A negative codec_id cannot be handled by the found_decoder API of
AVStream->info: if the codec_id is not recognized, found_decoder is set
to -codec_id, which has to be '<0' according to the API documentation.
This can cause NULL pointer dereferencing in try_decode_frame.
Also make sure the codec_type matches the expected one for codec_id.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
If size is zero, avio_get_str fails, leaving the buffer uninitialized.
This causes invalid reads in av_set_options_string.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Otherwise the ffm demuxer can return the same packets endlessly, if
there is no server attached.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
This should fix a infinite loop
Reviewed-by: Andreas Cadhalpun <andreas.cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If resyncing leads to the same position as previously, it will again
lead to a resync attempt, resulting in an infinite loop.
Thus don't seek back beyond the last syncpoint.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
A negative time base can trigger assertions.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If EOF is reached, while skipping bytes, avio_tell(pb) won't change
anymore, resulting in an infinite loop.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
They are used in a switch statement, but it is not guaranteed that the
COMM case (where they are set to 0) is reached before the other cases.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is a generic solution that will not reqiore modifications when new options are added.
This also fixes problem with current implementation when qmin or qmax=-1.
Only 8 bits was sent and read back as 255.
Fixes#1275Fixes#1461
Signed-off-by: Lukasz Marek <lukasz.m.luki2@gmail.com>
It has not been properly maintained for years and there is little hope
of that changing in the future.
It appears simpler to write a new replacement from scratch than
unbreaking it.
Currently FFM files generated with one versions of ffmpeg generally
cannot be read by another.
By spliting data into chunks, more fields can saftely be appended to
chunks as well as new chunks added.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This way avserver only depends on the data structures of the ffm
demuxer, which it already does, and not also on private functions
being exported by the library.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
wmaenc: use float planar sample format
(e)ac3enc: use planar sample format
aacenc: use planar sample format
adpcmenc: use planar sample format for adpcm_ima_wav and adpcm_ima_qt
adpcmenc: move 'ch' variable to higher scope
adpcmenc: fix 3 instances of variable shadowing
adpcm_ima_wav: simplify encoding
libvorbis: use planar sample format
libmp3lame: use planar sample formats
vorbisenc: use float planar sample format
ffm: do not write or read the audio sample format
parseutils: fix parsing of invalid alpha values
doc/RELEASE_NOTES: update for the 9 release.
smoothstreamingenc: Add a more verbose error message
smoothstreamingenc: Ignore the return value from mkdir
smoothstreamingenc: Try writing a manifest when opening the muxer
smoothstreamingenc: Move the output_chunk_list and write_manifest functions up
smoothstreamingenc: Properly return errors from ism_flush to the caller
smoothstreamingenc: Check the output UrlContext before accessing it
Conflicts:
doc/RELEASE_NOTES
libavcodec/aacenc.c
libavcodec/ac3enc_template.c
libavcodec/wmaenc.c
tests/ref/lavf/ffm
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: (58 commits)
amrnbdec: check frame size before decoding.
cscd: use negative error values to indicate decode_init() failures.
h264: prevent overreads in intra PCM decoding.
FATE: do not decode audio in the nuv test.
dxa: set audio stream time base using the sample rate
psx-str: do not allow seeking by bytes
asfdec: Do not set AVCodecContext.frame_size
vqf: set packet parameters after av_new_packet()
mpegaudiodec: use DSPUtil.butterflies_float().
FATE: add mp3 test for sample that exhibited false overreads
fate: add cdxl test for bit line plane arrangement
vmnc: return error on decode_init() failure.
libvorbis: add/update error messages
libvorbis: use AVFifoBuffer for output packet buffer
libvorbis: remove unneeded e_o_s check
libvorbis: check return values for functions that can return errors
libvorbis: use float input instead of s16
libvorbis: do not flush libvorbis analysis if dsp state was not initialized
libvorbis: use VBR by default, with default quality of 3
libvorbis: fix use of minrate/maxrate AVOptions
...
Conflicts:
Changelog
doc/APIchanges
libavcodec/avcodec.h
libavcodec/dpxenc.c
libavcodec/libvorbis.c
libavcodec/vmnc.c
libavformat/asfdec.c
libavformat/id3v2enc.c
libavformat/internal.h
libavformat/mp3enc.c
libavformat/utils.c
libavformat/version.h
libswscale/utils.c
tests/fate/video.mak
tests/ref/fate/nuv
tests/ref/fate/prores-alpha
tests/ref/lavf/ffm
tests/ref/vsynth1/prores
tests/ref/vsynth2/prores
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
x86: cabac: replace explicit memory references with "m" operands
avplay: don't request a stereo downmix
wmapro: use av_float2int()
lavc: avoid invalid memcpy() in avcodec_default_release_buffer()
lavu: replace int/float punning functions
lavfi: install libavfilter/vsrc_buffer.h
Remove extraneous semicolons
sdp: Restore the original mp4 format h264 extradata if converted
rtpenc: Add support for mp4 format h264
rtpenc: Simplify code by introducing a separate end pointer
movenc: Use the actual converted sample for RTP hinting
Fix a bunch of common typos.
Conflicts:
doc/developer.texi
doc/eval.texi
doc/filters.texi
doc/protocols.texi
ffmpeg.c
ffplay.c
libavcodec/mpegvideo.h
libavcodec/x86/cabac.h
libavfilter/Makefile
libavformat/avformat.h
libavformat/cafdec.c
libavformat/flvdec.c
libavformat/flvenc.c
libavformat/gxfenc.c
libavformat/img2.c
libavformat/movenc.c
libavformat/mpegts.c
libavformat/rtpenc_h264.c
libavformat/utils.c
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The existing functions defined in intfloat_readwrite.[ch] are
both slow and incorrect (infinities are not handled).
This introduces a new header with fast, inline conversion
functions using direct union punning assuming an IEEE-754
system, an assumption already made throughout the code.
The one use of Intel/Motorola extended 80-bit format is
replaced by simpler code sufficient under the present
constraints (positive normal values).
The old functions are marked deprecated and retained for
compatibility.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* 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>
