While not explicitly stated in the specs, the original author
has stated that S_TEXT/UTF-8 is expected to be text using Subrip
markup, but without Subrip in-band timing.
So, now that we have a decoder that conforms to this expectation,
let's use it.
Note that this change will impact tools that use libavformat. If
they expect srt subtitles to have CODEC_ID_TEXT, they must be
adjusted to expect CODEC_ID_SUBRIP. The actual content is, obviously,
unchanged.
Signed-off-by: Philip Langdale <philipl@overt.org>
* qatar/master:
doc/APIchanges: add an entry for codec descriptors.
vorbisenc: set AVCodecContext.bit_rate to 0
vorbisenc: fix quality parameter
FATE: add ALAC encoding tests
lpc: fix alignment of windowed samples for odd maximum LPC order
alacenc: use s16p sample format as input
alacenc: remove unneeded sample_fmt check
alacenc: fix max_frame_size calculation for the final frame
adpcm_swf: Use correct sample offsets when using trellis.
rtmp: support strict rtmp servers
mjpegdec: support AVRn interlaced
x86: remove FASTDIV inline asm
Conflicts:
doc/APIchanges
libavcodec/mjpegdec.c
libavcodec/vorbisenc.c
libavutil/x86/intmath.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In order to send or receive a stream FCPublish, FCSubscribe and _checkbw
are completely optional and often not implemented. releaseStream over a
non-existen stream might report an error instead of being silent.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The special cases in demuxers and decoders are a mess otherwise (and more
would be needed to support it fully)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mpegvideo_enc: don't use deprecated avcodec_encode_video().
cmdutils: refactor -codecs option.
avconv: make -shortest a per-output file option.
lavc: add avcodec_descriptor_get_by_name().
lavc: add const to AVCodec* function parameters.
swf(dec): replace CODEC_ID with AV_CODEC_ID
dvenc: don't use deprecated AVCODEC_MAX_AUDIO_FRAME_SIZE
rtmpdh: Do not generate the same private key every time when using libnettle
rtp: remove ff_rtp_get_rtcp_file_handle().
rtsp.c: use ffurl_get_multi_file_handle() instead of ff_rtp_get_rtcp_file_handle()
avio: add (ff)url_get_multi_file_handle() for getting more than one fd
h264: vdpau: fix crash with unsupported colorspace
amrwbdec: Decode the fr_quality bit properly
Conflicts:
Changelog
cmdutils.c
cmdutils_common_opts.h
doc/ffmpeg.texi
ffmpeg.c
ffmpeg.h
ffmpeg_opt.c
libavcodec/h264.c
libavcodec/options.c
libavcodec/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Replace mpz_random by mpz_urandomb with a random state initialization in
order to improve the randomness.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
libvpxenc: use the default bitrate if not set
utvideo: Rename utvideo.c to utvideodec.c
doc: Fix syntax errors in sample Emacs config
mjpegdec: more meaningful return values
configure: clean up Altivec detection
getopt: Remove an unnecessary define
rtmp: Use int instead of ssize_t
getopt: Add missing includes
rtmp: Add support for receiving incoming streams
Add missing includes for code relying on external libraries
Conflicts:
libavcodec/libopenjpegenc.c
libavcodec/libvpxenc.c
libavcodec/mjpegdec.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Not all compilers support ssize_t (MSVC doesn't), and none of these
variables need to be larger than 32 bit.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
Fix even more missing includes after the common.h removal
build: Factor out rangecoder dependencies to CONFIG_RANGECODER
build: Factor out error resilience dependencies to CONFIG_ERROR_RESILIENCE
x86: avcodec: Consistently name all init files
Add more missing includes after removing the implicit common.h
Add some more missing includes after removing the implicit common.h
Don't include common.h from avutil.h
rtmp: Automatically compute the hash for SWFVerification
Conflicts:
configure
doc/APIchanges
doc/examples/decoding_encoding.c
libavcodec/Makefile
libavcodec/assdec.c
libavcodec/audio_frame_queue.c
libavcodec/avpacket.c
libavcodec/dv_profile.c
libavcodec/dwt.c
libavcodec/libtheoraenc.c
libavcodec/rawdec.c
libavcodec/rv40dsp.c
libavcodec/tiff.c
libavcodec/tiffenc.c
libavcodec/v210dec.h
libavcodec/vc1dsp.c
libavcodec/x86/Makefile
libavfilter/asrc_anullsrc.c
libavfilter/avfilter.c
libavfilter/buffer.c
libavfilter/formats.c
libavfilter/vf_ass.c
libavfilter/vf_drawtext.c
libavfilter/vf_fade.c
libavfilter/vf_select.c
libavfilter/video.c
libavfilter/vsrc_testsrc.c
libavformat/version.h
libavutil/audioconvert.c
libavutil/error.h
libavutil/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Unsurprisingly, if a timing-less subrip decoder is desireable, an
encoder is as well. With this in place, we can move on to remove
the use of the old encoder/decoder with embedded timing and move
all timing handling the (de)muxer where they belong.
Signed-off-by: Philip Langdale <philipl@overt.org>
* 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>
