* qatar/master:
rtmp: Add support for SWFVerification
api-example: use new video encoding API.
x86: avcodec: Appropriately name files containing only init functions
mpegvideo_mmx_template: drop some commented-out cruft
libavresample: add mix level normalization option
w32pthreads: Add missing #includes to make header compile standalone
rtmp: Gracefully ignore _checkbw errors by tracking them
rtmp: Do not send _checkbw calls as notifications
prores: interlaced ProRes encoding
Conflicts:
doc/examples/decoding_encoding.c
libavcodec/proresenc_kostya.c
libavcodec/w32pthreads.h
libavcodec/x86/Makefile
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Specifies how the server verifies client SWF files before allowing the
files to connect to an application. Verifying SWF files is a security
measure that prevents someone from creating their own SWF files that can
attempt to stream your resources.
Signed-off-by: Martin Storsjö <martin@martin.st>
The _checkbw calls were changed to use transactionId 0 in commit
82613564 so that servers would not return _result/_error about it.
While this is the strict interpretation of the spec, there are
servers that return _error about it, even if transactionId was 0.
The latest version of EvoStream Media Server (the commercial version
of crtmpserver) behaves properly as described, i.e. returning an
_error normally but not returning anything when using transactionId
0. The latest version of crtmpserver (right now at least) doesn't
behave like this though, it returns an error even if transactionId
was 0.
There are also other servers that return errors even if transactionId
is set to 0. Therefore set a proper transaction id so that the invoke
can be tracked and the error properly ignored instead.
Signed-off-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Michael Bradshaw <mbradshaw@sorensonmedia.com>
Reviewed-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The resolution is in the packets, so decoding must happen.
Since most other formats do not set the dimension, make it
a special case for PGS. If other codecs were to have the
same requirement, using a CODEC_CAP would be preferred.
This also changes behavior as the descriptor table is more complete than
the switch/case it replaces. As well as considering all non video as
intra only
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavf: Detect discontinuities in timestamps for framerate/analyzeduration calculation
lavf: Initialize the stream info timestamps in avformat_new_stream
id3v2: Match PIC mimetype/format case-insensitively
configure: Rename check_asm() to more fitting check_inline_asm()
fate: Only test enabled filters
avresample: De-doxygenize some comments where Doxygen is not appropriate
rtmp: split chunk_size var into in_chunk_size and out_chunk_size
rtmp: Factorize the code by adding find_tracked_method
Conflicts:
configure
Merged-by: Michael Niedermayer <michaelni@gmx.at>
These are normally initialized to AV_NOPTS_VALUE at the start
of avformat_find_stream_info, but if a new stream is found while
this function is running (e.g. like in mpegts), the newly added
AVStreams didn't have these values properly initalized, leading
to avformat_find_stream_info terminating too soon (when the
first timestamps are far from 0).
Signed-off-by: Martin Storsjö <martin@martin.st>
Some files' embedded art seems to have the mimetype 'image/JPG' instead
of 'image/jpg'. Libav fails to parse those because it matches
case-sensitively.
Use av_strncasecmp() to fix this behaviour.
Signed-off-by: Mohammad Alsaleh <msal@tormail.org>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
g723.1: fix addition overflow
g723.1: simplify and fix multiplication overflow
g723.1: deobfuscate an expression
g723.1: remove unused #includes
ARM: add missing "cc" clobber in av_clipl_int32_arm()
rtmp: Factorize the code by adding handle_invoke_error
rtmp: Factorize the code by adding handle_invoke_status
rtmp: Factorize the code by adding handle_invoke_result
libavutil: remove unused av_abort() macro
ffmenc: replace if/abort with assert()
libavutil: drop offsetof() fallback definition
libavutil: drop fallback definitions of INTxx_MIN/MAX
configure: Check for a sctp struct instead of just the header
configure: suncc: Add -xc99 to dependency flags, required on Solaris
doxygen: Fix function parameter names to match the code
doc: Drop obsolete shared libs cflags hint to workaround Cygwin gcc bugs
swf: Move shared table out of the header file
swf: Move swf_audio_codec_tags table to the only place it is used
fate: add G.723.1 decoder tests
Conflicts:
configure
doc/platform.texi
libavformat/Makefile
libavutil/arm/intmath.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This adds a function to retrieve the number of entries in a
dictionary and updates the places directly accessing what should
be an opaque struct to use this new function instead.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The condition is trivially true, but keeping the assert() is
sensible to avoid FFM_HEADER_SIZE ever getting out of sync with
the actual code.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
mpegvideo: reduce excessive inlining of mpeg_motion()
mpegvideo: convert mpegvideo_common.h to a .c file
build: factor out mpegvideo.o dependencies to CONFIG_MPEGVIDEO
Move MASK_ABS macro to libavcodec/mathops.h
x86: move MANGLE() and related macros to libavutil/x86/asm.h
x86: rename libavutil/x86_cpu.h to libavutil/x86/asm.h
aacdec: Don't fall back to the old output configuration when no old configuration is present.
rtmp: Add message tracking
rtsp: Support mpegts in raw udp packets
rtsp: Support receiving plain data over UDP without any RTP encapsulation
rtpdec: Remove an unused include
rtpenc: Remove an av_abort() that depends on user-supplied data
vsrc_movie: discourage its use with avconv.
avconv: allow no input files.
avconv: prevent invalid reads in transcode_init()
avconv: rename OutputStream.is_past_recording_time to finished.
Conflicts:
configure
doc/filters.texi
ffmpeg.c
ffmpeg.h
libavcodec/Makefile
libavcodec/aacdec.c
libavcodec/mpegvideo.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (23 commits)
build: cosmetics: Reorder some lists in a more logical fashion
x86: pngdsp: Fix assembly for OS/2
fate: add test for RTjpeg in nuv with frameheader
rtmp: send check_bw as notification
g723_1: clip argument for 15-bit version of normalize_bits()
g723_1: use all LPC vectors in formant postfilter
id3v2: Support v2.2 PIC
avplay: fix build with lavfi disabled.
avconv: split configuring filter configuration to a separate file.
avconv: split option parsing into a separate file.
mpc8: do not leave padding after last frame in buffer for the next decode call
mpegaudioenc: list supported channel layouts.
mpegaudiodec: don't print an error on > 1 frame in a packet.
api-example: update to new audio encoding API.
configure: add --enable/disable-random option
doc: cygwin: Update list of FATE package requirements
build: Remove all installed headers and header directories on uninstall
build: change checkheaders to use regular build rules
rtmp: Add a new option 'rtmp_subscribe'
rtmp: Add support for subscribing live streams
...
Conflicts:
Makefile
common.mak
configure
doc/examples/decoding_encoding.c
ffmpeg.c
libavcodec/g723_1.c
libavcodec/mpegaudiodec.c
libavcodec/x86/pngdsp.asm
libavformat/version.h
library.mak
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Allow to override the default 'glob_sequence' value, which is deprecated
in favor of the new 'glob' and 'sequence' options.
The new pattern types should be easier on the user since they are more
predictable than 'glob_sequence', and do not require awkward escaping.
This is limited to the chars that arent filtered by av_log() already
we might filter more aggressively if theres some case where this becomes
needed.
Fixes Ticket1181
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
id3 v2.2 uses image format ("JPG","PNG") instead of mimetypes.
Currently, the attached picture is skipped because the format string
does not match a known picture mimetype.
This patch fixes this behaviour.
Signed-off-by: Mohammad Alsaleh <msal@tormail.org>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
When streaming live streams using the Akamai, Edgecast or Limelight CDN,
players cannot simply connect to the live stream. Instead, they have to
subscribe to it, by sending an FC Subscribe call to the server.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'f5d2c597e99af218b0d4d1cf9737c7e68ee934e4':
build: fix library installation on cygwin
mpc8: add a flush function
mpc8: set packet duration and stream start time instead of tracking frames
Conflicts:
libavformat/mpc8.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
At this place, the normal way of initializing a struct works
fine, there's no need for a struct literal.
Signed-off-by: Martin Storsjö <martin@martin.st>
The previous method of having to initialize it outside lead
to incorrect code: even if it was initialized, it usually was
only initialized once, thus a packet that could not be matched
to any stream would just be processed with the return values
from the previous call.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Also slightly more correct behaviour in case streams_left for
some reason is 0 from the start.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Fixes crash based on a uninitialized array index read.
If the read does not crash then out of array writes based
on the same index might have been triggered afterwards.
Found-by: inferno@chromium.org
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The new name seems more consistent with the assumed logic.
"start_index" represents the minimum accepted value as first index, and
not the maximum value as implicitely assumed by the previous name.
The current demuxer does not bother to write packet durations,
which makes it impossible to remux into a new format.
Signed-off-by: Philip Langdale <philipl@overt.org>
As packet duration is not stored inherently in MPEG4 containers,
subtitles have their duration expressed by storing an additional
empty packet with a pts matching the desired end time of the real
subtitle. Additionally, it is generally expected that all streams
start at time = 0, so an empty packet needs to be inserted at the
beginning of the stream, before the first real subtitle.
Unfortunately, ffmpeg lacks a proper way to express that a subtitle
might map to multiple packets, so the muxer is the only place we
can handle this.
Signed-off-by: Philip Langdale <philipl@overt.org>
This is almost a revert of: (the file from the report still works)
commit 80e58c6153
Author: Benoit Fouet <benoit.fouet@free.fr>
Date: Wed Feb 11 11:09:36 2009 +0000
Allow demuxing of audio substreams stored as 0x06 type.
Fixes issue 725: MPEG2 PS with PCM audio.
On behalf of Jai.
Originally committed as revision 17150 to svn://svn.ffmpeg.org/ffmpeg/trunk
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
dca: Switch dca_sample_rates to avpriv_ prefix; it is used across libs
ARM: use =const syntax instead of explicit literal pools
ARM: use standard syntax for all LDRD/STRD instructions
fft: port FFT/IMDCT 3dnow functions to yasm, and disable on x86-64.
dct-test: allow to compile without HAVE_INLINE_ASM.
x86/dsputilenc: bury inline asm under HAVE_INLINE_ASM.
dca: Move tables used outside of dcadec.c to a separate file.
dca: Rename dca.c ---> dcadec.c
x86: h264dsp: Remove unused variable ff_pb_3_1
apetag: change a forgotten return to return 0
Conflicts:
libavcodec/Makefile
libavcodec/dca.c
libavcodec/x86/fft_3dn.c
libavcodec/x86/fft_3dn2.c
libavcodec/x86/fft_mmx.asm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mpc8: return more meaningful error codes.
mpc: return more meaningful error codes.
wv,mpc8: don't return apetag data in packets.
rtmp: do not warn about receiving metadata packets
x86: h264dsp: Adjust YASM #ifdefs
x86: yadif: Mark mmxext optimizations as such
h264: convert loop filter strength dsp function to yasm.
Improve descriptiveness of a number of codec and container long names
Conflicts:
libavcodec/flvdec.c
libavcodec/libopenjpegdec.c
libavformat/apetag.c
libavformat/mp3dec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
avformat: Drop pointless "format" from container long names
swscale: bury one more piece of inline asm under HAVE_INLINE_ASM.
wv: K&R formatting cosmetics
configure: Add missing descriptions to help output
h264_ps: declare array of colorspace strings on its own line.
fate: amix: specify f32 sample format for comparison
tiny_psnr: support 32-bit float samples
eamad/eatgq/eatqi: call special EA IDCT directly
eamad: remove use of MpegEncContext
mpegvideo: remove unnecessary inclusions of faandct.h
af_asyncts: avoid overflow in out_size with large delta values
af_asyncts: add first_pts option
Conflicts:
configure
libavcodec/eamad.c
libavcodec/h264_ps.c
libavformat/crcenc.c
libavformat/ffmdec.c
libavformat/ffmenc.c
libavformat/framecrcenc.c
libavformat/md5enc.c
libavformat/nutdec.c
libavformat/rawenc.c
libavformat/yuv4mpeg.c
tests/tiny_psnr.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
flvdec: remove spurious use of stream id
lavf: deprecate r_frame_rate.
lavf: round estimated average fps to a "standard" fps.
Conflicts:
ffmpeg.c
ffprobe.c
libavformat/avformat.h
libavformat/electronicarts.c
libavformat/flvdec.c
libavformat/rawdec.c
libavformat/utils.c
tests/ref/fate/iv8-demux
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'fe1c1198e670242f3cf9e3e1eef27cff77f3ee23':
lavf: use dts difference instead of AVPacket.duration in find_stream_info()
avf: introduce nobuffer option
fate: make yadif tests consistent across systems
vf_hqdn3d: support 9 and 10bit colordepth
vf_hqdn3d: reduce intermediate precision
vf_hqdn3d: simplify and optimize
factor identical ff_inplace_start_frame out of two filters
vf_hqdn3d: cosmetics
avprobe/avconv: fix tentative declaration compile errors on MSVS.
Conflicts:
doc/APIchanges
ffmpeg.c
ffprobe.c
libavformat/avformat.h
libavformat/options_table.h
libavformat/utils.c
libavformat/version.h
tests/fate/filter.mak
tests/ref/fate/filter-yadif-mode0
tests/ref/fate/filter-yadif-mode1
Merged-by: Michael Niedermayer <michaelni@gmx.at>
According to its description, it is supposed to be the LCM of all the
frame durations. The usability of such a thing is vanishingly small,
especially since we cannot determine it with any amount of reliability.
Therefore get rid of it after the next bump.
Replace it with the average framerate where it makes sense.
FATE results for the wtv and xmv demux tests change. In the wtv case
this is caused by the file being corrupted (or possibly badly cut) and
containing invalid timestamps. This results in lavf estimating the
framerate wrong and making up wrong frame durations.
In the xmv case the file contains pts jumps, so again the estimated
framerate is far from anything sane and lavf again makes up different
frame durations.
In some other tests lavf starts making up frame durations from different
frame.
AVPacket.duration is mostly made up and thus completely useless, this is
especially true for video streams.
Therefore use dts difference for framerate estimation and
the max_analyze_duration check.
The asyncts test now needs -analyzeduration, because the default is 5
seconds and the audio stream in the sample appears at ~10 seconds.
Useful in cases where a significant analyzeduration is
still needed, while minimizing buffering before output.
An example is processing low-latency streams where all
media types won't necessarily come in if the
analyzeduration is small.
Additional changes by Josh Allmann <joshua.allmann@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master: (35 commits)
h264_idct_10bit: port x86 assembly to cpuflags.
x86inc: clip num_args to 7 on x86-32.
x86inc: sync to latest version from x264.
fft: rename "z" to "zc" to prevent name collision.
wv: return meaningful error codes.
wv: return AVERROR_EOF on EOF, not EIO.
mp3dec: forward errors for av_get_packet().
mp3dec: remove a pointless local variable.
mp3dec: remove commented out cruft.
lavfi: bump minor to mark stabilizing the ABI.
FATE: add tests for yadif.
FATE: add a test for delogo video filter.
FATE: add a test for amix audio filter.
audiogen: allow specifying random seed as a commandline parameter.
vc1dec: Override invalid macroblock quantizer
vc1: avoid reading beyond the last line in vc1_draw_sprites()
vc1dec: check that coded slice positions and interlacing match.
vc1dec: Do not ignore ff_vc1_parse_frame_header_adv return value
configure: Move parts that should not be user-selectable to CONFIG_EXTRA
lavf: remove commented out cruft in avformat_find_stream_info()
...
Conflicts:
Makefile
configure
libavcodec/vc1dec.c
libavcodec/x86/h264_deblock.asm
libavcodec/x86/h264_deblock_10bit.asm
libavcodec/x86/h264dsp_mmx.c
libavfilter/version.h
libavformat/mp3dec.c
libavformat/utils.c
libavformat/wv.c
libavutil/x86/x86inc.asm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Previously, we returned any error code except AVERROR_EOF to the
caller - only if AVERROR_EOF or 0 was returned, we proceeded to
the next segment.
With some setups of web servers, using Connection: close in https
and GnuTLS, we don't get a clean error code at the end of segments.
In those cases, just proceed to the next segment.
Tested-by: Antti Seppälä <a.seppala@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
OpenSSL returns 0 when the peer has closed the connection. GnuTLS
doesn't return that though, but returns
GNUTLS_E_UNEXPECTED_PACKET_LENGTH if the connection simply is closed
without a clean close notify packet.
Tested-by: Antti Seppälä <a.seppala@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
proresdsp: port x86 assembly to cpuflags.
lavr: x86: improve non-SSE4 version of S16_TO_S32_SX macro
lavfi: better channel layout negotiation
alac: check for truncated packets
alac: reverse lpc coeff order, simplify filter
lavr: add x86-optimized mixing functions
x86: add support for fmaddps fma4 instruction with abstraction to avx/sse
tscc2: fix typo in array index
build: use COMPILE template for HOSTOBJS
build: do full flag handling for all compiler-type tools
eval: fix printing of NaN in eval fate test.
build: Rename aandct component to more descriptive aandcttables
mpegaudio: bury inline asm under HAVE_INLINE_ASM.
x86inc: automatically insert vzeroupper for YMM functions.
rtmp: Check the buffer length of ping packets
rtmp: Allow having more unknown data at the end of a chunk size packet without failing
rtmp: Prevent reading outside of an allocate buffer when receiving server bandwidth packets
Conflicts:
Makefile
configure
libavcodec/x86/proresdsp.asm
libavutil/eval.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Without this fix the last sample was missing from the packet.
Signed-off-by: Marton Balint <cus@passwd.hu>
Reviewed-by: Matthieu Bouron <matthieu.bouron@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
libopenjpeg: support YUV and deep RGB pixel formats
Fix typo in v410 decoder.
vf_yadif: unset cur_buf on the input link.
vf_overlay: ensure the overlay frame does not get leaked.
vf_overlay: prevent premature freeing of cur_buf
Support urlencoded http authentication credentials
rtmp: Return an error when the client bandwidth is incorrect
rtmp: Return proper error code in handle_server_bw
rtmp: Return proper error code in handle_client_bw
rtmp: Return proper error codes in handle_chunk_size
lavr: x86: add missing vzeroupper in ff_mix_1_to_2_fltp_flt()
vp8: Replace x*155/100 by x*101581>>16.
vp3: don't use calls to inline asm in yasm code.
x86/dsputil: put inline asm under HAVE_INLINE_ASM.
dsputil_mmx: fix incorrect assembly code
rtmp: Factorize the code by adding handle_invoke
rtmp: Factorize the code by adding handle_chunk_size
rtmp: Factorize the code by adding handle_ping
rtmp: Factorize the code by adding handle_client_bw
rtmp: Factorize the code by adding handle_server_bw
Conflicts:
libavcodec/libopenjpegdec.c
libavcodec/x86/dsputil_mmx.c
libavfilter/vf_overlay.c
libavformat/Makefile
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
By moving it to a later point relative and unknown timestamps
are more likely to have been corrected
similar patch reviewed-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Conflicts:
libavformat/utils.c
It should be possible to specify usernames in http requests containing
urlencoded characters. This patch adds support for decoding the auth
strings.
Signed-off-by: Antti Seppälä <a.seppala@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
rtmp: Add a new option 'rtmp_pageurl'
doc: Update the description of the rtmp_tcurl option
rtmp: Make the description of the rtmp_tcurl option more generic
libfdk-aacenc: add LATM/LOAS encapsulation support
sctp: add port missing error message
tcp: add port missing error message
avfilter: Fix printf format string conversion specifier
Conflicts:
libavcodec/version.h
libavfilter/avfilter.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Without this patch a user a bit absent-minded may not notice that
the connection doesn't work because the port is missing.
Signed-off-by: Martin Storsjö <martin@martin.st>
Without this patch a user a bit absent-minded may not notice that
the connection doesn't work because the port is missing.
Signed-off-by: Martin Storsjö <martin@martin.st>
The native decoder and MPlayer's binary decoder only need the
APRG atom, QuickTime at least requires also the ARES atom and
four additional 0 bytes padding at the end of stsd.
Attached patch (together with demuxing patch) allows WMP/msacm G723.1 codec decode files encoded by FFmpeg.
Tested with both 6400 and 5333 mode
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtmp: Add credit/copyright to librtmp authors for parts of the RTMPE code
rtmp: Move the CONFIG_ condition into the if conditions
aac: Mention abbreviation as well in long_name
build: Skip compiling rtmpdh.h if ffrtmpcrypt protocol is not enabled
doc: Add Git configuration section
configure: Add a dependency on https for rtmpts
rtp: Only choose static payload types if the sample rate and channels are right
Conflicts:
doc/git-howto.texi
libavformat/rtmpproto.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Attached patch fixes remuxing of G723.1 in wav, so the output is playable by WMP.
(It's still not enough for encoding - probably some extradata should be added to the output file
to make it playable by WMP/win codec)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This makes sure these calls are removed by dead code elimination
even if optimization is disabled. This fixes building without
crypto libraries without optimization.
Signed-off-by: Martin Storsjö <martin@martin.st>
If using a different sample rate or number of channels, use a dynamic
payload type instead, where the parameters are passed in the SDP.
G722 is a special case where the normal rules don't apply.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
v410dec: Implement explode mode support
zerocodec: fix direct rendering.
wav: init st to NULL to avoid a false-positive warning.
wavpack: set bits_per_raw_sample for S32 samples to properly identify 24-bit
h264: refactor NAL decode loop
RTMPTE protocol support
RTMPE protocol support
rtmp: Add ff_rtmp_calc_digest_pos()
rtmp: Rename rtmp_calc_digest to ff_rtmp_calc_digest and make it global
swscale: add missing HAVE_INLINE_ASM check.
lavfi: place x86 inline assembly under HAVE_INLINE_ASM.
vc1: Add a test for interlaced field pictures
swscale: Mark all init functions as av_cold
swscale: x86: Drop pointless _mmx suffix from filenames
lavf: use conditional notation for default codec in muxer declarations.
swscale: place inline assembly bilinear scaler under HAVE_INLINE_ASM.
dsputil: ppc: cosmetics: pretty-print
dsputil: x86: add SHUFFLE_MASK_W macro
configure: respect CC_O setting in check_cc
Conflicts:
Changelog
configure
libavcodec/v410dec.c
libavcodec/zerocodec.c
libavformat/asfenc.c
libavformat/version.h
libswscale/utils.c
libswscale/x86/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If st is NULL, it means no 'fmt ' tag is found, but 'data' tag (which
needs a previous 'fmt ' tag to be parsed correctly and st initialized)
check will make sure st is never dereferenced in that case.
Fixes warning:
libavformat/wav.c: In function ‘wav_read_header’:
libavformat/wav.c:499:44: warning: ‘st’ may be used uninitialized in this function [-Wmaybe-uninitialized]
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This adds two protocols, but one of them is an internal implementation
detail just used as an abstraction layer/generalization in the code. The
RTMPE protocol implementation uses ffrtmpcrypt:// as an alternative to the
tcp:// protocol. This allows moving most of the lower level logic out
from the higher level generic rtmp code.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
FATE: fix the asyncts test
build: Drop gcc-specific warning flag from header compilation rule
FATE: add a test for the asyncts audio filter.
matroskadec: return more correct error code on read error.
buffersrc: check ff_get_audio_buffer() for errors.
lavfi: check all ff_get_video_buffer() calls for errors.
lavfi: check all avfilter_ref_buffer() calls for errors.
vf_select: avoid an unnecessary avfilter_ref_buffer().
buffersrc: avoid creating unnecessary buffer reference
lavfi: use avfilter_unref_bufferp() where appropriate.
vf_fps: add more error checks.
vf_fps: fix a memleak on malloc failure.
lavfi: check all ff_start_frame/draw_slice/end_frame calls for errors
lavfi: add error handling to end_frame().
lavfi: add error handling to draw_slice().
lavfi: add error handling to start_frame().
Conflicts:
Makefile
ffplay.c
libavfilter/buffersrc.c
libavfilter/vf_boxblur.c
libavfilter/vf_drawtext.c
libavfilter/vf_fade.c
libavfilter/vf_frei0r.c
libavfilter/vf_hflip.c
libavfilter/vf_overlay.c
libavfilter/vf_pad.c
libavfilter/vf_scale.c
libavfilter/video.c
libavfilter/vsrc_color.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
x86: swscale: Place inline assembly code under appropriate #ifdefs
rtsp: remove terminal comma in FF_RTP_FLAG_OPTS macro.
configure: Remove redundant RTMPT/RTMPTS dependencies
configure: add filtering of host cflags/ldflags
configure: initialise all flag filters at the same place
configure: add filtering of linker flags
configure: name some variables more consistently
configure: remove filter_cppflags
configure: set icc_version where it is needed
mpegenc: remove disabled code
Conflicts:
configure
libavformat/movenc.c
libswscale/x86/swscale_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This happens with for example mplayer.
Fixing it in ffmpeg allows new ffmpeg to be compiled with older mplayer
which would not be possible if the fix was just in mplayer.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If skip_samples is set and timestamps are synthesized using durations,
make them start at -skip_samples (rescaled) instead of 0,
so that the timestamp of the first undiscarded sample is 0.
* qatar/master: (38 commits)
alac: cosmetics: general pretty-printing and comment clean up
alac: calculate buffer size outside the loop in allocate_buffers()
alac: change some data types to plain int
alac: cosmetics: rename some variables and function names
alac: multi-channel decoding support
alac: split element parsing into a separate function
alac: support a read sample size of up to 32
alac: output in planar sample format
alac: add 32-bit decoding support
alac: simplify channel interleaving
alac: use AVPacket fields directly in alac_decode_frame()
alac: fix check for valid max_samples_per_frame
alac: use get_sbits() to read LPC coefficients instead of casting
alac: move the current samples per frame to the ALACContext
alac: avoid using a double-negative when checking if the frame is compressed
alac: factor out output_size check in predictor_decompress_fir_adapt()
alac: factor out loading of next decoded sample in LPC prediction
alac: use index into buffer_out instead of incrementing the pointer
alac: simplify lpc coefficient adaptation
alac: reduce the number of local variables needed in lpc prediction
...
Conflicts:
libavcodec/alac.c
libavformat/cafdec.c
libavformat/mov.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
