fix ticket: 7305
vs->sequence - hls->start_sequence - vs->nb_entries is the
after_init_list_dur fragment numbers
fix the wrong compute way vs->sequence - vs->nb_entries
Signed-off-by: Steven Liu
---
libavformat/hlsenc.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/libavfor
Fixed using uninitialized value "global_work[0]" when calling "av_log".
---
libavfilter/vf_avgblur_opencl.c | 8
1 file changed, 4 insertions(+), 4 deletions(-)
diff --git a/libavfilter/vf_avgblur_opencl.c b/libavfilter/vf_avgblur_opencl.c
index d1d3eb1..772b14e 100644
--- a/libavfilter
Thank you for your comments Michael and apologies if my commit message was
inadequate (I am new to this forum and this is my first patch).
The bug can be reproduced when downmixing audio with more than 8 channels,
for example:
./ffmpeg -i input_9ch.wav -filter:a:0
pan="6c|c0=0.166*c0+0.166*c6|c1=
Rematrixing supports up to 64 channels but there is only a limited number of
channel layouts defined. Currently, in/out channel count is obtained from the
channel layout so if the channel layout is undefined (e.g. for 9, 10, 11
channels etc.) the in/out channel count will be 0 and the rematrixin
Marcin Gorzel (2018-07-09):
> Subject: Re: [FFmpeg-devel] [PATCH] Use channel count if channel layout is
> undefined
Please remember to mention "lswr" or "libswresample" in the first line
of the commit message.
> Rematrixing supports up to 64 channels but there is only a limited
> number of chan
On Tue, Jul 3, 2018 at 5:59 PM Michael Niedermayer
wrote:
>
> On Tue, Jul 03, 2018 at 12:14:19PM -0700, Jacob Trimble wrote:
> > On Mon, Jul 2, 2018 at 6:07 PM Michael Niedermayer
> > wrote:
> > >
> > > On Mon, Jun 25, 2018 at 04:03:32PM -0700, Jacob Trimble wrote:
> > > > Signed-off-by: Jacob Tr
Hi Nicolas,
> Please remember to mention "lswr" or "libswresample" in the first line
> of the commit message.
>
Apologies, I will update the commit message.
> > Rematrixing supports up to 64 channels but there is only a limited
> > number of channel layouts defined. Currently, in/out channel c
Marcin Gorzel (2018-07-09):
> ./ffmpeg -i input_9ch.wav -filter:a:0
> pan="6c|c0=0.166*c0+0.166*c6|c1=c1|c2=c2|c3=c3|c4=c4|c5=c5" -y
> output_6ch.wav
>
> Without the patch, the output in the first channel (c0) is noise. After
> applying the patch, I can verify that two sine tones are mixed togethe
Global header OBUs can be exported straight out of containers like
ISOBMFF and Matroska, but for the rest (IVF, AV1 Annex-B
encapsulation, and of course raw AV1 bitstreams) this is required.
Signed-off-by: James Almer
---
libavcodec/Makefile| 2 +-
libavcodec/av1.h
aom_codec_get_global_headers() is not implemented as of libaom 1.0.0
for AV1, so we're forced to extract the relevant header OBUs from the
first packet and propagate them as packet side data for now.
Signed-off-by: James Almer
---
This is far from ideal. Whereas the mp4 muxer can handle extradata
Signed-off-by: James Almer
---
I talked with Mark Thompson about AV1 parsing code in libavcodec and
he told me he prefered to keep everything within his upcoming CBS
implementation. I very much agree the eventual decoder and avparser
should use it given they will require deep bitstream parsing, bu
Signed-off-by: James Almer
---
libavformat/isom.c | 1 +
libavformat/mov.c | 31 +++
2 files changed, 32 insertions(+)
diff --git a/libavformat/isom.c b/libavformat/isom.c
index 2792371c25..ce66d1bcd4 100644
--- a/libavformat/isom.c
+++ b/libavformat/isom.c
@@ -188,
Signed-off-by: James Almer
---
ff_av1_filter_obus() could eventually be replaced by an autoinserted
filter_units bsf, assuming it doesn't slow down the muxing process
too much (CBS is fast reading packets, but not so much assembling and
writing packets).
ff_isom_write_av1c() however can't be repla
What the subject says.
There are a few comments in some of the patches as well.
James Almer (5):
avcodec: add AV1 packet split API
avcodec/extract_extradata: add support for AV1
avcodec/libaomenc: export Sequence Header and Metadata OBUs as
extradata
avformat/mov: add support for AV1
On Mon, Jul 09, 2018 at 01:55:37PM +0100, Marcin Gorzel wrote:
> Thank you for your comments Michael and apologies if my commit message was
> inadequate (I am new to this forum and this is my first patch).
>
> The bug can be reproduced when downmixing audio with more than 8 channels,
> for example
On 7/9/2018 1:29 AM, Book Moons wrote:
> Hello,
>
> I have a Power optimized function in asm for an upcoming patch. Having some
> trouble figuring out how to wire it up to the makefiles / build system. Are
> there any docs on how to do that?
>
> The only other example of Power asm seems to be l
On Mon, Jul 09, 2018 at 03:26:54PM -0300, James Almer wrote:
> Signed-off-by: James Almer
> ---
> ff_av1_filter_obus() could eventually be replaced by an autoinserted
> filter_units bsf, assuming it doesn't slow down the muxing process
> too much (CBS is fast reading packets, but not so much assem
On 7/9/2018 6:08 PM, Michael Niedermayer wrote:
> On Mon, Jul 09, 2018 at 03:26:54PM -0300, James Almer wrote:
>> Signed-off-by: James Almer
>> ---
>> ff_av1_filter_obus() could eventually be replaced by an autoinserted
>> filter_units bsf, assuming it doesn't slow down the muxing process
>> too m
On Sun, Jul 08, 2018 at 12:54:25AM +0200, Michael Niedermayer wrote:
> The version 1 needs the channel count and would divide by 0
> Fixes: division by 0
> Fixes: fpe_movenc.c_1108_1.ogg
> Fixes: fpe_movenc.c_1108_2.ogg
> Fixes: fpe_movenc.c_1108_3.wav
>
> Found-by: #CHEN HONGXU#
> Signed-off-by:
Hi James,
I passed along your wish for aom_codec_get_global_headers().
Tom created this bug for tracking:
https://bugs.chromium.org/p/aomedia/issues/detail?id=2012
On Mon, Jul 9, 2018 at 11:29 AM James Almer wrote:
> aom_codec_get_global_headers() is not implemented as of libaom 1.0.0
> for AV1
Hi James,
On Mon, Jul 9, 2018 at 11:26 AM, James Almer wrote:
[...]
> @@ -5438,7 +5461,7 @@ int ff_mov_write_packet(AVFormatContext *s,
> AVPacket *pkt)
> av_log(s, AV_LOG_WARNING, "pts has no value\n");
> pkt->pts = pkt->dts;
> }
> -if (pkt->dts != pkt->pts)
> +
On 7/9/2018 7:23 PM, Baptiste Coudurier wrote:
> Hi James,
>
> On Mon, Jul 9, 2018 at 11:26 AM, James Almer wrote:
>
> [...]
>
>
>> @@ -5438,7 +5461,7 @@ int ff_mov_write_packet(AVFormatContext *s,
>> AVPacket *pkt)
>> av_log(s, AV_LOG_WARNING, "pts has no value\n");
>> pkt->
On Tue, Jul 10, 2018 at 12:29 AM Baptiste Coudurier
wrote:
>
> Hi James,
>
> On Mon, Jul 9, 2018 at 11:26 AM, James Almer wrote:
>
> [...]
>
>
> > @@ -5438,7 +5461,7 @@ int ff_mov_write_packet(AVFormatContext *s,
> > AVPacket *pkt)
> > av_log(s, AV_LOG_WARNING, "pts has no value\n");
> >
On 7/9/2018 6:53 PM, Niki Bowe wrote:
> Hi James,
> I passed along your wish for aom_codec_get_global_headers().
> Tom created this bug for tracking:
> https://bugs.chromium.org/p/aomedia/issues/detail?id=2012
Thanks a lot! It would simplify the libaom wrapper a lot and actually
allow us to write
Cleaning the build folder did the trick. Hadn't even thought of it. Prior
builds had an exploratory C version. Appreciate it.
On July 9, 2018 4:49 PM, James Almer wrote:
> On 7/9/2018 1:29 AM, Book Moons wrote:
>
> > Hello,
> >
> > I have a Power optimized function in asm for an upcoming patc
On 09-07-2018 10:06 AM, Gyan Doshi wrote:
Will push tonight if no objections.
Pushed as 0bd48ab2d9e463b75ef52c0eaa0cc00c4c385cce
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Hi,
I suffered a problem that when I decoded the HEVC WPP video with
ffmpeg 4.0 with the option "-thread_type slice", there was something wrong
with the output stream. Please find the attachment for the screenshot.
This is the command: ffmpeg.exe -thread_type slice -threads 16 -i
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