There is a minor bug in xbm encode which adds a trailing comma at the end
of data. This isn't a big problem, but it would be nicer to be more
technically true to an array of data (by not including the last comma).
This bug fixes the output from something like this (having 4 values):
static unsigned char image_bits[] = { 0x00, 0x11, 0x22, }
to C code that looks like this instead (having 3 values):
static unsigned char image_bits[] = { 0x00, 0x11, 0x22 }
which is the intended results.
Subject: [PATCH 1/3] avcodec/xbmenc: Do not add last comma into output array
xbm outputs c arrays of data.
Including a comma at the end means there is another value to be added.
This bug fix changes something like this:
static unsigned char image_bits[] = { 0x00, 0x11, 0x22, }
to C code like this:
static unsigned char image_bits[] = { 0x00, 0x11, 0x22 }
Signed-off-by: Joe Da Silva <digital@joescat.com>
If the edit lists remove parts of the output timeline, or add a
delay to it, this should be included in the mvhd/tkhd/mdhd durations,
which should correspond to the edit lists.
For tracks starting with pts < 0, the edit list trims out the segment
before pts=0. For tracks starting with pts > 0, a delay element is
added in the edit list, delaying the start of the track data.
In both cases, the practical effect is that the post-edit output
is as if the track had started with pts = 0. Thus calculate the range
from pts=0 to end_pts, for the purposes of mvhd/tkhd/mdhd, unless
edit lists explicitly are disabled.
mov_write_edts_tag needs to operate on the actual pts duration of
the track samples, not the duration that already takes the edit
list effect into account.
Signed-off-by: Martin Storsjö <martin@martin.st>
The AC3 encoder used to be a separate library called "Aften", which
got merged into libavcodec (literally, SVN commits and all).
The merge preserved as much features from the library as possible.
The code had two versions - a fixed point version and a floating
point version. FFmpeg had floating point DSP code used by other
codecs, the AC3 decoder including, so the floating-point DSP was
simply replaced with FFmpeg's own functions.
However, FFmpeg had no fixed-point audio code at that point. So
the encoder brought along its own fixed-point DSP functions,
including a fixed-point MDCT.
The fixed-point MDCT itself is trivially just a float MDCT with a
different type and each multiply being a fixed-point multiply.
So over time, it got refactored, and the FFT used for all other codecs
was templated.
Due to design decisions at the time, the fixed-point version of the
encoder operates at 16-bits of precision. Although convenient, this,
even at the time, was inadequate and inefficient. The encoder is noisy,
does not produce output comparable to the float encoder, and even
rings at higher frequencies due to the badly approximated winow function.
Enter MIPS (owned by Imagination Technologies at the time). They wanted
quick fixed-point decoding on their FPUless cores. So they contributed
patches to template the AC3 decoder so it had both a fixed-point
and a floating-point version. They also did the same for the AAC decoder.
They however, used 32-bit samples. Not 16-bits. And we did not have
32-bit fixed-point DSP functions, including an MDCT. But instead of
templating our MDCT to output 3 versions (float, 32-bit fixed and 16-bit fixed),
they simply copy-pasted their own MDCT into ours, and completely
ifdeffed our own MDCT code out if a 32-bit fixed point MDCT was selected.
This is also the status quo nowadays - 2 separate MDCTs, one which
produces floating point and 16-bit fixed point versions, and one
sort-of integrated which produces 32-bit MDCT.
MIPS weren't all that interested in encoding, so they left the encoder
as-is, and they didn't care much about the ifdeffery, mess or quality - it's
not their problem.
So the MDCT/FFT code has always been a thorn in anyone looking to clean up
code's eye.
Backstory over. Internally AC3 operates on 25-bit fixed-point coefficients.
So for the floating point version, the encoder simply runs the float MDCT,
and converts the resulting coefficients to 25-bit fixed-point, as AC3 is inherently
a fixed-point codec. For the fixed-point version, the input is 16-bit samples,
so to maximize precision the frame samples are analyzed and the highest set
bit is detected via ac3_max_msb_abs_int16(), and the coefficients are then
scaled up via ac3_lshift_int16(), so the input for the FFT is always at least 14 bits,
computed in normalize_samples(). After FFT, the coefficients are scaled up to 25 bits.
This patch simply changes the encoder to accept 32-bit samples, reusing
the already well-optimized 32-bit MDCT code, allowing us to clean up and drop
a large part of a very messy code of ours, as well as prepare for the future lavu/tx
conversion. The coefficients are simply scaled down to 25 bits during windowing,
skipping 2 separate scalings, as the hacks to extend precision are simply no longer
necessary. There's no point in running the MDCT always at 32 bits when you're
going to drop 6 bits off anyway, the headroom is plenty, and the MDCT rounds
properly.
This also makes the encoder even slightly more accurate over the float version,
as there's no coefficient conversion step necessary.
SIZE SAVINGS:
ARM32:
HARDCODED TABLES:
BASE - 10709590
DROP DSP - 10702872 - diff: -6.56KiB
DROP MDCT - 10667932 - diff: -34.12KiB - both: -40.68KiB
DROP FFT - 10336652 - diff: -323.52KiB - all: -364.20KiB
SOFTCODED TABLES:
BASE - 9685096
DROP DSP - 9678378 - diff: -6.56KiB
DROP MDCT - 9643466 - diff: -34.09KiB - both: -40.65KiB
DROP FFT - 9573918 - diff: -67.92KiB - all: -108.57KiB
ARM64:
HARDCODED TABLES:
BASE - 14641112
DROP DSP - 14633806 - diff: -7.13KiB
DROP MDCT - 14604812 - diff: -28.31KiB - both: -35.45KiB
DROP FFT - 14286826 - diff: -310.53KiB - all: -345.98KiB
SOFTCODED TABLES:
BASE - 13636238
DROP DSP - 13628932 - diff: -7.13KiB
DROP MDCT - 13599866 - diff: -28.38KiB - both: -35.52KiB
DROP FFT - 13542080 - diff: -56.43KiB - all: -91.95KiB
x86:
HARDCODED TABLES:
BASE - 12367336
DROP DSP - 12354698 - diff: -12.34KiB
DROP MDCT - 12331024 - diff: -23.12KiB - both: -35.46KiB
DROP FFT - 12029788 - diff: -294.18KiB - all: -329.64KiB
SOFTCODED TABLES:
BASE - 11358094
DROP DSP - 11345456 - diff: -12.34KiB
DROP MDCT - 11321742 - diff: -23.16KiB - both: -35.50KiB
DROP FFT - 11276946 - diff: -43.75KiB - all: -79.25KiB
PERFORMANCE (10min random s32le):
ARM32 - before - 39.9x - 0m15.046s
ARM32 - after - 28.2x - 0m21.525s
Speed: -30%
ARM64 - before - 36.1x - 0m16.637s
ARM64 - after - 36.0x - 0m16.727s
Speed: -0.5%
x86 - before - 184x - 0m3.277s
x86 - after - 190x - 0m3.187s
Speed: +3%
Fixes a decoding regression introduced by e9a2a87773, and as a side effect also
fixes bogus values set to certain audio frames that had some samples discarded,
where the offsets added to pts, pkt_dts and pkt_duration were not reflected in
best_effort_timestamp.
Signed-off-by: James Almer <jamrial@gmail.com>
SMVJPEG stores frames as slices of a big JPEG image. The decoder is
implemented as a wrapper that instantiates a full internal MJPEG
decoder, then forwards the decoded frames with offset data pointers.
This is unnecessarily complex and fragile, not supporting useful decoder
capabilities like direct rendering.
Re-implement the decoder inside the MJPEG decoder, which is accomplished
by returning each decoded frame multiple times, setting cropping
information appropriately on each instance.
One peculiar aspect of the previous design is that since
- the smvjpeg decoder returns one frame per input packet
- there are multiple frames in each packets (the aformentioned slices)
the demuxer needs to return each packet multiple times.
This is now also eliminated - the demuxer now returns each packet
exactly once, with the duration set to the number of frames it decodes
to.
This also removes one of the last remaining internal uses of the old
video decoding API.
It depends on the muxer generating the timestamps, which is deprecated
and scheduled for removal on next bump.
A bunch of tests change timestamps, because of ffmpeg.c is not
generating them correctly. This should be fixed later.
Factor out the code into a separate muxing-specific function.
Stop accessing the deprecated AVStream-embedded codec context, use the
average framerate (if specified) instead.
By using the frame counter (and the video time base) for audio pts we lose some
timestamp precision but we ensure that video and audio coming from the same DV
frame are always in sync.
This patch also makes timestamps after seek consistent and it should also fix
the timestamps when the audio clock is unlocked and have a completely
indpendent clock source. (E.g. runs on fixed 48009 Hz which should have been
exact 48000 Hz)
Fixes out of sync timestamps in ticket #8762.
Signed-off-by: Marton Balint <cus@passwd.hu>
The previous threshold, 4 KB, maybe was reasonable when it was set
(in 2010), but in today's settings and with typical network speeds
and data sizes, it's pretty small. 32 KB probably is a more reasonable
default now, regardless of input.
This changes the test references for two seek tests.
When using the normal seek function, which boils down to the lseek(2)
function, a seek to an out of bounds position doesn't return an error,
but that condition is only reported when doing the subsequent read
(which returns EOF). When doing more seeks by fast forwarding, the
fact that the seeked to destination is out of bounds is noticed and
reported sooner in these cases.
Signed-off-by: Martin Storsjö <martin@martin.st>
While the FATE suite contains a sample file for Musepack 8, it did not
use it to test the decoder; it is only used in the mpc8-demux test that
tests the demuxer via streamcopy. Therefore this commit adds an actual
encoder test.
The test uses the framecrc output, because Musepack SV8 is an encoder
that returns multiple frames for a single packet, so that timing
information in the test output is valueable. Output seeking has been
used in order to limit the size of the ref file as well as to test this
codepath for the first time.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
We now have the possibility of getting AVFrames here, and we should
not touch the muxer's codecpar after writing the header.
Results of FATE tests change as the MXF and Matroska muxers actually
write down the field/frame coding type of a stream in their
respective headers. Before this change, these values in codecpar
would only be set after the muxer was initialized. Now, the
information is also available for encoder and muxer initialization.
Additionally, reap the first rewards by being able to set the
color related encoding values based on the passed AVFrame.
The only tests that seem to have changed their results with this
change seem to be the MXF tests. There, the muxer writes the
limited/full range flag to the output container if the encoder
is not set to "unspecified".
This disallows the usage of ? and # in libavformat specific scheme options
(e.g. subfile,,start,32815239,end,0,,:video.ts) but this change was considered
acceptable.
Signed-off-by: ruiquan.crq <caihaoning83@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
These conversion appears to be exhibiting the same rounding error as the rgbf32 formats where.
I seperated the rounding value from the 16 and 128 offsets, I think it makes it a little more clear.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
changes since v1:
- made into fate test
- fixed c90 warnings
- tests more intermediate formats
- tested on BE mips too
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If the average bit rate cannot be calculated, such as in the case
of streamed fragmented mp4, utilize various available parameters
in priority order.
Tests are updated where the esds or btrt or ISML manifest boxes'
output changes.
This is utilized by various media ingests to figure out the bit
rate of the content you are pushing towards it, so write it for
video, audio and subtitle tracks in case at least one nonzero value
is available. It is only mentioned for timed metadata sample
descriptions in QTFF, so limit it only to ISOBMFF (MODE_MP4) mode.
Updates the FATE tests which have their results changed due to the
20 extra bytes being written per track.
SMPTE 12M timecode can only count frames up to 39, because the tens-of-frames
value is stored in 2 bit. In order to resolve this 50/60 fps SMPTE timecode is
using the field bit (which is the same bit as the phase correction bit) to
signal the least significant bit of a 50/60 fps timecode. See SMPTE ST
12-1:2014 section 12.1.
Therefore we slightly change the format of the return value of
av_timecode_get_smpte_from_framenum and AV_FRAME_DATA_S12M_TIMECODE and start
using the previously unused Phase Correction bit as Field bit. (As the SMPTE
standard suggests)
We add 50/60 fps support to av_timecode_get_smpte_from_framenum by calling the
recently added av_timecode_get_smpte function in it which already handles this
properly.
This change affects the decklink indev and the DV and MXF muxers. MXF has no
fate test for 50/60fps content, DV does, therefore the changes.
MediaInfo (a recent version) confirms that half-frame timecode must be inserted
to DV. MXFInspect confirms valid timecode insertion to the System Item of MXF
files. For MXF, also see EBU R122.
Note that for DV the field flag is not used because in the HDV specs (SMPTE
370M) it is still defined as biphase mark polarity correction flag. So it
should not matter that the DV muxer overrides the field bit.
Signed-off-by: Marton Balint <cus@passwd.hu>
This AV1 decoder is currently only used for hardware accelerated decoding.
It can be extended into a native decoder in the future, so set its name to
"av1" and temporarily give it the lowest priority in the codec list.
Signed-off-by: Fei Wang <fei.w.wang@intel.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Allow to set the EOF timestamp.
Also: doc/filters/testsrc*: specify the rounding of the duration option.
The changes in the ref files are right.
For filter-fps-down, the graph is testsrc2=r=7:d=3.5,fps=3.
3.5=24.5/7, so the EOF of testsrc2 will have PTS 25/7.
25/7=(10+5/7)/3, so the EOF PTS for fps should be 11/7,
and the output should contain a frame at PTS 10.
For filter-fps-up, the graph is testsrc2=r=3:d=2,fps=7,
for filter-fps-up-round-down and filter-fps-up-round-up
it is the same with explicit rounding options.
But there is no rounding: testsrc2 produces exactly 6 frames
and 2 seconds, fps converts it into exactly 14 frames.
The tests should probably be adjusted to restore them to
a useful coverage.
Paul B Mahol
Jose Da Silva
Marton Balint
Martin Storsjö
Lynne
Anton Khirnov
James Almer
Harry Mallon
Mark Reid
Mohammad Izadi
Thierry Foucu
Limin Wang
Zane van Iperen
Andreas Rheinhardt
Jan Ekström
ruiquan.crq
Michael Niedermayer
Tomas Härdin
Fei Wang
Nicolas George