Re: [Libav-user] Pushing RTMPS stream to a streaming server

[black copper]

>
>
> I want to push an RTMPS stream to Wowza streaming server, from inside an
> ARM device (IP Camera).
>
> I can push RTMP stream like this:
>
> ffmpeg  -rtsp_transport tcp -i rtsp://127.0.0.1/channel1 -c copy -f flv
> rtmp://[ip]:1935/live/rtmp_test
>
> This works perfectly fine. Now, I have read in FFmpeg docs:
>
> https://www.ffmpeg.org/ffmpeg-protocols.html#tls
>
> that I can actually push an stream using a certificate.
>
> I would like to know if anyone has already done this before, as with
> documentation so far (both Wowza and FFmpeg), I am still struggling to get
> this working.
>
>

Hi All

*bump email*
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[Libav-user] Pushing RTMPS stream to a streaming server

[black copper]

Hi

I want to push an RTMPS stream to Wowza streaming server, from inside an
ARM device (IP Camera).

I can push RTMP stream like this:

ffmpeg  -rtsp_transport tcp -i rtsp://127.0.0.1/channel1 -c copy -f flv
rtmp://[ip]:1935/live/rtmp_test

This works perfectly fine. Now, I have read in FFmpeg docs:

https://www.ffmpeg.org/ffmpeg-protocols.html#tls

that I can actually push an stream using a certificate.

I would like to know if anyone has already done this before, as with
documentation so far (both Wowza and FFmpeg), I am still struggling to get
this working.

Thanks
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Re: [Libav-user] Issue trying to build FFmpeg for ARM with librtmp support

[black copper]

>
>
> It appears you have an issue cross-compiling librtmp,
> maybe you should ask there?
>
>
Hi:

Yes, I have already asked same on librtmp mailing list + forum. Just
thought maybe someone in this mailing list has already experienced, so
could guide better.

Thanks
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[Libav-user] Issue trying to build FFmpeg for ARM with librtmp support

[black copper]

Hi,



I am trying to compile librtmp so I can build FFmpeg with RTMP support for
ARM processor.



I already have the toolchain, and solo build of FFmpeg was also successful,
and testing from inside the ARM processor was success as well.



My understanding:

-  Ffmpeg

o   Librtmp

§  Openssl

§  zlib

This hierarchy is required to build FFmepg.



So far I have built  openssl for ARM, and zlib for ARM, and, I can see it
is located in right ARM output folder.



Prerequisites:



export LD_LIBRARY_PATH=/opt/toolchain_gnueabi-4.4.0_ARMv5TE/usr/lib/

export CCPREFIX="/opt/toolchain_gnueabi-4.4.0_ARMv5TE/usr/bin/arm-
unknown-linux-uclibcgnueabi-"

export CFLAGS="-I/opt/toolchain_gnueabi-4.4.0_ARMv5TE/usr/include"

export LDFLAGS="-L/opt/toolchain_gnueabi-4.4.0_ARMv5TE/usr/lib/"



1-  Steps to build zlib:



CC=arm-linux-gcc



./configure --prefix=/opt/toolchain_gnueabi-4.4.0_ARMv5TE/usr

Make

Make install



2-  Steps to build openssl:



export cross=arm-linux-

./Configure dist --prefix=/opt/toolchain_gnueabi-4.4.0_ARMv5TE/usr

make CC="${cross}gcc" AR="${cross}ar r" RANLIB="${cross}ranlib"

make install





3-  Steps to build librtmp:



make CROSS_COMPILE=arm-linux-
INC=-I/opt/toolchain_gnueabi-4.4.0_ARMv5TE/usr/include
LIB=-L/opt/toolchain_gnueabi-4.4.0_ARMv5TE/usr/lib



above 1,2 steps are successful, with 3rd, I get this:



make CROSS_COMPILE=arm-linux-
INC=-I/opt/toolchain_gnueabi-4.4.0_ARMv5TE/usr/include
LIB=-L/opt/toolchain_gnueabi-4.4.0_ARMv5TE/usr/lib

make[1]: Entering directory '/home/user/Downloads/ip_code/rtmpdump/librtmp'

arm-linux-gcc -shared -Wl,-soname,librtmp.so.1  -o librtmp.so.1 rtmp.o
log.o amf.o hashswf.o parseurl.o  -lssl -lcrypto -lz

/opt/toolchain_gnueabi-4.4.0_ARMv5TE/usr/bin/../lib/gcc/arm-
unknown-linux-uclibcgnueabi/4.4.0/../../../../arm-unknown-linux-uclibcgnueabi/bin/ld:
cannot find -lssl

/opt/toolchain_gnueabi-4.4.0_ARMv5TE/usr/bin/../lib/gcc/arm-
unknown-linux-uclibcgnueabi/4.4.0/../../../../arm-unknown-linux-uclibcgnueabi/bin/ld:
cannot find -lcrypto

/opt/toolchain_gnueabi-4.4.0_ARMv5TE/usr/bin/../lib/gcc/arm-
unknown-linux-uclibcgnueabi/4.4.0/../../../../arm-unknown-linux-uclibcgnueabi/bin/ld:
cannot find -lz

collect2: ld returned 1 exit status

Makefile:92: recipe for target 'librtmp.so.1' failed

make[1]: *** [librtmp.so.1] Error 1

make[1]: Leaving directory '/home/user/Downloads/ip_code/rtmpdump/librtmp'

Makefile:76: recipe for target 'librtmp/librtmp.a' failed

make: *** [librtmp/librtmp.a] Error 2





but in the output folder I can see the right files are there:







[user@localhost rtmpdump]$ cd /opt/toolchain_gnueabi-4.4.0_ARMv5TE/usr/lib

[user@localhost lib]$ ls

bin  libavcodec.a  libgmp.so.10.0.2  libz.so

certslibavdevice.a libiberty.a   libz.so.1

engines  libavfilter.a libmpfr.lalibz.so.1.2.11

gcc  libavformat.a libmpfr.soman

include  libavutil.a   libmpfr.so.4  misc

ldscriptslibcrypto.a   libmpfr.so.4.0.1  openssl.cnf

lib  libfakeroot-0.so  libpostproc.a pkgconfig

libaacplus.a libfakeroot.lalibssl.a  private

libaacplus.lalibfakeroot.solibswresample.a   share

libaacplus.solibgmp.la libswscale.a

libaacplus.so.2  libgmp.so libx264.a

libaacplus.so.2.0.2  libgmp.so.10  libz.a

[user@localhost lib]$



Any idea how to compile?



Thanks
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Re: [Libav-user] MP4 concatenation with MP42 major_brand

[black copper]

Le decadi 30 frimaire, an CCXXV, black copper a écrit :
> I have MP4 source videos, that contain:
>
> video: H264
> audio: G.711 (mulaw)
>
> I want to use concatenation on source videos to get one MP4 file.
>
> I used concat filter by first creating a list: mylist.txt file as follows:
>
> file 'v1.mp4'
> file 'v2.mp4'
> file 'v3.mp4'
>
> then envoked ffmpeg command like this:
>
> ffmpeg -f concat -i mylist.txt -c copy op.mp4 -y
>
> This worked only with files that are mp41 compatible.
>
> With mp42 files that I mentioned above, I get this error:
>
> [mp4 @ 0x24ca780] Could not find tag for codec pcm_mulaw in stream #1,
> codec not currently supported in container
> Could not write header for output file #0 (incorrect codec parameters ?):
> Invalid argument

This message is about output, not input. Please test remuxing a single
file, without concat, using the first file in each sequence:

ffmpeg -i input.mp4 -c copy output.mp4


>> yes, its giving the same error...

If, as I suspect, it fails the same way, your problem is exactly what is
written in the second line of the error message.

Also, note that you posted on the wrong mailing-list. Adding ffmpeg-user
as recipient, please reply only there.

>> did that, awaiting moderator approval - thanks,
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[Libav-user] MP4 concatenation with MP42 major_brand

[black copper]

I have MP4 source videos, that contain:

video: H264
audio: G.711 (mulaw)

I want to use concatenation on source videos to get one MP4 file.

I used concat filter by first creating a list: mylist.txt file as follows:

file 'v1.mp4'
file 'v2.mp4'
file 'v3.mp4'

then envoked ffmpeg command like this:

ffmpeg -f concat -i mylist.txt -c copy op.mp4 -y

This worked only with files that are mp41 compatible.

With mp42 files that I mentioned above, I get this error:

[mp4 @ 0x24ca780] Could not find tag for codec pcm_mulaw in stream #1,
codec not currently supported in container
Could not write header for output file #0 (incorrect codec parameters ?):
Invalid argument

Full command line:


ffmpeg -f concat -i mylist.txt -c copy op.mp4 -y
ffmpeg version N-82760-g55affd9 Copyright (c) 2000-2016 the FFmpeg
developers
  built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3)
  configuration: --pkg-config-flags=--static
--extra-cflags=-I/home/ubuntu/ffmpeg_build/include
--extra-ldflags=-L/home/ubuntu/ffmpeg_build/lib --enable-gpl
--enable-libass --enable-libfdk-aac --enable-libfreetype
--enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis
--enable-libvpx --enable-libx264 --enable-libx265 --enable-nonfree
  libavutil  55. 41.101 / 55. 41.101
  libavcodec 57. 66.109 / 57. 66.109
  libavformat57. 58.101 / 57. 58.101
  libavdevice57.  2.100 / 57.  2.100
  libavfilter 6. 68.100 /  6. 68.100
  libswscale  4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc54.  2.100 / 54.  2.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x23ed7c0] Auto-inserting h264_mp4toannexb
bitstream filter
Guessed Channel Layout for Input Stream #0.0 : mono
Input #0, concat, from 'mylist.txt':
  Duration: N/A, start: 0.00, bitrate: 518 kb/s
Stream #0:0(eng): Audio: pcm_mulaw (ulaw / 0x77616C75), 8000 Hz, mono,
s16, 64 kb/s
Metadata:
  creation_time   : 2016-12-20T03:41:14.00Z
  handler_name: ?Apple Sound Media Handler
Stream #0:1(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p,
1920x1080, 454 kb/s, 27.70 fps, 27.70 tbr, 90k tbn, 180k tbc
Metadata:
  creation_time   : 2016-12-20T03:41:14.00Z
  handler_name: ?Apple Video Media Handler
  encoder : H.264
[mp4 @ 0x2413780] Could not find tag for codec pcm_mulaw in stream #1,
codec not currently supported in container
Could not write header for output file #0 (incorrect codec parameters ?):
Invalid argument
Stream mapping:
  Stream #0:1 -> #0:0 (copy)
  Stream #0:0 -> #0:1 (copy)
Last message repeated 1 times


I'm on Ubuntu 14.04 OS.

Is there anyway I can get this functionality to work with mp42 compatible
files? any help is much appreciated,

Thanks,
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Re: [Libav-user] FFmpeg restream MP4 video automatically

[black copper]

Thank you,

-stream_loop works as suggested!...

On Fri, Sep 16, 2016 at 3:15 PM, Paul B Mahol  wrote:

> On 9/16/16, Igor Gulyaev  wrote:
> > Unfortunately ffmpeg doesn't have loop option for video files, but
> desired
> > behavior can be accomplished using filters.
>
> There is -stream_loop.
>
> >
> > Please check this ling
> > http://video.stackexchange.com/questions/12905/repeat-
> loop-input-video-with-ffmpeg
> >
>
> Mentioned above too.
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[Libav-user] FFmpeg restream MP4 video automatically

[black copper]

I am streaming RTSP with command line something like this:

ffmpeg -re -i ~/Downloads/test.mp4 -c copy -f rtsp rtsp://
127.0.0.1:1935/livetest/stream

Original video length is about 15 minutes.

I want to automatically restart once file is finished - sort of like in a
continuous loop.

Does FFmpeg command line give me this option? Or do I need to implement
some other way?

Thanks
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Re: [Libav-user] Convert audio to G711 raw format

[black copper]

>
>
> $ ffmpeg -i input -f alaw out1
> $ ffmpeg -i input -f mulaw out2
>
> or:
>
> $ ffmpeg -i input out.al
> $ ffmpeg -i input out.ul
>
> Carl Eugen
>
>
Thanks - just what I wanted!
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[Libav-user] Reading from IP Camera (RTSP) using FFmpeg results in 5xx Server Error

[black copper]

I have a scenario where I am trying to read from IP camera using RTSP
method.

On my development PC everything is working fine (rtsp url is something
like: rtsp://admin:12345@192.168.1.60:554, but when I try to test it over a
friend's place, he has different camera (Axis IP camera), and the url fails
to open.

self generated log is something like:

Error: avformat_open_input()
error message: Server returned 5XX Server Error reply

I tried opening it using HTTP tunneling, but this time I'm getting a
different error:

error message: Error number -5 occurred

The latter error message even appears on my dev PC (just to mention that I
tried this as well); so HTTP tunnel is no good.

Any one knows how to fix 5XX Server Error?

Thanks,
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Re: [Libav-user] avformat_alloc_output_context2 with WEBM defaults to VP9/Opus now - need to change

[black copper]

Sorry for the delay:


> The output context only has suggested default codecs. You can just
> change them to whatever you want, you don't have to follow the default
> (ie. just don't  listen to AVCodec->video_codec/audio_codec, but set
> your own.
>

I have patched my code with something like:

avformat_alloc_output_context2(>ocVid, opfmt, NULL,
this->filenameVid.c_str());
if ( ocVid->oformat->audio_codec == AV_CODEC_ID_OPUS )
{
ocVid->oformat->audio_codec = AV_CODEC_ID_VORBIS;
}
if ( ocVid->oformat->video_codec == AV_CODEC_ID_VP9 )
{
ocVid->oformat->video_codec = AV_CODEC_ID_VP8;
}
this->fmtVid = this->ocVid->oformat;

This did the work for me; however, default VLC or Media Player Classic is
unable to play it properly - however, it does playback fine on firefox
browser, and also with FFPlay.

Both VLC and MPC should not show this problem, as they were working
properly before... I wonder what is wrong..
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Re: [Libav-user] libx264 library direct usage vs. FFmpeg wrapper routines

[black copper]

Hi,

Has anyone tried a similar exercise? Would really appreciate any cues from
everyone...

On Thu, Oct 15, 2015 at 4:38 PM, black copper <blackcopp...@gmail.com>
wrote:

> I want to experiment with libx264 encoding routines (using its library).
> Since it is already present inside FFmpeg libraries; I would like to know
> is there any benefit I could expect to achieve by compiling and using
> libx264 directly; instead of using it via FFmpeg libs?
>
> Thanks,
>
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Re: [Libav-user] avformat_alloc_output_context2 with WEBM defaults to VP9/Opus now - need to change

[black copper]

I posted a question yesterday; posting again as reminder - maybe someone
has a quick solution:

Hi,

I am using FFmpeg Zeranoe builds for Windows. I took a snapshot of their
builds few months ago, and just recently decided to upgrade it.

As with previous build, when I called avformat_alloc_output_context2()
function with something like:

avformat_alloc_output_context2(, NULL, NULL, "test.webm")

I always got the right codec: WebM with VP8 video and Vorbis audio.

Now, with the latest build, it is giving me VP9 video and Opus audio. Due
to some reasons, Opus just don't seem to work with me, giving me error with
function:

avcodec_open2()

with some text like: "Invalid argument" (when seen with av_strerror()
function ).

I just want to use VP8/Vorbis; any ideas how I can
force avformat_alloc_output_context2() to use this?

Thanks,

On Tue, Oct 6, 2015 at 5:42 PM, black copper <blackcopp...@gmail.com> wrote:

> Hi,
>
> I am using FFmpeg Zeranoe builds for Windows. I took a snapshot of their
> builds few months ago, and just recently decided to upgrade it.
>
> As with previous build, when I called avformat_alloc_output_context2()
> function with something like:
>
> avformat_alloc_output_context2(, NULL, NULL, "test.webm")
>
> I always got the right codec: WebM with VP8 video and Vorbis audio.
>
> Now, with the latest build, it is giving me VP9 video and Opus audio. Due
> to some reasons, Opus just don't seem to work with me, giving me error with
> function:
>
> avcodec_open2()
>
> with some text like: "Invalid argument" (when seen with av_strerror()
> function ).
>
> I just want to use VP8/Vorbis; any ideas how I can
> force avformat_alloc_output_context2() to use this?
>
> Thanks,
>
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[Libav-user] avformat_alloc_output_context2 with WEBM defaults to VP9/Opus now - need to change

[black copper]

Hi,

I am using FFmpeg Zeranoe builds for Windows. I took a snapshot of their
builds few months ago, and just recently decided to upgrade it.

As with previous build, when I called avformat_alloc_output_context2()
function with something like:

avformat_alloc_output_context2(, NULL, NULL, "test.webm")

I always got the right codec: WebM with VP8 video and Vorbis audio.

Now, with the latest build, it is giving me VP9 video and Opus audio. Due
to some reasons, Opus just don't seem to work with me, giving me error with
function:

avcodec_open2()

with some text like: "Invalid argument" (when seen with av_strerror()
function ).

I just want to use VP8/Vorbis; any ideas how I can
force avformat_alloc_output_context2() to use this?

Thanks,
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