From: Andriy Gelman
The fp_format option was incorrectly declared, meaning that
it could not be set on the cli (via recommended settings
raw/compressed/base64). This is fixed in the commit.
---
libavformat/chromaprint.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git
From: Andriy Gelman
As of commit 21b2442f in the chromaprint library, selecting "-algorithm 2" via
the ffmpeg cli creates a null pointer dereference. This can be replicated by:
./ffmpeg -f lavfi -i sine=d=20,asetnsamples=n=1000 -f chromaprint -algorithm 2 -
Until this issue is resolved, this
From: Andriy Gelman
Setting silence_threshold requires that -algorithm 3 option is also set.
Add this to the logging message.
---
libavformat/chromaprint.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/libavformat/chromaprint.c b/libavformat/chromaprint.c
index
From: Andriy Gelman
Silence detection can only be set with the -algorithm 3 option, so
update the documentation.
---
doc/muxers.texi | 10 ++
1 file changed, 6 insertions(+), 4 deletions(-)
diff --git a/doc/muxers.texi b/doc/muxers.texi
index b3da8bf12e..cad376aaeb 100644
---
From: Andriy Gelman
The pointer fp after the call to chromaprint_get_raw_fingerpoint() points to an
array of uint32_t whereas the current code assumed just a char stream. Thus
when writing the raw fingerprint, the output would be truncated by a factor of
4. This is fixed in the commit.
For
Fixes: left shift of 1 by 31 places cannot be represented in type 'int'.
Affected the FATE-tests vsynth1-cinepak, vsynth2-cinepak and
vsynth_lena-cinepak. Also fixes ticket #8220.
Signed-off-by: Andreas Rheinhardt
---
I am resending this, because in the meantime ticket #8220 has been
opened
The flac parser uses a fifo to buffer its data. Consequently, when
searching for sync codes of flac packets, one needs to take care of
the possibility of wraparound. This is done by using an optimized start
code search that works on each of the continuous buffers separately and
by explicitly
When flushing, MAX_FRAME_HEADER_SIZE bytes (always zero) are supposed to
be written to the fifo buffer in order to be able to check the rest of
the buffer for frame headers. It was intended to write these by writing
a small buffer of size MAX_FRAME_HEADER_SIZE to the buffer. But the way
it was
Signed-off-by: Andreas Rheinhardt
---
libavcodec/flac_parser.c | 35 +--
1 file changed, 17 insertions(+), 18 deletions(-)
diff --git a/libavcodec/flac_parser.c b/libavcodec/flac_parser.c
index 7ff7683c2e..9280246af2 100644
--- a/libavcodec/flac_parser.c
+++
FLAC sync codes contain a byte equal to 0xFF and so the function that
searches for sync codes first searched for this byte. It did this by
checking four bytes at once; these bytes have been read via AV_RB32, but
the test works just as well with native endianness.
Signed-off-by: Andreas Rheinhardt
The FLACHeaderMarker structure contained a pointer to an array of int;
said array was always allocated and freed at the same time as its
referencing FLACHeaderMarker; the pointer was never modified to point to
a different array and each FLACHeaderMarker had its own unique array.
Furthermore, all
For a parser, the input buffer is always != NULL: In case of flushing,
the indicated size of the input buffer will be zero and the input buffer
will point to a zeroed buffer of size 0 + AV_INPUT_BUFFER_PADDING.
Therefore one does not need to check for whether said buffer is NULL or
not.
Signed-off-by: Andreas Rheinhardt
---
libavcodec/flac_parser.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/libavcodec/flac_parser.c b/libavcodec/flac_parser.c
index 376ba2bcfc..7ff7683c2e 100644
--- a/libavcodec/flac_parser.c
+++ b/libavcodec/flac_parser.c
@@ -734,6 +734,7 @@ static void
Only decrement the number of buffered headers if a header has actually
been freed.
Signed-off-by: Andreas Rheinhardt
---
libavcodec/flac_parser.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/libavcodec/flac_parser.c b/libavcodec/flac_parser.c
index 197f234e61..8c61f3a88c 100644
---
Put an AVIOContext whose lifetime doesn't extend beyond the function
where it is allocated on the stack instead of allocating and freeing it.
Signed-off-by: Andreas Rheinhardt
---
libavformat/flac_picture.c | 10 +++---
1 file changed, 3 insertions(+), 7 deletions(-)
diff --git
Put an AVIOContext whose lifetime doesn't extend beyond the function where
it is allocated on the stack instead of allocating and freeing it.
Signed-off-by: Andreas Rheinhardt
---
libavformat/mpjpegdec.c | 10 +++---
1 file changed, 3 insertions(+), 7 deletions(-)
diff --git
Fix #7620
In the case tee muxer with both "bsf" and "use_fifo" parameters
wil trigger this bug. Tee muxer will first steal parameters (like "f",
"select"...) and then "use_fifo" will try reading out remaining options
and pass them to fifo as option "format_options".
Current code miss the part of
On 10/3/2019 8:52 PM, James Almer wrote:
> The rescaling can be done at muxing/encoding time, for formats that require
> it.
>
> Signed-off-by: James Almer
> ---
> libavformat/matroskadec.c | 33 ++---
> 1 file changed, 10 insertions(+), 23 deletions(-)
>
> diff
On 10/5/2019 6:41 PM, Michael Niedermayer wrote:
> Fixes: Timeout (17sec ->281ms)
> Fixes:
> 17833/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_H264_fuzzer-5638346914660352
>
> Found-by: continuous fuzzing process
> https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
>
Fixes: signed integer overflow: 2147483647 + 511 cannot be represented in type
'int'
Fixes:
17899/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TAK_fuzzer-5719753322135552
Found-by: continuous fuzzing process
https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by:
Fixes:
17886/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5728165124636672
Found-by: continuous fuzzing process
https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer
---
libavcodec/apedec.c | 4
1 file changed, 4 insertions(+)
Fixes: Timeout (22 -> 100 ms)
Fixes:
15173/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_HQX_fuzzer-5662556846292992
Fixes:
17896/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_HQX_fuzzer-5679312077848576
Found-by: continuous fuzzing process
Fixes: left shift of negative value -1
Fixes:
17890/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_RALF_fuzzer-5643307467669504
Found-by: continuous fuzzing process
https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer
---
libavcodec/ralf.c | 6
Fixes: Timeout (? -> 2sec)
Fixes:
17886/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5728165124636672
Found-by: continuous fuzzing process
https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer
---
libavcodec/apedec.c | 4
1 file
Fixes: Timeout (17sec ->281ms)
Fixes:
17833/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_H264_fuzzer-5638346914660352
Found-by: continuous fuzzing process
https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer
---
libavcodec/h2645_parse.c | 6
On Tue, Sep 24, 2019 at 4:31 PM Elliott Karpilovsky
wrote:
>
> Current default is 200kbps, which produces inconsistent
> results (too high for low-res, too low for hi-res). Use
> CRF instead, which will adapt. Affects VP9. Also have
> VP8 use a default bitrate of 256kbps.
> ---
>
On Thu, Oct 03, 2019 at 09:53:16 +0800, Jun Zhao wrote:
> From: Jun Zhao
> Subject: [PATCH V1 2/2] lavfi/hqdn3d: add slice thread optionmation
Nit: big typo in the first line of your commit message.
Moritz
___
ffmpeg-devel mailing list
On Thu, Oct 03, 2019 at 09:53:15AM +0800, Jun Zhao wrote:
> From: Mengye Lv
>
> When used ROUNDED_DIV(a,b), if a is unsigned integer zero, it's
> will lead to an underflow issue(it called unsigned integer
> wrapping).
>
> Fixes #8062
>
> Signed-off-by: Mengye Lv
> Signed-off-by: Jun Zhao
>
On Fri, Oct 04, 2019 at 09:36:54PM +0800, Jun Zhao wrote:
> From: Jun Zhao
>
> Correct the flags for AVCodecContext.flags2.
>
> Signed-off-by: Jun Zhao
> ---
> libavcodec/options_table.h |4 ++--
> 1 files changed, 2 insertions(+), 2 deletions(-)
LGTM
thx
[...]
--
Michael GnuPG
On 2019-10-04T10:04:46+0200, Paul B Mahol wrote:
> +Apply Normalized Least-Mean-Squares algorithm to first audio stream using
> second audio stream.
Apply Normalized Least-Mean-Squares algorithm to [the] first audio stream using
[the] second audio stream.
> +This is adaptive filter used to
From: Andriy Gelman
The unit member was incorrectly set to NULL for fp_format, meaning that
this option could not be set on the cli (via recommended settings -
raw/compressed/base64). This is fixed in the commit.
---
libavformat/chromaprint.c | 2 +-
1 file changed, 1 insertion(+), 1
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