Re: [FFmpeg-user] LATM demuxer

[Carl Eugen Hoyos]

2017-10-31 15:54 GMT+01:00 Andy Furniss :
> Sometime this year I thought I saw mention of ffmpeg and a
> LATM standalone demuxer.

I committed it six years ago.
(The technical name of the file format is "loas", "latm" is one
of the things you can put into loas.)

Your question was possibly: "Why does fdk not decode latm?"
The answer is related to a decision FFmpeg developers made
long ago (they knew it wasn't a very good idea), not
implementing a bitstream filter that converts latm to aac and
allowing to decode the result with any aac decoder but
instead implementing an assigned latm decoder that cannot
be replaced so easily.

Carl Eugen

PS: Concerning "clean-up": Instead of fixing issues like the
above, known for nearly a decade, for example configure
was changed so that on some systems, building takes
(literally!) hours instead of minutes now.
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Re: [FFmpeg-user] Read ffmpeg segment on stdout

[Carl Eugen Hoyos]

2017-10-31 23:27 GMT+01:00 Moritz Barsnick :
> On Tue, Oct 31, 2017 at 01:55:49 +0100, Alberto Esposito wrote:

>> 2) Otherwise, how can I detect a new keyframe just by
>> reading the byte stream?
>
> You mean from a non-segmented stream? First, the important
> question: What for? (The answer to your question is: It
> depends on the codec, possibly on the container, I would say.)

And valid, playable streams without keyframes exist.

Carl Eugen
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Re: [FFmpeg-user] Read ffmpeg segment on stdout

[Moritz Barsnick]

On Tue, Oct 31, 2017 at 01:55:49 +0100, Alberto Esposito wrote:

> but if I enable segmentation it doesn't work:
[...]
> Output #0, segment, to 'pipe:1':
> Output file #0 does not contain any stream

For some reason, the segment muxer requires you to map something
explicitly. Add "-map 0" (or whatever streams you want) and that error
should go away.

> 1) Can segmented files be written to stdout? Is there a special character
> for file separation?

Eh, no. The point of the segment muxer is to split up the stream into
separate files. It takes a filename template to name those files, and
can therefore not write to special files such as pipes. (Protocols such
as ftp should work, I believe that was changed a while ago.)

What does your browser do? Do you have a special JavaScript (or
something) implementation which reads the segments and displays them?

What are you actually trying to do?

> 2) Otherwise, how can I detect a new keyframe just by reading the byte
> stream?

You mean from a non-segmented stream? First, the important question:
What for? (The answer to your question is: It depends on the codec,
possibly on the container, I would say.)

For streaming to a browser, I would say HLS is the best solution. Serve
up the listfile (m3u8) and the segments from a werserver, or directly
from disk.

Moritz
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Re: [FFmpeg-user] Compiling ffmpeg

[Gonzalo Garramuño]

El 30/10/17 a las 16:09, Legault, Phillip [GTSUS] escribió:

./configure --enable-gpl --enable-libfdk_aac --enable-libmp3lame 
--enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 
--enable-nonfree --disable-static --enable-shared
getting error
ERROR: libtheora not found


As the message says, you need libtheora.  On Ubuntu, for example, you'd do:

$ sudo apt-get install libtheora0

and possibly:

$ sudo apt-get install libtheora-dev

--
Gonzalo Garramuño

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Re: [FFmpeg-user] QSV Hevc to HLS not segmenting?

[Markku Vainio]

>> https://software.intel.com/en-us/forums/intel-media-sdk/topic/746642
>>
>> https://github.com/libav/libav/commit/98afe3fb71afd4a18009924aaba56bc
>> 577bbd400.patch
>
> Will you send this patch to the development mailing list?
https://lists.ffmpeg.org/pipermail/ffmpeg-devel/2017-July/213272.html

i've send a patch some month ago, maybe Mark have some better way to fix it, 
so. Markku can test on newest version.

>
> Please cut your quotes, Carl Eugen

https://github.com/libav/libav/commit/98afe3fb71afd4a18009924aaba56bc577bbd400.patch

I tested this one (manually edited qsvenc_hevc.c) from Intel site since it 
makes HEVC_QSV IDR placement uniform with other hevc encoders and doesn’t 
require user input.

Works like a charm for me on Intel Skull Canyon with Centos 7

Markku.
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[FFmpeg-user] Read ffmpeg segment on stdout

[Alberto Esposito]

Hello everybody,

I'm trying to segment a video file to play it in the browser. Ideally, I
would like to read the stdout and pipe it to the browser, but if I enable
segmentation it doesn't work:


$ ffmpeg -i test.mp4 -f segment -segment_format mp4 -segment_format_options
movflags=+faststart+frag_keyframe+empty_moov+default_base_moof pipe:1
...
Output #0, segment, to 'pipe:1':
Output file #0 does not contain any stream


1) Can segmented files be written to stdout? Is there a special character
for file separation?

2) Otherwise, how can I detect a new keyframe just by reading the byte
stream?

thank you guys :)
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Re: [FFmpeg-user] LATM demuxer

[Andy Furniss]

Andy Furniss wrote:
Sometime this year I thought I saw mention of ffmpeg and a LATM 
standalone demuxer. I may mis-recall/ may have misread, does anyone

know anything about this?

The reason I would need it is that currently feeding latm to decode
with fdk seems impossible.

Re-coding seems to lose the normal drc and the dvb (drc and mixdown) 
meta that fdk seems to have options to handle.


In the absence of being able to do -c:a copy = the AAC re-coding
case. Is there is a way not to loose meta data?

TIA


Hmm, it's years since I looked at this last and then IIRC it was stated
that fdk lib didn't do latm.
Looking at latest git there are files named like it possibly does now.

Whatever, it doesn't work, so here's some output working without then
failing with libfdk_aac.

ffmpeg -i ~/bp-part-c1.ts -f null -
ffmpeg version N-88204-g7f9de7b Copyright (c) 2000-2017 the FFmpeg 
developers

  built with gcc 5.3.0 (GCC)
  configuration: --prefix=/usr --disable-doc --enable-gpl 
--enable-nonfree --enable-libvpx --enable-libx265 --enable-libx264 
--enable-gnutls --enable-libdrm --enable-libfreetype --enable-libfdk-aac

  libavutil  56.  0.100 / 56.  0.100
  libavcodec 58.  0.101 / 58.  0.101
  libavformat58.  0.101 / 58.  0.101
  libavdevice58.  0.100 / 58.  0.100
  libavfilter 7.  0.100 /  7.  0.100
  libswscale  5.  0.100 /  5.  0.100
  libswresample   3.  0.100 /  3.  0.100
  libpostproc55.  0.100 / 55.  0.100
[h264 @ 0x389cc80] mmco: unref short failure
Last message repeated 1 times
[mpegts @ 0x38983a0] PES packet size mismatch
Input #0, mpegts, from '/home/andy/bp-part-c1.ts':
  Duration: 00:24:18.20, start: 68053.963267, bitrate: 6403 kb/s
  Program 17540
Metadata:
  service_name: BBC ONE HD
  service_provider:
Stream #0:0[0x19c9]: Video: h264 (High) ([27][0][0][0] / 0x001B), 
yuv420p(tv, bt709, progressive), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 
25 tbr, 90k tbn, 50 tbc
Stream #0:1[0x19ca](eng): Audio: aac_latm (LC) ([17][0][0][0] / 
0x0011), 48000 Hz, 5.1, fltp
Stream #0:2[0x19ce](eng): Audio: aac_latm (HE-AACv2) ([17][0][0][0] 
/ 0x0011), 48000 Hz, stereo, fltp
Stream #0:3[0x19cd](eng): Subtitle: dvb_subtitle ([6][0][0][0] / 
0x0006)

Stream mapping:
  Stream #0:0 -> #0:0 (h264 (native) -> wrapped_avframe (native))
  Stream #0:1 -> #0:1 (aac_latm (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[h264 @ 0x38cd2c0] mmco: unref short failure
[h264 @ 0x391f660] mmco: unref short failure
Output #0, null, to 'pipe:':
  Metadata:
encoder : Lavf58.0.101
Stream #0:0: Video: wrapped_avframe, yuv420p, 1920x1080 [SAR 1:1 
DAR 16:9], q=2-31, 200 kb/s, 25 fps, 25 tbn, 25 tbc

Metadata:
  encoder : Lavc58.0.101 wrapped_avframe
Stream #0:1(eng): Audio: pcm_s16le, 48000 Hz, 5.1, s16, 4608 kb/s
Metadata:
  encoder : Lavc58.0.101 pcm_s16le
frame=  936 fps=267 q=-0.0 Lsize=N/A time=00:00:38.84 bitrate=N/A 
speed=11.1x
video:483kB audio:21492kB subtitle:0kB other streams:0kB global 
headers:0kB muxing overhead: unknown




ffmpeg -c:a libfdk_aac -i ~/bp-part-c1.ts -f null -
ffmpeg version N-88204-g7f9de7b Copyright (c) 2000-2017 the FFmpeg 
developers

  built with gcc 5.3.0 (GCC)
  configuration: --prefix=/usr --disable-doc --enable-gpl 
--enable-nonfree --enable-libvpx --enable-libx265 --enable-libx264 
--enable-gnutls --enable-libdrm --enable-libfreetype --enable-libfdk-aac

  libavutil  56.  0.100 / 56.  0.100
  libavcodec 58.  0.101 / 58.  0.101
  libavformat58.  0.101 / 58.  0.101
  libavdevice58.  0.100 / 58.  0.100
  libavfilter 7.  0.100 /  7.  0.100
  libswscale  5.  0.100 /  5.  0.100
  libswresample   3.  0.100 /  3.  0.100
  libpostproc55.  0.100 / 55.  0.100
[h264 @ 0x2d73c80] mmco: unref short failure
Last message repeated 1 times
[libfdk_aac @ 0x2d74d80] aacDecoder_DecodeFrame() failed: 4002
[libfdk_aac @ 0x2d74d80] aacDecoder_DecodeFrame() failed: 4007
[libfdk_aac @ 0x2d74d80] aacDecoder_DecodeFrame() failed: 4006
[libfdk_aac @ 0x2d74d80] aacDecoder_DecodeFrame() failed: 400a
[libfdk_aac @ 0x2d74d80] aacDecoder_DecodeFrame() failed: 1001
[libfdk_aac @ 0x2d74d80] aacDecoder_DecodeFrame() failed: 4006
Last message repeated 1 times
[libfdk_aac @ 0x2d74d80] aacDecoder_DecodeFrame() failed: 4002
Last message repeated 1 times
[libfdk_aac @ 0x2d74d80] aacDecoder_DecodeFrame() failed: 1001
[libfdk_aac @ 0x2d74d80] Multiple frames in a packet.
[libfdk_aac @ 0x2d74d80] aacDecoder_DecodeFrame() failed: 4002
[libfdk_aac @ 0x2d74d80] aacDecoder_DecodeFrame() failed: 4007
[libfdk_aac @ 0x2d74d80] aacDecoder_DecodeFrame() failed: 4002
[libfdk_aac @ 0x2d74d80] aacDecoder_DecodeFrame() failed: 1001
[libfdk_aac @ 0x2d74d80] aacDecoder_DecodeFrame() failed: 4007
[libfdk_aac @ 0x2d74d80] aacDecoder_DecodeFrame() failed: 1001
[libfdk_aac @ 0x2d74d80] aacDecoder_DecodeFrame() failed: 4002
[libfdk_aac @ 0x2d74d80] aacDecoder_DecodeFrame() failed: 1001

[FFmpeg-user] LATM demuxer

[Andy Furniss]

Sometime this year I thought I saw mention of ffmpeg and a LATM
standalone demuxer.
I may mis-recall/ may have misread, does anyone know anything about this?

The reason I would need it is that currently feeding latm to decode with
fdk seems impossible.

Re-coding seems to lose the normal drc and the dvb (drc and mixdown)
meta that fdk seems to have options to handle.

In the absence of being able to do -c:a copy = the AAC re-coding case.
Is there is a way not to loose meta data?

TIA
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Re: [FFmpeg-user] Problem in concatenation

[Gabriele Greco]

>
>
>
> This video was written with an old version of FFmpeg.
>
> I suggest you test current FFmpeg git head and report back.
>
>
Current git head works (but the one i had before do not), and release/3.4
works too, any older 3.x versions do not work (the second video is broken).

I'm trying to move my base to release/3.4, but I still do not know what
kind of problem it was! I hope not to introduce too many regressions,
thanks for the support!

--
Bye,
 Gabry
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