All tests are run through the fate-run.sh script which already
sets up redirections. Using the outputs set there simplifies
things somewhat.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
ac3enc: use correct alignment and length in channel coupling dsp functions.
ffmpeg: don't abuse a global for passing framerate from input to output
ffmpeg: don't abuse a global for passing channels from input to output
ffmpeg: don't abuse a global for passing samplerate from input to output
ARM: update ff_h264_idct8_add4_neon for 4:4:4 changes
swscale: use SwsContext for av_log when available
swscale: Remove HAVE_MMX from files that are only compiled with MMX enabled.
swscale: Fix compilation with --disable-mmx2.
Conflicts:
ffmpeg.c
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
configure: Add -U__STRICT_ANSI__ to CPPFLAGS on Cygwin and DOS.
aacdec: fix typo in scalefactor clipping check
fate: fix fate-h264-conformance-frext-pph10i4-panasonic-a crcs.
fate: update 9/10bit refs.
h264: Properly set coded_{width, height} when parsing H.264.
x86 asm: Add SECTION_TEXT to dct32_sse.asm.
Fix 9/10 bit in swscale.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
configure: make executable again
LATM/AAC: Free previously initialized context on reinit.
configure: Do not unconditionally add -Wall to host CFLAGS.
configure: Set OS/2 objformat to a.out.
Add support for a.out object format to assembler macros.
fate: disable threading for encoding
fate: add comment field
fate: allow overriding default build and install dirs
mpegtsenc: Add an AVClass pointer to the private data
mpegaudio: clean up #includes
mpegaudio: move all header parsing to mpegaudiodecheader.[ch]
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This explicitly disables threading for encoding as slices are otherwise
automatically activated. This should be dropped once option resetting
between files is fully implemented.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This adds a comment field to the report header, suitable for
extra information not covered by the automatic fields.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
APIchanges: fill in date and commit for request_sample_fmt
Add floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders.
Add support for request_sample_format in ffmpeg and ffplay.
Add APIchanges entry for request_sample_fmt.
Add request_sample_fmt field to AVCodecContext.
Add float_interleave() to FmtConvertContext with x86-optimized versions.
Remove unused make variable SEEK_REFFILE
fate: remove redundant aref and vref references
fate: remove do_ffmpeg_nocheck function
fate: do not collect -benchmark output
mpegaudiodec: remove decode_end() function
fate: run aref and vref as regular tests
mpegaudio: sanitise compute_antialias_* names
mpeg12: add slice-threading checks to slice-threading initializers.
h264: copy pixel_shift between slice threading contexts.
mdec: enable frame-level multithreading.
mdec.c: fix overread.
Conflicts:
libavcodec/aacdec.c
libavcodec/ac3dec.c
libavcodec/avcodec.h
libavcodec/dca.c
libavcodec/h264.c
libavcodec/mdec.c
libavcodec/mpeg12.c
libavcodec/options.c
libavcodec/version.h
libavcodec/vorbisdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This function is essentially an alias for run_ffmpeg and is only
used in one place. This patch removes the function and replaces
the call with the equivalent (simpler) run_ffmpeg call.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The old regtest scripts pass -benchmark and collect the utime values.
As these values are never used, this machinery can be removed.
Signed-off-by: Mans Rullgard <mans@mansr.com>
These tests create reference files used for psnr calculation in
the other codec tests. Treating them as (mostly) regular tests
simplifies the makefile and makes them visible in the fate reports.
The latter makes errors in these runs easier to identify.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Make the iff demuxer send the whole audio chunk to the decoder as a
single packet, move stereo interleaving from the iff demuxer to the
decoder, and introduce an 8svx_raw decoder which performs
stereo interleaving.
This is required for handling stereo data correctly, indeed samples
are stored like:
LLLLLL....RRRRRR
that is all left samples are at the beginning of the chunk, all right
samples at the end, so it is necessary to store and process the whole
buffer in order to decode each frame. Thus the decoder needs all the
audio chunk before it can return interleaved data.
Fix decoding of files 8svx_exp.iff and 8svx_fib.iff, fix trac issue #169.
Also remove code that overwrites the C versions of functions in
sws_init_swScale_altivec(), so that it uses the C functions of files
if no altivec-optimized version exists.
* qatar/master: (33 commits)
rtpdec_qdm2: Don't try to parse data packet if no configuration is received
ac3enc: put the counting of stereo rematrixing bits in the same place to make the code easier to understand.
ac3enc: clean up count_frame_bits() and count_frame_bits_fixed()
mpegvideo: make FF_DEBUG_DCT_COEFF output coeffs via av_log() instead of just via AVFrame.
srtdec: make sure we don't write past the end of buffer
wmaenc: improve channel count and bitrate error handling in encode_init()
matroskaenc: make sure we don't produce invalid file with no codec ID
matroskadec: check that pointers were initialized before accessing them
lavf: fix function name in compute_pkt_fields2 av_dlog message
lavf: fix av_find_best_stream when providing a wanted stream.
lavf: fix av_find_best_stream when decoder_ret is given and using a related stream.
ffmpeg: factorize quality calculation
tiff: add support for SamplesPerPixel tag in tiff_decode_tag()
tiff: Prefer enum TiffCompr over int for TiffContext.compr.
mov: Support edit list atom version 1.
configure: Enable libpostproc automatically if GPL code is enabled.
Cosmetics: fix prototypes in oggdec
oggdec: fix memleak with continuous streams.
matroskaenc: add missing new line in av_log() call
dnxhdenc: add AVClass in private context.
...
swscale changes largely rewritten by me or replaced by baptsites due to lots of bugs in ronalds code.
Above code is also just in case its not obvios to a large extended duplicates that where cherry picked
from ffmpeg.
Conflicts:
configure
ffmpeg.c
libavformat/matroskaenc.c
libavutil/pixfmt.h
libswscale/ppc/swscale_template.c
libswscale/swscale.c
libswscale/swscale_template.c
libswscale/utils.c
libswscale/x86/swscale_template.c
tests/fate/h264.mak
tests/ref/lavfi/pixdesc_le
tests/ref/lavfi/pixfmts_copy_le
tests/ref/lavfi/pixfmts_null_le
tests/ref/lavfi/pixfmts_scale_le
tests/ref/lavfi/pixfmts_vflip_le
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (32 commits)
10-bit H.264 x86 chroma v loopfilter asm
Port SMPTE S302M audio decoder from FFmbc 0.3. [Copyright headers corrected]
Fix crash of interlaced MPEG2 decoding
h264pred: fix one more aliasing violation.
doc/APIchanges: fill in missing hashes and dates.
flacenc: use proper initializers for AVOption default values.
lavc: deprecate named constants for deprecated antialias_algo.
aac: workaround for compilation on cygwin
swscale: extend YUV422p support to 10bits depth
tiff: add support for inverted FillOrder for uncompressed data
Remove unused softfloat implementation.
h264pred: fix aliasing violations.
rotozoom: Eliminate French variable name.
rotozoom: Check return value of fread().
rotozoom: Return an error value instead of calling exit().
rotozoom: Make init_demo() return int and check for errors on invocation.
rotozoom: Drop silly UINT8 typedef.
rotozoom: Drop some unnecessary parentheses.
rotozoom: K&R coding style cosmetics
rtsp: Only do keepalive using GET_PARAMETER if the server supports it
...
Conflicts:
Changelog
cmdutils.c
doc/APIchanges
doc/general.texi
ffmpeg.c
ffplay.c
libavcodec/h264pred_template.c
libavcodec/resample.c
libavutil/pixfmt.h
libavutil/softfloat.c
libavutil/softfloat.h
tests/rotozoom.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fix handling of input if not in native endianness, and add support for
9/10-bit output. This allows us to force endianness of YUV420P 9/10bit
in the H264/10bit fate tests, which should fix them on big-endian
systems.
The file does not decode correctly yet the checksums match this wrongly
decoded file. Thus the checksums must be wrong.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
framecrc returns different values when one swiches endianness,
this apparently has been missed by "the fork" who added the 10bit fate
tests. Sorry for missing this during the merge.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes the warnings:
tests/rotozoom.c:252: warning: ignoring return value of ‘fread’, declared with attribute warn_unused_result
tests/rotozoom.c:254: warning: ignoring return value of ‘fread’, declared with attribute warn_unused_result
* qatar/master: (30 commits)
AVOptions: make default_val a union, as proposed in AVOption2.
arm/h264pred: add missing argument type.
h264dsp_mmx: place bracket outside #if/#endif block.
lavf/utils: fix ff_interleave_compare_dts corner case.
fate: add 10-bit H264 tests.
h264: do not print "too many references" warning for intra-only.
Enable decoding of high bit depth h264.
Adds 8-, 9- and 10-bit versions of some of the functions used by the h264 decoder.
Add support for higher QP values in h264.
Add the notion of pixel size in h264 related functions.
Make the h264 loop filter bit depth aware.
Template dsputil_template.c with respect to pixel size, etc.
Template h264idct_template.c with respect to pixel size, etc.
Preparatory patch for high bit depth h264 decoding support.
Move some functions in dsputil.c into a new file dsputil_template.c.
Move the functions in h264idct into a new file h264idct_template.c.
Move the functions in h264pred.c into a new file h264pred_template.c.
Preparatory patch for high bit depth h264 decoding support.
Add pixel formats for 9- and 10-bit yuv420p.
Choose h264 chroma dc dequant function dynamically.
...
Conflicts:
doc/APIchanges
ffmpeg.c
ffplay.c
libavcodec/alpha/dsputil_alpha.c
libavcodec/arm/dsputil_init_arm.c
libavcodec/arm/dsputil_init_armv6.c
libavcodec/arm/dsputil_init_neon.c
libavcodec/arm/dsputil_iwmmxt.c
libavcodec/arm/h264pred_init_arm.c
libavcodec/bfin/dsputil_bfin.c
libavcodec/dsputil.c
libavcodec/h264.c
libavcodec/h264.h
libavcodec/h264_cabac.c
libavcodec/h264_cavlc.c
libavcodec/h264_loopfilter.c
libavcodec/h264_ps.c
libavcodec/h264_refs.c
libavcodec/h264dsp.c
libavcodec/h264idct.c
libavcodec/h264pred.c
libavcodec/mlib/dsputil_mlib.c
libavcodec/options.c
libavcodec/ppc/dsputil_altivec.c
libavcodec/ppc/dsputil_ppc.c
libavcodec/ppc/h264_altivec.c
libavcodec/ps2/dsputil_mmi.c
libavcodec/sh4/dsputil_align.c
libavcodec/sh4/dsputil_sh4.c
libavcodec/sparc/dsputil_vis.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/x86/dsputil_mmx.c
libavformat/options.c
libavformat/utils.c
libavutil/pixfmt.h
libswscale/swscale.c
libswscale/swscale_internal.h
libswscale/swscale_template.c
tests/ref/seek/lavf_avi
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Looks better for some cases, worse for others, overall not much difference.
Its more correct though.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This should fix behavior introduced by commit
96573c0d76. Av_rescale_rnd() is not
lossless so if two timestamps are equal after being rescaled they are
not always actually identical. This patch use av_compare_ts() to get
always a correct result.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
