Re: [FFmpeg-devel] Experiences in using ffmpeg to transcode broadcast video

[Mika Raento]

Hi

Thanks for the feedback.

I'll try to capture at least some of these in tickets, and reply on this
thread as I go along.

Async not working with damaged audio input is
https://trac.ffmpeg.org/ticket/2693 (existing ticket)

Sub2video not working with negative timestamps is
https://trac.ffmpeg.org/ticket/4062 (new ticket)

Mika

On 27 October 2014 01:00, Michael Niedermayer  wrote:

> On Wed, Sep 24, 2014 at 10:43:40AM +0300, Mika Raento wrote:
> > Dear all
> >
> > This mail is meant mainly as a note to other potential users, but
> > possibly as input to development - time allowing I might be able to pick
> > up some of the pieces myself. It's also a thank-you for all the hard
> > work in ffmpeg.
> >
> > I've successfully implemented a transcoding pipeline for producing
> > multi-bitrate fragmented mp4 files from broadcast DVB input. More
> > concretely, I'm taking in broadcast TS captures with either mpeg2 video
> > and mp2 audio (SD, varying aspect ratio) or h264 video and aac audio,
> > both potentially with dvbsub. From that I'm producing ISMV output with
> > multiple bitrate h264 video at fixed 16:9 aspect ratio and multiple
> > bitrate aac audio, with burned subtitles. The ISMV outputs are
> > post-processed with tools/ismindex and with the hls muxer.
> >
> > There are number of limitations in ffmpeg that I've had to work around:
> >
> > - I haven't gotten sub2video or async working without reasonably
> >   monotonous DTS. Broadcast TS streams can easily contain backward jumps
> >   in time (e.g., a cheapo source that plays mp4 files and starts each
> file
> >   at 0). The fix is to cut the TS into pieces at timestamp jump locations
> >   and using '-itsoffset' to rewrite the timestamps and then concatenate.
> >   I'm using the segment and concat muxers for that.
> > - Sub2video doesn't work with negative timestamps, so I use '-itsoffset'
> >   to get positive timestamps
> > - For HD streams, I need to scale up the sub2video results from SD to
> >   HD. Sub2video doesn't handle the HD subtitle geometries. I'm not
> >   enough of an expert to know whether that's the issue, or whether that's
> >   just the way it's supposed to work with SD subs (typical) with HD
> video.
> > - For columnboxing, I use the scale, pad and setdar video filters. These
> >   work fine, but their parameters are only evaluated at start, so I need
> >   to cut the video into pieces with a single aspect ratio first and
> >   concatenate later.
> > - Audio sync (using aresample) gets confused if the input contains
> >   errors, so I need to first re-encode audio (with '-copyts') and only
> >   after that synchronize.
> > - The TS PIDs are not kept over the segment muxer, so I given them on
> >   the command line with '-streamid'.
>
> a bit late reply but
> bug reports or feature requests for all these are welcome unless there
> are already tickets for them assuming these issues still exist
>
> thx
>
> [...]
>
> --
> Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
>
> Why not whip the teacher when the pupil misbehaves? -- Diogenes of Sinope
>
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Re: [FFmpeg-devel] Experiences in using ffmpeg to transcode broadcast video

[Mika Raento]

segment muxer PIDs were fixed in
https://github.com/FFmpeg/FFmpeg/commit/502fc3b3d4b36015562d19d74f27d0a4ff835c4e
by me

On 29 October 2014 10:17, Mika Raento  wrote:

> Hi
>
> Thanks for the feedback.
>
> I'll try to capture at least some of these in tickets, and reply on this
> thread as I go along.
>
> Async not working with damaged audio input is
> https://trac.ffmpeg.org/ticket/2693 (existing ticket)
>
> Sub2video not working with negative timestamps is
> https://trac.ffmpeg.org/ticket/4062 (new ticket)
>
> Mika
>
> On 27 October 2014 01:00, Michael Niedermayer  wrote:
>
>> On Wed, Sep 24, 2014 at 10:43:40AM +0300, Mika Raento wrote:
>> > Dear all
>> >
>> > This mail is meant mainly as a note to other potential users, but
>> > possibly as input to development - time allowing I might be able to pick
>> > up some of the pieces myself. It's also a thank-you for all the hard
>> > work in ffmpeg.
>> >
>> > I've successfully implemented a transcoding pipeline for producing
>> > multi-bitrate fragmented mp4 files from broadcast DVB input. More
>> > concretely, I'm taking in broadcast TS captures with either mpeg2 video
>> > and mp2 audio (SD, varying aspect ratio) or h264 video and aac audio,
>> > both potentially with dvbsub. From that I'm producing ISMV output with
>> > multiple bitrate h264 video at fixed 16:9 aspect ratio and multiple
>> > bitrate aac audio, with burned subtitles. The ISMV outputs are
>> > post-processed with tools/ismindex and with the hls muxer.
>> >
>> > There are number of limitations in ffmpeg that I've had to work around:
>> >
>> > - I haven't gotten sub2video or async working without reasonably
>> >   monotonous DTS. Broadcast TS streams can easily contain backward jumps
>> >   in time (e.g., a cheapo source that plays mp4 files and starts each
>> file
>> >   at 0). The fix is to cut the TS into pieces at timestamp jump
>> locations
>> >   and using '-itsoffset' to rewrite the timestamps and then concatenate.
>> >   I'm using the segment and concat muxers for that.
>> > - Sub2video doesn't work with negative timestamps, so I use '-itsoffset'
>> >   to get positive timestamps
>> > - For HD streams, I need to scale up the sub2video results from SD to
>> >   HD. Sub2video doesn't handle the HD subtitle geometries. I'm not
>> >   enough of an expert to know whether that's the issue, or whether
>> that's
>> >   just the way it's supposed to work with SD subs (typical) with HD
>> video.
>> > - For columnboxing, I use the scale, pad and setdar video filters. These
>> >   work fine, but their parameters are only evaluated at start, so I need
>> >   to cut the video into pieces with a single aspect ratio first and
>> >   concatenate later.
>> > - Audio sync (using aresample) gets confused if the input contains
>> >   errors, so I need to first re-encode audio (with '-copyts') and only
>> >   after that synchronize.
>> > - The TS PIDs are not kept over the segment muxer, so I given them on
>> >   the command line with '-streamid'.
>>
>> a bit late reply but
>> bug reports or feature requests for all these are welcome unless there
>> are already tickets for them assuming these issues still exist
>>
>> thx
>>
>> [...]
>>
>> --
>> Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
>>
>> Why not whip the teacher when the pupil misbehaves? -- Diogenes of Sinope
>>
>> ___
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>> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>>
>>
>
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Re: [FFmpeg-devel] [RFC]Print a warning if a subtitle demuxer changes utf16 to utf8.

[Carl Eugen Hoyos]

Nicolas George  nsup.org> writes:

> Le septidi 7 brumaire, an CCXXIII, Carl Eugen Hoyos a écrit :
> > The user should be told by libavformat if it does something 
> > unexpected.
> > Related to ticket #4059.
> 
> I do not object to this quick fix

The warning was merged by Michael.

Thank you, Carl Eugen

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Re: [FFmpeg-devel] [RFC]Do not set the lame quality if the user didn't set it

[Carl Eugen Hoyos]

Michael Niedermayer  gmx.at> writes:

> > Attached patch makes FFmpeg mp3 output more similar to lame's.
> > It appears to me that 5 is not the default if vbr is used.

> LGTM

Merged by you.

Thank you, Carl Eugen

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Re: [FFmpeg-devel] [PATCH]Mention in the documentation that fieldmatch needs cfr input

[Carl Eugen Hoyos]

Clément Bœsch  pkh.me> writes:

> > +The filter only works for constant frame rate input. If your input
> > +has mixed telecined and progressive content with changing framerate,
> > +try the   ref{pullup} filter.
> 
> Well I don't mind much but then... how is pullup making 
> any difference here actually?

I removed the pullup suggestion and the patch was merged 
by Michael.

As discussed on irc, I believe that this should be fixed 
in decimate: It can detect the framerate change and not 
drop any frames for the progressive parts.

Carl Eugen

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Re: [FFmpeg-devel] [PATCH] Use v4l2 input format automatically if filename starts with "/dev/video"

[Carl Eugen Hoyos]

Andrey Utkin  gmail.com> writes:

> +if (av_strstart(p->filename, "/dev/video", NULL))
> +return AVPROBE_SCORE_MAX;

I took the freedom to apply this patch with MAX - 1.

Carl Eugen

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Re: [FFmpeg-devel] [PATCH 4/6] dv: fix weight table for 2x4x8 transform

[Christophe Gisquet]

Hi,

2014-10-25 20:32 GMT+02:00 Christophe Gisquet :
> What I meant is I would need someone to:
> 1) provide a command line to swap fields to produce an artificially
> interlaced image
> 2) confirm that -flags ildct sets CODEC_FLAG_INTERLACED_DCT and does
> what is needed

Unfortunately, I haven't been able to create a sample out of lena with
libavfilter that triggers this (and I'm unwilling to spend more time
on that).

Anyway, most libavfilters require a GPL build, so this contrives the
conditions to perform the test. This is potentially an issue in the
interlaced encoding decision.

The alternative is to just add a fate sample.

-- 
Christophe
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[FFmpeg-devel] [PATCH] avformat: remove messages that are useful for ffmpeg.c only

[wm4]

Other applications don't have this problem. The warning is useful for
ffmpeg.c only, so it should be in ffmpeg.c.

Note that the same problem exists for e.g. Matroska, which always uses
UTF-8 for subtitles. If anyone wants to solve this properly, something
somewhere else should check whether the packets are already in UTF-8,
or add a "this is UTF-8" flag to the demuxer. Another possibility would
be rejecting any conversion from UTF-16, since that can never happen.

For now, this message is annoying and useless for non-ffmpeg.c users,
and as I understand, libav* is supposed to serve these users too.

This partially reverts commit 19a6431ec247e4842236292cc5f8cfc8f87da11e.
---
 libavformat/subtitles.c | 3 ---
 libavformat/subtitles.h | 1 -
 2 files changed, 4 deletions(-)

diff --git a/libavformat/subtitles.c b/libavformat/subtitles.c
index 7f4bdef..b796df3 100644
--- a/libavformat/subtitles.c
+++ b/libavformat/subtitles.c
@@ -45,9 +45,6 @@ void ff_text_init_avio(void *s, FFTextReader *r, AVIOContext 
*pb)
 r->buf_pos += 3;
 }
 }
-if (s && (r->type == FF_UTF16LE || r->type == FF_UTF16BE))
-av_log(s, AV_LOG_WARNING,
-   "UTF16 is automatically converted to UTF8, do not specify a 
character encoding\n");
 }
 
 void ff_text_init_buf(FFTextReader *r, void *buf, size_t size)
diff --git a/libavformat/subtitles.h b/libavformat/subtitles.h
index eb719ea..f2279b6 100644
--- a/libavformat/subtitles.h
+++ b/libavformat/subtitles.h
@@ -49,7 +49,6 @@ typedef struct {
  * Initialize the FFTextReader from the given AVIOContext. This function will
  * read some bytes from pb, and test for UTF-8 or UTF-16 BOMs. Further accesses
  * to FFTextReader will read more data from pb.
- * If s is not NULL, the user will be warned if a UTF-16 conversion takes 
place.
  *
  * The purpose of FFTextReader is to transparently convert read data to UTF-8
  * if the stream had a UTF-16 BOM.
-- 
2.1.1

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[FFmpeg-devel] [PATCH]Improve the fieldmatch documentation about mixed telecined content

[Carl Eugen Hoyos]

Hi!

Attached patch is meant to improve the warning that mixed 
telecined and progressive content is not handled well by 
fieldmatch and decimate.

I believe this paragraph is more useful in the fieldmatch 
section of the documentation, if it should be moved to 
the decimate section, please say so.

Thank you, Carl Eugen
diff --git a/doc/filters.texi b/doc/filters.texi
index 01eb893..7be29de 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -4447,9 +4447,9 @@ and VIVTC/VFM (VapourSynth project). The later is a light 
clone of TFM from
 which @code{fieldmatch} is based on. While the semantic and usage are very
 close, some behaviour and options names can differ.
 
-The filter currently only works for constant frame rate input. Do not use it
-if your input has mixed telecined and progressive content with changing
-framerate.
+The @ref{decimate} filter currently only works for constant frame rate input.
+Do not use @code{fieldmatch} and @ref{decimate} if your input has mixed
+telecined and progressive content with changing framerate.
 
 The filter accepts the following options:
 
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Re: [FFmpeg-devel] [PATCH] avformat: remove messages that are useful for ffmpeg.c only

[Carl Eugen Hoyos]

wm4  googlemail.com> writes:

> For now, this message is annoying and useless for 
> non-ffmpeg.c users, and as I understand, libav* is 
> supposed to serve these users too.

Why don't you reduce the log level instead?

> This partially reverts commit 19a6431e

And why a partial revert?

Carl Eugen

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Re: [FFmpeg-devel] [PATCH] avformat: remove messages that are useful for ffmpeg.c only

[wm4]

On Wed, 29 Oct 2014 09:16:51 + (UTC)
Carl Eugen Hoyos  wrote:

> wm4  googlemail.com> writes:
> 
> > For now, this message is annoying and useless for 
> > non-ffmpeg.c users, and as I understand, libav* is 
> > supposed to serve these users too.
> 
> Why don't you reduce the log level instead?

For what purpose?

> > This partially reverts commit 19a6431e
> 
> And why a partial revert?

A logging context may or may not be useful in the future, e.g. on
recoding errors.
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Re: [FFmpeg-devel] [PATCH]Improve the fieldmatch documentation about mixed telecined content

[Carl Eugen Hoyos]

Carl Eugen Hoyos  ag.or.at> writes:

> Attached patch is meant to improve the warning that mixed 
> telecined and progressive content is not handled well by 
> fieldmatch and decimate.

Ok'ed by Clément on irc and pushed.

Thank you, Carl Eugen

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Re: [FFmpeg-devel] [PATCH] avformat/mxfenc: AVC Intra support

[tomas . hardin]

On 2014-10-28 23:38, Thomas Mundt wrote:

Carl Eugen, I changed the indentations and attached a unified diff.
Since I´m on windows I used turtoise git. I hope that´s okay.

Regards,
Thomas


Looks good to me

/Tomas
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Re: [FFmpeg-devel] [PATCH] avformat: remove messages that are useful for ffmpeg.c only

[Carl Eugen Hoyos]

wm4  googlemail.com> writes:

> > > For now, this message is annoying and useless for 
> > > non-ffmpeg.c users, and as I understand, libav* is 
> > > supposed to serve these users too.
> > 
> > Why don't you reduce the log level instead?
> 
> For what purpose?

I was under the impression that a lot of messages 
are currently output by the libraries and that you 
are against seeing one more with warning level.
But I am apparently wrong?

The problem with reverting the patch is that a 
user simply has no chance to understand why he 
must not provide the "right" and documented 
option to libavcodec in the utf-16 case as 
explained in the ticket.
Do you see another solution for this problem?

Carl Eugen

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Re: [FFmpeg-devel] Problem uploading results to f...@fate.ffmpeg.org

[Michael Niedermayer]

On Wed, Oct 29, 2014 at 03:53:04AM +0100, Ngassa Finjap wrote:
> Hello,
> 
> I have ran the FATE test suite on my Fedora host using this: ( make V=2 -j2
> SAMPLES=/var/fate/samples THREADS=2 fate) and I can't seem to authenticate
> and upload the results.  I had previously sent a key to for authentication.
> But I had some problems uploading my test results. I have generated a new
> key for my system and sent to the admin for authentication. I"ll like to

key added to server


> have some more assistance in submitting my results successfully since I am
> having problems using the wiki.
> 
> Thanks,
> Amalia
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Re: [FFmpeg-devel] [PATCH] avformat/mxfenc: AVC Intra support

[Michael Niedermayer]

On Wed, Oct 29, 2014 at 10:27:25AM +0100, tomas.har...@codemill.se wrote:
> On 2014-10-28 23:38, Thomas Mundt wrote:
> >Carl Eugen, I changed the indentations and attached a unified diff.
> >Since I´m on windows I used turtoise git. I hope that´s okay.
> >
> >Regards,
> >Thomas
> 
> Looks good to me

applied

thanks

[...]
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Re: [FFmpeg-devel] [PATCH 0/6] dv: of inverse weight tables

[Christophe Gisquet]

Hi,

2014-10-25 13:29 GMT+02:00 Reimar Döffinger :
> Maybe as a patch that puts it into tools?
> I don't know if that's the right place, but I would kind of prefer to
> have these kind of things around.
> Probably overkill, but we have libavcodec/tableprint.h that has helpers
> to generate files with tables.

I see dv_tablegen.c which sounds like a good place. You probably have
a better idea seeing how several files serving a similar purpose were
created by you. On the other hand, that's just 2*128 bytes.

Otherwise, the mailing list is there to forever archive that program
(hence my sending it).

-- 
Christophe
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Re: [FFmpeg-devel] Experiences in using ffmpeg to transcode broadcast video

[Mika Raento]

Tickets for sub2video and async not working with non-monotonic input, using
-copyts:

https://trac.ffmpeg.org/ticket/4062
https://trac.ffmpeg.org/ticket/4064

Hm. Lots of my issues seem to relate to using -copyts. I had trouble
keeping all video and all audio if there were periods where one or the
other were missing (esp. in the beginning). Maybe I could make this simpler
if I'd relax that requirement (I don't know how much happier my users are
if they get audio but no video, rather than nothing).

Mika

On 29 October 2014 10:38, Mika Raento  wrote:

> segment muxer PIDs were fixed in
> https://github.com/FFmpeg/FFmpeg/commit/502fc3b3d4b36015562d19d74f27d0a4ff835c4e
> by me
>
> On 29 October 2014 10:17, Mika Raento  wrote:
>
>> Hi
>>
>> Thanks for the feedback.
>>
>> I'll try to capture at least some of these in tickets, and reply on this
>> thread as I go along.
>>
>> Async not working with damaged audio input is
>> https://trac.ffmpeg.org/ticket/2693 (existing ticket)
>>
>> Sub2video not working with negative timestamps is
>> https://trac.ffmpeg.org/ticket/4062 (new ticket)
>>
>> Mika
>>
>> On 27 October 2014 01:00, Michael Niedermayer  wrote:
>>
>>> On Wed, Sep 24, 2014 at 10:43:40AM +0300, Mika Raento wrote:
>>> > Dear all
>>> >
>>> > This mail is meant mainly as a note to other potential users, but
>>> > possibly as input to development - time allowing I might be able to
>>> pick
>>> > up some of the pieces myself. It's also a thank-you for all the hard
>>> > work in ffmpeg.
>>> >
>>> > I've successfully implemented a transcoding pipeline for producing
>>> > multi-bitrate fragmented mp4 files from broadcast DVB input. More
>>> > concretely, I'm taking in broadcast TS captures with either mpeg2 video
>>> > and mp2 audio (SD, varying aspect ratio) or h264 video and aac audio,
>>> > both potentially with dvbsub. From that I'm producing ISMV output with
>>> > multiple bitrate h264 video at fixed 16:9 aspect ratio and multiple
>>> > bitrate aac audio, with burned subtitles. The ISMV outputs are
>>> > post-processed with tools/ismindex and with the hls muxer.
>>> >
>>> > There are number of limitations in ffmpeg that I've had to work around:
>>> >
>>> > - I haven't gotten sub2video or async working without reasonably
>>> >   monotonous DTS. Broadcast TS streams can easily contain backward
>>> jumps
>>> >   in time (e.g., a cheapo source that plays mp4 files and starts each
>>> file
>>> >   at 0). The fix is to cut the TS into pieces at timestamp jump
>>> locations
>>> >   and using '-itsoffset' to rewrite the timestamps and then
>>> concatenate.
>>> >   I'm using the segment and concat muxers for that.
>>> > - Sub2video doesn't work with negative timestamps, so I use
>>> '-itsoffset'
>>> >   to get positive timestamps
>>> > - For HD streams, I need to scale up the sub2video results from SD to
>>> >   HD. Sub2video doesn't handle the HD subtitle geometries. I'm not
>>> >   enough of an expert to know whether that's the issue, or whether
>>> that's
>>> >   just the way it's supposed to work with SD subs (typical) with HD
>>> video.
>>> > - For columnboxing, I use the scale, pad and setdar video filters.
>>> These
>>> >   work fine, but their parameters are only evaluated at start, so I
>>> need
>>> >   to cut the video into pieces with a single aspect ratio first and
>>> >   concatenate later.
>>> > - Audio sync (using aresample) gets confused if the input contains
>>> >   errors, so I need to first re-encode audio (with '-copyts') and only
>>> >   after that synchronize.
>>> > - The TS PIDs are not kept over the segment muxer, so I given them on
>>> >   the command line with '-streamid'.
>>>
>>> a bit late reply but
>>> bug reports or feature requests for all these are welcome unless there
>>> are already tickets for them assuming these issues still exist
>>>
>>> thx
>>>
>>> [...]
>>>
>>> --
>>> Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
>>>
>>> Why not whip the teacher when the pupil misbehaves? -- Diogenes of Sinope
>>>
>>> ___
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>>>
>>
>
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Re: [FFmpeg-devel] [PATCH] avformat: remove messages that are useful for ffmpeg.c only

[wm4]

On Wed, 29 Oct 2014 09:35:32 + (UTC)
Carl Eugen Hoyos  wrote:

> wm4  googlemail.com> writes:
> 
> > > > For now, this message is annoying and useless for 
> > > > non-ffmpeg.c users, and as I understand, libav* is 
> > > > supposed to serve these users too.
> > > 
> > > Why don't you reduce the log level instead?
> > 
> > For what purpose?
> 
> I was under the impression that a lot of messages 
> are currently output by the libraries and that you 
> are against seeing one more with warning level.
> But I am apparently wrong?

I output all messages starting from warnings (messages with higher
verbosity level are usually even more useless).

> The problem with reverting the patch is that a 
> user simply has no chance to understand why he 
> must not provide the "right" and documented 
> option to libavcodec in the utf-16 case as 
> explained in the ticket.
> Do you see another solution for this problem?

Read the commit message: either add a flag to avformat that subtitle
packets are always in UTF-8, or reject conversions from UTF-16, since
they can never happen.
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Re: [FFmpeg-devel] [PATCH 2/2] avcodec/libutvideodec: Support YUV422P10

[Michael Niedermayer]

On Wed, Oct 29, 2014 at 06:54:36AM +0100, Jan Ehrhardt wrote:
> Michael Niedermayer in gmane.comp.video.ffmpeg.devel (Sat, 25 Oct 2014
> 00:14:14 +0200):
> >patch applied
> 
> FFmpeg does not (cross)compile anymore on Ubuntu (GCC 4.9.1) after this
> patch:
> 
> libavcodec/libutvideodec.cpp: In function 'int
> utvideo_decode_init(AVCodecContext*)':
> libavcodec/libutvideodec.cpp:97:19: error: 'UTVF_v210' was not declared
> in this scope
>  if (format == UTVF_v210)
>^
> libavcodec/libutvideodec.cpp: In function 'int

should be fixed

[...]
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Re: [FFmpeg-devel] Experiences in using ffmpeg to transcode broadcast video

[Mika Raento]

Created a feature request to support columnboxing input that changes
aspect-ratio mid-stream.

https://trac.ffmpeg.org/ticket/4065

On 29 October 2014 13:49, Mika Raento  wrote:

> Tickets for sub2video and async not working with non-monotonic input,
> using -copyts:
>
> https://trac.ffmpeg.org/ticket/4062
> https://trac.ffmpeg.org/ticket/4064
>
> Hm. Lots of my issues seem to relate to using -copyts. I had trouble
> keeping all video and all audio if there were periods where one or the
> other were missing (esp. in the beginning). Maybe I could make this simpler
> if I'd relax that requirement (I don't know how much happier my users are
> if they get audio but no video, rather than nothing).
>
> Mika
>
> On 29 October 2014 10:38, Mika Raento  wrote:
>
>> segment muxer PIDs were fixed in
>> https://github.com/FFmpeg/FFmpeg/commit/502fc3b3d4b36015562d19d74f27d0a4ff835c4e
>> by me
>>
>> On 29 October 2014 10:17, Mika Raento  wrote:
>>
>>> Hi
>>>
>>> Thanks for the feedback.
>>>
>>> I'll try to capture at least some of these in tickets, and reply on this
>>> thread as I go along.
>>>
>>> Async not working with damaged audio input is
>>> https://trac.ffmpeg.org/ticket/2693 (existing ticket)
>>>
>>> Sub2video not working with negative timestamps is
>>> https://trac.ffmpeg.org/ticket/4062 (new ticket)
>>>
>>> Mika
>>>
>>> On 27 October 2014 01:00, Michael Niedermayer  wrote:
>>>
 On Wed, Sep 24, 2014 at 10:43:40AM +0300, Mika Raento wrote:
 > Dear all
 >
 > This mail is meant mainly as a note to other potential users, but
 > possibly as input to development - time allowing I might be able to
 pick
 > up some of the pieces myself. It's also a thank-you for all the hard
 > work in ffmpeg.
 >
 > I've successfully implemented a transcoding pipeline for producing
 > multi-bitrate fragmented mp4 files from broadcast DVB input. More
 > concretely, I'm taking in broadcast TS captures with either mpeg2
 video
 > and mp2 audio (SD, varying aspect ratio) or h264 video and aac audio,
 > both potentially with dvbsub. From that I'm producing ISMV output with
 > multiple bitrate h264 video at fixed 16:9 aspect ratio and multiple
 > bitrate aac audio, with burned subtitles. The ISMV outputs are
 > post-processed with tools/ismindex and with the hls muxer.
 >
 > There are number of limitations in ffmpeg that I've had to work
 around:
 >
 > - I haven't gotten sub2video or async working without reasonably
 >   monotonous DTS. Broadcast TS streams can easily contain backward
 jumps
 >   in time (e.g., a cheapo source that plays mp4 files and starts each
 file
 >   at 0). The fix is to cut the TS into pieces at timestamp jump
 locations
 >   and using '-itsoffset' to rewrite the timestamps and then
 concatenate.
 >   I'm using the segment and concat muxers for that.
 > - Sub2video doesn't work with negative timestamps, so I use
 '-itsoffset'
 >   to get positive timestamps
 > - For HD streams, I need to scale up the sub2video results from SD to
 >   HD. Sub2video doesn't handle the HD subtitle geometries. I'm not
 >   enough of an expert to know whether that's the issue, or whether
 that's
 >   just the way it's supposed to work with SD subs (typical) with HD
 video.
 > - For columnboxing, I use the scale, pad and setdar video filters.
 These
 >   work fine, but their parameters are only evaluated at start, so I
 need
 >   to cut the video into pieces with a single aspect ratio first and
 >   concatenate later.
 > - Audio sync (using aresample) gets confused if the input contains
 >   errors, so I need to first re-encode audio (with '-copyts') and only
 >   after that synchronize.
 > - The TS PIDs are not kept over the segment muxer, so I given them on
 >   the command line with '-streamid'.

 a bit late reply but
 bug reports or feature requests for all these are welcome unless there
 are already tickets for them assuming these issues still exist

 thx

 [...]

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>>>
>>
>
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[FFmpeg-devel] [Patch]support more AVC Intra formats without SPS/PPS header

[Thomas Mundt]

Hi, this patch will add support for AVC Intra 50 720p and 1080p without SPS/PPS 
header in mxf and mov demuxers. I got the SPS/PPS tables from libbmx.

Regards,
Thomas


avci50.patch
Description: Binary data
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Re: [FFmpeg-devel] [PATCH] lavd/v4l2: don't use avpriv_ prefix for internal functions

[Giorgio Vazzana]

2014-10-21 12:26 GMT+02:00 Stefano Sabatini :
> On date Tuesday 2014-10-21 03:56:06 -0300, James Almer encoded:
>> No need to keep the old symbols around until a major bump since lavd 
>> functions
>> with the avpriv_ prefix were never exposed.
>>
>> Signed-off-by: James Almer 
>
> LGTM.

Ping. Any news about this? Can it be applied?
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Re: [FFmpeg-devel] [PATCH] lavd/v4l2: don't use avpriv_ prefix for internal functions

[Michael Niedermayer]

On Wed, Oct 29, 2014 at 03:20:02PM +0100, Giorgio Vazzana wrote:
> 2014-10-21 12:26 GMT+02:00 Stefano Sabatini :
> > On date Tuesday 2014-10-21 03:56:06 -0300, James Almer encoded:
> >> No need to keep the old symbols around until a major bump since lavd 
> >> functions
> >> with the avpriv_ prefix were never exposed.
> >>
> >> Signed-off-by: James Almer 
> >
> > LGTM.
> 
> Ping. Any news about this? Can it be applied?

applied

thx

[...]
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Re: [FFmpeg-devel] [Patch]support more AVC Intra formats without SPS/PPS header

[tomas . hardin]

On 2014-10-29 14:04, Thomas Mundt wrote:

Hi, this patch will add support for AVC Intra 50 720p and 1080p
without SPS/PPS header in mxf and mov demuxers. I got the SPS/PPS
tables from libbmx.

Regards,
Thomas


Looks OK to me.

/Tomas
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Re: [FFmpeg-devel] FFMPEG : Redirecting Matroska muxed data to socket

[Calvin Walton]

On Wed, 2014-10-29 at 11:26 +0530, Parth Shah wrote:
> Hi all,
> 
> I am using FFMPEG library to mux H.264 and AAC frames to Matroska 
> (.mkv)
> file. I can do that both using command line and C program.
> 
> Now, instead of writing the muxed matroska data into file I want to
> write these muxed data directly on to socket or pipe. My actual goal 
> is
> to write a C program that send muxed data to socket and server will
> receive this muxed data.
> 
> I tried using protocol tcp. They are working with the matroska 
> format.
> So, My C program is able to send muxed data successfully over socket 
> and
> server is able to receive this muxed data.
> 
> But when I apply ffprobe command over the received file, I am getting
> duration and bitrate field N/A. and when I tried to play this file 
> with
> vlc i am unable to seek the file and getting garbage duration.

When ffmpeg writes matroska directly to a file, it writes all of the 
data for the media streams out, then seeks back to the start of the 
file to fill in the duration field.

But you're sending the matroska over a tcp stream - effectively a live 
stream. As a result, ffmpeg can't go back and update these fields once 
the file is completely written; so they're left unset.

This isn't normally a problem - it means that players will not show 
the correct duration while playing the file, but seeking should still 
work (and does in e.g. mplayer - it sounds like you might be hitting a 
vlc bug if seeking doesn't work there).

To fix the received file, you'll just have to remux the mkv, e.g. by 
doing
ffmpeg -i received.mkv -c copy remuxed.mkv


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Re: [FFmpeg-devel] [PATCH]Mention in the documentation that fieldmatch needs cfr input

[Calvin Walton]

On Sat, 2014-10-25 at 12:22 +, Carl Eugen Hoyos wrote:
> Clément Bœsch  pkh.me> writes:
> 
> > > +The filter only works for constant frame rate input. If your 
> > > input
> > > +has mixed telecined and progressive content with changing 
> > > framerate,
> > > +try the   ref{pullup} filter.
> > 
> > Well I don't mind much but then... how is pullup making
> > any difference here actually
> 
> Just to sum it up:
> pullup works fine and is fast for all samples I have seen,
> it definitely misses many merging opportunities if the
> horizontal motion is very low (meaning some frames with
> artefacts remain for every real-world input).
> fieldmatch is very slow, it is apparently able to
> produce perfect output for badly cut input with constant
> (telecined) framerate but it fails completely for mixed
> content (as found on many DVD's).
> 
> Imo, fps=3/1001,fieldmatch,decimate should fix this
> but decimate unfortunately does not drop the frame that
> fps inserted but a random (?) frame.
> (Or fieldmatch finds matches in progressive input?)

For content that was in mpeg2 with field flags set appropriate for 
display on an interlaced TV - which basically accounts for all DVD 
content - mplayer had a 'softpulldown' filter that used these flags to 
turn the video stream from mixed progressive+telecined+interlaced 
content with variable framerate (in NTSC, mixed 24/1001 frames/s and 
60/1001 fields/s) into a telecined+interlaced video with constant 
framerate (60/1001 fields/s).

I wonder if something similar would be useful in ffmpeg, to provide a 
constant fps stream that a detelecine filter could use as input.

-- 
Calvin Walton 
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Re: [FFmpeg-devel] [PATCH] avformat: remove messages that are useful for ffmpeg.c only

[Carl Eugen Hoyos]

wm4  googlemail.com> writes:

> > > > Why don't you reduce the log level instead?
> > > 
> > > For what purpose?
> > 
> > I was under the impression that a lot of messages 
> > are currently output by the libraries and that you 
> > are against seeing one more with warning level.
> > But I am apparently wrong?
> 
> I output all messages starting from warnings (messages 
> with higher verbosity level are usually even more useless).

Great!

Then please reduce the log level for the offensive 
message.

Carl Eugen

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Re: [FFmpeg-devel] [PATCH]Mention in the documentation that fieldmatch needs cfr input

[Carl Eugen Hoyos]

Calvin Walton  kepstin.ca> writes:

> For content that was in mpeg2 with field flags set 
> appropriate for display on an interlaced TV 

This is unrelated to this issue:
FFmpeg simply ignores the flag and interprets the 
input as progressive.

> which basically accounts for all DVD content

I live in PAL-country but from user reports and 
samples (like the one in question) I know that 
this unfortunately isn't true;-(

This whole issue is about vob files that contain 
both progressive 24000/1001 and (hard-) telecined 
3/1001 video in the same stream (with actual 
switching frame rates).
It appears to me that decimate should detect the 
frame rate changes and not drop anything for the 
progressive content.

Carl Eugen

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Re: [FFmpeg-devel] [PATCH] lavd/v4l2: don't use avpriv_ prefix for internal functions

[James Almer]

On 29/10/14 11:20 AM, Giorgio Vazzana wrote:
> 2014-10-21 12:26 GMT+02:00 Stefano Sabatini :
>> On date Tuesday 2014-10-21 03:56:06 -0300, James Almer encoded:
>>> No need to keep the old symbols around until a major bump since lavd 
>>> functions
>>> with the avpriv_ prefix were never exposed.
>>>
>>> Signed-off-by: James Almer 
>>
>> LGTM.
> 
> Ping. Any news about this? Can it be applied?

Wow, somehow missed this thread getting replies.
Sorry for the delay, and thanks Michael for pushing it.
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[FFmpeg-devel] [PATCH] avformat/subtitles: reduce log level of UTF-16 warning

[wm4]

Applications can use this to silence the message.
---
As suggested. Not a great solution, but does the job to me.
---
 libavformat/subtitles.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/libavformat/subtitles.c b/libavformat/subtitles.c
index 7f4bdef..67624fc 100644
--- a/libavformat/subtitles.c
+++ b/libavformat/subtitles.c
@@ -46,7 +46,7 @@ void ff_text_init_avio(void *s, FFTextReader *r, AVIOContext 
*pb)
 }
 }
 if (s && (r->type == FF_UTF16LE || r->type == FF_UTF16BE))
-av_log(s, AV_LOG_WARNING,
+av_log(s, AV_LOG_INFO,
"UTF16 is automatically converted to UTF8, do not specify a 
character encoding\n");
 }
 
-- 
2.1.1

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Re: [FFmpeg-devel] [PATCH] lavu/atomic: add support for the new memory model aware gcc built-ins

[James Almer]

On 28/10/14 11:16 PM, Michael Niedermayer wrote:
> On Mon, Oct 27, 2014 at 11:25:01PM -0300, James Almer wrote:
>> __sync built-ins are considered legacy and will be deprecated.
>> These new memory model aware built-ins have been available since GCC 4.7.0
>>
>> Use them by default when available except for __atomic_compare_exchange_n(),
>> which is slower, and is instead implemented as a fallback for when and if gcc
>> removes the legacy __sync built-ins.
>>
>> Signed-off-by: James Almer 
>> ---
>>  configure  |  4 +++-
>>  libavutil/atomic_gcc.h | 17 +
>>  2 files changed, 20 insertions(+), 1 deletion(-)
> 
> LGTM

Pushed, thanks.

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Re: [FFmpeg-devel] [PATCH]Mention in the documentation that fieldmatch needs cfr input

[Nicholas Robbins]

> On Wednesday, October 29, 2014 12:25 PM, Carl Eugen Hoyos  
> wrote:

> > Calvin Walton  kepstin.ca> writes:
> 
>>  For content that was in mpeg2 with field flags set 
>>  appropriate for display on an interlaced TV 
> 
> This is unrelated to this issue:
> FFmpeg simply ignores the flag and interprets the 
> input as progressive.
> 
>>  which basically accounts for all DVD content
> 
> I live in PAL-country but from user reports and 
> samples (like the one in question) I know that 
> this unfortunately isn't true;-(
> 
> This whole issue is about vob files that contain 
> both progressive 24000/1001 and (hard-) telecined 
> 3/1001 video in the same stream (with actual 
> switching frame rates).
> It appears to me that decimate should detect the 
> frame rate changes and not drop anything for the 
> progressive content.
> 
> Carl Eugen


I've also seen mixed hard telecined (24->29.97) and interlaced (29.97) in the 
same stream, or hard telecined and 29.97 progressive. This whole thing makes me 
wish we used PAL here. Also metric paper, but that is a different gripe.

Not now, but later, I would be interested in helping to adapt decimate to do 
what we want.
A detectfps filter might be useful too, and would be a subset of the work.

Nick
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Re: [FFmpeg-devel] [PATCH] avformat/subtitles: reduce log level of UTF-16 warning

[Carl Eugen Hoyos]

wm4  googlemail.com> writes:

>  if (s && (r->type == FF_UTF16LE || r->type == FF_UTF16BE))
> -av_log(s, AV_LOG_WARNING,
> +av_log(s, AV_LOG_INFO,

Patch applied.

Thank you, Carl Eugen

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Re: [FFmpeg-devel] [PATCH]Mention in the documentation that fieldmatch needs cfr input

[Carl Eugen Hoyos]

Nicholas Robbins  ffmpeg.org> writes:

> Not now, but later, I would be interested in helping to 
> adapt decimate to do what we want. A detectfps filter 
> might be useful too, and would be a subset of the work.

I believe if decimate just detects 23.9 -> 29.97 and 
29.97 -> 23.9 it will fix nearly all use cases.

Carl Eugen

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Re: [FFmpeg-devel] [PATCH]Mention in the documentation that fieldmatch needs cfr input

[Nicholas Robbins]

> On Wednesday, October 29, 2014 1:33 PM, Carl Eugen Hoyos  
> wrote:

> > Nicholas Robbins  ffmpeg.org> writes:
> 
>>  Not now, but later, I would be interested in helping to 
>>  adapt decimate to do what we want. A detectfps filter 
>>  might be useful too, and would be a subset of the work.
> 
> I believe if decimate just detects 23.9 -> 29.97 and 
> 29.97 -> 23.9 it will fix nearly all use cases.
> 
> Carl Eugen

If it is running after fieldmatch, then it needs to detect

23.9
29.9 (w 1/5 duplicates duplicates, ouput of fieldmatch on TC material)
29.9 (w/o duplicates, interlaced or progressive, but not TC)

A bonus would be to differentiate 29.9 that was originally 23.9 (Film, TC with 
3:2) or originally 24.9 (PAL, TC with 3:3:2:2:2)

If the input to -vf "fieldmatch,decimate" is mixed 30fps progressive and the 
result of TC then it decimate will always see 30fps frame rate, but some frames 
will be duplicated, or am I missing something?

-Nick
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Re: [FFmpeg-devel] [PATCH] avformat/subtitles: reduce log level of UTF-16 warning

[wm4]

On Wed, 29 Oct 2014 17:30:47 + (UTC)
Carl Eugen Hoyos  wrote:

> wm4  googlemail.com> writes:
> 
> >  if (s && (r->type == FF_UTF16LE || r->type == FF_UTF16BE))
> > -av_log(s, AV_LOG_WARNING,
> > +av_log(s, AV_LOG_INFO,
> 
> Patch applied.
> 
> Thank you, Carl Eugen

Thanks.

Would a patch adding a "this is UTF-8" flag to the subtitle codec
headers be accepted? What should it be named? (There's already lots of
stuff in AVCodecContext...)
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Re: [FFmpeg-devel] Experiences in using ffmpeg to transcode broadcast video

[Mika Raento]

Created a feature request to automatically scale subtitles for HD video

https://trac.ffmpeg.org/ticket/4066


On 29 October 2014 14:25, Mika Raento  wrote:

> Created a feature request to support columnboxing input that changes
> aspect-ratio mid-stream.
>
> https://trac.ffmpeg.org/ticket/4065
>
> On 29 October 2014 13:49, Mika Raento  wrote:
>
>> Tickets for sub2video and async not working with non-monotonic input,
>> using -copyts:
>>
>> https://trac.ffmpeg.org/ticket/4062
>> https://trac.ffmpeg.org/ticket/4064
>>
>> Hm. Lots of my issues seem to relate to using -copyts. I had trouble
>> keeping all video and all audio if there were periods where one or the
>> other were missing (esp. in the beginning). Maybe I could make this simpler
>> if I'd relax that requirement (I don't know how much happier my users are
>> if they get audio but no video, rather than nothing).
>>
>> Mika
>>
>> On 29 October 2014 10:38, Mika Raento  wrote:
>>
>>> segment muxer PIDs were fixed in
>>> https://github.com/FFmpeg/FFmpeg/commit/502fc3b3d4b36015562d19d74f27d0a4ff835c4e
>>> by me
>>>
>>> On 29 October 2014 10:17, Mika Raento  wrote:
>>>
 Hi

 Thanks for the feedback.

 I'll try to capture at least some of these in tickets, and reply on
 this thread as I go along.

 Async not working with damaged audio input is
 https://trac.ffmpeg.org/ticket/2693 (existing ticket)

 Sub2video not working with negative timestamps is
 https://trac.ffmpeg.org/ticket/4062 (new ticket)

 Mika

 On 27 October 2014 01:00, Michael Niedermayer  wrote:

> On Wed, Sep 24, 2014 at 10:43:40AM +0300, Mika Raento wrote:
> > Dear all
> >
> > This mail is meant mainly as a note to other potential users, but
> > possibly as input to development - time allowing I might be able to
> pick
> > up some of the pieces myself. It's also a thank-you for all the hard
> > work in ffmpeg.
> >
> > I've successfully implemented a transcoding pipeline for producing
> > multi-bitrate fragmented mp4 files from broadcast DVB input. More
> > concretely, I'm taking in broadcast TS captures with either mpeg2
> video
> > and mp2 audio (SD, varying aspect ratio) or h264 video and aac audio,
> > both potentially with dvbsub. From that I'm producing ISMV output
> with
> > multiple bitrate h264 video at fixed 16:9 aspect ratio and multiple
> > bitrate aac audio, with burned subtitles. The ISMV outputs are
> > post-processed with tools/ismindex and with the hls muxer.
> >
> > There are number of limitations in ffmpeg that I've had to work
> around:
> >
> > - I haven't gotten sub2video or async working without reasonably
> >   monotonous DTS. Broadcast TS streams can easily contain backward
> jumps
> >   in time (e.g., a cheapo source that plays mp4 files and starts
> each file
> >   at 0). The fix is to cut the TS into pieces at timestamp jump
> locations
> >   and using '-itsoffset' to rewrite the timestamps and then
> concatenate.
> >   I'm using the segment and concat muxers for that.
> > - Sub2video doesn't work with negative timestamps, so I use
> '-itsoffset'
> >   to get positive timestamps
> > - For HD streams, I need to scale up the sub2video results from SD to
> >   HD. Sub2video doesn't handle the HD subtitle geometries. I'm not
> >   enough of an expert to know whether that's the issue, or whether
> that's
> >   just the way it's supposed to work with SD subs (typical) with HD
> video.
> > - For columnboxing, I use the scale, pad and setdar video filters.
> These
> >   work fine, but their parameters are only evaluated at start, so I
> need
> >   to cut the video into pieces with a single aspect ratio first and
> >   concatenate later.
> > - Audio sync (using aresample) gets confused if the input contains
> >   errors, so I need to first re-encode audio (with '-copyts') and
> only
> >   after that synchronize.
> > - The TS PIDs are not kept over the segment muxer, so I given them on
> >   the command line with '-streamid'.
>
> a bit late reply but
> bug reports or feature requests for all these are welcome unless there
> are already tickets for them assuming these issues still exist
>
> thx
>
> [...]
>
> --
> Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
>
> Why not whip the teacher when the pupil misbehaves? -- Diogenes of
> Sinope
>
> ___
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> ffmpeg-devel@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>
>

>>>
>>
>
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Re: [FFmpeg-devel] [Patch]support more AVC Intra formats without SPS/PPS header

[Michael Niedermayer]

On Wed, Oct 29, 2014 at 04:35:34PM +0100, tomas.har...@codemill.se wrote:
> On 2014-10-29 14:04, Thomas Mundt wrote:
> >Hi, this patch will add support for AVC Intra 50 720p and 1080p
> >without SPS/PPS header in mxf and mov demuxers. I got the SPS/PPS
> >tables from libbmx.
> >
> >Regards,
> >Thomas
> 
> Looks OK to me.

applied

thx

[...]
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Rewriting code that is poorly written but fully understood is good.
Rewriting code that one doesnt understand is a sign that one is less smart
then the original author, trying to rewrite it will not make it better.


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Re: [FFmpeg-devel] [Patch]support more AVC Intra formats without SPS/PPS header

[Thomas Mundt]

Thanks
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[FFmpeg-devel] mxfenc: fix indentation after last commit

[Thomas Mundt]

Should have done this as second part of my AVCI Patch. Sorry!

Regards,
Thomas


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[FFmpeg-devel] "OPW Qualification Task: Enable daemon mode for FFserver"

[Binathi Bingi]

>From 0fb7dcf1f126bd137e2b2025c5cd6cff4af65801 Mon Sep 17 00:00:00 2001
From: Binathi Bingi 
Date: Thu, 30 Oct 2014 01:14:08 +0530
Subject: [PATCH] ffserver: enable back deamon mode

---
 ffserver.c| 34 ++
 ffserver_config.c |  2 --
 2 files changed, 30 insertions(+), 6 deletions(-)

diff --git a/ffserver.c b/ffserver.c
index ea2a2ae..96e312e 100644
--- a/ffserver.c
+++ b/ffserver.c
@@ -236,7 +236,7 @@ static int rtp_new_av_stream(HTTPContext *c,
  HTTPContext *rtsp_c);

 static const char *my_program_name;
-
+static int ffserver_daemon;
 static int no_launch;
 static int need_to_start_children;

@@ -3671,6 +3671,7 @@ static void handle_child_exit(int sig)
 static void opt_debug(void)
 {
 config.debug = 1;
+ffserver_daemon = 1;
 snprintf(config.logfilename, sizeof(config.logfilename), "-");
 }

@@ -3694,7 +3695,7 @@ int main(int argc, char **argv)
 {
 struct sigaction sigact = { { 0 } };
 int ret = 0;
-
+ffserver_daemon = 1;
 config.filename = av_strdup("/etc/ffserver.conf");

 parse_loglevel(argc, argv, options);
@@ -3736,10 +3737,35 @@ int main(int argc, char **argv)
 build_feed_streams();

 compute_bandwidth();
-
+if(ffserver_daemon){
+int ffserver_id = 0;
+pid_t sid = 0;
+
+ffserver_id = fork(); // Create child process
+
+if (ffserver_id < 0){
+printf("fork failed!\n"); //Indication of failure
+exit(1);
+}
+
+if(ffserver_id > 0){ //Parent process need to kill
+exit(0);
+}
+
+sid = setsid(); //set new session
+if(sid < 0){
+exit(1); //return failure
+}
+
+open ("/dev/null", O_RDWR);
+
+if (strcmp(config.logfilename, "-") != 0) {
+close(1);
+}
+}
 /* signal init */
 signal(SIGPIPE, SIG_IGN);
-
+
 if (http_server() < 0) {
 http_log("Could not start server\n");
 exit(1);
diff --git a/ffserver_config.c b/ffserver_config.c
index e44cdf7..e2c78d8 100644
--- a/ffserver_config.c
+++ b/ffserver_config.c
@@ -358,8 +358,6 @@ static int ffserver_parse_config_global(FFServerConfig
*config, const char *cmd,
 ffserver_get_arg(arg, sizeof(arg), p);
 if (resolve_host(&config->http_addr.sin_addr, arg) != 0)
 ERROR("%s:%d: Invalid host/IP address: %s\n", arg);
-} else if (!av_strcasecmp(cmd, "NoDaemon")) {
-WARNING("NoDaemon option has no effect, you should remove it\n");
 } else if (!av_strcasecmp(cmd, "RTSPPort")) {
 ffserver_get_arg(arg, sizeof(arg), p);
 val = atoi(arg);
-- 
1.9.1


Cheers,
Binathi
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Re: [FFmpeg-devel] "OPW Qualification Task: Enable daemon mode for FFserver"

[Michael Niedermayer]

On Thu, Oct 30, 2014 at 02:14:59AM +0530, Binathi Bingi wrote:
> From 0fb7dcf1f126bd137e2b2025c5cd6cff4af65801 Mon Sep 17 00:00:00 2001
> From: Binathi Bingi 
> Date: Thu, 30 Oct 2014 01:14:08 +0530
> Subject: [PATCH] ffserver: enable back deamon mode
> 
> ---
>  ffserver.c| 34 ++
>  ffserver_config.c |  2 --
>  2 files changed, 30 insertions(+), 6 deletions(-)
> 
[...]
> diff --git a/ffserver_config.c b/ffserver_config.c
> index e44cdf7..e2c78d8 100644
> --- a/ffserver_config.c
> +++ b/ffserver_config.c
> @@ -358,8 +358,6 @@ static int ffserver_parse_config_global(FFServerConfig
> *config, const char *cmd,
>  ffserver_get_arg(arg, sizeof(arg), p);
>  if (resolve_host(&config->http_addr.sin_addr, arg) != 0)
>  ERROR("%s:%d: Invalid host/IP address: %s\n", arg);
> -} else if (!av_strcasecmp(cmd, "NoDaemon")) {
> -WARNING("NoDaemon option has no effect, you should remove it\n");
>  } else if (!av_strcasecmp(cmd, "RTSPPort")) {
>  ffserver_get_arg(arg, sizeof(arg), p);
>  val = atoi(arg);

this would break configuration files which contain the NoDaemon
option

[...]
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The real ebay dictionary, page 1
"Used only once"- "Some unspecified defect prevented a second use"
"In good condition" - "Can be repaird by experienced expert"
"As is" - "You wouldnt want it even if you were payed for it, if you knew ..."


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Re: [FFmpeg-devel] mxfenc: fix indentation after last commit

[Michael Niedermayer]

On Wed, Oct 29, 2014 at 08:26:25PM +, Thomas Mundt wrote:
> Should have done this as second part of my AVCI Patch. Sorry!
> 
> Regards,
> Thomas

>  mxfenc.c |   18 +-
>  1 file changed, 9 insertions(+), 9 deletions(-)
> bd730a65f3d562a29fd14a84d79e1104547a262b  indent.patch
>  libavformat/mxfenc.c | 18 +-
>  1 file changed, 9 insertions(+), 9 deletions(-)

applied

thanks

[...]

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[FFmpeg-devel] [PATCH 1/3] ffplay: implement separete audio decoder thread

[Marton Balint]

Signed-off-by: Marton Balint 
---
 ffplay.c | 265 ---
 1 file changed, 153 insertions(+), 112 deletions(-)

diff --git a/ffplay.c b/ffplay.c
index a979164..24bcae2 100644
--- a/ffplay.c
+++ b/ffplay.c
@@ -121,7 +121,8 @@ typedef struct PacketQueue {
 
 #define VIDEO_PICTURE_QUEUE_SIZE 3
 #define SUBPICTURE_QUEUE_SIZE 16
-#define FRAME_QUEUE_SIZE FFMAX(VIDEO_PICTURE_QUEUE_SIZE, SUBPICTURE_QUEUE_SIZE)
+#define SAMPLE_QUEUE_SIZE 9
+#define FRAME_QUEUE_SIZE FFMAX(SAMPLE_QUEUE_SIZE, 
FFMAX(VIDEO_PICTURE_QUEUE_SIZE, SUBPICTURE_QUEUE_SIZE))
 
 typedef struct AudioParams {
 int freq;
@@ -196,6 +197,7 @@ typedef struct Decoder {
 typedef struct VideoState {
 SDL_Thread *read_tid;
 SDL_Thread *video_tid;
+SDL_Thread *audio_tid;
 AVInputFormat *iformat;
 int no_background;
 int abort_request;
@@ -217,6 +219,7 @@ typedef struct VideoState {
 
 FrameQueue pictq;
 FrameQueue subpq;
+FrameQueue sampq;
 
 Decoder auddec;
 Decoder viddec;
@@ -242,8 +245,6 @@ typedef struct VideoState {
 unsigned int audio_buf1_size;
 int audio_buf_index; /* in bytes */
 int audio_write_buf_size;
-int audio_buf_frames_pending;
-int audio_last_serial;
 struct AudioParams audio_src;
 #if CONFIG_AVFILTER
 struct AudioParams audio_filter_src;
@@ -252,7 +253,6 @@ typedef struct VideoState {
 struct SwrContext *swr_ctx;
 int frame_drops_early;
 int frame_drops_late;
-AVFrame *frame;
 
 enum ShowMode {
 SHOW_MODE_NONE = -1, SHOW_MODE_VIDEO = 0, SHOW_MODE_WAVES, 
SHOW_MODE_RDFT, SHOW_MODE_NB
@@ -712,12 +712,29 @@ static Frame *frame_queue_peek_writable(FrameQueue *f)
 return &f->queue[f->windex];
 }
 
+static Frame *frame_queue_peek_readable(FrameQueue *f)
+{
+/* wait until we have a readable a new frame */
+SDL_LockMutex(f->mutex);
+while (f->size - f->rindex_shown <= 0 &&
+   !f->pktq->abort_request) {
+SDL_CondWait(f->cond, f->mutex);
+}
+SDL_UnlockMutex(f->mutex);
+
+if (f->pktq->abort_request)
+return NULL;
+
+return &f->queue[(f->rindex + f->rindex_shown) % f->max_size];
+}
+
 static void frame_queue_push(FrameQueue *f)
 {
 if (++f->windex == f->max_size)
 f->windex = 0;
 SDL_LockMutex(f->mutex);
 f->size++;
+SDL_CondSignal(f->cond);
 SDL_UnlockMutex(f->mutex);
 }
 
@@ -1280,6 +1297,7 @@ static void stream_close(VideoState *is)
 
 /* free all pictures */
 frame_queue_destory(&is->pictq);
+frame_queue_destory(&is->sampq);
 frame_queue_destory(&is->subpq);
 SDL_DestroyCond(is->continue_read_thread);
 #if !CONFIG_AVFILTER
@@ -2100,6 +2118,93 @@ end:
 }
 #endif  /* CONFIG_AVFILTER */
 
+static int audio_thread(void *arg)
+{
+VideoState *is = arg;
+AVFrame *frame = av_frame_alloc();
+Frame *af;
+#if CONFIG_AVFILTER
+int last_serial = -1;
+int64_t dec_channel_layout;
+int reconfigure;
+#endif
+int got_frame = 0;
+AVRational tb;
+int ret = 0;
+
+if (!frame)
+return AVERROR(ENOMEM);
+
+do {
+if ((got_frame = decoder_decode_frame(&is->auddec, frame, NULL)) < 0)
+goto the_end;
+
+if (got_frame) {
+tb = (AVRational){1, frame->sample_rate};
+
+#if CONFIG_AVFILTER
+dec_channel_layout = 
get_valid_channel_layout(frame->channel_layout, av_frame_get_channels(frame));
+
+reconfigure =
+cmp_audio_fmts(is->audio_filter_src.fmt, 
is->audio_filter_src.channels,
+   frame->format, 
av_frame_get_channels(frame))||
+is->audio_filter_src.channel_layout != dec_channel_layout 
||
+is->audio_filter_src.freq   != frame->sample_rate 
||
+is->auddec.pkt_serial   != last_serial;
+
+if (reconfigure) {
+char buf1[1024], buf2[1024];
+av_get_channel_layout_string(buf1, sizeof(buf1), -1, 
is->audio_filter_src.channel_layout);
+av_get_channel_layout_string(buf2, sizeof(buf2), -1, 
dec_channel_layout);
+av_log(NULL, AV_LOG_DEBUG,
+   "Audio frame changed from rate:%d ch:%d fmt:%s 
layout:%s serial:%d to rate:%d ch:%d fmt:%s layout:%s serial:%d\n",
+   is->audio_filter_src.freq, 
is->audio_filter_src.channels, 
av_get_sample_fmt_name(is->audio_filter_src.fmt), buf1, last_serial,
+   frame->sample_rate, av_frame_get_channels(frame), 
av_get_sample_fmt_name(frame->format), buf2, is->auddec.pkt_serial);
+
+is->audio_filter_src.fmt= frame->format;
+is->audio_filter_src.channels   = 
av_frame_get_channels(frame);
+is->audio_filter_src.channel_layout = dec_channel_layout;
+is->audio_filter_src.freq 

[FFmpeg-devel] [PATCH 2/3] ffplay: fix indentation after last commit

[Marton Balint]

Signed-off-by: Marton Balint 
---
 ffplay.c | 182 +++
 1 file changed, 89 insertions(+), 93 deletions(-)

diff --git a/ffplay.c b/ffplay.c
index 24bcae2..a3b34fd 100644
--- a/ffplay.c
+++ b/ffplay.c
@@ -2424,105 +2424,101 @@ static int audio_decode_frame(VideoState *is)
 int wanted_nb_samples;
 Frame *af;
 
-{
-if (is->paused)
-return -1;
+if (is->paused)
+return -1;
 
-do {
-if (!(af = frame_queue_peek_readable(&is->sampq)))
+do {
+if (!(af = frame_queue_peek_readable(&is->sampq)))
+return -1;
+frame_queue_next(&is->sampq);
+} while (af->serial != is->audioq.serial);
+
+data_size = av_samples_get_buffer_size(NULL, 
av_frame_get_channels(af->frame),
+   af->frame->nb_samples,
+   af->frame->format, 1);
+
+dec_channel_layout =
+(af->frame->channel_layout && av_frame_get_channels(af->frame) == 
av_get_channel_layout_nb_channels(af->frame->channel_layout)) ?
+af->frame->channel_layout : 
av_get_default_channel_layout(av_frame_get_channels(af->frame));
+wanted_nb_samples = synchronize_audio(is, af->frame->nb_samples);
+
+if (af->frame->format!= is->audio_src.fmt||
+dec_channel_layout   != is->audio_src.channel_layout ||
+af->frame->sample_rate   != is->audio_src.freq   ||
+(wanted_nb_samples   != af->frame->nb_samples && !is->swr_ctx)) {
+swr_free(&is->swr_ctx);
+is->swr_ctx = swr_alloc_set_opts(NULL,
+ is->audio_tgt.channel_layout, 
is->audio_tgt.fmt, is->audio_tgt.freq,
+ dec_channel_layout,   
af->frame->format, af->frame->sample_rate,
+ 0, NULL);
+if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) {
+av_log(NULL, AV_LOG_ERROR,
+   "Cannot create sample rate converter for conversion of %d 
Hz %s %d channels to %d Hz %s %d channels!\n",
+af->frame->sample_rate, 
av_get_sample_fmt_name(af->frame->format), av_frame_get_channels(af->frame),
+is->audio_tgt.freq, 
av_get_sample_fmt_name(is->audio_tgt.fmt), is->audio_tgt.channels);
+swr_free(&is->swr_ctx);
+return -1;
+}
+is->audio_src.channel_layout = dec_channel_layout;
+is->audio_src.channels   = av_frame_get_channels(af->frame);
+is->audio_src.freq = af->frame->sample_rate;
+is->audio_src.fmt = af->frame->format;
+}
+
+if (is->swr_ctx) {
+const uint8_t **in = (const uint8_t **)af->frame->extended_data;
+uint8_t **out = &is->audio_buf1;
+int out_count = (int64_t)wanted_nb_samples * is->audio_tgt.freq / 
af->frame->sample_rate + 256;
+int out_size  = av_samples_get_buffer_size(NULL, 
is->audio_tgt.channels, out_count, is->audio_tgt.fmt, 0);
+int len2;
+if (out_size < 0) {
+av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size() 
failed\n");
+return -1;
+}
+if (wanted_nb_samples != af->frame->nb_samples) {
+if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - 
af->frame->nb_samples) * is->audio_tgt.freq / af->frame->sample_rate,
+wanted_nb_samples * is->audio_tgt.freq 
/ af->frame->sample_rate) < 0) {
+av_log(NULL, AV_LOG_ERROR, "swr_set_compensation() failed\n");
 return -1;
-frame_queue_next(&is->sampq);
-} while (af->serial != is->audioq.serial);
-
-{
-data_size = av_samples_get_buffer_size(NULL, 
av_frame_get_channels(af->frame),
-   af->frame->nb_samples,
-   af->frame->format, 1);
-
-dec_channel_layout =
-(af->frame->channel_layout && av_frame_get_channels(af->frame) 
== av_get_channel_layout_nb_channels(af->frame->channel_layout)) ?
-af->frame->channel_layout : 
av_get_default_channel_layout(av_frame_get_channels(af->frame));
-wanted_nb_samples = synchronize_audio(is, af->frame->nb_samples);
-
-if (af->frame->format!= is->audio_src.fmt||
-dec_channel_layout   != is->audio_src.channel_layout ||
-af->frame->sample_rate   != is->audio_src.freq   ||
-(wanted_nb_samples   != af->frame->nb_samples && 
!is->swr_ctx)) {
-swr_free(&is->swr_ctx);
-is->swr_ctx = swr_alloc_set_opts(NULL,
- is->audio_tgt.channel_layout, 
is->audio_tgt.fmt, is->audio_tgt.freq,
- dec_channel_

[FFmpeg-devel] [PATCH 3/3] ffplay: only output null packet once on EOF

[Marton Balint]

The generic decoder properly flushes the codecs with one null packet as well.

Signed-off-by: Marton Balint 
---
 ffplay.c | 22 ++
 1 file changed, 10 insertions(+), 12 deletions(-)

diff --git a/ffplay.c b/ffplay.c
index a3b34fd..e04f622 100644
--- a/ffplay.c
+++ b/ffplay.c
@@ -3110,27 +3110,25 @@ static int read_thread(void *arg)
 goto fail;
 }
 }
-if (eof) {
-if (is->video_stream >= 0)
-packet_queue_put_nullpacket(&is->videoq, is->video_stream);
-if (is->audio_stream >= 0)
-packet_queue_put_nullpacket(&is->audioq, is->audio_stream);
-if (is->subtitle_stream >= 0)
-packet_queue_put_nullpacket(&is->subtitleq, 
is->subtitle_stream);
-SDL_Delay(10);
-eof=0;
-continue;
-}
 ret = av_read_frame(ic, pkt);
 if (ret < 0) {
-if (ret == AVERROR_EOF || avio_feof(ic->pb))
+if ((ret == AVERROR_EOF || avio_feof(ic->pb)) && !eof) {
+if (is->video_stream >= 0)
+packet_queue_put_nullpacket(&is->videoq, is->video_stream);
+if (is->audio_stream >= 0)
+packet_queue_put_nullpacket(&is->audioq, is->audio_stream);
+if (is->subtitle_stream >= 0)
+packet_queue_put_nullpacket(&is->subtitleq, 
is->subtitle_stream);
 eof = 1;
+}
 if (ic->pb && ic->pb->error)
 break;
 SDL_LockMutex(wait_mutex);
 SDL_CondWaitTimeout(is->continue_read_thread, wait_mutex, 10);
 SDL_UnlockMutex(wait_mutex);
 continue;
+} else {
+eof = 0;
 }
 /* check if packet is in play range specified by user, then queue, 
otherwise discard */
 stream_start_time = ic->streams[pkt->stream_index]->start_time;
-- 
1.8.4.5

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Re: [FFmpeg-devel] "OPW Qualification Task: Enable daemon mode for FFserver"

[Binathi Bingi]

If you check the latest ffserver.conf file on GIT [
http://git.videolan.org/?p=ffmpeg.git
],
there is NoDaemon option in it. So the patch is written as per latest
version. But if desired can include NoDameon option in ffserver.conf and
make futher changes.

Binathi

On Thu, Oct 30, 2014 at 3:43 AM, Michael Niedermayer 
wrote:

> On Thu, Oct 30, 2014 at 02:14:59AM +0530, Binathi Bingi wrote:
> > From 0fb7dcf1f126bd137e2b2025c5cd6cff4af65801 Mon Sep 17 00:00:00 2001
> > From: Binathi Bingi 
> > Date: Thu, 30 Oct 2014 01:14:08 +0530
> > Subject: [PATCH] ffserver: enable back deamon mode
> >
> > ---
> >  ffserver.c| 34 ++
> >  ffserver_config.c |  2 --
> >  2 files changed, 30 insertions(+), 6 deletions(-)
> >
> [...]
> > diff --git a/ffserver_config.c b/ffserver_config.c
> > index e44cdf7..e2c78d8 100644
> > --- a/ffserver_config.c
> > +++ b/ffserver_config.c
> > @@ -358,8 +358,6 @@ static int
> ffserver_parse_config_global(FFServerConfig
> > *config, const char *cmd,
> >  ffserver_get_arg(arg, sizeof(arg), p);
> >  if (resolve_host(&config->http_addr.sin_addr, arg) != 0)
> >  ERROR("%s:%d: Invalid host/IP address: %s\n", arg);
> > -} else if (!av_strcasecmp(cmd, "NoDaemon")) {
> > -WARNING("NoDaemon option has no effect, you should remove
> it\n");
> >  } else if (!av_strcasecmp(cmd, "RTSPPort")) {
> >  ffserver_get_arg(arg, sizeof(arg), p);
> >  val = atoi(arg);
>
> this would break configuration files which contain the NoDaemon
> option
>
> [...]
> --
> Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
>
> The real ebay dictionary, page 1
> "Used only once"- "Some unspecified defect prevented a second use"
> "In good condition" - "Can be repaird by experienced expert"
> "As is" - "You wouldnt want it even if you were payed for it, if you knew
> ..."
>
> ___
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> ffmpeg-devel@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>
>
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Re: [FFmpeg-devel] [PATCH] movenc: Always write the trun data_offset

[Michael Niedermayer]

On Tue, Oct 28, 2014 at 11:05:54PM +, Bryan Huh wrote:
> The MOV_TRUN_DATA_OFFSET flag was always getting set, so data_offset is
> being expected and read. The offset was computed correctly anyway during
> fragment-flush. Alternatively, I could have selectively set the flag
> only when base-data-offset is defined, but certain players (Chrome)
> seem to fail when no data-offset is provided in the trun.
> ---
>  libavformat/movenc.c | 9 ++---
>  1 file changed, 2 insertions(+), 7 deletions(-)
> 
> diff --git a/libavformat/movenc.c b/libavformat/movenc.c
> index a43752a..923cf92 100644
> --- a/libavformat/movenc.c
> +++ b/libavformat/movenc.c
> @@ -3187,13 +3187,8 @@ static int mov_write_trun_tag(AVIOContext *pb, 
> MOVMuxContext *mov,
>  avio_wb24(pb, flags);
>  
>  avio_wb32(pb, track->entry); /* sample count */
> -if (mov->flags & FF_MOV_FLAG_OMIT_TFHD_OFFSET &&
> -!(mov->flags & FF_MOV_FLAG_SEPARATE_MOOF) &&
> -track->track_id != 1)
> -avio_wb32(pb, 0); /* Later tracks follow immediately after the 
> previous one */
> -else
> -avio_wb32(pb, moof_size + 8 + track->data_offset +
> -  track->cluster[0].pos); /* data offset */
> +avio_wb32(pb, moof_size + 8 + track->data_offset +
> +  track->cluster[0].pos); /* data offset */
>  if (flags & MOV_TRUN_FIRST_SAMPLE_FLAGS)
>  avio_wb32(pb, get_sample_flags(track, &track->cluster[0]));

martin made some changes to this code, which conflict with this
patch, is this patch still needed ?

[...]
-- 
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB

The real ebay dictionary, page 3
"Rare item" - "Common item with rare defect or maybe just a lie"
"Professional" - "'Toy' made in china, not functional except as doorstop"
"Experts will know" - "The seller hopes you are not an expert"


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Re: [FFmpeg-devel] "OPW Qualification Task: Enable daemon mode for FFserver"

[Michael Niedermayer]

On Thu, Oct 30, 2014 at 07:49:10AM +0530, Binathi Bingi wrote:
> If you check the latest ffserver.conf file on GIT [
> http://git.videolan.org/?p=ffmpeg.git
> ],
> there is NoDaemon option in it. So the patch is written as per latest
> version. But if desired can include NoDameon option in ffserver.conf and
> make futher changes.

ffserver must work with any config file it worked with previously
not only the ffserver.conf file from git

[...]
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Those who are best at talking, realize last or never when they are wrong.


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Re: [FFmpeg-devel] [PATCH 2/2] avcodec/libutvideodec: Support YUV422P10

[Jan Ehrhardt]

Michael Niedermayer in gmane.comp.video.ffmpeg.devel (Wed, 29 Oct 2014
13:20:58 +0100):
>should be fixed

Ack. Thanks.

Jan

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Re: [FFmpeg-devel] [PATCH] movenc: Always write the trun data_offset

[Bryan Huh]

Yes. Even with his change to the if-statement, I am proposing the same
change: to remove the if-statement.

It would be nice to get this sanity-checked by Martin though.

On Wed, Oct 29, 2014 at 7:24 PM, Michael Niedermayer 
wrote:

> On Tue, Oct 28, 2014 at 11:05:54PM +, Bryan Huh wrote:
> > The MOV_TRUN_DATA_OFFSET flag was always getting set, so data_offset is
> > being expected and read. The offset was computed correctly anyway during
> > fragment-flush. Alternatively, I could have selectively set the flag
> > only when base-data-offset is defined, but certain players (Chrome)
> > seem to fail when no data-offset is provided in the trun.
> > ---
> >  libavformat/movenc.c | 9 ++---
> >  1 file changed, 2 insertions(+), 7 deletions(-)
> >
> > diff --git a/libavformat/movenc.c b/libavformat/movenc.c
> > index a43752a..923cf92 100644
> > --- a/libavformat/movenc.c
> > +++ b/libavformat/movenc.c
> > @@ -3187,13 +3187,8 @@ static int mov_write_trun_tag(AVIOContext *pb,
> MOVMuxContext *mov,
> >  avio_wb24(pb, flags);
> >
> >  avio_wb32(pb, track->entry); /* sample count */
> > -if (mov->flags & FF_MOV_FLAG_OMIT_TFHD_OFFSET &&
> > -!(mov->flags & FF_MOV_FLAG_SEPARATE_MOOF) &&
> > -track->track_id != 1)
> > -avio_wb32(pb, 0); /* Later tracks follow immediately after the
> previous one */
> > -else
> > -avio_wb32(pb, moof_size + 8 + track->data_offset +
> > -  track->cluster[0].pos); /* data offset */
> > +avio_wb32(pb, moof_size + 8 + track->data_offset +
> > +  track->cluster[0].pos); /* data offset */
> >  if (flags & MOV_TRUN_FIRST_SAMPLE_FLAGS)
> >  avio_wb32(pb, get_sample_flags(track, &track->cluster[0]));
>
> martin made some changes to this code, which conflict with this
> patch, is this patch still needed ?
>
> [...]
> --
> Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
>
> The real ebay dictionary, page 3
> "Rare item" - "Common item with rare defect or maybe just a lie"
> "Professional" - "'Toy' made in china, not functional except as doorstop"
> "Experts will know" - "The seller hopes you are not an expert"
>
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Re: [FFmpeg-devel] [PATCH] movenc: Always write the trun data_offset

[Michael Niedermayer]

On Wed, Oct 29, 2014 at 08:49:11PM -0700, Bryan Huh wrote:
> Yes. Even with his change to the if-statement, I am proposing the same
> change: to remove the if-statement.
> 
> It would be nice to get this sanity-checked by Martin though.

ok, cc-ing martin


> 
> On Wed, Oct 29, 2014 at 7:24 PM, Michael Niedermayer 
> wrote:
> 
> > On Tue, Oct 28, 2014 at 11:05:54PM +, Bryan Huh wrote:
> > > The MOV_TRUN_DATA_OFFSET flag was always getting set, so data_offset is
> > > being expected and read. The offset was computed correctly anyway during
> > > fragment-flush. Alternatively, I could have selectively set the flag
> > > only when base-data-offset is defined, but certain players (Chrome)
> > > seem to fail when no data-offset is provided in the trun.
> > > ---
> > >  libavformat/movenc.c | 9 ++---
> > >  1 file changed, 2 insertions(+), 7 deletions(-)
> > >
> > > diff --git a/libavformat/movenc.c b/libavformat/movenc.c
> > > index a43752a..923cf92 100644
> > > --- a/libavformat/movenc.c
> > > +++ b/libavformat/movenc.c
> > > @@ -3187,13 +3187,8 @@ static int mov_write_trun_tag(AVIOContext *pb,
> > MOVMuxContext *mov,
> > >  avio_wb24(pb, flags);
> > >
> > >  avio_wb32(pb, track->entry); /* sample count */
> > > -if (mov->flags & FF_MOV_FLAG_OMIT_TFHD_OFFSET &&
> > > -!(mov->flags & FF_MOV_FLAG_SEPARATE_MOOF) &&
> > > -track->track_id != 1)
> > > -avio_wb32(pb, 0); /* Later tracks follow immediately after the
> > previous one */
> > > -else
> > > -avio_wb32(pb, moof_size + 8 + track->data_offset +
> > > -  track->cluster[0].pos); /* data offset */
> > > +avio_wb32(pb, moof_size + 8 + track->data_offset +
> > > +  track->cluster[0].pos); /* data offset */
> > >  if (flags & MOV_TRUN_FIRST_SAMPLE_FLAGS)
> > >  avio_wb32(pb, get_sample_flags(track, &track->cluster[0]));
> >
> > martin made some changes to this code, which conflict with this
> > patch, is this patch still needed ?
> >
> > [...]
> > --
> > Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
> >
> > The real ebay dictionary, page 3
> > "Rare item" - "Common item with rare defect or maybe just a lie"
> > "Professional" - "'Toy' made in china, not functional except as doorstop"
> > "Experts will know" - "The seller hopes you are not an expert"
> >
> ___
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> ffmpeg-devel@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
> 

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When the tyrant has disposed of foreign enemies by conquest or treaty, and
there is nothing more to fear from them, then he is always stirring up
some war or other, in order that the people may require a leader. -- Plato


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[FFmpeg-devel] Encode2() function in libavcodec/avcodec.h

[greeshma]

Hi,

the attributes has been changed to (AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr);

what is the exact change made in the function.
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