* qatar/master:
mov: Use defines for sample flags in fragments
mov: Use defines for trun flags
mov: Use defines for tfhd flags
proresenc: force bitrate not to exceed given limit
vc1parse: call vc1_init_common().
wma: don't return 0 on invalid packets.
asf: prevent packet_size_left from going negative if hdrlen > pktlen.
mjpegb: don't return 0 at the end of frame decoding.
rtpdec: Identify incorrectly signalled H263
vp8dsp: split long line.
aiff: don't skip block_align==0 check on COMM-after-SSND files.
dpcm: ignore extra unpaired bytes in stereo streams.
mp3on4: require a minimum framesize.
mpc7: assign an error level + context to av_log() msg.
huffyuv: error out on bit overrun.
dct-test: Add the missing ff_ prefix to the altivec functions
dct-test: Remove a stray declaration of a nonexistent function
movenc: Write the unknown duration as 64 bit fields in ismv
movenc: Write track durations with all bits set if duration is unknown
Conflicts:
libavcodec/dct-test.c
libavcodec/wmadec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is so that TS fragments produced by
http://code.google.com/p/httpsegmenter/
would be compatible with JW Player.
A new member variable prev_payload_key was added to MpegTSWriteStream
to help detect transition from non-key to key frame, so that
PAT/PMT would not be produced for every keyframe in intra-only videos.
Signed-off-by: Pavel Koshevoy <pkoshevoy@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This changes a number of FATE results, since before this commit, the
timestamps in all tests using rawenc were made up by lavf.
In most cases, the previous timestamps were completely bogus.
In some other cases -- raw formats, mostly h264 -- the new timestamps
are bogus as well. The only difference is that timestamps invented by
the muxer are replaced by timestamps invented by the demuxer.
cscd -- avconv sets output codec timebase from r_frame_rate
and r_frame_rate is in this case some guessed number 31.42 (377/12),
which is not accurate enough to represent all timestamps. This results
in some frames having duplicate pts. Therefore, vsync 0 needs to be
changed to vsync 2 and avconv drops two frames. A proper fix in the
future would be to set output timebase to something saner in avconv.
nuv -- previous timestamps for video were wrong AND the cscd
comment applies, one frame is dropped.
vp8-signbias -- the file contains two frames with identical timestamps,
so -vsync 0 needs to be removed/changed to -vsync 2 and avconv drops one
frame.
vc1-ism -- apparrently either the demuxer lies about timestamps or the
file is broken, since dts == pts on all packets, but reordering clearly
takes place.
Current code compares the desired recording time with InputStream.pts,
which has a very unclear meaning. Change the code to use actual
timestamps of the frames passed to the encoder.
In several tests, one less frame is encoded, which is more correct.
In the idroq test one more frame is encoded, which is again more
correct.
Behavior with stream copy should be unchanged.
* qatar/master:
smacker: Sanity check huffman tables found in the headers.
smacker: remove dead store
qdm2: Check data block size for bytes to bits overflow.
mxfdec: Fix files with essence containers larger than 2 GiB.
mxfdec: Employ correct printf conversion specifiers for POSIX int types.
vc1: always read the bfraction element for interlaced fields
fate: add XWD image regression test
lavf: prevent infinite loops while flushing in avformat_find_stream_info
matroskadec: Pad AAC extradata.
ismindex: Fix build on mingw
Conflicts:
libavformat/mxfdec.c
libavformat/utils.c
tests/lavf-regression.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (22 commits)
rv34: frame-level multi-threading
mpegvideo: claim ownership of referenced pictures
aacsbr: prevent out of bounds memcpy().
ipmovie: fix pts for CODEC_ID_INTERPLAY_DPCM
sierravmd: fix audio pts
bethsoftvideo: Use bytestream2 functions to prevent buffer overreads.
bmpenc: support for PIX_FMT_RGB444
swscale: fix crash in fast_bilinear code when compiled with -mred-zone.
swscale: specify register type.
rv34: use get_bits_left()
avconv: reinitialize the filtergraph on resolution change.
vsrc_buffer: error on changing frame parameters.
avconv: fix -copyinkf.
fate: Update file checksums after the mov muxer change in a78dbada55
movenc: Don't store a nonzero creation time if nothing was set by the caller
bmpdec: support for rgb444 with bitfields compression
rgb2rgb: allow conversion for <15 bpp
doc: fix stray reference to FFmpeg
v4l2: use C99 struct initializer
v4l2: poll the file descriptor
...
Conflicts:
avconv.c
libavcodec/aacsbr.c
libavcodec/bethsoftvideo.c
libavcodec/kmvc.c
libavdevice/v4l2.c
libavfilter/vsrc_buffer.c
libswscale/swscale_unscaled.c
libswscale/x86/input.asm
tests/ref/acodec/alac
tests/ref/acodec/pcm_s16be
tests/ref/acodec/pcm_s24be
tests/ref/acodec/pcm_s32be
tests/ref/acodec/pcm_s8
tests/ref/lavf/mov
tests/ref/vsynth1/dnxhd_1080i
tests/ref/vsynth1/mpeg4
tests/ref/vsynth1/qtrle
tests/ref/vsynth1/svq1
tests/ref/vsynth2/dnxhd_1080i
tests/ref/vsynth2/mpeg4
tests/ref/vsynth2/qtrle
tests/ref/vsynth2/svq1
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
isom: sort and pretty-print codec_movaudio_tags[]
isom: remove pointless comments in codec_movaudio_tags[]
isom: remove commented-out tag for vorbis
movenc: write 'chan' tag for AC-3 in MOV
mov: add support for reading and writing the 'chan' tag
audioconvert: add some additional channel and channel layout macros
audioconvert: change 7.1 "wide" layout to use side surround channels
movenc: simplify handling of pcm vs. adpcm vs. other compressed codecs
doc: update documentation to use avconv
doc: update demuxers section
doc: extend external library coverage
doc: split platform specific information
doc: port the git-howto to texinfo
doc: provide fallback css and customize @float
doc: document fate in a texinfo
doxy: change hue value to match our green
Conflicts:
doc/fate.txt
doc/ffserver.texi
doc/general.texi
doc/muxers.texi
doc/protocols.texi
doc/t2h.init
libavformat/isom.c
libavformat/mov.c
libavutil/avutil.h
tests/ref/acodec/pcm_s16be
tests/ref/acodec/pcm_s24be
tests/ref/acodec/pcm_s32be
tests/ref/acodec/pcm_s8
tests/ref/lavf/mov
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Use Sound Sample Description Version 2 for all MOV files.
Updated FATE references accordingly.
Note that ADPCM is treated as compressed audio in version 2.
The qatar implementation makes no sense.
a muxer without timestamps is constant fps thus needs vsync.
the crc/mp5 are special cases that have timestamps yet allow any
nonsensical timestamps.
raw (yuv/rgb) video is constant fps thus needs vsync too.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
libavutil: add utility functions to simplify allocation of audio buffers.
libavutil: add planar sample formats and av_sample_fmt_is_planar()
avconv: fix segfault at EOF with delayed pictures
pcmdec: remove unneeded resetting of samples pointer
avconv: remove a now unused parameter from output_packet().
avconv: formatting fixes in output_packet()
avconv: declare some variables in blocks where they are used
avconv: use the same behavior when decoding audio/video/subs
bethsoftvideo: return proper consumed size for palette packets.
cdg: skip packets that don't contain a cdg command.
crcenc: add flags
avconv: use vsync 0 for AVFMT_NOTIMESTAMPS formats.
tiffenc: add a private option for selecting compression algorithm
md5enc: add flags
ARM: remove needless .text/.align directives
Conflicts:
doc/APIchanges
libavcodec/tiffenc.c
libavutil/avutil.h
libavutil/samplefmt.c
libavutil/samplefmt.h
tests/ref/fate/bethsoft-vid
tests/ref/fate/cdgraphics
tests/ref/fate/film-cvid-pcm-stereo-8bit
tests/ref/fate/mpeg2-field-enc
tests/ref/fate/nuv
tests/ref/fate/tiertex-seq
tests/ref/fate/tscc-32bit
tests/ref/fate/vmnc-32bit
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (22 commits)
configure: add check for w32threads to enable it automatically
rtmp: do not hardcode invoke numbers
cinepack: return non-generic errors
fate-lavf-ts: use -mpegts_transport_stream_id option.
Add an APIchanges entry and a minor bump for avio changes.
avio: Mark the old interrupt callback mechanism as deprecated
avplay: Set the new interrupt callback
avconv: Set new interrupt callbacks for all AVFormatContexts, use avio_open2() everywhere
cinepak: remove redundant coordinate checks
cinepak: check strip_size
cinepak, simplify, use AV_RB24()
cinepak: simplify, use FFMIN()
cinepak: Fix division by zero, ask for sample if encoded_buf_size is 0
applehttp: Fix seeking in streams not starting at DTS=0
http: Don't use the normal http proxy mechanism for https
tls: Handle connection via a http proxy
http: Reorder two code blocks
http: Add a new protocol for opening connections via http proxies
http: Split out the non-chunked buffer reading part from http_read
segafilm: add support for raw videos
...
Conflicts:
avconv.c
configure
doc/APIchanges
libavcodec/cinepak.c
libavformat/applehttp.c
libavformat/version.h
tests/lavf-regression.sh
tests/ref/lavf/ts
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
proresdsp: fix function prototypes.
prores-idct: fix overflow in c code.
fate: update prores-alpha ref after changing pix_fmt to yuv444p10le
prores: add missing feature warning for alpha
mov: 10l: Terminate string with 0 not '0'
mov: Prevent illegal writes when chapter titles are very short.
prores: add appropriate -fix_fmt parameter to FATE command
riff: always generate a proper WAVEFORMATEX structure in ff_put_wav_header
lavc: add a flag-based error_recognition field to AVCodecContext and deprecate non-flag-based ER field
lavc: rename deprecation symbol FF_API_VERY_AGGRESSIVE to FF_API_ER
Conflicts:
libavcodec/avcodec.h
libavformat/mov.c
tests/fate/prores.mak
tests/ref/acodec/g726
tests/ref/fate/prores-alpha
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The cbSize field should be included in all cases, even with PCM where
its value is ignored.
Fixes encoding PCM audio in Matroska for some players which insist on
a full WAVEFORMATEX structure for A_MS/ACM audio.
Since fate uses wav files for the audio test a larger number of tests
has changed checksums or shifted positions due to the 2 byte longer
wave header.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
Similar to libswscale this does resampling and format convertion, just for audio
instead of video.
changing sampling rate, sample formats, channel layouts and sample packing all
in one with a very simple public interface.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavfi: add select filter
oggdec: fix out of bound write in the ogg demuxer
movenc: create an alternate group for each media type
lavd: add libcdio-paranoia input device for audio CD grabbing
rawdec: refactor private option for raw video demuxers
pcmdec: use unique classes for all pcm demuxers.
rawdec: g722 is always 1 channel/16kHz
Conflicts:
Changelog
configure
doc/filters.texi
libavdevice/avdevice.h
libavfilter/avfilter.h
libavfilter/vf_select.c
tests/ref/lavf/mov
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>
* commit 'b5849f77095439e994b11c25e6063d443b36c228': (21 commits)
ac3enc: merge AC3MDCTContext with AC3EncodeContext.
ac3enc: prefer passing AC3EncodeContext rather than AVCodecContext
ac3enc: fix memleak
mpeg1video: add CODEC_CAP_SLICE_THREADS.
lavf: fix segfault in av_open_input_stream()
mpegtsenc: set Random Access indicator on keyframe start packets
lavf: Cleanup try_decode_frame() logic.
Replace some gotos that lead to single return statements by direct return.
build: move tests/seek_test.c to libavformat and reuse generic build rules
mxfenc: include needed header for ff_iso8601_to_unix_time() prototype
Add a check for strptime().
lavf: factor out conversion of ISO8601 string to unix time
wav: parse 'bext' metadata
wav: keep parsing until EOF if the input is seekable and we know the size of the data tag
wav: Refactor the tag checking into a switch statement
wav: make sure neither data_size nor sample_count is negative.
wav: refactor the 'fmt ' tag search and parsing.
wav: add an option for writing BEXT chunk
ffmpeg: get rid of a pointless limit on number of streams.
ffmpeg: remove an unused define.
...
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* sws_32bit_integration:
regtests/sws: update checksums for recent changes
sws: dont mess with XInc when the code needing it isnt used
sws: Fix chroma init for 32bit buffers.
swscale: error dithering for 16/9/10-bit to 8-bit.
swscale: fix overflow in 16-bit vertical scaling.
swscale: fix crash in 8-bpc bilinear output without alpha.
swscale: fix 16-bit scaling when output is 8-bits.
sws: fix non native endian 9-15 bit input with 16bit out
sws: disable scale16 when int32 is used
sws: fix rgb -> 16bit
sws: fix uv overwrite in 32bt
sws: fix gray16_1
sws:ix yuv2rgb48_1_c_template()
sws: fix 16/32 bug from merge
swscale: for >8bit scaling, read in native bit-depth.
swscale: fix another yuv range conversion overflow in 16bit scaling. (cherry picked from commit 81cc7d0bd1)
swscale: fix yuv range correction when using 16-bit scaling. (cherry picked from commit e0b8fff6c7)
swscale: implement >8bit scaling support.
Merged-by: Michael Niedermayer <michaelni@gmx.at>