From: Jan Sebechlebsky
This fixes ticket #5487 - mjpeg2jpeg bitstream filter causes
segmentation fault with header-less mjpeg.
Signed-off-by: Jan Sebechlebsky
---
libavcodec/mjpeg2jpeg_bsf.c | 4 +++-
1 file changed, 3 insertions(+), 1
On Sun, May 1, 2016 at 8:30 PM, James Almer wrote:
> With the samples you shared and with a random lbr-in-wav mono sample i found
> in the wild i get the following when i try to do a codec copy.
> Core and every other DTS extension in contrast seem to set timestamps just
>
From: Jan Sebechlebsky
This fixes ticket #5487 - mjpeg2jpeg bitstream filter causes
segmentation fault with header-less mjpeg.
Signed-off-by: Jan Sebechlebsky
---
libavcodec/mjpeg2jpeg_bsf.c | 2 ++
1 file changed, 2 insertions(+)
diff
From: Jan Sebechlebsky
TeeSlave.bsfs is array of pointers to AVBitStreamFilterContext,
so element size should be really size of a pointer, not size
of TeeSlave structure.
Signed-off-by: Jan Sebechlebsky
---
libavformat/tee.c | 2 +-
1 file
On 5/1/2016 12:42 PM, foo86 wrote:
> Also add actual speaker pair definitions.
> ---
> libavcodec/dca.h | 28
> libavcodec/dca_exss.c | 9 ++---
> 2 files changed, 30 insertions(+), 7 deletions(-)
Applied.
___
On 5/1/2016 12:41 PM, foo86 wrote:
> ---
> libavcodec/Makefile |2 +-
> libavcodec/dca_core.c | 70 +---
> libavcodec/dca_core.h |2 +-
> libavcodec/dcadata.c |9 -
> libavcodec/dcadata.h |2 -
> libavcodec/dcadec.c |4 +-
> libavcodec/dcahuff.c | 1099
>
From: Jan Sebechlebsky
Replace av_copy_packet and deprecated av_dup_packet by
creating reference using av_packet_ref.
Signed-off-by: Jan Sebechlebsky
---
This should be effectively the same as calling av_packet_clone,
but without dynamic
On 5/1/16, Christophe Gisquet wrote:
> Hi,
>
> 2016-05-01 15:33 GMT+02:00 Christophe Gisquet
> :
>> +fate-lossless-wma24-2: CMD = md5 -i
>> $(TARGET_SAMPLES)/lossless-audio/Mega_Weird_Audio_Test_24bit.wma -f s24le
>
> The recent fixes
Hi!
Attached patch stops setting bits_per_raw_sample if it makes no sense as for
example in the wmall24 -> pcm_s16 case:
Stream #0:0: Audio: pcm_s16le, 96000 Hz, stereo, s16 (24 bit), 3072 kb/s
Mostly tested with audio.
Please comment, Carl Eugen
diff --git a/ffmpeg.c b/ffmpeg.c
index
> > VIDC.m702=digivcap.dll Matrox Offline HD
It looks like I managed to create samples for this:
"This codec renders video to a proxy HD video format for video editing purposes.
Resolutions supported: 320x180 for 720p projects, and 480x270 for 1080i/p
projects"
but the problem is that there
> > > > > Improved version attached.
> >
> > I found another bug, this time in lossy mode - the file encoded as "joint
> > stereo" decodes after a few seconds into noise:
> >
> > https://www.datafilehost.com/d/c2e8b332
>
> here are a few more samples:
>
>
Hi
On Thu, Apr 28, 2016 at 08:44:50PM -0700, Todd Volkert wrote:
> Resolves https://trac.ffmpeg.org/ticket/4209
> ---
> libavformat/movenc.c | 2 ++
> 1 file changed, 2 insertions(+)
comments in movenc indicate that this style of metadata caused problems
with gtkpod / ipod
has this combination
On 5/1/2016 12:27 PM, foo86 wrote:
> This adds decoder for DTS Express (LBR) format that is typically used for
> secondary audio tracks on BDs.
>
> Changes since the previous version:
> - dropped merged patches
> - changed avpriv_ prefix to ff_ on inline function in a header
> - replaced
---
libavcodec/Makefile |2 +-
libavcodec/dca_core.c | 70 +---
libavcodec/dca_core.h |2 +-
libavcodec/dcadata.c |9 -
libavcodec/dcadata.h |2 -
libavcodec/dcadec.c |4 +-
libavcodec/dcahuff.c | 1099 +
Also add actual speaker pair definitions.
---
libavcodec/dca.h | 28
libavcodec/dca_exss.c | 9 ++---
2 files changed, 30 insertions(+), 7 deletions(-)
diff --git a/libavcodec/dca.h b/libavcodec/dca.h
index ccb02af..a1ac763 100644
--- a/libavcodec/dca.h
+++
2016-05-01 15:54 GMT+02:00 Paul B Mahol :
> There where 2 distinct issues: 32bit instead of 16bit integers and
> wrong handling of raw pcm.
> The 96k is about the first one, last decoded frame md5 differs for example.
Added a test for the file with raw pcm tiles then.
--
This adds decoder for DTS Express (LBR) format that is typically used for
secondary audio tracks on BDs.
Changes since the previous version:
- dropped merged patches
- changed avpriv_ prefix to ff_ on inline function in a header
- replaced LOCAL_ALIGNED(32, ...) with LOCAL_ALIGNED_32(...)
2016-05-01 15:33 GMT+02:00 Christophe Gisquet :
> This is done by actually handling the "prev_values" in the cascaded LMS data
> as if it were int16_t, thus requiring switching at various locations the
> computations.
Patch update since Michael's fix, which was
> > > Improved version attached.
>
> I found another bug, this time in lossy mode - the file encoded as "joint
> stereo" decodes after a few seconds into noise:
>
> https://www.datafilehost.com/d/c2e8b332
here are a few more samples:
https://www.datafilehost.com/d/e4825eb4
quantization: 1.15
On 5/1/2016 9:57 AM, Michael Niedermayer wrote:
> On Sun, May 01, 2016 at 12:24:06PM +0200, Carl Eugen Hoyos wrote:
>> Hi!
>>
>> Debian uses the following as a regression test for aac in mpegts:
>> $ ffmpeg -f lavfi -i sine=d=0.1 -acodec aac -strict -2 out.ts
>> $ ffmpeg -i out.ts
>> (possibly
Hi,
2016-05-01 15:33 GMT+02:00 Christophe Gisquet :
> +fate-lossless-wma24-2: CMD = md5 -i
> $(TARGET_SAMPLES)/lossless-audio/Mega_Weird_Audio_Test_24bit.wma -f s24le
The recent fixes actually changed the crc for that file.
Is
Hi!
Debian uses the following as a regression test for aac in mpegts:
$ ffmpeg -f lavfi -i sine=d=0.1 -acodec aac -strict -2 out.ts
$ ffmpeg -i out.ts
(possibly simplified)
This worked in 2.8 when the default aac bitrate was 128k and the
length of the output file 2632 bytes. Since f0a82124 the
Hi!
Debian uses the following as a regression test for aac in mpegts:
$ ffmpeg -f lavfi -i sine=d=0.1 -acodec aac -strict -2 out.ts
$ ffmpeg -i out.ts
(possibly simplified)
This worked in 2.8 when the default aac bitrate was 128k and the
length of the output file 2632 bytes. Since f0a82124 the
On 4/30/16, Christophe Gisquet wrote:
> Patch 2 is the squashing of several previous commits, as there were
> no opinion on their contents nor the way to go.
>
> The SSE4 one is the final version from its last thread.
>
> The last patch in this set is new, and
On Sat, Apr 30, 2016 at 10:17:33PM +0200, Marton Balint wrote:
> Signed-off-by: Marton Balint
> ---
> ffplay.c | 7 +++
> 1 file changed, 7 insertions(+)
I have a nagging feeling someone with better knowledge
of ALSA and how we handle it might find a better solution,
but
From: Thomas Volkert
---
Changelog | 2 +-
MAINTAINERS| 1 +
libavformat/Makefile | 1 +
libavformat/rtpenc.c | 15 +
libavformat/rtpenc.h | 1 +
libavformat/rtpenc_vc2hq.c | 134
On Sun, May 01, 2016 at 12:24:06PM +0200, Carl Eugen Hoyos wrote:
> Hi!
>
> Debian uses the following as a regression test for aac in mpegts:
> $ ffmpeg -f lavfi -i sine=d=0.1 -acodec aac -strict -2 out.ts
> $ ffmpeg -i out.ts
> (possibly simplified)
>
> This worked in 2.8 when the default aac
On 4/30/16, Christophe Gisquet wrote:
> 16bits samples with CDLMS orders of 8 are currently unsupported, but have
> never
> been encountered before.
>
> However, 8 seems to be the most frequent, if not the only order used for
> 24bits.
> In that case, the dsp
This is done by actually handling the "prev_values" in the cascaded LMS data
as if it were int16_t, thus requiring switching at various locations the
computations.
---
libavcodec/wmalosslessdec.c | 109 +++-
1 file changed, 58 insertions(+), 51 deletions(-)
The unique user so far is wmalossless 24bits. The few samples tested show an
order of 8, so more unrolling or an avx2 version do not make sense.
Timings: 68 -> 49 cycles
---
libavcodec/x86/lossless_audiodsp.asm| 33 +
libavcodec/x86/lossless_audiodsp_init.c |
Hi,
i found this codec, which was in adndroid vesrsion of Mirillis Action -
libmirillis-nativeaudio.so.
Maybe it will help for supporting of playing through this propriare mpg4 under
linux.
--
S pozdravom / mit freundlichen Grüßen / best regards
Ing. Attila Tóth
PhD student
Email adress :
Due to the changes to the cascaded LMS coefficients, most of the code
needed a rewrite.
In particular, the SSE4 madd32 code is no longer that similar to be
shared inside a macro.
Christophe Gisquet (4):
fate: wma: add lossless 24bits test
wmalossless: allow calling madd_int16
x86: lossless
The loops are guaranteed to be at least multiples of 8, so this
unrolling is safe but allows exploiting execution ports.
For int32 version: 68 -> 58c.
---
libavcodec/lossless_audiodsp.c | 12
1 file changed, 8 insertions(+), 4 deletions(-)
diff --git
On Sun, May 01, 2016 at 11:21:34AM +0200, Paul B Mahol wrote:
> On 4/30/16, Christophe Gisquet wrote:
> > ---
> > tests/fate/lossless-audio.mak | 5 -
> > tests/ref/fate/lossless-wma24-1 | 1 +
> > tests/ref/fate/lossless-wma24-2 | 1 +
> > 3 files changed, 6
On Sun, May 01, 2016 at 13:42:58 +0200, Thomas Volkert wrote:
> + "Please set -f_strict experimental in order to enable
> it.\n");
"-f_strict"?
"deprecated; use strict, save via avconv" is how it's documented (not
that I understand the last part).
Moritz
On Mon, May 02, 2016 at 02:51:25AM +0300, sebechlebsky...@gmail.com wrote:
> From: Jan Sebechlebsky
>
> This fixes ticket #5487 - mjpeg2jpeg bitstream filter causes
> segmentation fault with header-less mjpeg.
>
> Signed-off-by: Jan Sebechlebsky
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