* 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>
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.
* qatar/master:
avcodec: add a cook parser to get subpacket duration
FATE: allow lavf tests to alter input parameters
FATE: replace the acodec-pcm_s24daud test with an enc_dec_pcm checksum test
FATE: replace the acodec-g726 test with 4 new encode/decode tests
FATE: replace current g722 encoding tests with an encode/decode test
FATE: add a pattern rule for generating asynth wav files
FATE: optionally write a WAVE header in audiogen
avutil: add audio fifo buffer
Conflicts:
doc/APIchanges
libavcodec/version.h
libavutil/avutil.h
tests/Makefile
tests/codec-regression.sh
tests/fate/voice.mak
tests/lavf-regression.sh
tests/ref/acodec/g722
tests/ref/acodec/g726
tests/ref/acodec/pcm_s24daud
tests/ref/lavf/dv_fmt
tests/ref/lavf/gxf
tests/ref/lavf/mxf
tests/ref/lavf/mxf_d10
tests/ref/seek/lavf_dv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (35 commits)
fix space type in Changelog
ZeroCodec Decoder
RealAudio Lossless decoder
rtpenc: Use AVFormatContext.packet_size instead of a private option
url: Document the expected behaviour of url_read
libavformat: Use AVFormatContext.probesize in init_input
docs: Fix a stray reference to tags in the generic doxy on dicts
cosmetics: Align some AVInput/OutputFormat declarations
zmbv: check decompress result
zmbv: correct indentation
adpcm: convert adpcm_thp to bytestream2.
adpcm: convert adpcm_yamaha to bytestream2.
adpcm: convert adpcm_swf to bytestream2.
adpcm: convert adpcm_sbpro to bytestream2.
adpcm: convert adpcm_ct to bytestream2.
adpcm: convert adpcm_ima_amv/smjpeg to bytestream2.
adpcm: convert adpcm_ea_xas to bytestream2.
adpcm: convert adpcm_ea_r1/2/3 to bytestream2.
adpcm: convert ea_maxis_xa to bytestream2.
adpcm: convert adpcm_ea to bytestream2.
...
Conflicts:
Changelog
libavcodec/Makefile
libavcodec/adpcm.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/version.h
libavcodec/zerocodec.c
libavcodec/zmbv.c
libavformat/riff.c
libavformat/url.h
tests/ref/fate/truemotion1-15
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (29 commits)
amrwb: remove duplicate arguments from extrapolate_isf().
amrwb: error out early if mode is invalid.
h264: change underread for 10bit QPEL to overread.
matroska: check buffer size for RM-style byte reordering.
vp8: disable mmx functions with sse/sse2 counterparts on x86-64.
vp8: change int stride to ptrdiff_t stride.
wma: fix invalid buffer size assumptions causing random overreads.
Windows Media Audio Lossless decoder
rv10/20: Fix slice overflow with checked bitstream reader.
h263dec: Disallow width/height changing with frame threads.
rv10/20: Fix a buffer overread caused by losing track of the remaining buffer size.
rmdec: Honor .RMF tag size rather than assuming 18.
g722: Fix the QMF scaling
r3d: don't set codec timebase.
electronicarts: set timebase for tgv video.
electronicarts: parse the framerate for cmv video.
ogg: don't set codec timebase
electronicarts: don't set codec timebase
avs: don't set codec timebase
wavpack: Fix an integer overflow
...
Conflicts:
libavcodec/arm/vp8dsp_init_arm.c
libavcodec/fraps.c
libavcodec/h264.c
libavcodec/mpeg4videodec.c
libavcodec/mpegvideo.c
libavcodec/msmpeg4.c
libavcodec/pnmdec.c
libavcodec/qpeg.c
libavcodec/rawenc.c
libavcodec/ulti.c
libavcodec/vcr1.c
libavcodec/version.h
libavcodec/wmalosslessdec.c
libavformat/electronicarts.c
libswscale/ppc/yuv2rgb_altivec.c
tests/ref/acodec/g722
tests/ref/fate/ea-cmv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
dxva2: don't check for DXVA_PictureParameters->wDecodedPictureIndex
img2: split muxer and demuxer into separate files
rm: prevent infinite loops for index parsing.
aac: fix infinite loop on end-of-frame with sequence of 1-bits.
mov: Add more HDV and XDCAM FourCCs.
lavf: don't set AVCodecContext.has_b_frames in compute_pkt_fields().
rmdec: when using INT4 deinterleaving, error out if sub_packet_h <= 1.
cdxl: correctly synchronize video timestamps to audio
mlpdec_parser: fix a few channel layouts.
Add channel names to channel_names[] array for channels added in b2890f5
movenc: Buffer the mdat for the initial moov fragment, too
flvdec: Ignore the index if the ignidx flag is set
flvdec: Fix indentation
movdec: Don't parse all fragments if ignidx is set
movdec: Restart parsing root-level atoms at the right spot
prores: use natural integer type for the codebook index
mov: Add support for MPEG2 HDV 720p24 (hdv4)
swscale: K&R formatting cosmetics (part I)
swscale: variable declaration and placement cosmetics
Conflicts:
configure
libavcodec/aacdec.c
libavcodec/mlp_parser.c
libavformat/flvdec.c
libavformat/img2.c
libavformat/isom.h
libavformat/mov.c
libavformat/movenc.c
libswscale/rgb2rgb.c
libswscale/rgb2rgb_template.c
libswscale/yuv2rgb.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Specifically, prevent jumping back in the file for the next index, since
this can lead to infinite loops where we jump between indexes referring
to each other, and don't read indexes that don't fit in the file.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
We read sub_packet_h / 2 packets per line of data (during deinterleaving),
which equals zero if sub_packet_h <= 1, thus causing us to not read any
data, leading to an infinite loop.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
* qatar/master: (29 commits)
cabac: Move code only used within the CABAC test program into the test program.
vp56: Drop unnecessary cabac.h #include.
h264-test: Initialize AVCodecContext.av_class.
build: Skip compiling network.h and rtsp.h if networking is not enabled.
cosmetics: drop some pointless parentheses
Disable annoying warning without changing behavior
faq: Solutions for common problems with sample paths when running FATE.
avcodec: attempt to clarify the CODEC_CAP_DELAY documentation
avcodec: fix avcodec_encode_audio() documentation.
FATE: xmv-demux test; exercise the XMV demuxer without decoding the perceptual codecs inside.
vqf: recognize more metadata chunks
FATE test: BMV demuxer and associated video and audio decoders.
FATE: indeo4 video decoder test.
FATE: update xxan-wc4 test to a sample with more code coverage.
Change the recent h264_mp4toannexb bitstream filter test to output to an elementary stream rather than a program stream.
g722enc: validate AVCodecContext.trellis
g722enc: set frame_size, and also handle an odd number of input samples
g722enc: split encoding into separate functions for trellis vs. no trellis
mpegaudiodec: Use clearer pointer math
tta: Fix returned error code at EOF
...
Conflicts:
libavcodec/h264.c
libavcodec/indeo3.c
libavcodec/interplayvideo.c
libavcodec/ivi_common.c
libavcodec/libxvidff.c
libavcodec/mpegvideo.c
libavcodec/ppc/mpegvideo_altivec.c
libavcodec/tta.c
libavcodec/utils.c
libavfilter/vsrc_buffer.c
libavformat/Makefile
tests/fate/indeo.mak
tests/ref/acodec/g722
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtpdec: Templatize the code for different g726 bitrate variants
rv40: move loop filter to rv34dsp context
lavf: make av_set_pts_info private.
rtpdec: Add support for G726 audio
rtpdec: Add an init function that can do custom codec context initialization
avconv: make copy_tb on by default.
matroskadec: don't set codec timebase.
rmdec: don't set codec timebase.
avconv: compute next_pts from input packet duration when possible.
lavf: estimate frame duration from r_frame_rate.
avconv: update InputStream.pts in the streamcopy case.
Conflicts:
avconv.c
libavdevice/alsa-audio-dec.c
libavdevice/bktr.c
libavdevice/fbdev.c
libavdevice/libdc1394.c
libavdevice/oss_audio.c
libavdevice/v4l.c
libavdevice/v4l2.c
libavdevice/vfwcap.c
libavdevice/x11grab.c
libavformat/au.c
libavformat/eacdata.c
libavformat/flvdec.c
libavformat/mpegts.c
libavformat/mxfenc.c
libavformat/rtpdec_g726.c
libavformat/wtv.c
libavformat/xmv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Also reduce verbosity for the unsupported stream message, use
an AVFormatContext for av_log and and print the tag of the
unknown stream.
Improves ticket #672.
* qatar/master:
avformat: Avoid a warning about mixed declarations and code
BMV demuxer and decoder
matroskaenc: Make sure the seekhead struct is freed even on seek failure
mpeg12enc: Remove write-only variables.
mpeg12enc: Don't set up run-level info for level 0.
msmpeg4: Don't set up run-level info for level 0.
avformat: Warn about using network functions without calling avformat_network_init
avformat: Revise wording
rdt: Set AVFMT_NOFILE on ff_rdt_demuxer
rdt: Check the return value of avformat_open
rtsp: Discard the dynamic handler, if it has an alloc function which failed
dsputil: use cpuflags in x86 versions of vector_clip_int32()
Conflicts:
libavcodec/avcodec.h
libavcodec/version.h
libavformat/Makefile
libavformat/allformats.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This makes rdt work again, which has been broken since
603b8bc2a1. This commit made
opening a demuxer without a file (or in this case, with a filename
which can't be opened) fail, unless the demuxer actually declared
AVFMT_NOFILE.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
presets: rename presets directory
lavc: make avcodec_get_context_defaults3 "officially" public
lavf: replace av_new_stream->avformat_new_stream part II.
lavf,lavd: replace av_new_stream->avformat_new_stream part I.
lavf: add avformat_new_stream as a replacement for av_new_stream.
Use correct scaling table for bwd-pred MVs in second B-field
Ut Video decoder
Makefile: change presets extension to .avpreset
lavfi: add rgbtestsrc source, ported from MPlayer libmpcodecs
lavfi: add testsrc source
AVOptions: add documentation.
presets: update libx264 ffpresets
Conflicts:
Changelog
doc/APIchanges
doc/ffmpeg.texi
ffpresets/libx264-ipod320.ffpreset
ffpresets/libx264-ipod640.ffpreset
ffserver.c
libavcodec/avcodec.h
libavcodec/options.c
libavcodec/version.h
libavdevice/libdc1394.c
libavfilter/avfilter.h
libavfilter/vsrc_testsrc.c
libavformat/flvdec.c
libavformat/riff.c
libavformat/version.h
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
movenc: fix NULL reference in mov_write_tkhd_tag
rmdec: Reject invalid deinterleaving parameters
rv34: Fix potential overreads
rv34: Fix buffer size used for MC of B frames after a resolution change
rv34: Avoid NULL dereference on corrupted bitstream
rv10: Reject slices that does not have the same type as the first one
vf_yadif: add an option to enable/disable deinterlacing based on src frame "interlaced" flag
vsrc_color: set output pos values to -1
vsrc_color: add @file doxy
vsrc_buffer: remove duplicated file description
eval: implement not() expression
eval: add sqrt function for computing the square root
rmdec: use the deinterleaving mode and not the codec when creating audio packets.
lavf: Fix context pointer in av_open_input_stream when avformat_open_input fails
Conflicts:
doc/eval.texi
doc/filters.texi
libavcodec/rv10.c
libavfilter/vsrc_color.c
libavformat/rmdec.c
libavutil/avutil.h
libavutil/eval.c
tests/ref/fate/eval
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Use deinterleavers for demangling audio packets in RealMedia.
vf_scale: don't leak SWS context.
doxygen: drop another pointless star from pointer variable name
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Unlike other containers RealMedia stores its audio packets in scrambled form,
with interleaver ID preceeding audio codec ID. Currently deinterleaving
decision is tied to the codec while it's possible to have non-default
deinterleaver with audio codec (like Int0 deinterleaver instead of specific
one for Sipro).
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master: (23 commits)
h264: hide reference frame errors unless requested
swscale: split hScale() function pointer into h[cy]Scale().
Move clipd macros to x86util.asm.
avconv: reindent.
avconv: rescue poor abused start_time global.
avconv: rescue poor abused recording_time global.
avconv: merge two loops in output_packet().
avconv: fix broken indentation.
avconv: get rid of the arbitrary MAX_FILES limit.
avconv: get rid of the output_streams_for_file vs. ost_table schizophrenia
avconv: add a wrapper for output AVFormatContexts and merge output_opts into it
avconv: make itsscale syntax consistent with other options.
avconv: factor out adding input streams.
avconv: Factorize combining auto vsync with format.
avconv: Factorize video resampling.
avconv: Don't unnecessarily convert ipts to a double.
ffmpeg: remove unsed variable nopts
RV3/4 parser: remove unused variable 'off'
add XMV demuxer
rmdec: parse FPS in RealMedia properly
...
Conflicts:
avconv.c
libavformat/version.h
libswscale/swscale.c
tests/ref/fate/lmlm4-demux
Merged-by: Michael Niedermayer <michaelni@gmx.at>
First, container stores only DTS and not PTS as it was believed.
Second, multiple frames in a packet store timestamp instead of position
after the frame length.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
Fix NASM include directive
dsputil_mmx: Honor HAVE_AMD3DNOW
lavf,lavd: remove all usage of AVFormatParameters from demuxers.
jack: add 'channels' private option.
VC-1: fix reading of custom PAR.
Remove redundant and dubious video codec detection by its extradata
mpeg12: remove repeat-field code disabled since May 2002
patch checklist: suggest fate instead of regression tests
Turn on resampling on sudden size change instead of bailing out during recode.
avtools: reinitialise filter chain when input video stream changes dimensions
Conflicts:
Makefile
avconv.c
doc/developer.texi
ffplay.c
libavcodec/x86/dsputil_mmx.c
libavdevice/libdc1394.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
ffmpeg: fix some indentation
ffmpeg: fix operation with --disable-avfilter
simple_idct: remove disabled code
motion_est: remove disabled code
vc1: remove disabled code
fate: separate lavf-mxf_d10 test from lavf-mxf
cabac: Move code only used in the cabac test program to cabac.c.
ffplay: warn that -pix_fmt is no longer working, suggest alternative
ffplay: warn that -s is no longer working, suggest alternative
lavf: rename enc variable in utils.c:has_codec_parameters()
lavf: use designated initialisers for all (de)muxers.
wav: remove a use of deprecated AV_METADATA_ macro
rmdec: remove useless ap parameter from rm_read_header_old()
dct-test: remove write-only variable
des: fix #if conditional around P_shuffle
Use LOCAL_ALIGNED in ff_check_alignment()
Conflicts:
ffmpeg.c
libavformat/avidec.c
libavformat/matroskaenc.c
libavformat/mp3enc.c
libavformat/oggenc.c
libavformat/utils.c
tests/ref/lavf/mxf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
* 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>
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)
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>
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)
in its place.
av_metadata_set() is going to be dropped at the next major bump.
Originally committed as revision 22961 to svn://svn.ffmpeg.org/ffmpeg/trunk
AVERROR(ENOMEM).
AVERROR_NOMEM is deprecated and will be dropped at the next libavutil
major bump.
Originally committed as revision 22791 to svn://svn.ffmpeg.org/ffmpeg/trunk
word, so treat it this way instead of extracting different parts from 32-bit
value.
Originally committed as revision 20820 to svn://svn.ffmpeg.org/ffmpeg/trunk
instead of many strcmp() on always four-byte strings.
Idea borrowed from RM demuxer in FFmbc by Baptiste Coudurier.
Originally committed as revision 20819 to svn://svn.ffmpeg.org/ffmpeg/trunk
This avoids a crash when the next slice is not a start slice and thus
pkt->data is still NULL.
This probably only happens with broken or unsupported files like
http://samples.mplayerhq.hu/real/multirate/JustaSpa1937_64kb.rm
that need further fixes, but keeping vst state consistent is still a good idea.
Originally committed as revision 19830 to svn://svn.ffmpeg.org/ffmpeg/trunk
used to return packet data, which might update the flags/timestamp to be
used for the next packet data returned by the demuxer. However, that was
separated out into a new function, and the flags/timestamp are thus never
updated within ff_rm_parse_packet() anymore, and thus do not need to be
a pointer.
Originally committed as revision 19539 to svn://svn.ffmpeg.org/ffmpeg/trunk
Has been tested against streamed / non-seekable input and passes make
seektest. See "[PATCH] rmdec.c: parse INDX chunk" thread on mailinglist.
Originally committed as revision 18013 to svn://svn.ffmpeg.org/ffmpeg/trunk
as requested by Kostya). See "[PATCH] rmdec.c: remove cache access
duplication".
Originally committed as revision 18010 to svn://svn.ffmpeg.org/ffmpeg/trunk
cache, since this can already be accessed through ff_rm_retrieve_cache().
See "[PATCH] rmdec.c: remove cache access duplication" thread.
Originally committed as revision 18009 to svn://svn.ffmpeg.org/ffmpeg/trunk
the newer (.rm, audio/video) files. See "[PATCH] rmdec.c: merge old/new
packet reading code" thread on mailinglist.
Originally committed as revision 18005 to svn://svn.ffmpeg.org/ffmpeg/trunk
discussion in the ML thread "[PATCH] rmdec.c: merge old/new packet reading
code".
Over time, this code broke somewhat, e.g. seq was never actually written
into (and was thus always 1, therefore the seq condition was always true),
whereas it was supposed to be set to the sequence number of the video slice
in case the video frame is divided over multiple RM packets (slices). The
problem of this is that packets other than those containing the beginning
of a video frame would be indexed as well.
Secondly, flags&2 is supposed to be true for video keyframes and for these
audio packets containing the start of a block. For some codecs (e.g. AAC),
that is every single packet, whereas for others (e.g. cook), that is the
packet containing the first of a series of scrambled packets that are to be
descrambled together. Indexing any of the following would lead to incomplete
and thus useless frames. Problem here is that flags would be reset to 2 to
indicate that the first packet is ready to be returned, and in addition if
no data was left to be returned (which is always true for the first packet),
then we wouldn't actually write the index entry anyway.
All in all, the idea was good and it probably worked at some point, but that
is long ago. This patch should at the very least make it likely for this code
to be executed again at the right times, i.e. the way it was originally
intended to be used.
Originally committed as revision 17993 to svn://svn.ffmpeg.org/ffmpeg/trunk
and if the size is broken (20 bytes, header-only), calculate the expected
size and skip the index entries anyway. See "[PATCH] rmdec.c: correctly
skip indexes" thread.
Originally committed as revision 17924 to svn://svn.ffmpeg.org/ffmpeg/trunk
has two possible outcomes: either len and rm->remaining_len are the same, in
which case we care about the outcome and it is zero, or rm->remaining_len is
currently not in use and we don't care about the outcome. In that case, len
is positive and rm->remaining_len is zero, which leads to a negative result.
This is confusing and could eventually lead to a sign-flip if we skip a lot
of packets (unlikely, but still). Therefore, just always set it to zero.
Originally committed as revision 17919 to svn://svn.ffmpeg.org/ffmpeg/trunk
has two possible outcomes: either len and rm->remaining_len are the same, in
which case we care about the outcome and it is zero, or rm->remaining_len is
currently not in use and we don't care about the outcome. In that case, len
is positive and rm->remaining_len is zero, which leads to a negative result.
This is confusing and could eventually lead to a sign-flip if we skip a lot
of packets (unlikely, but still). Therefore, just always set it to zero.
Originally committed as revision 17910 to svn://svn.ffmpeg.org/ffmpeg/trunk
ff_rm_parse_packet(). See "[PATCH] Make RM demuxer behave better with -an
option" thread, which sort-of turned into an aggregate of unrelated rmdec.c
cleanups.
Originally committed as revision 17909 to svn://svn.ffmpeg.org/ffmpeg/trunk
rm->audio_pkt_cnt in case multiple packets should be read before the next
syncpoint in the file, so that ffplay -an on a file containing AAC audio
works. See "[PATCH] Make RM demuxer behave better with -an option" thread
on mailinglist.
Originally committed as revision 17908 to svn://svn.ffmpeg.org/ffmpeg/trunk
libavformat/rmdec.c:550: warning: assignment makes pointer from integer
Patch by Dominique Leuenberger (dominique-ffmpeg-devel A leuenberger D net)
Originally committed as revision 16489 to svn://svn.ffmpeg.org/ffmpeg/trunk
the (larger) allocated size. (prevents segfaults due to later failures
from 900MB-sized packets, yes fuzzed file not a valid one)
Originally committed as revision 16404 to svn://svn.ffmpeg.org/ffmpeg/trunk
of allocated slices matches the actual.
Audio still does a copy (marked with FIXME in the code so this is not missed).
Originally committed as revision 16402 to svn://svn.ffmpeg.org/ffmpeg/trunk
"rmdec.c: double free" discussion on mailinglist, patch with suggestions
from Reimar Doffinger.
Originally committed as revision 16378 to svn://svn.ffmpeg.org/ffmpeg/trunk
AVStreams can be used to call these public rmdec.c functions as well, as is
the case for RDT/RTSP streams. See mailinglist discussion in "[PATCH] rdt.c:
don't reuse the same AVStream in both RTSP and RM demuxer" thread.
Originally committed as revision 16366 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allows multiple video or audio streams per .rm file. See mailinglist
thread "[PATCH] rmdec.c: implement RMVideo/AudioStream".
Originally committed as revision 16365 to svn://svn.ffmpeg.org/ffmpeg/trunk
"[PATCH] rmdec.c: use get_buffer and skip_bytes instead of loops of get_byte".
Originally committed as revision 16139 to svn://svn.ffmpeg.org/ffmpeg/trunk
Reason for this is that there are no shared entries in the demuxer/muxer
context, making it a mystery as to why it was shared between the two. See
"[PATCH] clean rmdemux/muxcontext" patch on mailinglist.
Originally committed as revision 16111 to svn://svn.ffmpeg.org/ffmpeg/trunk
ff_rm_parse_packet() to indicate whether more audio packets are available
in the demuxer from the last RM frame, and save that in the RDT parsing
context. See patch/discussion in "[PATCH] rdt.c: don't access RMContext"
on ML.
Originally committed as revision 16110 to svn://svn.ffmpeg.org/ffmpeg/trunk
specify the data source as function argument instead of in s->pb before
calling the function. Discussed in ML thread "[PATCH] fix small memleak
in rdt.c".
Originally committed as revision 15849 to svn://svn.ffmpeg.org/ffmpeg/trunk
in rtpdec.c, so that they can be shared and used in the same way in rtsp.c.
The handlers, since they are specific for RDT, are registered in rdt.c and
a new registration function is thus called from allformats.c.
The dynamic payload handler also implements RDT-specific SDP-line parsing for
OpaqueData and StartTime, which are specific for RDT and needed for proper
playback. OpaqueData contains one or a list ("MLTI") of "MDPR" chunks that
can be parsed by the rmdec.c function ff_rm_read_mdpr_codecdata(). To use
this function, we create a new rdt_demuxer, which has the same private data
as the rm_demuxer. The resulting AVFormatContext created with _open_stream()
can thus be used to call functions in the RM demuxer.
See discussion in "Realmedia patch" thread on ML.
Originally committed as revision 15234 to svn://svn.ffmpeg.org/ffmpeg/trunk
shared between the RM demuxer and the RTSP/RDT parser; both use the same
timebase. See discussion in "[PATCH] rmdec.c: move av_set_pts_info()" on ML.
Originally committed as revision 15164 to svn://svn.ffmpeg.org/ffmpeg/trunk
