Add checksums specific to BE machines after last swscale update.
Signed-off-by: Michael Niedermayer <firstname.lastname@example.org>
Merge remote branch 'qatar/master'
movenc: fix yuv range in avid atoms used by dnxhd.
yuv range: full 1 / normal 2
Signed-off-by: Anton Khirnov <email@example.com>
swscale: fix YUV420P 9/10bit support.
Fix handling of input if not in native endianness, and add support for9/10-bit output. This allows us to force endianness of YUV420P 9/10bitin the H264/10bit fate tests, which should fix them on big-endiansystems.
regtests: add grayscale qtrle
h264-fate: Fix 10bit H264 tests on big endian.
framecrc returns different values when one swiches endianness,this apparently has been missed by "the fork" who added the 10bit fatetests. Sorry for missing this during the merge.Signed-off-by: Michael Niedermayer <firstname.lastname@example.org>
error_concealment: Use previous pictures motion vectors when the current ones have been lost.
Looks better for some cases, worse for others, overall not much difference.Its more correct though.Signed-off-by: Michael Niedermayer <email@example.com>
lavf/utils: fix ff_interleave_compare_dts corner case.
This should fix behavior introduced by commit96573c0d7605672d69b42ae1dcf18764ce47c71a. Av_rescale_rnd() is notlossless so if two timestamps are equal after being rescaled they arenot always actually identical. This patch use av_compare_ts() to get...
fate: add 10-bit H264 tests.
swscale: implement Nbit->non native endian 16bit. Fixes v210.Signed-off-by: Michael Niedermayer <firstname.lastname@example.org>
fate: add 9/10 BE pixdesc checksumsSigned-off-by: Michael Niedermayer <email@example.com>
v210dec: switch to PIX_FMT_422P10Signed-off-by: Michael Niedermayer <firstname.lastname@example.org>
swscale: 9,10 bits pixel format output supportSigned-off-by: Michael Niedermayer <email@example.com>
adpcmenc: fix QT IMA ADPCM encoderSigned-off-by: Michael Niedermayer <firstname.lastname@example.org>
adpcmdec: Fix QT IMA ADPCM decoderSigned-off-by: Michael Niedermayer <email@example.com>
vc1: make overlap filter for I-frames bit-exact.
msvideo1 regression test
Make DV (sub) demuxer set proper pkt->pos values.
This makes the avi demuxer create packets with proper pos valueswith the file from ticket #140.
lavf: inspect more frames for fps when container time base is coarse
As per issue2629, most 23.976fps matroska H.264 files are incorrectlydetected as 24fps, as the matroska timestamps usually have onlymillisecond precision.
Fix that by doubling the amount of timestamps inspected for frame rate...
flashsv2enc: regression test.Signed-off-by: Michael Niedermayer <firstname.lastname@example.org>
Checksum update due to (should make fate green again) ffmpeg | branch: master | Anton Khirnov <email@example.com> | Tue Apr 26 09:59:07 2011 +0000| [f8fec0505294a4c05e5cfd9323e04258db465314] | committer: Anton Khirnov
mpegtsenc: make PMT PID really start on pmt_start_pid...
Try to fix big endian fateSigned-off-by: Michael Niedermayer <firstname.lastname@example.org>
mpegtsenc: make PMT PID really start on pmt_start_pid
hflip: make the filter accept PIX_FMT_BGR48LE and PIX_FMT_BGR48BE pixel formats
crop: make the filter accept PIX_FMT_BGR48LE and PIX_FMT_BGR48BE pixel formats
libswcale: PIX_FMT_BGR48LE and PIX_FMT_BGR48BE scaler implementation
ac3enc: correct the flipped sign in the ac3_fixed encoder
avi: try to synchronize the points in time of the starts of streams after seeking.Signed-off-by: Michael Niedermayer <email@example.com>
10l, commit that should have been stashed into the merge.Signed-off-by: Michael Niedermayer <firstname.lastname@example.org>
Update regtest checksums after revision 6001dad.
The string "FFmpeg" was replaced by "Libav" in metadata thatgot encoded in file headers.
In mov muxer, compute avg bitrate in esds
lavf/utils.c: Order packets with identical PTS by stream index.
This allows for more reproducible results when using multi-threading.
Signed-off-by: Ronald S. Bultje <email@example.com>
matroskaenc: don't write an empty Cues element.
ac3enc: select bandwidth based on bit rate, sample rate, and number offull-bandwidth channels.
This reduces high-frequency artifacts and improves the quality of the lowerfrequency audio at low bit rates.
ac3enc: use generic fixed-point mdct
This makes the AC3 encoder use the shared fixed-point MDCT ratherthan its own implementation. The checksum changes are due todifferent rounding in the MDCT.
Signed-off-by: Mans Rullgard <firstname.lastname@example.org>
Fix yuvj420p scaling artefact, issue1108.
In ipod/mov/mp4 muxer, always write esds descriptor length using 4 bytes,ipod shuffle doesn't support anything else.
In mov muxer, fix yuv range in avid atoms used by dnxhd.
Merge remote-tracking branch 'newdev/master'
Conflicts: libavcodec/dsputil.c libavcodec/mpegvideo.c libavcodec/snow.c...
Split fate-psx-str-v3 into a video-only and audio-only test.
Make the hflip filter accept PIX_FMT_BGR48LE and PIX_FMT_BGR48BE pixel formats
Make the crop filter accept PIX_FMT_BGR48LE and PIX_FMT_BGR48BE pixel formats
Add apply_window_int16() to DSPContext with x86-optimized versions and use itin the ac3_fixed encoder.
fate: update wmv8-drm reference
This updates the wmv8-drm reference after c47d383.
vc1: make P-frame deblock filter bit-exact.
asf: update seek test reference
This updates the seek test reference to match de11ee9. Before thischange, most of the seeks requested positions before the supposedstart of the file (the preroll time), resulting in the first packetbeing returned. With the preroll subtracted, some of these seeks...
ac3enc: do not right-shift fixed-point coefficients in the final MDCT stage.
This increases the accuracy of coefficients, leading to improved quality.Rescaling of the coefficients to full 25-bit accuracy is done rather thanoffsetting the exponent values. This requires coefficient scaling to be done...
bink: prevent overflows within binkidct by using int-sized intermediate array
vmdaudio: output 8-bit audio as AV_SAMPLE_FMT_U8.
There is no need to expand to 16-bits. Just use memcpy() to copy the raw data.
vmdaudio: output audio samples for standalone silent blocks.
ac3enc: fix bug in stereo rematrixing decision.
The rematrixing strategy reuse flags are not reset between frames, so theyneed to be initialized for all blocks, not just block 0.
ac3enc: change default floor code to 7.
This is to match the value in every (E-)AC-3 file from commercial sources.It has a negligible effect on audio quality.
Fix qtrle regression test, actually test qtrle.
ac3enc: Change EXP_DIFF_THRESHOLD to 500.
This patch changes the exponent difference threshold in the exponentstrategy decision function of the AC-3 encoder. I tested lowering inincrements of 100. From 1000 down to 500 generally increased in qualitywith each step, but 400 was generally much worse....
Update mpegts test reference
The output was changed by a7827a17c6b3388322350456d445c94b3a82cd25.
mpegtsenc: set reserved bits to 1 in PCR field
The reserved bits between PCR base and extension fields must beset to 1.
fate: add h264 test for extreme cases in planar prediction
fate: add lossless h264 test
fate: make lavfi tests output only md5
Instead of saving huge raw files, use the md5: output pseudo-protocolto calculate the checksum of the file directly. This is especiallyuseful when testing on remote targets as it avoids transferring 3.6GBover the network.
Add regression test for stereo s16le in voc.
Update smc fate ref due to r26310
Originally committed as revision 26342 to svn://svn.ffmpeg.org/ffmpeg/trunk
Add stereo rematrixing support to the AC-3 encoders.This improves the audio quality significantly for stereo source with both thefixed-point and floating-point AC-3 encoders.Update acodec-ac3_fixed and seek-ac3_rm test references.
Originally committed as revision 26271 to svn://svn.ffmpeg.org/ffmpeg/trunk
Add a FATE test for Playstation STR version 3
Originally committed as revision 26231 to svn://svn.ffmpeg.org/ffmpeg/trunk
Change the AC-3 encoder to use floating-point.Fixed-point AC-3 encoder renamed to ac3_fixed.Regression test acodec-ac3 renamed to acodec-ac3_fixed.Regression test lavf-rm changed to use ac3_fixed encoder.
Originally committed as revision 26209 to svn://svn.ffmpeg.org/ffmpeg/trunk
Change the default dB-per-bit code from 2 to 3.This gives slightly better quality in PEAQ tests.Code 3 gives a dBpb value of 2816 = -132dB (128 psd units = -6dB), whichcorresponds to 22 bits. Since the exponents have an offset applied, the16-bit source looks like 24-bit source to the bit allocation routine....
add SubRip decoder
Originally committed as revision 26119 to svn://svn.ffmpeg.org/ffmpeg/trunk
Add copy filter, useful for testing the avfilter_draw_slice() copycode.
Originally committed as revision 26112 to svn://svn.ffmpeg.org/ffmpeg/trunk
Change FIX15 back to clipping to -32767..32767.This avoids a 16-bit overflow in mdct512() due to a -32768 value in costab.References updated for acodec-ac3, lavf-rm, and seek-ac3_rm tests.Thanks to Måns Rullgård for finding the bug.
Originally committed as revision 26071 to svn://svn.ffmpeg.org/ffmpeg/trunk
Discard partial packet of last frame for fate-wmv8-drm to avoid test failsdue to VC-1 decoder overreads resulting in different output.
Originally committed as revision 26055 to svn://svn.ffmpeg.org/ffmpeg/trunk
Add test for ASF -cryptokey that tests only demuxing, but both audio and videoto complement the existing video-only decode test.
Originally committed as revision 26054 to svn://svn.ffmpeg.org/ffmpeg/trunk
Change ASF demuxer to return incomplete last packets.Whether the behaviour for streams using scrambling makes senseis unclear.
Originally committed as revision 26053 to svn://svn.ffmpeg.org/ffmpeg/trunk
Update the test references for lavf-rm and seek-ac3_rm.The references changed due to r25956.
Originally committed as revision 26004 to svn://svn.ffmpeg.org/ffmpeg/trunk
Simplify fix15().Turn it into 2 macros, and use av_clip_int16() and lrintf().This matches the int16 to float sample conversion in audioconvert.c.The regression test output is different due to lrintf() rounding.
Originally committed as revision 25956 to svn://svn.ffmpeg.org/ffmpeg/trunk
Add a FLAC parser.Seek test reference updated because FLAC seeking now works properly.Fixes roundup issue 1150.
Patch by Michael Chinen [mchinen at gmail]
Originally committed as revision 25914 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fix h264-conformance-frext-frext_mmco4_sony_b conformance test.
This includes a revert of r25840
Originally committed as revision 25842 to svn://svn.ffmpeg.org/ffmpeg/trunk
Update fate h264 test due to r25824, this file has 2 frames delay
Originally committed as revision 25840 to svn://svn.ffmpeg.org/ffmpeg/trunk
Make DNxHD encoder produce files that are strictly VC-3 compatible
Originally committed as revision 25756 to svn://svn.ffmpeg.org/ffmpeg/trunk
Remove now unused file (should have been part of commit r25735)
Originally committed as revision 25736 to svn://svn.ffmpeg.org/ffmpeg/trunk
Test 4XM decoding (and not only demuxing) in FATE tests
Originally committed as revision 25735 to svn://svn.ffmpeg.org/ffmpeg/trunk
Add test for cropping of interlaced H.264.
Originally committed as revision 25677 to svn://svn.ffmpeg.org/ffmpeg/trunk
Update 24 bpp TM1 reference for decoder fixes.
Originally committed as revision 25664 to svn://svn.ffmpeg.org/ffmpeg/trunk
Avoid negative SCR in mpeg ps muxer.Fixes a scr issue reported with dvdauthor ([FFmpeg-user] FFMPEG encoded MPEG-2 video causes error in DVDAuthor)
Originally committed as revision 25512 to svn://svn.ffmpeg.org/ffmpeg/trunk
Update rv30 FATE reference after last commit
The rm demuxer has timestamp bugs, so this test is sensitive to changes intimestamp correction. The previous commit did not make output any better or worseon this test, just different.
See https://roundup.ffmpeg.org/issue2288 for details....
Update gxf regression tests because of r25399
Originally committed as revision 25400 to svn://svn.ffmpeg.org/ffmpeg/trunk
In gxf muxer, fix flt entry offset, patch by Reuben Martin, reuben dot m at gmail dot com
Originally committed as revision 25395 to svn://svn.ffmpeg.org/ffmpeg/trunk
Init SCR in mpeg muxer based on first DTS.This fixes issues if the first DTS is far away from 0.
Originally committed as revision 25383 to svn://svn.ffmpeg.org/ffmpeg/trunk
Rename fate-gsm test to the more accurate fate-gsm-msand add a test for regular GSM as fate-gsm.
Fixes a 8kHz sample from issue 113.
Originally committed as revision 25313 to svn://svn.ffmpeg.org/ffmpeg/trunk
Update rv20 seek test reference
Originally committed as revision 25204 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fix rv20 encoding so the binary decoder can decode it.
Originally committed as revision 25203 to svn://svn.ffmpeg.org/ffmpeg/trunk
Set a constant frame size for encoding G.726 audio.
Originally committed as revision 25107 to svn://svn.ffmpeg.org/ffmpeg/trunk