On 2018/1/3 7:12, Mark Thompson wrote:
> This allows implementing sources as well as filters.
> ---
> libavfilter/opencl.c | 39 +--
> 1 file changed, 33 insertions(+), 6 deletions(-)
>
> diff --git a/libavfilter/opencl.c b/libavfilter/opencl.c
> index 005ad08
maybe you should return 1, not return ret
On Wed, Jan 3, 2018 at 12:54 PM, wrote:
> From: Jiejun Zhang
>
> This fixes #6940
>
> Although undocumented, AudioToolbox seems to require the data supplied
> by the callback (i.e. ffat_encode_callback) being unchanged until the
> next time the callbac
On Wed, Jan 3, 2018 at 1:00 PM, Aman Gupta wrote:
> On Tue, Jan 2, 2018 at 7:27 PM Wu Zhiqiang wrote:
>
> > On Wed, Jan 3, 2018 at 2:03 AM, Aman Gupta wrote:
> >
> > > On Tue, Jan 2, 2018 at 3:05 AM Steven Liu
> > wrote:
> > >
> > > > 2018-01-02 18:28 GMT+08:00 :
> > > > > From: Wu Zhiqiang
On Tue, Jan 2, 2018 at 7:27 PM Wu Zhiqiang wrote:
> On Wed, Jan 3, 2018 at 2:03 AM, Aman Gupta wrote:
>
> > On Tue, Jan 2, 2018 at 3:05 AM Steven Liu
> wrote:
> >
> > > 2018-01-02 18:28 GMT+08:00 :
> > > > From: Wu Zhiqiang
> > > >
> > > > HLS demuxer seeking use dts instead of pts.
> > > > D
From: Jiejun Zhang
This fixes #6940
Although undocumented, AudioToolbox seems to require the data supplied
by the callback (i.e. ffat_encode_callback) being unchanged until the
next time the callback is called. In the old implementation, the
AVBuffer backing the frame is recycled after the frame
On 1/3/2018 1:02 AM, Jiejun Zhang wrote:
> On Wed, Jan 3, 2018 at 10:02 AM, James Almer wrote:
>> On 1/2/2018 1:24 PM, zhangjiejun1...@gmail.com wrote:
>>> From: Jiejun Zhang
>>>
>>> This fixes #6940
>>>
>>> Although undocumented, AudioToolbox seems to require the data supplied
>>> by the callbac
On Wed, Jan 3, 2018 at 10:02 AM, James Almer wrote:
> On 1/2/2018 1:24 PM, zhangjiejun1...@gmail.com wrote:
>> From: Jiejun Zhang
>>
>> This fixes #6940
>>
>> Although undocumented, AudioToolbox seems to require the data supplied
>> by the callback (i.e. ffat_encode_callback) being unchanged unti
From: Wu Zhiqiang
HLS demuxer seeking use dts instead of pts.
Demuxer skip some frame when dts is before pts in special case.
And it is impossible to re-seek back to start time after playing.
---
libavformat/hls.c | 18 ++
1 file changed, 14 insertions(+), 4 deletions(-)
diff --
From: Wu Zhiqiang
Calculate first_timestamp based on first packet timestamp.
Some m3u8 have streams that second one has smaller timestamp
in first packet of this stream.
Start/seek from start time may fail due to EIO error.
It should be based on start_time of AvFormatContext.
Signed-off-by: Wu Z
On Wed, Jan 3, 2018 at 2:03 AM, Aman Gupta wrote:
> On Tue, Jan 2, 2018 at 3:05 AM Steven Liu wrote:
>
> > 2018-01-02 18:28 GMT+08:00 :
> > > From: Wu Zhiqiang
> > >
> > > HLS demuxer seeking use dts instead of pts.
> > > Demuxer skip some frame when dts is before pts in special case.
> > > An
On 1/2/2018 1:24 PM, zhangjiejun1...@gmail.com wrote:
> From: Jiejun Zhang
>
> This fixes #6940
>
> Although undocumented, AudioToolbox seems to require the data supplied
> by the callback (i.e. ffat_encode_callback) being unchanged until the
> next time the callback is called. In the old implem
On 2 January 2018 at 22:34, Michael Niedermayer
wrote:
> Signed-off-by: Michael Niedermayer
> ---
> libavcodec/opus.c | 4 ++--
> 1 file changed, 2 insertions(+), 2 deletions(-)
>
> diff --git a/libavcodec/opus.c b/libavcodec/opus.c
> index 9cbf4aed92..d00a17a7f1 100644
> --- a/libavcodec/opus.
---
libavdevice/alldevices.c | 6 +-
libavformat/allformats.c | 44 +++-
libavformat/avformat.h | 4
3 files changed, 48 insertions(+), 6 deletions(-)
diff --git a/libavdevice/alldevices.c b/libavdevice/alldevices.c
index 6b2a512..041eb85 10064
---
configure| 27 ++--
libavdevice/alldevices.c | 172 +++
libavdevice/avdevice.c | 46 -
libavdevice/avdevice.h | 28
libavdevice/version.h| 4 ++
5 files changed, 184 insertions(+), 93 deletions(-)
This fixes the fate-fifo-muxer test.
---
libavformat/Makefile | 2 +-
libavformat/allformats.c | 1 +
libavformat/fifo_test.c| 150 +
libavformat/tests/fifo_muxer.c | 115 +--
4 files changed, 154 inser
---
libavformat/allformats.c | 4 --
libavformat/rdt.c| 8 +--
libavformat/rdt.h| 3 +
libavformat/rtpdec.c | 157 ++-
libavformat/rtpdec.h | 29 -
libavformat/version.h| 4 +-
6 files changed, 136 insertions(
---
configure| 6 +-
doc/APIchanges | 5 +
libavformat/allformats.c | 870 ---
libavformat/avformat.h | 31 ++
libavformat/format.c | 56 +--
libavformat/version.h| 3 +
6 files changed, 564 insertions(+), 407
Also replace linked list with an array.
---
configure | 12 +-
doc/APIchanges |4 +
libavcodec/allcodecs.c | 1473
libavcodec/avcodec.h | 31 +
libavcodec/parser.c| 87 ++-
libavcodec/utils.c | 105
lib
On Tue, 2 Jan 2018, Michael Niedermayer wrote:
On Sun, Dec 10, 2017 at 11:11:18PM +0100, Marton Balint wrote:
Signed-off-by: Marton Balint
---
libavfilter/vf_framerate.c | 54 ++
1 file changed, 7 insertions(+), 47 deletions(-)
This broke fate-fi
This allows implementing sources as well as filters.
---
libavfilter/opencl.c | 39 +--
1 file changed, 33 insertions(+), 6 deletions(-)
diff --git a/libavfilter/opencl.c b/libavfilter/opencl.c
index 005ad089e2..37afc41f8b 100644
--- a/libavfilter/opencl.c
+++
---
E.g. with Beignet + i965:
./ffmpeg_g -y -init_hw_device vaapi=va:/dev/dri/renderD129 -init_hw_device
opencl=ocl@va -hwaccel vaapi -hwaccel_output_format vaapi -i in.mp4 -an
-filter_hw_device ocl -filter_complex '[0:v]hwmap[i1];
openclsrc=source=test.cl:kernel=ramp:w=1920:h=1080:format=nv12:
On 1/2/2018 10:34 PM, Michael Niedermayer wrote:
> Signed-off-by: Michael Niedermayer
> ---
> libavcodec/opus.c | 4 ++--
> 1 file changed, 2 insertions(+), 2 deletions(-)
Looks reasonable.
- Derek
___
ffmpeg-devel mailing list
ffmpeg-devel@ffmpeg.org
Signed-off-by: Michael Niedermayer
---
libavcodec/opus.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/libavcodec/opus.c b/libavcodec/opus.c
index 9cbf4aed92..d00a17a7f1 100644
--- a/libavcodec/opus.c
+++ b/libavcodec/opus.c
@@ -566,12 +566,12 @@ void ff_celt_bitalloc(Ce
On 1/2/2018 10:16 PM, Clément Bœsch wrote:
> I don't think you'll be much off by always assuming ENOMEM here when
> getting a NULL out frame, I think that's the common idiom when a function
> supposed to return allocated stuff returns NULL.
>
> But that's not very important, feel free to push as
On Tue, Jan 02, 2018 at 10:02:25PM +, Derek Buitenhuis wrote:
> On 1/2/2018 9:52 PM, Clément Bœsch wrote:
> > not exactly sure why you haven't just checked if `out` wasn't NULL, but it
> > should be fine that way too if you prefer it.
> >
> > I guess that's only to provide a finer grain error
On 1/2/2018 9:52 PM, Clément Bœsch wrote:
> not exactly sure why you haven't just checked if `out` wasn't NULL, but it
> should be fine that way too if you prefer it.
>
> I guess that's only to provide a finer grain error handling? It would make
> sense if ff_get_video_buffer was returning a meani
On 1/2/2018 9:53 PM, Clément Bœsch wrote:
> That's some weird ownership semantic for the error-path, but Nicolas knows
> better this API so I'll trust him on this one.
Yeah it was weird for me to, but I looked into its implementation to find out.
- Derek
__
On Mon, Jan 01, 2018 at 11:28:36AM -0500, Derek Buitenhuis wrote:
> This fixes a segfault caused by passing NULL to ff_filter_frame
> when an error occurs.
>
> Signed-off-by: Derek Buitenhuis
> ---
> libavfilter/vf_paletteuse.c | 25 -
> 1 file changed, 16 insertions(+),
On Mon, Jan 01, 2018 at 11:28:37AM -0500, Derek Buitenhuis wrote:
> This fixes a double free in he error case.
>
> Signed-off-by: Derek Buitenhuis
> ---
> This does fix the double free, but I am unsure if it is the correct free
> to removed to fix it. Comments welcome.
> ---
> libavfilter/vf_pal
On Mon, Jan 01, 2018 at 08:32:22PM +1100, mi...@brew.sh wrote:
> From: Maxim Poliakovski
>
> Changes by Misty De Meo :
>
> atrac3plus_parser: remove return statements for invalid data
>
> atrac3plus_parser: use libavcodec's oma
>
> atrac3plus_parser: pass along unexpected data unaltered
>
> a
On Sun, Dec 31, 2017 at 05:46:05PM +0800, mi...@brew.sh wrote:
> From: Misty De Meo
>
> MPS files are MPEG files used on PSP Video discs. They lack
> the PSMF header used by .pms files, and so the special casing
> in the original patch fails to support their audio. This patch
> fixes this by unco
On Tue, 2 Jan 2018 19:41:39 +0100
"Cyber Sinh" wrote:
> Nobody has an idea?
>
> Thanks.
>
> Checking AVMEDIA_TYPE seems to be the best way to split audio and video
> codecs.
> But what is the best way to exclude demuxers and parsers which have no sense
> for audio (because they are intended
On Sun, Dec 31, 2017 at 05:46:02PM +0800, mi...@brew.sh wrote:
> From: Misty De Meo
>
> ---
> tests/fate/atrac.mak | 4
> 1 file changed, 4 insertions(+)
>
> diff --git a/tests/fate/atrac.mak b/tests/fate/atrac.mak
> index acf79a539c..1707373890 100644
> --- a/tests/fate/atrac.mak
> +++ b/
On Mon, Jan 01, 2018 at 08:29:31PM +1100, mi...@brew.sh wrote:
> From: Misty De Meo
>
> ---
> tests/Makefile | 1 +
> tests/fate/psmf.mak | 23 +++
> 2 files changed, 24 insertions(+)
> create mode 100644 tests/fate/psmf.mak
LGTM & works
thx
[...]
--
Michael Gn
Hi,
On Tue, Jan 2, 2018 at 1:41 PM, Cyber Sinh wrote:
> Nobody has an idea?
Please don't top-post.
I think you know how to do codecs. As for muxers, that's up to you, there's
no generic way and I don't think we'd want one on our side. Whether you
generate the enable/disable list for audio-onl
Nobody has an idea?
Thanks.
-Message d'origine-
De : ffmpeg-devel [mailto:ffmpeg-devel-boun...@ffmpeg.org] De la part de Cyber
Sinh
Envoyé : samedi 30 décembre 2017 21:03
À : 'FFmpeg development discussions and patches'
Objet : Re: [FFmpeg-devel] Global options to compile FFmpeg with on
On Tue, Jan 2, 2018 at 3:05 AM Steven Liu wrote:
> 2018-01-02 18:28 GMT+08:00 :
> > From: Wu Zhiqiang
> >
> > HLS demuxer seeking use dts instead of pts.
> > Demuxer skip some frame when dts is before pts in special case.
> > And it is impossible to re-seek back to start time after playing.
> >
On 1/2/2018 2:48 PM, Rostislav Pehlivanov wrote:
> On 2 January 2018 at 17:01, wm4 wrote:
>
>> On Tue, 2 Jan 2018 16:47:14 +
>> Rostislav Pehlivanov wrote:
>>
>>> On 30 December 2017 at 14:59, Rostislav Pehlivanov
>>> wrote:
>>>
Deprecate the entire library. Merged years ago to provide
On Wed, Dec 20, 2017 at 4:31 PM, wm4 wrote:
> On Wed, 20 Dec 2017 15:10:43 -0800
> Jacob Trimble wrote:
>
>> From 1508d19e9f7acf43d76010ce54d59ff204613601 Mon Sep 17 00:00:00 2001
>> From: Jacob Trimble
>> Date: Tue, 5 Dec 2017 14:52:22 -0800
>> Subject: [PATCH] avcodec/avcodec.h: Add encryption
On 2 January 2018 at 17:01, wm4 wrote:
> On Tue, 2 Jan 2018 16:47:14 +
> Rostislav Pehlivanov wrote:
>
> > On 30 December 2017 at 14:59, Rostislav Pehlivanov
> > wrote:
> >
> > > Deprecate the entire library. Merged years ago to provide compatibility
> > > with Libav, it remained unmaintain
On 1/2/2018 4:45 PM, Amit Gandhi wrote:
> Does anyone happen to know why inband FEC option is not supported by libopus
> wrapper in ffmpeg while the packet_loss option is supported?
Probably no other reason than the fact that nobody mapped it to
an AVOption yet.
- Derek
_
Hi,
Does anyone happen to know why inband FEC option is not supported by libopus
wrapper in ffmpeg while the packet_loss option is supported?
Looking into the code of encoder wrapper ~/libavcodec/libopusenc.c the line
ret = opus_multistream_encoder_ctl(enc,
OPUS_SET_PACKET_LOSS_PERC(opts->packe
From: Author Name
Apple's AIFF protocol clearly states that each chucnk which is odd sized a
padding should be added.
In the old version of aiffdec adding of padding was done in `get_meta`. And in
case of unknown chunk name it was done in defalut case.
The new version has deleted the padding in
On Tue, 2 Jan 2018 16:47:14 +
Rostislav Pehlivanov wrote:
> On 30 December 2017 at 14:59, Rostislav Pehlivanov
> wrote:
>
> > Deprecate the entire library. Merged years ago to provide compatibility
> > with Libav, it remained unmaintained by the FFmpeg project and duplicated
> > functionali
On 30 December 2017 at 14:59, Rostislav Pehlivanov
wrote:
> Deprecate the entire library. Merged years ago to provide compatibility
> with Libav, it remained unmaintained by the FFmpeg project and duplicated
> functionality provided by libswresample.
>
> In order to improve consistency and reduce
On 2 January 2018 at 16:18, Paul B Mahol wrote:
> Signed-off-by: Paul B Mahol
> ---
> doc/filters.texi | 14 +++
> libavfilter/Makefile | 1 +
> libavfilter/af_aiir.c| 232 ++
> +
> libavfilter/allfilters.c | 1 +
> 4 files change
From: Jiejun Zhang
This fixes #6940
Although undocumented, AudioToolbox seems to require the data supplied
by the callback (i.e. ffat_encode_callback) being unchanged until the
next time the callback is called. In the old implementation, the
AVBuffer backing the frame is recycled after the frame
It makes no sense to return an error after the first reconnect, and then
somehow resume the next time it's called. Usually this will lead to
demuxer errors. Make reconnecting block instead, until it has either
successfully reconnected, or given up.
Also make the wait reasonably interruptible. Sinc
>
> Can't you instead create a new reference for the frame buffer? Or will
> making it non writable break things further into the process? It would
> save you a memcpy per frame.
Great idea. It works. Making it non-writable should be enough. I'm
submitting v3. Please take a look.
_
---
libavformat/http.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/libavformat/http.c b/libavformat/http.c
index 5eff87f8bb..eb029e33a0 100644
--- a/libavformat/http.c
+++ b/libavformat/http.c
@@ -1455,7 +1455,7 @@ static int http_read_stream(URLContext *h, uint8_t *buf,
i
Signed-off-by: Paul B Mahol
---
doc/filters.texi | 14 +++
libavfilter/Makefile | 1 +
libavfilter/af_aiir.c| 232 +++
libavfilter/allfilters.c | 1 +
4 files changed, 248 insertions(+)
create mode 100644 libavfilter/af_aiir.c
dif
On 1/2/2018 12:03 PM, zhangjiejun1...@gmail.com wrote:
> From: Jiejun Zhang
>
> This fixes #6940
>
> Although undocumented, AudioToolbox seems to require the data supplied
> by the callback (i.e. ffat_encode_callback) being unchanged until the
> next time the callback is called. In the old imple
From: Jiejun Zhang
This fixes #6940
Although undocumented, AudioToolbox seems to require the data supplied
by the callback (i.e. ffat_encode_callback) being unchanged until the
next time the callback is called. In the old implementation, the
AVBuffer backing the frame is recycled after the frame
On Tue, Jan 2, 2018 at 10:37 PM, wm4 wrote:
> On Tue, 2 Jan 2018 22:27:49 +0800
> Jiejun Zhang wrote:
>
>> On Tue, Jan 2, 2018 at 8:03 PM, Carl Eugen Hoyos wrote:
>> > 2018-01-02 8:52 GMT+01:00 :
>> >
>> >> @@ -565,6 +579,10 @@ static av_cold int ffat_close_encoder(AVCodecContext
>> >> *avctx)
what is the mean: min_size + min_size / 16 + 32
On Tue, Jan 2, 2018 at 9:00 AM, Carl Eugen Hoyos wrote:
> 2018-01-02 1:14 GMT+01:00 Michael Niedermayer :
> > On Mon, Jan 01, 2018 at 11:10:57PM +0100, Carl Eugen Hoyos wrote:
> >> Hi!
> >>
> >> Similar reason as last mem.c patch: av_fast_realloc
On Tue, 2 Jan 2018 22:27:49 +0800
Jiejun Zhang wrote:
> On Tue, Jan 2, 2018 at 8:03 PM, Carl Eugen Hoyos wrote:
> > 2018-01-02 8:52 GMT+01:00 :
> >
> >> @@ -565,6 +579,10 @@ static av_cold int ffat_close_encoder(AVCodecContext
> >> *avctx)
> >> ff_bufqueue_discard_all(&at->frame_queue);
On Tue, Jan 2, 2018 at 8:03 PM, Carl Eugen Hoyos wrote:
> 2018-01-02 8:52 GMT+01:00 :
>
>> @@ -565,6 +579,10 @@ static av_cold int ffat_close_encoder(AVCodecContext
>> *avctx)
>> ff_bufqueue_discard_all(&at->frame_queue);
>> ff_bufqueue_discard_all(&at->used_frame_queue);
>> ff_af
Oh, very true, I must have missed that -- thank you!
On 2 January 2018 at 14:08, James Almer wrote:
> On 1/2/2018 10:59 AM, Thomas Köppe wrote:
> > Hello,
> >
> > Friendly ping, can I interest you in this patch? It seems like it fixes
> an
> > oversight that would be annoying for anyone trying t
On 1/2/2018 10:59 AM, Thomas Köppe wrote:
> Hello,
>
> Friendly ping, can I interest you in this patch? It seems like it fixes an
> oversight that would be annoying for anyone trying to build FFMPEG for
> DJGPP.
>
> Thanks!
This seems to have been applied already. See commit
53c492640c6b46907157
Hello,
Friendly ping, can I interest you in this patch? It seems like it fixes an
oversight that would be annoying for anyone trying to build FFMPEG for
DJGPP.
Thanks!
On 14 November 2017 at 18:26, Thomas Köppe wrote:
> The macro was added in 43171a2a738f5114768d34a7278e56e5fde714bc, but I
> f
Signed-off-by: Nicolas George
---
libavfilter/framesync.h | 3 +++
1 file changed, 3 insertions(+)
diff --git a/libavfilter/framesync.h b/libavfilter/framesync.h
index 9fdc4d1ae2..abf3bf552b 100644
--- a/libavfilter/framesync.h
+++ b/libavfilter/framesync.h
@@ -286,6 +286,9 @@ int ff_framesync_i
Signed-off-by: Nicolas George
---
libavfilter/framesync.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
Not actually experienced a problem, but similar to Derek's fix.
diff --git a/libavfilter/framesync.c b/libavfilter/framesync.c
index 82d715750c..da12c58a61 100644
--- a/libavfilter/
Derek Buitenhuis (2018-01-01):
> This fixes a double free in he error case.
>
> Signed-off-by: Derek Buitenhuis
> ---
> This does fix the double free, but I am unsure if it is the correct free
> to removed to fix it. Comments welcome.
> ---
> libavfilter/vf_paletteuse.c | 1 -
> 1 file changed,
On Tue, 2 Jan 2018 03:55:18 +0100
Michael Niedermayer wrote:
> On Tue, Dec 26, 2017 at 01:50:08AM +, wm4 wrote:
> > ffmpeg | branch: master | wm4 | Thu Dec 21 20:23:14
> > 2017 +0100| [9b121dfc32810250938021952aab4172a988cb56] | committer: wm4
> >
> > w32pthreads: always use Vista+ API, dr
On Tue, Jan 02, 2018 at 02:14:15PM +0100, Hendrik Leppkes wrote:
> On Tue, Jan 2, 2018 at 2:05 PM, Michael Niedermayer
> wrote:
> > On Tue, Jan 02, 2018 at 10:36:44AM +0100, Hendrik Leppkes wrote:
> >> On Tue, Jan 2, 2018 at 4:28 AM, Michael Niedermayer
> >> wrote:
> >> > On Tue, Jan 02, 2018 at
On Tue, Jan 2, 2018 at 2:05 PM, Michael Niedermayer
wrote:
> On Tue, Jan 02, 2018 at 10:36:44AM +0100, Hendrik Leppkes wrote:
>> On Tue, Jan 2, 2018 at 4:28 AM, Michael Niedermayer
>> wrote:
>> > On Tue, Jan 02, 2018 at 03:58:14AM +0100, Michael Niedermayer wrote:
>> >> On Tue, Jan 02, 2018 at 03
On Tue, Jan 02, 2018 at 10:36:44AM +0100, Hendrik Leppkes wrote:
> On Tue, Jan 2, 2018 at 4:28 AM, Michael Niedermayer
> wrote:
> > On Tue, Jan 02, 2018 at 03:58:14AM +0100, Michael Niedermayer wrote:
> >> On Tue, Jan 02, 2018 at 03:55:18AM +0100, Michael Niedermayer wrote:
> >> > On Tue, Dec 26,
On 02/01/18 11:27, wm4 wrote:
> On Mon, 1 Jan 2018 23:35:42 +
> Mark Thompson wrote:
>
>> On 29/12/17 07:06, Zhong Li wrote:
>>> Add the flag "AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX" to indicate
>>> AVCodecContext.hw_device_ctx should be set before calling
>>> avcodec_open2() for qsv decodin
On 02/01/18 09:15, Li, Zhong wrote:
>> From: ffmpeg-devel [mailto:ffmpeg-devel-boun...@ffmpeg.org] On Behalf
>> Of Mark Thompson
>> Sent: Tuesday, January 2, 2018 7:36 AM
>> To: ffmpeg-devel@ffmpeg.org
>> Subject: Re: [FFmpeg-devel] [PATCH] lavc/qsvdec: hw device should be set
>>
>> On 29/12/17 07:
2018-01-02 8:52 GMT+01:00 :
> @@ -565,6 +579,10 @@ static av_cold int ffat_close_encoder(AVCodecContext
> *avctx)
> ff_bufqueue_discard_all(&at->frame_queue);
> ff_bufqueue_discard_all(&at->used_frame_queue);
> ff_af_queue_close(&at->afq);
> +if (at->audio_data_buf_size > 0) {
2018-01-02 16:59 GMT+08:00 :
> From: Jiejun Zhang
>
> This fixes #6940
> ---
> libavcodec/audiotoolboxenc.c | 34 +-
> 1 file changed, 29 insertions(+), 5 deletions(-)
>
> diff --git a/libavcodec/audiotoolboxenc.c b/libavcodec/audiotoolboxenc.c
> index 71885d1530.
On Mon, 1 Jan 2018 23:35:42 +
Mark Thompson wrote:
> On 29/12/17 07:06, Zhong Li wrote:
> > Add the flag "AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX" to indicate
> > AVCodecContext.hw_device_ctx should be set before calling
> > avcodec_open2() for qsv decoding.
> > It is consistent with examples
2018-01-02 18:28 GMT+08:00 :
> From: Wu Zhiqiang
>
> HLS demuxer seeking use dts instead of pts.
> Demuxer skip some frame when dts is before pts in special case.
> And it is impossible to re-seek back to start time after playing.
> ---
> libavformat/hls.c | 18 ++
> 1 file chang
From: Wu Zhiqiang
HLS demuxer seeking use dts instead of pts.
Demuxer skip some frame when dts is before pts in special case.
And it is impossible to re-seek back to start time after playing.
---
libavformat/hls.c | 18 ++
1 file changed, 14 insertions(+), 4 deletions(-)
diff --
On Tue, Jan 2, 2018 at 10:47 AM, Nicolas George wrote:
> Hendrik Leppkes (2018-01-02):
>> I don't think (old) Wine is something we should worry too much about,
>> considering you can just use native Linux builds, I doubt any real
>> usage occurs for FFmpeg through Wine.
>
> What about proprietary
Am 29.12.2017 um 19:12 schrieb Devin Heitmueller:
> To support the existing use case where multi-channel audio can be
> captured (i.e. 7.1)
Just to be clear, the current use case is NOT to capture multi-channel
audio like 7.1. It's just to capture all of the mono SDI channels into
one FFmpeg-inter
Hendrik Leppkes (2018-01-02):
> I don't think (old) Wine is something we should worry too much about,
> considering you can just use native Linux builds, I doubt any real
> usage occurs for FFmpeg through Wine.
What about proprietary applications that use FFmpeg (either by calling
the executable o
On Tue, Jan 2, 2018 at 4:28 AM, Michael Niedermayer
wrote:
> On Tue, Jan 02, 2018 at 03:58:14AM +0100, Michael Niedermayer wrote:
>> On Tue, Jan 02, 2018 at 03:55:18AM +0100, Michael Niedermayer wrote:
>> > On Tue, Dec 26, 2017 at 01:50:08AM +, wm4 wrote:
>> > > ffmpeg | branch: master | wm4
> Am 28.12.2017 um 19:20 schrieb Michael Niedermayer :
>>
>> +av_image_copy_to_buffer(pkt.data, pkt_buffer_size,
>> +(const uint8_t * const *) image_plane_data,
>> +image_linestrides, ctx->image_format,
>> +c
> From: ffmpeg-devel [mailto:ffmpeg-devel-boun...@ffmpeg.org] On Behalf
> Of Mark Thompson
> Sent: Tuesday, January 2, 2018 7:36 AM
> To: ffmpeg-devel@ffmpeg.org
> Subject: Re: [FFmpeg-devel] [PATCH] lavc/qsvdec: hw device should be set
>
> On 29/12/17 07:06, Zhong Li wrote:
> > Add the flag "AV_C
From: Jiejun Zhang
This fixes #6940
---
libavcodec/audiotoolboxenc.c | 34 +-
1 file changed, 29 insertions(+), 5 deletions(-)
diff --git a/libavcodec/audiotoolboxenc.c b/libavcodec/audiotoolboxenc.c
index 71885d1530..0c1e5feadc 100644
--- a/libavcodec/audiotoolb
From: Author Name
Apple's AIFF protocol clearly states that each chucnk which is odd sized a
padding should be added.
In the old version of aiffdec adding of padding was done in `get_meta`. And in
case of unknown chunk name it was done in defalut case.
The new version has deleted the padding in
On 1/2/18 1:41 PM, 刘歧 wrote:
>
>> On 2 Jan 2018, at 14:48, Karthick J wrote:
>>
>> From: Karthick Jeyapal
>>
>> ---
>> libavformat/dashenc.c | 11 +++
>> 1 file changed, 11 insertions(+)
>>
>> diff --git a/libavformat/dashenc.c b/libavformat/dashenc.c
>> index 3345b89..c4c112b 100644
>>
From: Karthick Jeyapal
---
libavformat/dashenc.c | 4
1 file changed, 4 insertions(+)
diff --git a/libavformat/dashenc.c b/libavformat/dashenc.c
index 3345b89..2e4ff67 100644
--- a/libavformat/dashenc.c
+++ b/libavformat/dashenc.c
@@ -1,6 +1,7 @@
/*
* MPEG-DASH ISO BMFF segmenter
* Co
From: Karthick Jeyapal
Currently http end of chunk is signalled implicitly in dashenc_io_open().
This mean playlists http writes would have to wait upto a segment duration to
signal end of chunk causing delays.
This patch will fix that problem and improve performance.
---
libavformat/dashenc.c
From: Author Name
Apple's AIFF protocol clearly states that each chucnk which is odd sized a
padding should be added.
In the old version of aiffdec adding of padding was done in `get_meta`. And in
case of unknown chunk name it was done in defalut case.
The new version has deleted the padding in
> On 2 Jan 2018, at 14:48, Karthick J wrote:
>
> From: Karthick Jeyapal
>
> ---
> libavformat/dashenc.c | 11 +++
> 1 file changed, 11 insertions(+)
>
> diff --git a/libavformat/dashenc.c b/libavformat/dashenc.c
> index 3345b89..c4c112b 100644
> --- a/libavformat/dashenc.c
> +++ b/liba
> On 2 Jan 2018, at 14:48, Karthick J wrote:
>
> From: Karthick Jeyapal
>
> Currently http end of chunk is signalled implicitly in dashenc_io_open().
> This mean playlists http writes would have to wait upto a segment duration to
> signal end of chunk causing delays.
> This patch will fix tha
2018-01-02 15:52 GMT+08:00 :
> From: Jiejun Zhang
>
> This fixes #6940
> ---
> libavcodec/audiotoolboxenc.c | 20 +++-
> 1 file changed, 19 insertions(+), 1 deletion(-)
>
> diff --git a/libavcodec/audiotoolboxenc.c b/libavcodec/audiotoolboxenc.c
> index 71885d1530..b70375a692 100
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