffer.
Originally committed as revision 31746 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
Log:
Add msmpeg4v1 regtest
Added:
trunk/tests/ref/fate/msmpeg4v1
Modified:
trunk/tests/fate2.mak
According to Mans, "make test" tests already msmpeg4v1.
Originally committed as revision 24260 to svn://svn.ffmpeg.org/ffmpeg/trunk
This makes it clear that we are specifying the aspect ratio, and not
the intended display size in pixels.
Originally committed as revision 24239 to svn://svn.ffmpeg.org/ffmpeg/trunk
of sub_id, this allows detecting when that information is not available and
just decode everything.
In addition extradata is required for many codecs and thus in contrast to
sub_id generally already passed on by any programs using libav*.
Also ask for a sample if we encounter a stream with multiple/changing IDs.
Originally committed as revision 24238 to svn://svn.ffmpeg.org/ffmpeg/trunk
These functions are not documented for other decoders and
should be obvious enough even without Doxygen.
patch by Thilo Borgmann, thilo.borgmann googlemail com
Originally committed as revision 24237 to svn://svn.ffmpeg.org/ffmpeg/trunk
Patch by Josh Allmann, joshua dot allmann at gmail, original code
by Ronald S Bultje.
Originally committed as revision 24236 to svn://svn.ffmpeg.org/ffmpeg/trunk
The timestamp of the first RTP packet forming the output AVPacket is
written back in *timestamp, which is used in later calculations in generic
rtpdec code (together with RTCP sync timestamps) to form the final pkt->pts
value.
Originally committed as revision 24235 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fix the MDCT line to frequency calculation for the first line in each sfb.
Use this value to calculate ATH.
Originally committed as revision 24231 to svn://svn.ffmpeg.org/ffmpeg/trunk
Only names consisting of letters, numbers, hyphens, and underscores
are allowed.
Originally committed as revision 24222 to svn://svn.ffmpeg.org/ffmpeg/trunk
libavcodec/wmavoice.c:1906: warning: format `%lu' expects type `long unsigned int', but argument 5 has type `unsigned int'
approved by Ronald and Mans on IRC
Originally committed as revision 24218 to svn://svn.ffmpeg.org/ffmpeg/trunk
The AAC decoder and ADTS-to-ASC BSF both require the header decoder
but not full parsing capabilities.
Originally committed as revision 24217 to svn://svn.ffmpeg.org/ffmpeg/trunk
Our general use case just requires HTML documentation, so skip the extra step.
Originally committed as revision 24216 to svn://svn.ffmpeg.org/ffmpeg/trunk
It created false positives on seeks and where the first frame is STOP or SHORT.
It failed to warn in illegal SHORT->LONG transitions. In general it created
much confusion and many junk bug reports from the users.
Originally committed as revision 24214 to svn://svn.ffmpeg.org/ffmpeg/trunk
That makes easier understand what went wrong.
In debug mode the whole reply gets printed.
Originally committed as revision 24212 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes warnings about wrong type being used, e.g.:
libswscale/yuv2rgb.c: In function ‘ff_yuv2rgb_c_init_tables’:
libswscale/yuv2rgb.c:778: warning: passing argument 4 of ‘fill_table’ from incompatible pointer type
libswscale/yuv2rgb.c:598: note: expected ‘uint8_t *’ but argument is of type ‘uint16_t *’
Originally committed as revision 31722 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
Remove check for an specific w32api version, checking instead if vfw.h
supports vfw capture. The defines in w32api 3.12 were wrong, so this must be
accounted for in the check.
Originally committed as revision 24203 to svn://svn.ffmpeg.org/ffmpeg/trunk
Other parts of FFmpeg use NE (native endian) rather than ME (machine).
This makes it consistent.
Originally committed as revision 24169 to svn://svn.ffmpeg.org/ffmpeg/trunk
The byte count printed excludes the header, and offsets are applied
after the the headers are skipped.
Reference files updated to reflect new output. Some stddev/psnr values
have changed slightly due to headers no longer being compared.
Originally committed as revision 24143 to svn://svn.ffmpeg.org/ffmpeg/trunk
This adds the 'oneoff' comparison method. It compares two s16le pcm
files with a max difference <= the fuzz factor treated as success.
Default fuzz is 1. On failure, the full tiny_psnr output is printed.
Originally committed as revision 24134 to svn://svn.ffmpeg.org/ffmpeg/trunk
This passes the following additional variables to the test script:
- CMP method to use comparing output with reference, default 'diff'
- REF reference file, default 'tests/ref/fate/${test}'
- FUZZ maximum fuzz factor in comparison, meaning depends on CMP
Originally committed as revision 24133 to svn://svn.ffmpeg.org/ffmpeg/trunk
Regression test reference updates are due to the extra output
from tiny_psnr.
Patch by Vitor Sessak
Originally committed as revision 24132 to svn://svn.ffmpeg.org/ffmpeg/trunk
The print_enabled() function prints all elements in a list which
are enabled.
Originally committed as revision 24119 to svn://svn.ffmpeg.org/ffmpeg/trunk
This is clearer than comparing against an unsigned number to force an unsigned
comparison that we need.
Originally committed as revision 24114 to svn://svn.ffmpeg.org/ffmpeg/trunk
This avoid recreating the ref files every time an individual test
is run from the command line.
Originally committed as revision 24113 to svn://svn.ffmpeg.org/ffmpeg/trunk
if any E-AC-3 frames have been detected instead of switching back and forth for
AC-3 core + dependent E-AC-3 substream(s).
Fixes Issue 2022.
Originally committed as revision 24103 to svn://svn.ffmpeg.org/ffmpeg/trunk
The old names were not av_ prefixed, and they were causing linking
failure on many platforms. The new names are also more descriptive.
Patch by Stefano Sabatini.
Originally committed as revision 24101 to svn://svn.ffmpeg.org/ffmpeg/trunk
This ensures code always goes into the .text section and avoids the
need to specify it explicitly after changing sections.
Originally committed as revision 24095 to svn://svn.ffmpeg.org/ffmpeg/trunk
The w/h values are wrong and extradata contains a fake seq header that causes
decoding to fail later, a valid seq header is in the stream itself.
Partially fixes issue658
Originally committed as revision 24082 to svn://svn.ffmpeg.org/ffmpeg/trunk
libavcodec/h264.h:1260: warning: ‘decode_mb_skip’ defined but not used
patch by Eli Friedman, eli.friedman gmail com
Originally committed as revision 24069 to svn://svn.ffmpeg.org/ffmpeg/trunk
Number of channels is restricted to 6 so the size is acceptable
for the stack.
Originally committed as revision 24068 to svn://svn.ffmpeg.org/ffmpeg/trunk
of the file, and put under #if CONFIG_PAD_FILTER the code specific to
the pad filter.
Simplify the inclusion of the color source.
Originally committed as revision 24067 to svn://svn.ffmpeg.org/ffmpeg/trunk
Author: bcoudurier
Date: Sat Jul 3 03:11:04 2010 +0000
Set graph swscale opts before parsing it, that way opts are available
when auto-adding scalers.
It changed the swscale flags used by the auto-added scalers, and so
the output video.
Originally committed as revision 24065 to svn://svn.ffmpeg.org/ffmpeg/trunk
This is the version available in Debian stable, so it should be a reasonable
baseline that can be expected to be present on all developer machines.
Originally committed as revision 24064 to svn://svn.ffmpeg.org/ffmpeg/trunk
to verify the sync word the extension fsize field should be compared to
the core data length field.
Patch by nick.nbrereton@net
Originally committed as revision 24054 to svn://svn.ffmpeg.org/ffmpeg/trunk
truehd demuxers
Patch by Michael Karcher <ffmpeg at mkarcher dot dialup dot fu-berlin dot de>
Originally committed as revision 24053 to svn://svn.ffmpeg.org/ffmpeg/trunk
Patch by Michael Karcher <ffmpeg at mkarcher dot dialup dot fu-berlin dot de>
Originally committed as revision 24052 to svn://svn.ffmpeg.org/ffmpeg/trunk
and update the manual page accordingly.
Based on a patch by John Calcote $(echo "<kpio.dbmdpuf@hnbjm.dpn>" | tr "b-za" "a-z").
Originally committed as revision 24049 to svn://svn.ffmpeg.org/ffmpeg/trunk
The recently added dummy rule for missing headers took precedence
over the tablegen rules. Listing the generated headers explicitly
overrides this. A cleaner solution would be preferable.
Originally committed as revision 24046 to svn://svn.ffmpeg.org/ffmpeg/trunk
On PPC a leaf function has a 288-byte red zone below the stack pointer,
sparing these functions the chore of setting up a full stack frame.
When a function call is disguised within an inline asm block, the
compiler might not adjust the stack pointer as required before a
function call, resulting in the red zone being clobbered.
Moving the entire function to pure asm avoids this problem and also
results in somewhat better code.
Originally committed as revision 24044 to svn://svn.ffmpeg.org/ffmpeg/trunk
Author: stefano
Date: Sat Jul 3 20:19:38 2010
New Revision: 24023
Log:
State in the guidelines that function and parameter descriptions in
the doxy must use impersonal verbal form.
This form is apparently favored by most English speaker developers,
and has the advantage of being easier to use than the third person
form.
This should hopefully put an end to the Third Person Holy Bikeshed
War.
The change was neither discussed on the FFmpeg ML, nor approved by the
project leader.
The Holy War will continue soon on FFmpeg-devel, stay tuned ;-).
Originally committed as revision 24039 to svn://svn.ffmpeg.org/ffmpeg/trunk
Patch by Alexei Svitkine print("%s.%s@gmail.%s" % ("alexei", "svitkine", "com")).
Originally committed as revision 24038 to svn://svn.ffmpeg.org/ffmpeg/trunk
Patch by Alexei Svitkine cout << name << "." << surname << "@" << "gmail.com".
See thread:
Subject: [FFmpeg-devel] new command-line option for ffplay
Date: Wed, 23 Jun 2010 09:13:50 -0400
Originally committed as revision 24037 to svn://svn.ffmpeg.org/ffmpeg/trunk
the options are deprecated and that -vf crop should be used instead.
Based on a patch by John Calcote $(echo "<j0hn.c4lc073@gm41l.c0m>" | tr "04731" "oatei").
Originally committed as revision 24036 to svn://svn.ffmpeg.org/ffmpeg/trunk
ffmpeg.c: In function 'new_audio_stream':
ffmpeg.c:665: warning: 'best' may be used uninitialized in this function
ffmpeg.c:665: note: 'best' was declared here
Originally committed as revision 24034 to svn://svn.ffmpeg.org/ffmpeg/trunk
ffmpeg.c: In function 'new_audio_stream':
ffmpeg.c:665: warning: 'best' may be used uninitialized in this function
ffmpeg.c:665: note: 'best' was declared here
Originally committed as revision 24033 to svn://svn.ffmpeg.org/ffmpeg/trunk
the doxy must use impersonal verbal form.
This form is apparently favored by most English speaker developers,
and has the advantage of being easier to use than the third person
form.
This should hopefully put an end to the Third Person Holy Bikeshed
War.
Originally committed as revision 24023 to svn://svn.ffmpeg.org/ffmpeg/trunk
and options. Keep old options until next major version bump.
Patch by Rodney Baker, rodney d baker a iinet d net d au
Originally committed as revision 24021 to svn://svn.ffmpeg.org/ffmpeg/trunk
This is the version available in Debian stable, so it should be a reasonable
baseline that can be expected to be available on all developer machines.
Originally committed as revision 23968 to svn://svn.ffmpeg.org/ffmpeg/trunk
Also make some small changes to saturation order of 4-tap SSSE3 MC to fix a
non-bitexactness bug.
Patch mostly by Eli Friedman <eli.friedman AT gmail DOT com>.
Originally committed as revision 23965 to svn://svn.ffmpeg.org/ffmpeg/trunk
Check for GNU assembler only when asm is enabled. Show a warning for
ppc/altivec only, and fail on ARM.
Originally committed as revision 23964 to svn://svn.ffmpeg.org/ffmpeg/trunk
Make sure that the *_tablegen.h header is listed in the dependencies, but
filter it out in the compile line.
Originally committed as revision 23963 to svn://svn.ffmpeg.org/ffmpeg/trunk
1.8x faster than altivec radix-2 on a G4
8% faster vorbis decoding
Patch (mostly) by Loren Merritt
Originally committed as revision 23956 to svn://svn.ffmpeg.org/ffmpeg/trunk
This checks which assembler syntax is supported and defines macros
for register names accordingly.
Originally committed as revision 23952 to svn://svn.ffmpeg.org/ffmpeg/trunk
Also put compute_antialias_integer() under !CONFIG_FLOAT and change
forward declarations to declare only the relevant one of these.
Fixes warnings about unused functions and pointer type mismatches.
Originally committed as revision 23950 to svn://svn.ffmpeg.org/ffmpeg/trunk
The mmx code is floating-point only, and this function does not know
from which decoder it is called. Without this change, the integer
decoder only "works" because the size of the context struct is smaller
in this case, and the mmx init function writes the function pointer
outside the allocated context.
Fixes issue 2054.
Originally committed as revision 23949 to svn://svn.ffmpeg.org/ffmpeg/trunk
prototype and also always call them with exactly those arguments.
The previous way seems to have worked on all supported platforms, however
it was not strictly valid C and would crash e.g. with a compiler that
uses stdcall by default.
Also avoids warnings if -Wstrict-prototypes is used.
Originally committed as revision 23939 to svn://svn.ffmpeg.org/ffmpeg/trunk
warning: the address of 'number' will always evaluate as 'true'
patch by Eli Friedman eli _dot_ friedman _at_ gmail _dot_ com
Originally committed as revision 23936 to svn://svn.ffmpeg.org/ffmpeg/trunk
The ADTS output files are not explicitly signaled because the format
does not support explicit signalling.
Originally committed as revision 23935 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allocates per-thread copies of some MpegEncContext.ac_val which
is used concurrently from the encoding threads.
Originally committed as revision 23933 to svn://svn.ffmpeg.org/ffmpeg/trunk
If both /dev/random and /dev/urandom failed to return data, an
uninitialised value might be returned. Since most systems have a
non-blocking /dev/urandom or have /dev/random with similar properties,
the chance of blocking is minimal, and the alternative of returning
non-random data is worse.
Originally committed as revision 23930 to svn://svn.ffmpeg.org/ffmpeg/trunk
Whose idea was it to have a CPU that didn't SIGILL on an invalid instruction?
Originally committed as revision 23927 to svn://svn.ffmpeg.org/ffmpeg/trunk
Necessary because of this GCC bug:
http://gcc.gnu.org/bugzilla/show_bug.cgi?id=44474
To do this, convert some, but not all (!) of the variables in VP56RangeCoder
into local variables.
If we convert c->high into a local variable, gcc gets the stupids and refuses
to use a conditional move for the unpredictable main branch.
TODO: dispense with this bullshit and write an asm version.
Originally committed as revision 23924 to svn://svn.ffmpeg.org/ffmpeg/trunk
This incantation causes gcc 4.3 to generate cmov on x86, a vastly better option
than a completely unpredictable branch.
Hopefully this carries over to newer versions and other CPUs with conditionals.
~5 cycles saved per call on a Core i7.
Originally committed as revision 23921 to svn://svn.ffmpeg.org/ffmpeg/trunk
Attempt to read from /dev/urandom and /dev/random with O_NONBLOCK set.
If neither succeeds, proceed with fallbacks.
Originally committed as revision 23903 to svn://svn.ffmpeg.org/ffmpeg/trunk
Makes it give correct results with e.g. gcc 4.4.
For unknown reasons the generate asm code also changes
on e.g. gcc 4.3, making the code a bit larger but also
a bit faster.
Originally committed as revision 23896 to svn://svn.ffmpeg.org/ffmpeg/trunk
"passing argument 1 of ‘av_memcpy_backptr’ from incompatible pointer type"
Patch by Eli Friedman, eli D friedman A gmail
Originally committed as revision 23895 to svn://svn.ffmpeg.org/ffmpeg/trunk
This function is called from the rv40_[hv]_loop_filter() wrappers
with some arguments constant, so it really needs to be inlined.
3% faster overall on Cortex-A8.
Originally committed as revision 23893 to svn://svn.ffmpeg.org/ffmpeg/trunk
This keeps the encoder from switching away from an wight short sequence
prematurely when attacks are detected two frames in a row. It also allows for
the legal and useful LONG_STOP_SEQUENCE to LONG_START_SEQUENCE transition.
Originally committed as revision 23870 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allows very large value strings, needed for xiph extradata.
Patch by Josh Allmann, joshua dot allmann at gmail
Originally committed as revision 23859 to svn://svn.ffmpeg.org/ffmpeg/trunk
16 subblocks (since we no longer need that), which should also lead to a
minor speedup.
Originally committed as revision 23854 to svn://svn.ffmpeg.org/ffmpeg/trunk
we apply them as 16x8/8x16/8x8 subblocks where possible. Since this allows
us to use width=8/16 instead of width=4 MC functions, we can now take more
advantage of SSE2/SSSE3 optimizations, leading to a total speedup for splitMV
filter of about 10%.
Originally committed as revision 23853 to svn://svn.ffmpeg.org/ffmpeg/trunk
This will be used for cleaning up code that is common among RTP depacketizers.
Patch by Josh Allmann, joshua dot allmann at gmail
Originally committed as revision 23847 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allows explicitly enabled things to _select others with the
intended result.
Originally committed as revision 23843 to svn://svn.ffmpeg.org/ffmpeg/trunk
byte for the alpha component, as it is never used, clarify the code
and prevent bad usage of it.
Originally committed as revision 23834 to svn://svn.ffmpeg.org/ffmpeg/trunk
ff_url_split() is retained as an alias, as it was used by ffserver,
to avoid breaking ABI compatibility with it.
Originally committed as revision 23822 to svn://svn.ffmpeg.org/ffmpeg/trunk
Two macros (WRITE_ARRAY and WRITE_ARRAY_2D) take the prefix (modifiers)
(not all tables are static, and they might not be constant either), the
type, and the name of the array. It'll be copied with same name and type,
and with the correct size of the currently-defined object.
Originally committed as revision 23821 to svn://svn.ffmpeg.org/ffmpeg/trunk
This drops one parameter from the functions' macros, and require structures
to be typedeffed, but ensures that it is possible to map 1-to-1 the type to
the function name.
Originally committed as revision 23820 to svn://svn.ffmpeg.org/ffmpeg/trunk
- MMXEXT, SSE2 and SSSE3 MC functions
- MMX and SSE4 IDCT dc_add functions
Patch by Jason Garrett-Glaser <darkshikari gmail com> and myself.
Originally committed as revision 23815 to svn://svn.ffmpeg.org/ffmpeg/trunk
gprof is far too intrusive to be of use as a profile for ffmpeg,
and it fails to build in many configurations. Oprofile is a
better tool for profiling on Linux.
Originally committed as revision 23807 to svn://svn.ffmpeg.org/ffmpeg/trunk
The TI compiler defines __eabi__ to signal that ARM EABI is in use.
We must check for this in addition to the gcc macro __ARM_EABI__.
Originally committed as revision 23804 to svn://svn.ffmpeg.org/ffmpeg/trunk
The maximum value of sub_blocks is 8, a safe size to always allocate on
stack.
Originally committed as revision 23797 to svn://svn.ffmpeg.org/ffmpeg/trunk
This isn't useful for the C functions, but will allow re-using H and V functions
for HV functions without adding separate H and V wrappers.
Originally committed as revision 23782 to svn://svn.ffmpeg.org/ffmpeg/trunk
Adding new libs to the front of the list allows them to resolve
symbols against previously added ones.
Originally committed as revision 23780 to svn://svn.ffmpeg.org/ffmpeg/trunk
The 'warn' function records a warning message for display after other
informational messages.
Originally committed as revision 23760 to svn://svn.ffmpeg.org/ffmpeg/trunk
Modify the asm accordingly.
GLOBAL is now no longoer necessary for PIC-compliant loads.
Originally committed as revision 23739 to svn://svn.ffmpeg.org/ffmpeg/trunk
OS/2 SMP support is rare, and a pthreads library exists.
No need to keep this code.
Originally committed as revision 23731 to svn://svn.ffmpeg.org/ffmpeg/trunk
For GCC, barcelona is just an alias for amdfam10, so simply add it in
there.
Originally committed as revision 23730 to svn://svn.ffmpeg.org/ffmpeg/trunk
Relevant BeOS variants support pthreads, so there is no need to
maintain the beos-native threads interface.
Originally committed as revision 23729 to svn://svn.ffmpeg.org/ffmpeg/trunk
'(' and ')' are used instead of '{' and '}' in the date specification
as the latter confound the texinfo interpreter.
Originally committed as revision 23726 to svn://svn.ffmpeg.org/ffmpeg/trunk
Also make the RTSP protocol use url_alloc and url_connect instead of relying
on the delay open behaviour.
Originally committed as revision 23710 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allows extending the URLProtocol struct without breaking binary
compatibility with code compiled with older definitions of the struct.
Originally committed as revision 23702 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes handling of totally silent packets during the encoding, that
also are 1 byte in size.
This fixes issue 2013
Originally committed as revision 23693 to svn://svn.ffmpeg.org/ffmpeg/trunk
@table @option.
Allow pod rendering, as texinfo multitables are not supported by
texi2pod.pl, also improve plain texinfo file readability.
Originally committed as revision 23688 to svn://svn.ffmpeg.org/ffmpeg/trunk
connection (e.g. a seek). This fixes the theoretical case where a server
sends a file first using chunked encoding, and then using non-chunked
encoding.
Originally committed as revision 23665 to svn://svn.ffmpeg.org/ffmpeg/trunk
them from being sent using chunked encoding (I don't think this ever happened,
but either way it would be wrong).
Originally committed as revision 23664 to svn://svn.ffmpeg.org/ffmpeg/trunk
It is already rejected by the "source >= 0 && source != ps->num_env" 0 envelope
case and is perfectly legally for the suppressed final envelope case.
Originally committed as revision 23661 to svn://svn.ffmpeg.org/ffmpeg/trunk
This removes some useless copying of handles, and simplifies error handling.
Patch by Josh Allmann, joshua dot allmann at gmail
Originally committed as revision 23648 to svn://svn.ffmpeg.org/ffmpeg/trunk
It allows VLD MPEG-2 decoding using DXVA2 (GPU assisted decoding API under
VISTA and Windows 7).
It is implemented by using AVHWAccel API.
Originally committed as revision 23644 to svn://svn.ffmpeg.org/ffmpeg/trunk
Since rtsp_hd isn't assigned to rt->rtsp_hd until after the setup phase,
the initialized URLContext could be leaked on failures.
Originally committed as revision 23643 to svn://svn.ffmpeg.org/ffmpeg/trunk
Earlier, the function only returned when the enough data to fill the
requested buffer was available. This lead to high latency when receiving
low-bandwidth streams.
Originally committed as revision 23642 to svn://svn.ffmpeg.org/ffmpeg/trunk
warning #188: enumerated type mixed with another type
enum PixelFormat pix_fmts[] = { c->pix_fmt, PIX_FMT_NONE };
^
Originally committed as revision 23640 to svn://svn.ffmpeg.org/ffmpeg/trunk
(extradata_size must be at least 5 bytes) the H.264 MP4 syntax check
in rtpenc.c
Originally committed as revision 23638 to svn://svn.ffmpeg.org/ffmpeg/trunk
"config.err". The former name was misleading, as the file contains
useful information not necessarily related to errors.
Originally committed as revision 23630 to svn://svn.ffmpeg.org/ffmpeg/trunk
The display definition segment is used to properly display SD DVB subtitles in
HD video streams.
Originally committed as revision 23626 to svn://svn.ffmpeg.org/ffmpeg/trunk
when reading a live feed, should retry regardless of whether any client has
opened the stream.
Originally committed as revision 23621 to svn://svn.ffmpeg.org/ffmpeg/trunk
Apparently the compiler sometimes bypasses the normal assembler, in
which case the asm label variant works. If the file contains any
inline asm, this bypassing is not done, and the assembler chokes on
invalid syntax. Inserting a dummy asm() statement in the test when
the compiler supports it makes sure no shortcuts are taken and the
test fails as it should.
Originally committed as revision 23616 to svn://svn.ffmpeg.org/ffmpeg/trunk
When symbol versioning is enabled, moving symbols from one library to
another breaks binary compatibility. This adds wrappers with the old
version tag for the av_*packet functions recently moved to lavc.
Originally committed as revision 23611 to svn://svn.ffmpeg.org/ffmpeg/trunk
Since the parsing of Vorbis/Theora fmtp headers is handled by the
parse_sdp_a_line function pointer now, the buffer in sdp_parse_fmtp
doesn't need to be this large any longer.
Patch by Josh Allmann, joshua dot allmann at gmail
Originally committed as revision 23599 to svn://svn.ffmpeg.org/ffmpeg/trunk
Also use the correct value for unknown size (instead of a truncated
and sign extended value).
Originally committed as revision 23591 to svn://svn.ffmpeg.org/ffmpeg/trunk
This interrupt parsing and goes one ebml level up when encountering
a cluster start inside an unknown sized element which is not supposed to
contain a cluster.
fix issue1991
Originally committed as revision 23589 to svn://svn.ffmpeg.org/ffmpeg/trunk
Now that we save the currently parsed ebml element ID after an interruption,
we don't need to special case the parsing of the first cluster.
Originally committed as revision 23588 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allows to interrupt parsing after reading an ID, and then properly
recover parsing.
Originally committed as revision 23587 to svn://svn.ffmpeg.org/ffmpeg/trunk
different magic in the header (mainly targeted to Sony's .oma/.aa3
format).
Patch by Michael Karcher, ffmpeg A mkarcher dialup fu-berlin de
Originally committed as revision 23583 to svn://svn.ffmpeg.org/ffmpeg/trunk
out of the block decoding loop. Indeo4 doesn't use any scale table but the quant
level itself as scale. Therefore access scale table only if its pointer != NULL.
Originally committed as revision 23569 to svn://svn.ffmpeg.org/ffmpeg/trunk
BeOS support has been broken for many years and the "maintainer" of the port
has not reacted to countless requests to get the port fixed.
approved by Mans
Originally committed as revision 23562 to svn://svn.ffmpeg.org/ffmpeg/trunk
rawdec.c: Only perform bits_per_coded_sample -> pix_fmt guesswork if pix_fmt hasn't been set
Originally committed as revision 23561 to svn://svn.ffmpeg.org/ffmpeg/trunk
ansi color code to colored_fputs(), and pass instead the error level so the
proper color code may be used.
Originally committed as revision 23553 to svn://svn.ffmpeg.org/ffmpeg/trunk
This makes url_fsize return correct values for delay opened connections
that have not yet been initialized.
This fixes using the image2 demuxer with http sources.
Originally committed as revision 23546 to svn://svn.ffmpeg.org/ffmpeg/trunk
generated document following the same order as that used in the texi
file.
Also allow rendering of *all* the sections, not only a limited
predefined subset.
Originally committed as revision 23543 to svn://svn.ffmpeg.org/ffmpeg/trunk
outputs the section if WORDS has been defined in the list of options
flags.
The meaning of this rule is not clear, and not used by the FFmpeg
documentation system anyway.
Originally committed as revision 23542 to svn://svn.ffmpeg.org/ffmpeg/trunk
them at the beginning of the file, for consistency with the final
ordering of the corresponding man pages.
Also create a separate @chapter for the "Options" section, as it is
done for the man pages, rather than merge it in the same section of
"Synopsis".
Originally committed as revision 23541 to svn://svn.ffmpeg.org/ffmpeg/trunk
the output codec context (instead of just copying a pointer to the extradata).
Originally committed as revision 23538 to svn://svn.ffmpeg.org/ffmpeg/trunk
If http_connect fails, we've already stored the new connection handle in s->hd,
so clear it so http_close won't double-free it.
10l to me for not spotting it during review
Originally committed as revision 23529 to svn://svn.ffmpeg.org/ffmpeg/trunk
Initialize the buffer av_strlcatf writes into.
Add the dash at the end of the Range header.
10l to me for not spotting it earlier
Originally committed as revision 23528 to svn://svn.ffmpeg.org/ffmpeg/trunk
The LFG is only used in one place, so there is no need to keep it in
the context.
Originally committed as revision 23527 to svn://svn.ffmpeg.org/ffmpeg/trunk
The connection is made on the first http_read, http_write or http_seek.
Patch by Josh Allmann, joshua dot allmann at gmail
Originally committed as revision 23525 to svn://svn.ffmpeg.org/ffmpeg/trunk
The default for frame_skip_threshold is 0, which thus disables the libvpx
frame dropping.
Originally committed as revision 23522 to svn://svn.ffmpeg.org/ffmpeg/trunk
This mode emits no keyframes at all, despite its name.
Patch by James Zern, jzern at google dot com
Originally committed as revision 23521 to svn://svn.ffmpeg.org/ffmpeg/trunk
Based on Han-Wen Hsu, et al. "Fast Complex Quadrature Mirror Filterbanks for
MPEG-4 HE-AAC"
Originally committed as revision 23519 to svn://svn.ffmpeg.org/ffmpeg/trunk
and bgr4_byte from the lavfi-pix_fmts test.
The formats are now supported by NUT.
Originally committed as revision 23510 to svn://svn.ffmpeg.org/ffmpeg/trunk
Done in preparation for RTSP over HTTP.
Patch by Josh Allmann, joshua dot allmann at gmail
Originally committed as revision 23494 to svn://svn.ffmpeg.org/ffmpeg/trunk
Start them on keyframes when reasonable, and delay writing audio packets
to help ensure that there's audio samples available for the first frame in
clusters.
Patch by James Zern <jzern at google>
Originally committed as revision 23473 to svn://svn.ffmpeg.org/ffmpeg/trunk
formats which are not currently supported by NUT and which cannot be
played by ffplay.
Originally committed as revision 23458 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes cases where ist->start_time wasn't initialized. This also makes
cur_pts calculated against the same origin for all streams.
Originally committed as revision 23450 to svn://svn.ffmpeg.org/ffmpeg/trunk
Since the buffer always was large enough, this bug didn't have any
effect in practice.
Originally committed as revision 23447 to svn://svn.ffmpeg.org/ffmpeg/trunk
Also replace "the HTML documentation of @file{ffmpeg}" with "the FFmpeg
HTML documentation", which is more generic.
Originally committed as revision 23444 to svn://svn.ffmpeg.org/ffmpeg/trunk
This makes the ordering of the sections in the file more consistent
with that of the final man output.
Originally committed as revision 23443 to svn://svn.ffmpeg.org/ffmpeg/trunk
unsigned int, as the corresponding variable set in
AVCodecContext.codec_tag is unsigned int.
Originally committed as revision 23441 to svn://svn.ffmpeg.org/ffmpeg/trunk
The offsets differ depending on configuration, so only check them when
they will actually be used. Presently, this is when NEON is enabled.
Originally committed as revision 23436 to svn://svn.ffmpeg.org/ffmpeg/trunk
AVFormatContext * in input rather than a NUTContext *, which allows to
log messages with av_log().
Originally committed as revision 23431 to svn://svn.ffmpeg.org/ffmpeg/trunk
The test for .align syntax was apparently unreliable with some compilers.
Using the .p2align directive instead avoids the need for this test.
Originally committed as revision 23428 to svn://svn.ffmpeg.org/ffmpeg/trunk
from 2 to 1, which is the actual value used in the spec. Fixes issue1978.
Path by John Wimer <john at god dot vtic dot net>.
Originally committed as revision 23414 to svn://svn.ffmpeg.org/ffmpeg/trunk
additionallym deprecate palette8torgb16 and its bgr variant without
replacement. These functions are not meant to be used by applications.
Discussed at: http://comments.gmane.org/gmane.comp.video.ffmpeg.devel/109340
Originally committed as revision 31301 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
unsupported and causes crashes when libavfilter is enabled.
Patch by Sebastian Vater <cdgs basty googlemail com>.
Originally committed as revision 23411 to svn://svn.ffmpeg.org/ffmpeg/trunk
command on its own line, and create a corresponding "TIPS" man page
section.
Note that such section is not displayed, as currently only sections
with pre-defined names are rendered.
Originally committed as revision 23409 to svn://svn.ffmpeg.org/ffmpeg/trunk
or texinfo command for specifying how to generate @item marks, and
make it use by default the mark symbol "*".
This is consistent with texinfo docs:
"If you don't specify a mark command, the default is `@bullet'."
Originally committed as revision 23408 to svn://svn.ffmpeg.org/ffmpeg/trunk
section, and make the whole rendered in the man output.
Simplify layout, and make it more consistent with that of the other
man pages. Also I cannot see a good reason for keeping split the two
sections.
Originally committed as revision 23407 to svn://svn.ffmpeg.org/ffmpeg/trunk
sws_setColorspaceDetails() to ff_yuv2rgb_c_init_tables().
Allow to factorize duplicated code.
Originally committed as revision 31300 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
const_name -> const_names
const_value -> const_values
func[12]_name -> func[12]_names
func[12] -> funcs[12]
All these parameters contain a list of values, using plural names for
them help understanding a little.
Originally committed as revision 23403 to svn://svn.ffmpeg.org/ffmpeg/trunk
containing an error code.
Allow these functions to convey the reason of the failure to the
calling function, failure which is not always due to a parsing error
but it may depend for example on a memory problem.
Also fix several potential memleaks.
Originally committed as revision 23402 to svn://svn.ffmpeg.org/ffmpeg/trunk
They contain exactly the same code as their 16bit variants, so this is
effectively code de-duplication.
Originally committed as revision 31298 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
We now have an LGPL replacement that is at least equally fast.
Originally committed as revision 31278 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
The license of libvpx is incompatible with the (L)GPL. As long as this is
the case, the only way to use it is by marking the result as nonfree.
Originally committed as revision 23371 to svn://svn.ffmpeg.org/ffmpeg/trunk
other audio codecs rather than only AC-3, so add some code that makes
word byte-swapping only happen for AC-3, not for all audio codecs.
Patch by Francesco Lavra <francescolavra interfree it>.
Originally committed as revision 23361 to svn://svn.ffmpeg.org/ffmpeg/trunk
packets are passed through and wont be marked as static which would
require them to be copied by av_dup_packet().
Originally committed as revision 23352 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes a regression introduced when libavfilter support was added to ffplay.
Originally committed as revision 23343 to svn://svn.ffmpeg.org/ffmpeg/trunk
Use a larger input audio buffer, refill it when it has less than 4 KB data
left.
Originally committed as revision 23323 to svn://svn.ffmpeg.org/ffmpeg/trunk
The minimal codebook to encode the band without clipping is used (as is done in the TLS).
Originally committed as revision 23318 to svn://svn.ffmpeg.org/ffmpeg/trunk
This still provides plenty of dynamic range, makes every move legal, and greatly reduces the search space.
Originally committed as revision 23317 to svn://svn.ffmpeg.org/ffmpeg/trunk
target regtest-lavfi_pix_fmts.
The lavfi_pix_fmts test is disabled, this because there are
many tests which are failing, and there are still some output files
which cannot be played by NUT/ffplay.
Originally committed as revision 23297 to svn://svn.ffmpeg.org/ffmpeg/trunk
CODEC_ID_RAWVIDEO <-> Y41B
CODEC_ID_RAWVIDEO <-> Y42B
CODEC_ID_RAWVIDEO <-> YUV9
CODEC_ID_RAWVIDEO <-> YVU9
These codec tags are listed in fourcc.org, and are already listed in
libavcodec/raw.c:ff_raw_pix_fmt_tags as AVI tags.
Originally committed as revision 23296 to svn://svn.ffmpeg.org/ffmpeg/trunk
This is probably what was originally intended; the codec pointers are all NULL.
Fix by Howard Chu, hyc at highlandsun dot com
Originally committed as revision 23295 to svn://svn.ffmpeg.org/ffmpeg/trunk
Don't allocate st->codec, it will be overwritten by the memcpy a few
lines further down.
Fix by Howard Chu, hyc at highlandsun dot com
Originally committed as revision 23291 to svn://svn.ffmpeg.org/ffmpeg/trunk
This isn't exactly semantically equivalent, but the field has already been
long abused to mean this, and writing it helps in determining a decent cfr
time base when transcoding from a mkv where the video codec stores none (VP8).
Originally committed as revision 23284 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes aspect ratio calculation for encoding from files with 0/0 stored,
common with ogg/theora
Originally committed as revision 23280 to svn://svn.ffmpeg.org/ffmpeg/trunk
remain valid. This fixes segfaults for those cases where data copy to
this invalid pointer is attempted.
Originally committed as revision 23264 to svn://svn.ffmpeg.org/ffmpeg/trunk
the input file.
This is required as Nut codec tags are not contained in
ff_codec_bmp_tags.
Originally committed as revision 23260 to svn://svn.ffmpeg.org/ffmpeg/trunk
Keep an old ff_ named function for binary compatibility until the
next major bump.
Originally committed as revision 23254 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes a regression on Windows introduced by r22769 in which the data read
from the file was not properly zero terminated. The file was read as text,
which caused the \r characters to be suppressed. Since the zero termination
happens at the end of the buffer, and there was one byte less read per line,
this caused the remaining space on the buffer to contain random data.
Originally committed as revision 23251 to svn://svn.ffmpeg.org/ffmpeg/trunk
pix_fmt_bps_mov.
The new names are consistent with the predominant FFmpeg style.
Originally committed as revision 23244 to svn://svn.ffmpeg.org/ffmpeg/trunk
This is usefull to keep track and display relations where things are a
bit more complex (like AVProtocols or demuxers used by demuxers and such)
Originally committed as revision 23199 to svn://svn.ffmpeg.org/ffmpeg/trunk
M_PI is defined by the included file libavutil/mathematics.h.
Originally committed as revision 31185 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
This is in preparation for RTP hinting in the MOV muxer, where
it needs to be able to create SDP fragments for each media stream.
Originally committed as revision 23160 to svn://svn.ffmpeg.org/ffmpeg/trunk
This is needed for subtitles where packets are infrequent.
Patch by Janne Grunau, janne-ffmpeg at jannau dot net.
Originally committed as revision 23157 to svn://svn.ffmpeg.org/ffmpeg/trunk
ff_parse_and_eval_expr(), place the names for constants/functions
before the corresponding values.
This looks more readable, as the user is expected to know the names
before the values.
Originally committed as revision 23149 to svn://svn.ffmpeg.org/ffmpeg/trunk
Set band info before determining scalefactors. Use the look ahead for
windowing decision.
Originally committed as revision 23132 to svn://svn.ffmpeg.org/ffmpeg/trunk
If the atom size is 0, qt-faststart currently hangs forever while scanning
the file.
Originally committed as revision 23129 to svn://svn.ffmpeg.org/ffmpeg/trunk
also attributes.h is public and external api and can thus not depend
on configure tested compiler support thus this part is removed. A
different solution must be found if this breaks for some compiler
which i hope it does not.
Originally committed as revision 23115 to svn://svn.ffmpeg.org/ffmpeg/trunk
handling of things such as masking/EHB/HAM for this type of data.
Patch by Sebastian Vater <cdgs basty googlemail com>.
Originally committed as revision 23112 to svn://svn.ffmpeg.org/ffmpeg/trunk
x86_64 / Mac OS X gcc 4.0.1
x86_64 / Linux icc (all)
x86_64 / Linux gcc 4.0.4
x86_64 / OpenBSD gcc 3.3.5
x86_64 / Linux suncc 5.10
and there are some reports of crashes.
Originally committed as revision 31170 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
this is 10-20cpu cycles faster on duron (whole is about 50-60 cpu cylses)
I wonder why gcc isnt doing this on its own ...
Originally committed as revision 23097 to svn://svn.ffmpeg.org/ffmpeg/trunk
Most coder functions read it. Carting this around in the context may be
suboptimal; a refactor should be considered.
Originally committed as revision 23094 to svn://svn.ffmpeg.org/ffmpeg/trunk
The latter does not exist and thus compilation fails.
patch by Janne Grunau, janne-ffmpeg jannau.net
Originally committed as revision 23087 to svn://svn.ffmpeg.org/ffmpeg/trunk
was encountered with a channel range that overlapped the previous substreams,
and the code assumed no such overlap was possible.
Patch by Nick Brereton <nick at nbrereton dot net>
Originally committed as revision 23084 to svn://svn.ffmpeg.org/ffmpeg/trunk
long was being incorrectly used as an x86-sized register, both for 32 and 64
bits, but this is not the case in win64.
Originally committed as revision 31153 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
such.
Also put the [3][0][0][0] codec tag, mapped to rgb565le, in a special
section. It needs to be specified *after* the nut RGB[16] codec tag,
otherwise it will be used by default when encoding normal non-flipped
rgb565le, and will be decoded like a flipped format (see
rawdec.c:raw_init_decoder()).
Originally committed as revision 23075 to svn://svn.ffmpeg.org/ffmpeg/trunk
The decodeplane8() function processes one byte of input less than
it should. Also, the for loop has an unusual style with side-effects
in the controlling expression; replaced with a more intuitive while
loop.
10l to Basty.
Originally committed as revision 23063 to svn://svn.ffmpeg.org/ffmpeg/trunk
Use the numeric value assigned to sws_flags for the sws_flags set in
the graph, rather than the string "bilinear", which is not even
parsable by the scale filter.
Originally committed as revision 23060 to svn://svn.ffmpeg.org/ffmpeg/trunk
swscale context.
This makes eventual warnings issued in case of swscale context
creation failure to be shown after the conversion information rather
than before, which is slightly less confusing.
Originally committed as revision 23056 to svn://svn.ffmpeg.org/ffmpeg/trunk
The old code is correct only when stride = 2*width.
Patch by Ronaldo Moura <ronaldo d moura monity com br>
Originally committed as revision 31142 to svn://svn.mplayerhq.hu/mplayer/trunk/libswscale
The Niagara/T1 supports only a subset of VIS, and even this is very slow.
Patch by Michael Kostylev <michael kostylev gmail>
Originally committed as revision 23045 to svn://svn.ffmpeg.org/ffmpeg/trunk
Should provide a meaningful error message for systems which do not
support strerror_r().
Fix roundup issue #1894.
Originally committed as revision 23032 to svn://svn.ffmpeg.org/ffmpeg/trunk
not defined.
This allows applications to check if av_strerror() cannot provide a
meaningful representation for the provided error code, without having
to actually check the filled string.
Originally committed as revision 23031 to svn://svn.ffmpeg.org/ffmpeg/trunk
@item @uref{http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/46_DDEncodingGuidelines.pdf,Dolby Digital Professional Encoding Guidelines}
@end itemize
@subsubsection Metadata Control Options
@table @option
@item -per_frame_metadata @var{boolean}
Allow Per-Frame Metadata. Specifies if the encoder should check for changing
metadata for each frame.
@table @option
@item 0
The metadata values set at initialization will be used for every frame in the
stream. (default)
@item 1
Metadata values can be changed before encoding each frame.
@end table
@end table
@subsubsection Downmix Levels
@table @option
@item -center_mixlev @var{level}
Center Mix Level. The amount of gain the decoder should apply to the center
channel when downmixing to stereo. This field will only be written to the
bitstream if a center channel is present. The value is specified as a scale
factor. There are 3 valid values:
@table @option
@item 0.707
Apply -3dB gain
@item 0.595
Apply -4.5dB gain (default)
@item 0.500
Apply -6dB gain
@end table
@item -surround_mixlev @var{level}
Surround Mix Level. The amount of gain the decoder should apply to the surround
channel(s) when downmixing to stereo. This field will only be written to the
bitstream if one or more surround channels are present. The value is specified
as a scale factor. There are 3 valid values:
@table @option
@item 0.707
Apply -3dB gain
@item 0.500
Apply -6dB gain (default)
@item 0.000
Silence Surround Channel(s)
@end table
@end table
@subsubsection Audio Production Information
Audio Production Information is optional information describing the mixing
environment. Either none or both of the fields are written to the bitstream.
@table @option
@item -mixing_level @var{number}
Mixing Level. Specifies peak sound pressure level (SPL) in the production
environment when the mix was mastered. Valid values are 80 to 111, or -1 for
unknown or not indicated. The default value is -1, but that value cannot be
used if the Audio Production Information is written to the bitstream. Therefore,
if the @code{room_type} option is not the default value, the @code{mixing_level}
option must not be -1.
@item -room_type @var{type}
Room Type. Describes the equalization used during the final mixing session at
the studio or on the dubbing stage. A large room is a dubbing stage with the
industry standard X-curve equalization; a small room has flat equalization.
This field will not be written to the bitstream if both the @code{mixing_level}
option and the @code{room_type} option have the default values.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx large
Large Room
@item 2
@itemx small
Small Room
@end table
@end table
@subsubsection Other Metadata Options
@table @option
@item -copyright @var{boolean}
Copyright Indicator. Specifies whether a copyright exists for this audio.
@table @option
@item 0
@itemx off
No Copyright Exists (default)
@item 1
@itemx on
Copyright Exists
@end table
@item -dialnorm @var{value}
Dialogue Normalization. Indicates how far the average dialogue level of the
program is below digital 100% full scale (0 dBFS). This parameter determines a
level shift during audio reproduction that sets the average volume of the
dialogue to a preset level. The goal is to match volume level between program
sources. A value of -31dB will result in no volume level change, relative to
the source volume, during audio reproduction. Valid values are whole numbers in
the range -31 to -1, with -31 being the default.
@item -dsur_mode @var{mode}
Dolby Surround Mode. Specifies whether the stereo signal uses Dolby Surround
(Pro Logic). This field will only be written to the bitstream if the audio
stream is stereo. Using this option does @b{NOT} mean the encoder will actually
apply Dolby Surround processing.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx off
Not Dolby Surround Encoded
@item 2
@itemx on
Dolby Surround Encoded
@end table
@item -original @var{boolean}
Original Bit Stream Indicator. Specifies whether this audio is from the
original source and not a copy.
@table @option
@item 0
@itemx off
Not Original Source
@item 1
@itemx on
Original Source (default)
@end table
@end table
@subsection Extended Bitstream Information
The extended bitstream options are part of the Alternate Bit Stream Syntax as
specified in Annex D of the A/52:2010 standard. It is grouped into 2 parts.
If any one parameter in a group is specified, all values in that group will be
written to the bitstream. Default values are used for those that are written
but have not been specified. If the mixing levels are written, the decoder
will use these values instead of the ones specified in the @code{center_mixlev}
and @code{surround_mixlev} options if it supports the Alternate Bit Stream
Syntax.
@subsubsection Extended Bitstream Information - Part 1
@table @option
@item -dmix_mode @var{mode}
Preferred Stereo Downmix Mode. Allows the user to select either Lt/Rt
(Dolby Surround) or Lo/Ro (normal stereo) as the preferred stereo downmix mode.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx ltrt
Lt/Rt Downmix Preferred
@item 2
@itemx loro
Lo/Ro Downmix Preferred
@end table
@item -ltrt_cmixlev @var{level}
Lt/Rt Center Mix Level. The amount of gain the decoder should apply to the
center channel when downmixing to stereo in Lt/Rt mode.
@table @option
@item 1.414
Apply +3dB gain
@item 1.189
Apply +1.5dB gain
@item 1.000
Apply 0dB gain
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain (default)
@item 0.500
Apply -6.0dB gain
@item 0.000
Silence Center Channel
@end table
@item -ltrt_surmixlev @var{level}
Lt/Rt Surround Mix Level. The amount of gain the decoder should apply to the
surround channel(s) when downmixing to stereo in Lt/Rt mode.
@table @option
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain
@item 0.500
Apply -6.0dB gain (default)
@item 0.000
Silence Surround Channel(s)
@end table
@item -loro_cmixlev @var{level}
Lo/Ro Center Mix Level. The amount of gain the decoder should apply to the
center channel when downmixing to stereo in Lo/Ro mode.
@table @option
@item 1.414
Apply +3dB gain
@item 1.189
Apply +1.5dB gain
@item 1.000
Apply 0dB gain
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain (default)
@item 0.500
Apply -6.0dB gain
@item 0.000
Silence Center Channel
@end table
@item -loro_surmixlev @var{level}
Lo/Ro Surround Mix Level. The amount of gain the decoder should apply to the
surround channel(s) when downmixing to stereo in Lo/Ro mode.
@table @option
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain
@item 0.500
Apply -6.0dB gain (default)
@item 0.000
Silence Surround Channel(s)
@end table
@end table
@subsubsection Extended Bitstream Information - Part 2
@table @option
@item -dsurex_mode @var{mode}
Dolby Surround EX Mode. Indicates whether the stream uses Dolby Surround EX
(7.1 matrixed to 5.1). Using this option does @b{NOT} mean the encoder will actually
apply Dolby Surround EX processing.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx on
Dolby Surround EX On
@item 2
@itemx off
Dolby Surround EX Off
@end table
@item -dheadphone_mode @var{mode}
Dolby Headphone Mode. Indicates whether the stream uses Dolby Headphone
encoding (multi-channel matrixed to 2.0 for use with headphones). Using this
option does @b{NOT} mean the encoder will actually apply Dolby Headphone
processing.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx on
Dolby Headphone On
@item 2
@itemx off
Dolby Headphone Off
@end table
@item -ad_conv_type @var{type}
A/D Converter Type. Indicates whether the audio has passed through HDCD A/D
conversion.
@table @option
@item 0
@itemx standard
Standard A/D Converter (default)
@item 1
@itemx hdcd
HDCD A/D Converter
@end table
@end table
@subsection Other AC-3 Encoding Options
@table @option
@item -stereo_rematrixing @var{boolean}
Stereo Rematrixing. Enables/Disables use of rematrixing for stereo input. This
is an optional AC-3 feature that increases quality by selectively encoding
the left/right channels as mid/side. This option is enabled by default, and it
is highly recommended that it be left as enabled except for testing purposes.
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