Callers always use a frame and cast it to AVPicture, change
ff_msrle_decode() to accept an AVFrame directly instead.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
This commit shall introduce the first step of adding support for the
Daala next generation video codec to FFmpeg. Although still in
development, the codec is showing good progress and exchanging work
through IETF drafts. The companies behind Daala are also participating
in the Alliance for Open Media, so it's likely that whatever the result
any of these collaborations produce it's probable that elements from
Daala could be used in them, or perhaps this codec itself could be the
result.
VP8E_UPD_ENTROPY, VP8E_UPD_REFERENCE, VP8E_USE_REFERENCE were removed
from libvpx and the remaining values were never used here
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Zern <jzern@google.com>
It was replaced by avpriv_ac3_parse_header2.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
The parser only reads the dca core sample rate, which is limited to a
maximum of 48000 Hz, while X96 and HD extensions can increase the sample
rate up to 192000 Hz.
This change prevents the parser and decoder fighting over the sample rate,
potentially confusing user applications. This also fixes sample rate
display of >48000Hz files with ffmpeg/ffprobe when using libdcadec.
Fixes ticket #4397
treat this the same as an over-sized superframe packet to break out of
the parser loop and allow the decoder to fail.
Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: James Zern <jzern@google.com>
Commit 3a0a2f33a6 claims large performance
advantages for AV_QSORT over libc's qsort. The reason is that I suspect
that libc's qsort (at least on non LTO builds, like the typical FFmpeg config)
can't inline the comparison callback:
https://stackoverflow.com/questions/5290695/is-there-any-way-a-c-c-compiler-can-inline-a-c-callback-function.
AV_QSORT has two things going for it:
1. The guaranteed inlining of qsort itself. This yields a negligible
boost that may be ignored.
2. The more serious possibility of potentially allowing the comparison
function to be inlined - this is likely responsible for the large boosts
reported.
There is a comment explaining that this is a place that could use some
performance improvement. Thus AV_QSORT is used to achieve that.
Benchmarks deemed unnecessary due to existing claims about AV_QSORT.
Tested with FATE.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
When the encoder is fed with less frames than its delay, the picture list looks like { NULL, NULL, ..., frame, frame, frame }. When flushing the encoder (input frame == NULL), we need to ensure the picture list is shifted enough so that we do not return an empty packet, which would mean the encoder has finished, while it has not encoded any frame.
Before the patch, the command:
'./ffmpeg_g -loglevel debug -f lavfi -i "testsrc=d=0.01" -bf 2 -vcodec mpeg2video out.mxf' prints:
Output stream #0:0 (video): 1 frames encoded; 0 packets muxed (0 bytes);
After:
Output stream #0:0 (video): 1 frames encoded; 1 packets muxed (8058 bytes);
Relates to ticket #4817.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
There were some errors in the calculation as well as an entire
unnecessary loop to find the gain coefficient. Merge the
two loops.
Thanks to @ubitux for the suggestions and testing.
Changes:
- strongly prefer dual filters to a single filter
- less strict about using 2 filters w.r.t. energy
- scrap the usage of threshold and spread, useless
- use odd-shaped windows to set the filter direction
- use 4 bits instead of 3 bits for short windows
- simplify and reduce the main loop to a single level
- add stricter regulations for short windows
All of this now makes the TNS implementation operate
as good as it can and it definitely shows. The frequency
thresholds are now even better defined by looking at
the spectrals and the overall sound has been improved at
the price of just a few bits that are well worth it.
Too much effort and work has been spent on such a simple function.
It simply refuses to work as the specifications say, the
transformation is NOT lossless and creates some crackling and
distortions.
Therefore disable it by default and add a couple of warnings to
scare people away from touching it or wasting their time the
way I did.
The decoder does this so I guess we better do that as well.
There's barely any difference between the autoregressive and
the moving average filters looking at spectrals though.
It didn't work out because of the exceptions that needed to be made
for the "-1" cases and was overall more confusing that just manually
checking and setting options for each profile.
Long Term Prediction allows for prediction of spectral coefficients
via the previously decoded time-dependent samples. This feature
works well with harmonic content 2 or more frames long, like speech,
human or non-human, piano music or any constant tones at very low
bitrates.
It should be noted that the current coder is highly efficient and
the rate control system is unable to encode files at extremely
low bitrates (less than 14kbps seems to be impossible) so this
extension isn't capable of optimum operation. Dramatic difference
is observable with some types of audio and speech but for the most
part the audiable differences are subtle. The spectrum looks better
however so the encoder is able to harvest the additional bits that
this feature provies, should the user choose to enable it. So
it's best to enable this feature only if encoding at the absolutely
lowest bitrate that the encoder is capable of.
Apparently it was set to be enabled by default but after the
profile commits it was reverted to be off by default because
I didn't notice.
Works well so (re)enable it.
The AVCodecContext.get_format callback is not only used for pixel format
negotiation with the API user, but also for hwaccel init. For the
latter, it's required that some codec parameters, in particular the
codec profile, are set when the callback is invoked.
This patch removes a get_format invocation where this is not guaranteed.
The codec parameters, including the profile, are really set further
below. (The same code path that sets the profile also calls get_format
properly too.)
This just happened to work by coincidence in most cases. For example, if
the API user just copied or reused the AVStream's AVCodecContext when
decoding, the profile would be set properly. But in some cases it
fails., such as with the sample WolfensteinTwitch.mp4 on the samples
server.
Remove the redundant get_format call. Apparently it serves no purpose
anymore, although it is possible that this was different at the time it
was added in commit ffd77f94a2.
This fixes hwaccel usage for API users which do not set the profile
when setting up the AVCodecContext (which is allowed).
This allows more efficient access to the array as the level and flags
are contiguous. Around 4% faster coefficient decoding.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This makes the h.264 decoder threadsafe to initialize.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Work on the AVFrame references directly.
Instead of setting up a flipped/swapped "view" on the pictures,
flip/swap them when returning decoded frames to the API user.
Rather than copying data buffers around, allocate a proper frame, and
use the standard AVFrame functions. This effectively makes the decoder
capable of direct rendering.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
This avoid going through constants.c while still sharing them
with proresdsp.asm
Reviewed-by: "Ronald S. Bultje" <rsbultje@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This patch fixes build of AAC encoder optimized for mips that was broken due
to some changes in generic code that were not propagated to the optimized code.
Also, some functions in the optimized code are basically duplicate of functions
from generic code. Since they do not bring enough improvement to the optimized
code to justify their existence, they are removed (which improves
maintainability of the optimized code).
Optimizations disabled in 97437bd are enabled again.
Signed-off-by: Nedeljko Babic <nedeljko.babic@imgtec.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
These aren't quite as helpful as the ones in 8bpp, since over there,
we can use pmulhrsw, but here the coefficients have too many bits to
be able to take advantage of pmulhrsw. However, we can still skip
cols for which all coefs are 0, and instead just zero the input data
for the row itx. This helps a few % on overall decoding speed.
The trouble with this function is that intermediates overflow 31+sign
bits, so I've added some helpers (that will also be used in 10/12bpp
8x8, 16x16 and 32x32) to make that easier, basically emulating a half-
assed pmaddqd using 2xpmaddwd. It's currently sse2-only, if anyone sees
potential in adding ssse3, I'd love to hear it.
On 12 frames of a 444p 12 bits DNxHR sequence, _put function:
C: 78902 decicycles in idct, 262071 runs, 73 skips
avx: 32478 decicycles in idct, 262045 runs, 99 skips
Difference between the 2:
stddev: 0.39 PSNR:104.47 MAXDIFF: 2
This is unavoidable and due to the scale factors used in the x86
version, which cannot match the C ones.
In addition, the trick of adding an initial bias to the input of a
pass can overflow, as the input coefficients are already 15bits,
which is the maximum this function can handle.
Overall, however, the omse on 12 bits samples goes from 0.16916 to
0.16883. Reducing rowshift by 1 improves to 0.0908, but causes
overflows.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Modeled from the prores version. Clips to [0;1023] and is bitexact.
Bitexactness requires to add offsets in different places compared to
prores or C, and makes the function approximately 2% slower.
For 16 frames of a DNxHD 4:2:2 10bits test sequence:
C: 60861 decicycles in idct, 1048205 runs, 371 skips
sse2: 27567 decicycles in idct, 1048216 runs, 360 skips
avx: 26272 decicycles in idct, 1048171 runs, 405 skips
The add version is not implemented, so the corresponding dsp
function is set to NULL to make it clear in a code executing it.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When the input of a pass has 15 or 16 bits of precision (in particular
the column pass), the addition of a bias to W4 may lead to overflows
in the input to pmaddwd.
This requires postponing the adding of the bias to after the first
butterfly. To do so, the fact that m15, unused although zeroed, is
exploited. In case the pass is safe, an address can be directly used,
and the number of xmm regs can be decreased. Otherwise, the 32bits bias
is loaded into it.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
omse goes from 0.03060703 (which fails for dct-test) to 0.01663750.
This also actually improve the error of decoding the sample generated
by fate-vsynth3-dnxhd1080i-10bit using simple_idct10 to FAANI, which
goes (when resampled to yuv422p) from:
stddev: 0.06 PSNR: 72.28 MAXDIFF: 1
to identical.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This should be reused for a generic simple_idct10 function.
Requires a bit of trickery to declare common constants in C.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Bare ampersand characters are still accepted, even though out-of-spec.
Also fixes adjacent tags not being parsed.
Fixes trac #4915
Signed-off-by: Ricardo Constantino <wiiaboo@gmail.com>
This commit adds the ability for a profile to set the default
options, as well as for the user to override such options
by simply stating them in the command line while still keeping
the same profile, as long as those options are still permitted by
the profile.
Example: setting the profile to aac_low (the default) will turn
PNS and IS on. They can be disabled by -aac_pns 0 and -aac_is 0,
respectively. Turning on -aac_pred 1 will cause the profile to be
elevated to aac_main, as long as no options forbidding aac_main
have been entered (like AAC-LTP, which will be pushed soon).
A useful feature is that by setting the profile to mpeg2_aac_low,
all MPEG4 features will be disabled and if the user tries to enable
them then the program will exit with an error. This profile is
signalled with the same bitstream as aac_low (MPEG4) but some devices
and decoders will fail if any MPEG4 features have been enabled.
This commit implements support for 7.1 channel audio. There's no
more predefined bitstream channel mappings so going beyond 8 channels
(and 7 channels exactly) will require programmable channel elements,
which is already underway.
The bulk of calls to quantize_band_cost are replaced
by a call to a version that memoizes, greatly improving
performance, since during coefficient search there is
a great deal of repeat work.
Memoization cannot always be applied, so do this in a
different function, and leave the original as-is.
Intermediate results can indeed violate SF delta. Instead of asserting
there, just make the code safe, and assert on the final result.
Also re-clamp SFs more often in short windows (which tend to violate
the restriction when encoding the switch from one window to the other)
It was merged with the iff_ilbm decoder in commit
929a24efff.
Define AV_CODEC_ID_IFF_BYTERUN1 as AV_CODEC_ID_IFF_ILBM for API
compatibility.
Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
This finalizes merging of the work in the patches in ticket #2686.
Improvements to twoloop and RC logic are extensive.
The non-exhaustive list of twoloop improvments includes:
- Tweaks to distortion limits on the RD optimization phase of twoloop
- Deeper search in twoloop
- PNS information marking to let twoloop decide when to use it
(turned out having the decision made separately wasn't working)
- Tonal band detection and priorization
- Better band energy conservation rules
- Strict hole avoidance
For rate control:
- Use psymodel's bit allocation to allow proper use of the bit
reservoir. Don't work against the bit reservoir by moving lambda
in the opposite direction when psymodel decides to allocate more/less
bits to a frame.
- Retry the encode if the effective rate lies outside a reasonable
margin of psymodel's allocation or the selected ABR.
- Log average lambda at the end. Useful info for everyone, but especially
for tuning of the various encoder constants that relate to lambda
feedback.
Psy:
- Do not apply lowpass with a FIR filter, instead just let the coder
zero bands above the cutoff. The FIR filter induces group delay,
and while zeroing bands causes ripple, it's lost in the quantization
noise.
- Experimental VBR bit allocation code
- Tweak automatic lowpass filter threshold to maximize audio bandwidth
at all bitrates while still providing acceptable, stable quality.
I/S:
- Phase decision fixes. Unrelated to #2686, but the bugs only surfaced
when the merge was finalized. Measure I/S band energy accounting for
phase, and prevent I/S and M/S from being applied both.
PNS:
- Avoid marking short bands with PNS when they're part of a window
group in which there's a large variation of energy from one window
to the next. PNS can't preserve those and the effect is extremely
noticeable.
M/S:
- Implement BMLD protection similar to the specified in
ISO-IEC/13818:7-2003, Appendix C Section 6.1. Since M/S decision
doesn't conform to section 6.1, a different method had to be
implemented, but should provide equivalent protection.
- Move the decision logic closer to the method specified in
ISO-IEC/13818:7-2003, Appendix C Section 6.1. Specifically,
make sure M/S needs less bits than dual stereo.
- Don't apply M/S in bands that are using I/S
Now, this of course needed adjustments in the compare targets and
fuzz factors of the AAC encoder's fate tests, but if wondering why
the targets go up (more distortion), consider the previous coder
was using too many bits on LF content (far more than required by
psy), and thus those signals will now be more distorted, not less.
The extra distortion isn't audible though, I carried extensive
ABX testing to make sure.
A very similar patch was also extensively tested by Kamendo2 in
the context of #2686.
This should fix the undefined behavior reported in:
https://trac.ffmpeg.org/ticket/4727.
I can reproduce this at runtime: simply stick in an abort call in
asym_quant to check if c < 0 and run FATE. I don't know ac3 so I can't
confirm if negative coefficients are intentional, but at the moment they
clearly are according to FATE.
This resolves the undefined behavior. Tested with FATE.
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
the pps offset is used to locate pps in the spspps_buf; however, the
current calc method is wrong because it is the offset of the original
avctx->extradata;
when there is only one sps in the avcc; the value is correct by
coincidence, however, it will fail in avcc with multi sps
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This fixes a warning observed on Clang 3.7:
"warning: attribute 'deprecated' is ignored, place it after "struct" to apply attribute to type declaration [-Wignored-attributes]"
and thus enables deprecation warning for the relevant struct.
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This should fix the first undefined behavior reported in:
https://trac.ffmpeg.org/ticket/4727.
I can't reproduce the runtime behavior reported in the ticket, hence I
can't confirm that this actually fixes the exact issue reported in the
ticket.
Regardless, I can confirm that this is a genuine issue, and that
negative shifts can (and do) occur, fixed by this.
Tested with FATE.
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
There is not much reason to generate such a small table at runtime.
Signed-off-by: Derek Buitenhuis <derekb@vimeo.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The MBAFF handling recently introduced on the decoder side shows that
the encoder does not support it correctly. Therefore, make the related
profile experimental.
Furthermore, current encoder logic treats it as unable to encode as
progressive, which isn't the case.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
MBAFF-like handling of interlaced content in CID 1260 is different from
the other CIDs, and in particular doesn't use the same syntax.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Profiles 1256 & 1270 (currently) signal at the frame header and MB
levels the colorspace used, either RGB or YUV. While a MB-level
varying colorspace is not supported, whether it is constant can be
tracked so as to determine the exact colorspace.
This requires having bitdepth and the ACT and 4:4:4 flags, in turn
needing the CID. Because setting those before having validated
enough things may result in invalid/unset DSP fucntions, setting
the bitdepth in the context is delayed.
It is not tested against a true RGB sequence, though.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
place primary audio coding header data into DCAAudioHeader
structure to make DCAContext clearer
and move channel related data to DCAChan structure to make
them easier to use by extensions
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Do not fail when original resolution is smaller than current one,
as the frame buffer is resized automatically.
Signed-off-by: Vittorio Giovara <vittorio.giovara at gmail.com>
Modified datatype of function argument (pitch from int32_t to ptrdiff_t)
Signed-off-by: Shivraj Patil <shivraj.patil@imgtec.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
