I/S energy, especially when it comes to phase cancellations,
needs to use signed coefficients as input, yet it was using
abs'd coefficients. That was a slight bug.
The relative error between two encoding strategies is the simple
difference of rate-distortion values, and not the absolute
difference. An absolute measure would allow worsening of the
quantization error as well as improving.
1. Fix sf_idx and band_type addressing to address only the first
subwindow in the group (others could hold garbage values)
2. Don't step on ms_mask when is_mask is set. I/S selection
already sets the ms_mask properly and shouldn't be overridden.
3. Use mid/sid cb/sf when computing coding error, as should be
since those are the cb/sfs that will eventually be set.
4. Fix distortion computation on multi-subwindow groups (was
subtracting the bits terms multiple times)
5. Clear ms_mask when one side uses PNS and the other doesn't.
When using PNS, ms_mask signals correlated noise, which can be
detected just like regular M/S detection, so we don't skip
noise bands, but when only one side uses PNS setting the flag
can confuse some encoders, so avoid that.
* commit 'aebf07075f4244caf591a3af71e5872fe314e87b':
dca: change the core to work with integer coefficients.
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
Fixes Ticket #5032
The samples in Ticket #5032 is using \r\r\n as line breaks. Since we
already are handling \r, or \n, or \r\n as line breaks, \r\n\n will be
considered as a double line breaks. This is an issue because
ff_subtitles_read_text_chunk() will as a result stop extracting a chunk
after just one line.
So instead of parsing the SRT by "chunks" (which means splitting every
double LB), this new parser is detecting timing lines, and split the
events on this basis. While this sounds safe and simple, it needs to
take into account the event number preceding the timing line while
handling situations such as:
- event number starting at 0 or actually any number instead of 1
- event numbers not being ordered at all
- event number being followed by text garbage (this really happened,
see Ticket #4898)
- event payload containing one or multiple number (a protagonist saying
a count-down, a date or whatever) which could be confused with a
chapter number
- event number being empty (see Ticket #2167)
- all kind of weird line breaks can appear randomly like wild pokémons
- untrustable line breaks (Ticket #5032)
The sample madness.srt tries to sum up most of this into one sample,
ticket5032-rrn.srt is the file containing \r\r\n line breaks. and
empty-events-2167.srt contains empty events.
The DCA core decoder converts integer coefficients read from the
bitstream to floats just after reading them (along with dequantization).
All the other steps of the audio reconstruction are done with floats
which makes the output for the DTS lossless extension (XLL)
actually lossy.
This patch changes the DCA core to work with integer coefficients
until QMF. At this point the integer coefficients are converted to floats.
The coefficients for the LFE channel (lfe_data) are not touched.
This is the first step for the really lossless XLL decoding.
This should fix this test failing on kfreebsd, a regression since
6e5dbe7, which decreased the CMP_TARGET by 1.
Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
With only 7 coefficients per short window at most the extra precision
makes a difference and seems to reduce crackling and stddev even
further.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This patch does 4 things, all of which interact and thus it
woudln't be possible to commit them separately without causing
either quality regressions or assertion failures.
Fate comparison targets don't all reflect improvements in
quality, yet listening tests show substantially improved quality
and stability.
1. Increase SF range utilization.
The spec requires SF delta values to be constrained within the
range -60..60. The previous code was applying that range to
the whole SF array and not only the deltas of consecutive values,
because doing so requires smarter code: zeroing or otherwise
skipping a band may invalidate lots of SF choices.
This patch implements that logic to allow the coders to utilize
the full dynamic range of scalefactors, increasing quality quite
considerably, and fixing delta-SF-related assertion failures,
since now the limitation is enforced rather than asserted.
2. PNS tweaks
The previous modification makes big improvements in twoloop's
efficiency, and every time that happens PNS logic needs to be
tweaked accordingly to avoid it from stepping all over twoloop's
decisions. This patch includes modifications of the sort.
3. Account for lowpass cutoff during PSY analysis
The closer PSY's allocation is to final allocation the better
the quality is, and given these modifications, twoloop is now
very efficient at avoiding holes. Thus, to compute accurate
thresholds, PSY needs to account for the lowpass applied
implicitly during twoloop (by zeroing high bands).
This patch makes twoloop set the cutoff in psymodel's context
the first time it runs, and makes PSY account for it during
threshold computation, making PE and threshold computations
closer to the final allocation and thus achieving better
subjective quality.
4. Tweaks to RC lambda tracking loop in relation to PNS
Without this tweak some corner cases cause quality regressions.
Basically, lambda needs to react faster to overall bitrate
efficiency changes since now PNS can be quite successful in
enforcing maximum bitrates, when PSY allocates too many bits
to the lower bands, suppressing the signals RC logic uses to
lower lambda in those cases and causing aggressive PNS.
This tweak makes PNS much less aggressive, though it can still
use some further tweaks.
Also update MIPS specializations and adjust fuzz
Also in lavc/mips/aacpsy_mips.h: remove trailing whitespace
"Fast seek" uses linear interpolation to find the position of the
requested seek time. For CBR this is more direct than using the
mp3 TOC and bypassing the TOC avoids problems with TOC precision.
(see https://crbug.com/545914#c13)
For VBR, fast seek is not precise, so continue to prefer the TOC
when available (the lesser of two evils).
Also, some re-ordering of the logic in mp3_seek to simplify and
give usetoc=1 precedence over fastseek flag.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
As noted in a comment, pe.min in the reference encoder
is centered around current pe. The bit reservoir algo
needs pe.min to be a local minimum, because it can only
account for local PE variations. If it's set to a global
minimum as was being done, bit reservoir logic doesn't
work as efficiently.
This patch tries to forget old minimums and converge to
a local minimum without losing the stability of the
previous solution. Listening tests until now suggest this
solves numerous RC issues.
* commit '823fa7004571cb8404ca5785f9fa6e85f0f9f3d3':
fate: Rework sgi tests into a suite and add the missing ones
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
All diferences in unit tests have been acounted for.
* commit '59e8ec0aa8ab174701d01a3bfe96fedd0b7fcead':
movenc: Add an API unit test for fragmenting options/calls
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Contrary to the normal fate tests that run via avconv, this tests
nontrivial call sequences that are only doable via the API
(mainly for different corner cases when using the muxer for
segmenting).
The test muxes fake packet data (with extradata that looks
enough like proper data to make the file be viewable with e.g.
boxdumper) and checks the hash of the produced files. The test also
verifies that fragments produced via different call sequences remain
identical (to avoid e.g. updating the output hashes and suddenly
having fragments that used to be identical suddenly diverging), for
fragments written with frag_discont and/or delay_moov.
Signed-off-by: Martin Storsjö <martin@martin.st>
The CMP variable seems to have been inherited from fate-api-seek which set it to null
the mxf reference needed a change due to c7e14a279f
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Similar to testsrc, but using drawutils and therefore
supporting a lot of pixel formats instead of just rgb24.
This allows using it as input for other tests without
requiring a format conversion.
It is also slightly faster than testsrc for some reason.
This fixes a fate failure after bumping the minor version
Its unknown why this is not needed for the other aac tests,
more investigation needed but for now i dont want to leave
it broken while its investigated
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.
The fate test command line is supposed to serve as an example. It's
nicer to explicitly state the profile rather than setting options
to force it for you.
GCC 3.4 miscompiles it on sunos. Date of release? The second of
August two thousand and five, anno Domini. That's ten years two
months and fourteen days ago. Three thousand seven hundred and
twenty seven days ago. One sixth of the average life expectancy
of a person living in a country with a human development index
of zero point eight hundred and eight, equality adjusted.
GCC 4.3 also miscompiles it, though not as bad.
The LTP encoding and the test is a bit slow currently, taking twice
the amount of time the other tests do, so in the future the
total time to encode might be cut down on that test.
It was useful to (accidentally?) spot an overflow in the column pass
of the x86 simple_idct10 implementation.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Includes escapes that should now be supported and a few features not yet
fully supported, like comments, regions, classes, ruby, and lang.
All were tested with https://quuz.org/webvtt/ for validation, except
regions because the validator doesn't support them yet, and I couldn't
find any other way to validate WebVTT.
Signed-off-by: Ricardo Constantino <wiiaboo@gmail.com>
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.
Currently only 2 profiles are evaluated because they are the only 2
with distributed test sequences.
- CID 1260: YUV 4:2:2 10 bits with block-adaptive interlace coding,
from ticket 4876;
- CID 1270: YUV 4:4:4 10 bits (HR), 1920x839, from ticket 4581.
They were generated from the ticket sequences by running the
following kind of command-line;
ffmpeg -i $INPUT -an -sn -vcodec copy -vframes 1 -y $OUTPUT.mov
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This patch tweaks search_for_pns to be both more
aggressive and more careful when applying PNS. On
the one side, it will again try to use PNS on zero
(or effectively zero) bands. For this, both zeroes
and band_type have to be checked (some ZERO bands
aren't marked in zeroes). On the other side, a more
accurate rate-distortion measure avoids using PNS
where it would cause audible distortion.
Also fixed a small bug in the computation of freq
that caused PNS usage on low-frequency bands during
8-short windows. This allows re-enabling PNS during
8-short.
The sample position is made weird and non-nominal to force catching
such issues as default values or specialized operations hiding
issues in corner cases.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This patch modifies the encode frame function to
retry encoding the frame when the resulting bit count
is too far off target, but only adjusting lambda
in small, incremental step. It also makes the logic
more conservative - otherwise it will contend with
bit reservoir-related variations in bit allocation,
and result in artifacts when frame have to be truncated
(usually at high bit rates transitioning from low
complexity to high complexity).
This patch refactors the AAC coders to reuse code
between the MIPS port and the regular, portable C code.
There were two main functions that had to use
hand-optimized versions of quantization code:
- search_for_quantizers_twoloop
- codebook_trellis_rate
Those two were split into their own template header
files so they can be inlined inside both the MIPS port
and the generic code. In each context, they'll link
to their specialized implementations, and thus be
optimized by the compiler.
This approach I believe is better than maintaining
several copies of each function. As past experience has
proven, having to keep those in sync was error prone.
In this way, they will remain in sync by default.
Also, an implementation of the dequantized output
argument for the optimized quantize_and_encode
functions is included in the patch. While the current
implementation of search_for_pred still isn't using
it, future iterations of main prediction probably will.
It should not imply any measurable performance hit while
not being used.
The recent commits change the value slightly. Even though it's
within the threshold it's better to risk as little as possible
especially when different systems, processors, FPUs and compilers
are involved.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit changes a few things about the noise substitution
logic:
- Brings back the quantization factor (reduced to 3) during
scalefactor index calculations.
- Rejects any zeroed bands. They should be inaudiable and it's
a waste transmitting the scalefactor indices for these.
- Uses swb_offsets instead of incrementing a 'start' with every
window group size.
- Rejects all PNS during short windows.
Overall improves quality. There was a plan to use the lfg system
to create the random numbers instead of using whatever the decoder
uses but for now this works fine. Entropy is far from important here.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit once again improves the PNS implementation by scaling the
thresholds with frequency. The thresholds get looser as the frequency
increases since higher frequencies are basically noise to human ears.
Also, this introduces quantization error correction for PNS. Should
the error be too much, no PNS will be used. The energy_ratio is used
to regulate the actual encoded PNS energy: if the generated PNS
energy is higher than the energy from the psy system, energy_ratio
is used to correct it so that hopefully once requantized and
transmitted the value in the decoder will be closer to what the
encoder has.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This was an oversight when the IS system was being first implemented.
The ener01 part was largely a result of trial and error and the fact
that the sum of coef0 and coef1 could result in a zero was
overlooked. Once ener01 turns to zero it's used to divide the left
channel energy which doesn't turn out so well as it fills IS[]
with -nan's and inf's which in turn confused the quantize_band_cost.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
TNS had both IS and PNS switched on when it makes more sense
to have them both off.
Prediction had a redundant argument.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
IS and PNS increase quality a ton so as a result the PSNR changed.
Disable the extensions and keep the tests separate such that there
will be no red herrings if one test fails.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Without this fate-filter-join failes with
FF_API_GET_CHANNEL_LAYOUT_COMPAT disabled.
Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
This fixes fate with FF_API_LAVF_BITEXACT disabled.
Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Tests fails on some ARM builds but it's close enough so it's okay.
NEON, half-precision floats, rounding errors, who knows.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit introduces a test for AAC-Main prediction
which was just reworked in this series of commits.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Works only for flv, h263 and huffyuv decoders.
Makes only one pass through the file (this should be changed to two passes)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This fixes fate with FF_API_REQUEST_CHANNELS disabled.
Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Works only with video stream.
First pass without seeking -- counts crcs of a frames and store it in an array.
After that it seeks a lot in different places and checks if crcs of these frames and crcs of frames in array are the same.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '58c3720a3cc71142b5d48d8ccdc9213f9a66cd33':
fate: Make sure a corner-case for ASF is covered
Adjusted fate ref to match the different timebase of the ffasf demuxer
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
Compute individual stream durations in matroska muxer.
Write them as string tags in the same format as mkvmerge tool does.
Signed-off-by: Sasi Inguva <isasi@google.com>
* commit 'a0797950527120c85263c910eb6ba08fddcfdcb3':
fate/mp3: specify the number of output samples instead of filesize
Merged-by: Michael Niedermayer <michael@niedermayer.cc>
Avoid clipping due to quantization noise to produce audible
artifacts, by detecting near-clipping signals and both attenuating
them a little and encoding escape-encoded bands (usually the
loudest) rounding towards zero instead of nearest, which tends to
decrease overall energy and thus clipping.
Currently fate tests measure numerical error so this change makes
tests using asynth (which are near clipping) report higher error
not less, because of window attenuation. Yet, they sound better,
not worse (albeit subtle, other samples aren't subtle at all).
Only measuring psychoacoustically weighted error would make for
a representative test, so that will be left for a future patch.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The test file they use needs avdevice to be created
Probably fixes Ticket 4455
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This change fixes a bug where a test that required a sample was being included
in the suite when SAMPLES was not set. It also improves the consistency of
variable names relating to the API tests.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'f91fe24e9bd6912c29bbb03d8afe878e045f9721':
g2meet: force simple idct for identical results over all fate configs
Conflicts:
tests/ref/fate/g2m3
tests/ref/fate/g2m4
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '4d1229dabf7a7e3b6a7b326afd79102256c3b008':
g2meet: Add FATE tests for all three G2M variants
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Most of the fate-dds-* and fate-txd-* tests already
output into the same pixel format regardless of
platform endianness, so there's no need to force
conversion to another format.
This fixes the tests fate-txd-16bpp, fate-txd-odd,
fate-dds-rgb16, fate-dds-rgb24 and fate-dds-xrgb on
big endian, where the tests seem to fail due to issues
with certain conversion codepaths in swscale.
Those conversion codepaths should of course be fixed, but
the individual decoder tests should use as little extra
conversion steps as possible.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '3ad678a85b96fc5fecd60e3d3a31ca5ffc89d67f':
fate: Update ac3 test to the new request_channel_layout option
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '441e8ae5efd681055e5af6f4317fb60110de9dd0':
FATE: drop the last truncated frame from the wmapro tests
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'd3ea79e8a65ddad4da11813bb43c46701295f68c':
FATE: drop the last truncated frame from the wma lossless test
Conflicts:
tests/fate/lossless-audio.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Result differs in pkt_duration and time_base.den for some reason.
Right now it tests only one example (adjusted to match the output).
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'c0b105756f61d253bdabcc2bb49453a2557e7c3b':
txd: Use the TextureDSP module for decoding
Conflicts:
configure
libavcodec/s3tc.c
libavcodec/s3tc.h
libavcodec/txd.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Using the internal DXTC routines brings support for non multiple of 4
textures. A new test is added to cover this feature. Hashes differ
since the decoding algorithm is different, though no visual changes
have been spotted.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
* commit 'c060d046aa2f89c0e601a2dcfbce53f0e36cf498':
af_resample: Set the number of samples in the last frame
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '6ec688e1bc76dd93151cbca1c340162ae4b10d77':
mp3: enable packed main_data decoding in MP4
Conflicts:
libavcodec/mpegaudiodec_template.c
Only the parts needed to support the available sample are merged
the remaining error checks are left in place
Merged-by: Michael Niedermayer <michaelni@gmx.at>
or if no rematrix and no resampling is performed and the input is 16bit
note reampling and rematrix itself always use more than 16bit internally
the "internal" sampling format is the format between these steps
Its unlikely the difference from this commit is audible in any case
unless there is some bug either before or after the change.
but multiple people prefer this and it slightly improves the precission
of computations.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
14496-3 suggests packing main_data of MP3 that is usually scattered
into multiple frames due to bit reservoir.
However, after packing main_data into a access unit, bitrate index
in the MPEG audio frame header doesn't match with actual frame size.
In order to accept this, this patch removes unnecessary frame size
checking on mp3 decoder.
Also, mov demuxer was changed to use MP3 parser only on special cases
(QT MOV with specific sample description) to avoid re-packetizing.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* commit '063f7467e4d14ab7fe01b2845dab60cc75df8b53':
rtmpdh: Add fate test for the DH handshake routine
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'd81fb63d87692765c004c19934b49427df434a07':
fate: Add a PICT test
Conflicts:
tests/fate/image.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This will test properly CRLF with make fate, make fate-subtitles and any
make fate-sub-* test. Before this commit, the rawdiff was triggered only
by make fate-subtitles.
Also make sure fate-sub-* only match the tests relying on fmtstdout
command, to at least avoid failing on MingW. See
https://ffmpeg.org/pipermail/ffmpeg-devel/2015-April/172395.html
These could be kept, but they are not overly useful. The only thing they
had over the remaining mp3 gapless test was seeking, which was incorrect
in the toc test, and only by chance correct in the notoc test.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
It's the most useful mode, because it seeks accurately, and does not
break features like gapless audio.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
"-usetoc 2" now invokes the generic seek and indexing mode. This mode
skips data until the seek target is reached, and this is exact. It also
makes gapless audio actually work if a seek past the start of the file
is involved.
Change the fate-gapless-mp3 test to use the new mode, and move the old
one to fate-gapless-mp3-toc (since the test forces use of the Xing TOC).
The new mode has a different result for the seek - this result is
actually correct.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Some players do not support setting minimumUpdatePeriod to zero.
This patch adds a new parameter that will let the users set any
value to this field. Also updates the test and the documentation.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This patch adds support for creating DASH manifests for WebM Live
Streams. It also updates the documentation and adds a fate test to
verify the behavior of the new muxer flag.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'acbe15a99f158dbb0edb837fb6557171dc4376d4':
fate: Add test for DCA XLL
Conflicts:
tests/fate/audio.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Regression test for the bug from trac ticket #4359 fixed in commit efff3854
Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: James Almer <jamrial@gmail.com>
* commit '62139b14e621f096d0f8ed90920d042b92867e40':
fate: Specify the idct to use for the aic-oddsize test
Conflicts:
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is a small change, but it does have a big impact on bit allocation.
all the regressions marked in the report have no audible
difference (I didn't check them all though), but the improvements can
be heard.
This affects mostly high bit rates. It's related to issue #2686.
In the report, A is the patched version, B is unpatched, all
comparisons show deltas in the form (A-B), so a positive pSNR delta
means a better quality in the patched version, and negative a
regression. Regressions are only considered for pSNR deltas below
-1db, they're considered serious below -6db.
All measurements were done with tiny_psnr.
The summary of the report inline for quick reading:
Files: 58
Bitrates: 6
Tests: 347
Serious Regressions: 0 (0%)
Regressions: 10 (2%)
Improvements: 54 (15%)
Big improvements: 26 (7%)
Worst regression - sine_tester.flac - 384k
- StdDev: 1.68 pSNR: -3.05 maxdiff: -178.00
Best improvement - 07 - Bound.flac - 384k
- StdDev: -1700.05 pSNR: 20.64 maxdiff: -29595.00
Average - StdDev: -55.67 pSNR: 1.20 maxdiff: -1593.00
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
