This avoids waiting for a count to increase which will always be 0 and may
reduce the startup delay for affected streams (rare)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Apparently, some live streams can delete segments too early, maybe
because the client is too far behind. In this case, it's better to skip
the segment, instead of returning EOF. (Yes, the HLS demuxer actually
returns AVERROR_EOF if opening the segment returns a 404 HTTP error.)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '1336bb06c9fbf9a14765e9f78616f2aad4f3a45a':
configure: Simplify avisynth check
Conflicts:
configure
No change as check_lib2 was already used
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '247aa7af7d8197247c181e3fbfe8d93d75e41b29':
avisynth: Simplify shared library name construction
Conflicts:
libavformat/avisynth.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Removing a bunch of questionable hacks makes it work. These hacks
apparently try to make concatenated mp3s with Lame headers seekable,
which doesn't make too much sense anyway. The main change is that we
trust the Xing header file size field now (the same field is used for
seeking with Xing TOC). Note that a mp3 might contain an unknown number
of unsupported additional tags, so we can't reliably compute this size
manually.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Return appropriate error codes and propagate the error codes from
helper functions to the outer calls. Also fix a potential leak in
call to av_realloc.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
It's obsolete after the addition of the pkg-config check.
See http://comments.gmane.org/gmane.comp.video.ffmpeg.devel/191983 for the
relevant discussion
Reviewed-by: Hendrik Leppkes <h.leppkes@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This fixes an invalid read if end is 0:
band_end = ff_ac3_bin_to_band_tab[end-1] + 1;
Depending on what is before the array, this can cause stack smashing,
when band_end becomes too large.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If band->thr is 0.0f, the division is undefined, making norm_fac not a
number or infinity, which causes psy_band->threshold to become NaN.
This is passed on to other variables until it finally reaches
sce->sf_idx and is converted to an integer (-2147483648).
This causes a segmentation fault when it is used as array index.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Reviewed-by: Claudio Freire <klaussfreire@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
ac may be NULL and then accessing ac->avctx results in a segmentation fault.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes segmentation faults, when pic->linesize[0] is negative.
In that case 'line * pic->linesize[0] + pixel_ptr' is treated as
unsigned and wraps around.
This reverts commit 7d78a964.
The problem was introduced in commit f7e1367f, which should obsolete
that commit.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
VP80 fourcc are writed for all contexts (without ctx->codec_tag)
how to reproduce the issue:
1) Get any vp9 video (for example http://base-n.de/webm/out9.webm)
2) ffmpeg -i out9.webm -vcodec copy out9.ivf
3) out9.ivf have VP80 fourcc at ivf header
The proposed fix solves this issue
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The spec madandate both time_scale and num_units_in_tick greater than 0,
however since they are not essential for decoding, just ignore the whole
block and try to finish parsing the VUI.
Related to Ticket4445.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
This commit implements the perceptual noise substitution AAC extension. This is a proof of concept
implementation, and as such, is not enabled by default. This is the fourth revision of this patch,
made after some problems were noted out. Any changes made since the previous revisions have been indicated.
In order to extend the encoder to use an additional codebook, the array holding each codebook has been
modified with two additional entries - 13 for the NOISE_BT codebook and 12 which has a placeholder function.
The cost system was modified to skip the 12th entry using an array to map the input and outputs it has. It
also does not accept using the 13th codebook for any band which is not marked as containing noise, thereby
restricting its ability to arbitrarily choose it for bands. The use of arrays allows the system to be easily
extended to allow for intensity stereo encoding, which uses additional codebooks.
The 12th entry in the codebook function array points to a function which stops the execution of the program
by calling an assert with an always 'false' argument. It was pointed out in an email discussion with
Claudio Freire that having a 'NULL' entry can result in unexpected behaviour and could be used as
a security hole. There is no danger of this function being called during encoding due to the codebook maps introduced.
Another change from version 1 of the patch is the addition of an argument to the encoder, '-aac_pns' to
enable and disable the PNS. This currently defaults to disable the PNS, as it is experimental.
The switch will be removed in the future, when the algorithm to select noise bands has been improved.
The current algorithm simply compares the energy to the threshold (multiplied by a constant) to determine
noise, however the FFPsyBand structure contains other useful figures to determine which bands carry noise more accurately.
Some of the sample files provided triggered an assertion when the parameter to tune the threshold was set to
a value of '2.2'. Claudio Freire reported the problem's source could be in the range of the scalefactor
indices for noise and advised to measure the minimal index and clip anything above the maximum allowed
value. This has been implemented and all the files which used to trigger the asserion now encode without error.
The third revision of the problem also removes unneded variabes and comparisons. All of them were
redundant and were of little use for when the PNS implementation would be extended.
The fourth revision moved the clipping of the noise scalefactors outside the second loop of the two-loop
algorithm in order to prevent their redundant calculations. Also, freq_mult has been changed to a float
variable due to the fact that rounding errors can prove to be a problem at low frequencies.
Considerations were taken whether the entire expression could be evaluated inside the expression
, but in the end it was decided that it would be for the best if just the type of the variable were
to change. Claudio Freire reported the two problems. There is no change of functionality
(except for low sampling frequencies) so the spectral demonstrations at the end of this commit's message were not updated.
Finally, the way energy values are converted to scalefactor indices has changed since the first commit,
as per the suggestion of Claudio Freire. This may still have some drawbacks, but unlike the first commit
it works without having redundant offsets and outputs what the decoder expects to have, in terms of the
ranges of the scalefactor indices.
Some spectral comparisons: https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/Original.png (original),
https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS_NO.png (encoded without PNS),
https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS1.2.png (encoded with PNS, const = 1.2),
https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/Difference1.png (spectral difference).
The constant is the value which multiplies the threshold when it gets compared to the energy, larger
values means more noise will be substituded by PNS values. Example when const = 2.2:
https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS_2.2.png
Reviewed-by: Claudio Freire <klaussfreire@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Softfloat will be used in implementation of AAC fixed point decoder.
This change is needed in order to more easily integrate ffmpegs softfloat in
already developed algorithm for AAC.
Signed-off-by: Nedeljko Babic <nedeljko.babic@imgtec.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
