Re: [FFmpeg-user] How to compile ffmpeg on Android

[Mohammed Bey Ahmed Khernache]

I am doing cross-compilation. I am running on Ubuntu x86_64. My target os
is: android.
I used the script below:
---
#!/bin/bash
#Change NDK to your Android NDK location

pushd .
cd ../ffmpeg
PLATFORM=$NDK/platforms/android-21/arch-arm64/
PREBUILT=$NDK/toolchains/aarch64-linux-android-4.9/prebuilt/linux-x86_64

GENERAL="\
--enable-small \
--enable-cross-compile \
--extra-libs="-lgcc" \
--arch=aarch64 \
--cc=$PREBUILT/bin/aarch64-linux-android-gcc \
--cross-prefix=$PREBUILT/bin/aarch64-linux-android- \
--nm=$PREBUILT/bin/aarch64-linux-android-nm \
--extra-cflags="-I../ffmpeg-build/armeabi-v8a/include" \
--extra-ldflags="-L../ffmpeg-build/armeabi-v8a/lib" "

MODULES="\
--enable-gpl"

function build_arm64
{
  ./configure \
  --logfile=conflog.txt \
  --target-os=android \
  --prefix=$PREFIX/armeabi-v8a \
  ${GENERAL} \
  --sysroot=$PLATFORM \
  --extra-cflags="" \
  $COMMON_FLAGS \
  --extra-ldflags="-Wl,-rpath-link=$PLATFORM/usr/lib -L$PLATFORM/usr/lib
-nostdlib -lc -lm -ldl -llog" \
  --enable-shared \
  --disable-static \
  --disable-doc \
  ${MODULES}

  make clean
  make
  make install
}

build_arm64


echo Android ARM64v8a builds finished

popd
---

The error I got is below:
---
WARNING:
/home/mohammedbey/Desktop/ffmpeg_sources/android-ndk-r17b/toolchains/aarch64-linux-android-4.9/prebuilt/linux-x86_64/bin/aarch64-linux-android-pkg-config
not found, library detection may fail.
config.h is unchanged
libavutil/avconfig.h is unchanged
libavfilter/filter_list.c is unchanged
libavcodec/codec_list.c is unchanged
libavcodec/parser_list.c is unchanged
libavcodec/bsf_list.c is unchanged
libavformat/demuxer_list.c is unchanged
libavformat/muxer_list.c is unchanged
libavdevice/indev_list.c is unchanged
libavdevice/outdev_list.c is unchanged
libavformat/protocol_list.c is unchanged
ffbuild/config.sh is unchanged
GEN libavutil/libavutil.version
GEN libswscale/libswscale.version
GEN libswresample/libswresample.version
GEN libpostproc/libpostproc.version
GEN libavcodec/libavcodec.version
GEN libavformat/libavformat.version
GEN libavfilter/libavfilter.version
GEN libavdevice/libavdevice.version
CC libavdevice/alldevices.o
In file included from ./libavformat/internal.h:24:0,
 from libavdevice/alldevices.c:23:
/home/mohammedbey/Desktop/ffmpeg_sources/android-ndk-r17b/toolchains/aarch64-linux-android-4.9/prebuilt/linux-x86_64/lib/gcc/aarch64-linux-android/4.9.x/include/stdint.h:9:26:
fatal error: stdint.h: No such file or directory
 # include_next 
  ^
compilation terminated.
ffbuild/common.mak:60: recipe for target 'libavdevice/alldevices.o' failed
make: *** [libavdevice/alldevices.o] Error 1
CC libavdevice/alldevices.o
In file included from ./libavformat/internal.h:24:0,
 from libavdevice/alldevices.c:23:
/home/mohammedbey/Desktop/ffmpeg_sources/android-ndk-r17b/toolchains/aarch64-linux-android-4.9/prebuilt/linux-x86_64/lib/gcc/aarch64-linux-android/4.9.x/include/stdint.h:9:26:
fatal error: stdint.h: No such file or directory
 # include_next 
  ^
compilation terminated.
ffbuild/common.mak:60: recipe for target 'libavdevice/alldevices.o' failed
make: *** [libavdevice/alldevices.o] Error 1
Android ARM64v8a builds finished
/home/mohammedbey/Desktop/ffmpeg4Android-master/ffmpeg-build-script
---


​

Mohammed BEY AHMED KHERNACHE
PhD student
Lab-STICC / CNRS UMR 6285
University of Southern Brittany, Lorient, France
Tel: +33-783-700-185
Email: mohammed.bey-ahmed-kherna...@univ-ubs.fr




On Thu, Jul 12, 2018 at 12:33 AM Rafael Lima  wrote:

> Are you jealous? did you at least tried to build?
>
> it is not outdated and not broken... as I've made it few weeks ago using
> the latest version from every lib i needed for my project...
>
> if you dont need all of them it's good you can simple remove from config
> file... a work that seams to hard for you to do
>
> On Wed, Jul 11, 2018 at 7:13 PM, Carl Eugen Hoyos 
> wrote:
>
> > 2018-07-12 0:01 GMT+02:00, Rafael Lima :
> > > https://github.com/bolds07/ffmpeg-for-android-for-dummies
> >
> > Yes, apart from being outdated and broken (as every other script
> > I have seen so far), this one has the great advantage that it
> > produces binaries that you cannot legally distribute - that must
> > help many users...
> >
> > Carl Eugen
> > 

[FFmpeg-user] Issue with connection FFmpeg and ffprobe for live stream transcoding

[Pedro Daniel Costa]

Hi guys

First of all let me mention, that am just typing this issue , after several 
hours of research in order to make something work that simply does not what to 
happen.


Here is my scenario.


I have 2 PC servers, one of them is running a multicast live transmition 
application, which stream live tv multicast, and on the other second server PC 
I have ffmpeg meant to be used for transcoding.


Before i was running all both applications  Multicast tool + transcoding on the 
same machine and everything was working fine


I am running ffmpeg 3.4.2 with ffserver and config files, everything like I 
said worked great on just one pc server, but because CPU cannot handle all the 
transcoding of the live channels I have decided to build a dedicated 
transcoding server machine.


So previously I was using   ffmpeg -i udp://239.106.xxx.xx:port  
http://ip-address-:port/channel-name.ffm


And then I could get the output live transcoding channel in  
http://ip-address:porta/channel-name.flv, everything was working fine..



So when I setup the 2nd PC server I have installed a specific Network card, 
just for running Multicast purposes, like I do have on the first server machine

So Server 1 uplinks multicast UDP to switch with multicast enabled, and I can 
watch all multicast channels connected with other devices on the same switch 
multicast enabled.

I have then connected PC2 transcoder multicast interface to the same switch..  
and I started receiving the multicast traffic..

When I use tcpdump tool on PC2 transcoder I can see the multicast traffic from 
PC1 server arriving ok..


But when I try to use ffmpeg -I with udp://239.106.xx.xx:port   I cannot see 
anything , if I try to use ffprobe also cannot pickup anything..


So my question is , is there any specific command to enable just one specific 
NIC for ffmpeg to run ?? because I am out of ideas

And I have both PC1 and PC2 servers on same multicast route add -n 224.0.0.0. 
xxx


This what comes out on the tcpdump on PC2 transcoder the server responsible for 
receiving the multicast traffic from PC1



root@encoder:~# tcpdump -i enp4s0
tcpdump: verbose output suppressed, use -v or -vv for full protocol decode
listening on enp4s0, link-type EN10MB (Ethernet), capture size 262144 bytes
20:48:13.261148 IP 170.144.137.108.47838 > 239.106.32.0.1000: UDP, length 1316
20:48:13.261156 IP 170.144.137.108.47838 > 239.106.32.0.1000: UDP, length 1316
20:48:13.261161 IP 170.144.137.108.47838 > 239.106.32.0.1000: UDP, length 1316
20:48:13.261351 IP 170.144.137.108.47838 > 239.106.32.0.1000: UDP, length 1316
20:48:13.261599 IP 170.144.137.108.47838 > 239.106.32.0.1000: UDP, length 1316
20:48:13.261607 IP 170.144.137.108.47838 > 239.106.32.0.1000: UDP, length 1316
20:48:13.261612 IP 170.144.137.108.47838 > 239.106.32.0.1000: UDP, length 1316
20:48:13.457230 IP 170.144.137.108.47838 > 239.106.224.0.1000: UDP, length 1316




but when I try to connect ffprobe or ffmpeg -I  nothing happens just stays like 
this below

root@encoder:16.04.york0.2 Copyright (c) 2007-2018 the FFmpeg develop ers
built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu116.04.york0.2' --toolchain=har dened 
--libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu 
--enable-gpl --disable-stripping --enable-avresample --enable-avisynth 
--enable- gnutls --enable-ladspa --enable-libass --enable-libbluray 
--enable-libbs2b --ena ble-libcaca --enable-libcdio --enable-libflite 
--enable-libfontconfig --enable-l ibfreetype --enable-libfribidi 
--enable-libgme --enable-libgsm --enable-libmp3la me --enable-libmysofa 
--enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse 
--enable-librubberband --enable-librsvg --enable-libshine --en able-libsnappy 
--enable-libsoxr --enable-libspeex --enable-libssh --enable-libth eora 
--enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack 
--enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid 
--enable-lib zmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl 
--enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 
--enable-chromaprint --e nable-frei0r --enable-libopencv --enable-libx264 
--enable-shared
libavutil 55. 78.100 / 55. 78.100
libavcodec 57.107.100 / 57.107.100
libavformat 57. 83.100 / 57. 83.100
libavdevice 57. 10.100 / 57. 10.100
libavfilter 6.107.100 / 6.107.100
libavresample 3. 7. 0 / 3. 7. 0
libswscale 4. 8.100 / 4. 8.100
libswresample 2. 9.100 / 2. 9.100
libpostproc 54. 7.100 / 54. 7.100


i am actually out of ideas

any help or comments will be appreciated


thanks
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Re: [FFmpeg-user] PSNR measure

[Carl Eugen Hoyos]

2018-07-13 2:39 GMT+02:00, José María Infanzón :
> Hi - I need to measure quality if a live stream, and I was
> thinking of measuring PSNR. So I was wondering how
> can I run this test since it is a live stream

> I do not have a refence stream or video.

Then what exactly do you want to measure?

> Should I look for a reference video of the same
> resolution and bitrate, like for example a Big buck
> bunny, and compare against It?

This makes only sense if the live stream shows Big Buck
Bunny (and if you can assure frame synchronisation).

Carl Eugen
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[FFmpeg-user] PSNR measure

[José María Infanzón]

Hi - I need to measure quality if a live stream, and I was thinking of
measuring PSNR. So I was wondering how can I run this test since it is a
live stream, I do not have a refence stream or video. Should I look for a
reference video of the same resolution and bitrate, like for example a Big
buck bunny, and compare against It?

Thanks
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Re: [FFmpeg-user] fMP4 generation speed

[Ronak Patel]

> On Jul 12, 2018, at 6:21 PM, Ronak Patel  
> wrote:
> 
> Hey Carl,
> 
> So I dug into this more today and I have root caused what's exactly happening 
> here. 
> 
> The problematic code is this: 
> https://github.com/FFmpeg/FFmpeg/blob/master/libavformat/hlsenc.c#L1368 
> 
> This is where the filename is set and the next line actually opens the file. 
> 
> The logic for this hls_window method is the following:
> 
> 1. Make a new temporary file.
> 2. Write out a new HLS manifest header.
> 3. Loop through all available segments and write out all of the entries for 
> them.
> 4. Close the temporary file when finished.
> 5. Rename the temporary file to the target file name.
> 6. Rinse and repeat for every single fragment.
> 
> Therefore, if you can imagine a 153 hour audio file, we write out a totally 
> new HLS manifest 550800 times (153 * 60 * 60 assuming a 1s fragment duration) 
> that gets progressively larger as each fragment is generated.
> 
> This is a classic O(N^2) algorithm implementation, instead of:
> 
> 1. Creating the destination file up front & write the manifest header.
> 2. Append the new segment to the file.
> 3. If this is the last segment, write out EXT-X-ENDLIST.
> 
> There's no looping involved, nor the need to make temporary files.
> 
> FYI that I've noticed the same sort of pattern being applied to MPEG DASH: 
> https://github.com/FFmpeg/FFmpeg/blob/master/libavformat/dashenc.c#L786 
> 
> 
> To implement something like this, looks like we'd have to significantly 
> re-engineer the code. Do you have any pointers on how to go about doing this? 
> Or, would you be able to help do this?
> 

I was thinking about this and I think it must have been implemented this way 
because of live hls streams? Not writing to the main manifest directly and 
keeping that stream open does release any file system locks that there would be 
otherwise. 

Is this the reason why this was done this way at all?


> Thanks for all your help,
> 
> Ronak
> 
>> On Jun 27, 2018, at 2:04 PM, Carl Zwanzig  wrote:
>> 
>> Hi,
>> 
>> I haven't traced it out completely, but take a look at the flag 
>> HLS_TEMP_FILE in libavformat/hlsenc.c.
>> 
>> Later,
>> 
>> z!
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Re: [FFmpeg-user] fMP4 generation speed

[Ronak Patel]

Hey Carl,

So I dug into this more today and I have root caused what's exactly happening 
here. 

The problematic code is this: 
https://github.com/FFmpeg/FFmpeg/blob/master/libavformat/hlsenc.c#L1368 

This is where the filename is set and the next line actually opens the file. 

The logic for this hls_window method is the following:

1. Make a new temporary file.
2. Write out a new HLS manifest header.
3. Loop through all available segments and write out all of the entries for 
them.
4. Close the temporary file when finished.
5. Rename the temporary file to the target file name.
6. Rinse and repeat for every single fragment.

Therefore, if you can imagine a 153 hour audio file, we write out a totally new 
HLS manifest 550800 times (153 * 60 * 60 assuming a 1s fragment duration) that 
gets progressively larger as each fragment is generated.

This is a classic O(N^2) algorithm implementation, instead of:

1. Creating the destination file up front & write the manifest header.
2. Append the new segment to the file.
3. If this is the last segment, write out EXT-X-ENDLIST.

There's no looping involved, nor the need to make temporary files.

FYI that I've noticed the same sort of pattern being applied to MPEG DASH: 
https://github.com/FFmpeg/FFmpeg/blob/master/libavformat/dashenc.c#L786 


To implement something like this, looks like we'd have to significantly 
re-engineer the code. Do you have any pointers on how to go about doing this? 
Or, would you be able to help do this?

Thanks for all your help,

Ronak

> On Jun 27, 2018, at 2:04 PM, Carl Zwanzig  wrote:
> 
> Hi,
> 
> I haven't traced it out completely, but take a look at the flag HLS_TEMP_FILE 
> in libavformat/hlsenc.c.
> 
> Later,
> 
> z!
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Re: [FFmpeg-user] How to select audio stream in filter_complex with concat?

[Moritz Barsnick]

On Thu, Jul 12, 2018 at 16:23:58 +0200, Dmitri wrote:
> I've also tried to add same file again and map it's audio stream, but it
> gives error:
> "Filter concat:out:a0 has an unconnected output"
> 
> ffmpeg -i "D:\videos\intro_1.mov" -ss 00:00:00 -to 00:00:40 -i
> "D:\videos\logo_1.mp4" -i "D:\videos\logo_2.mp4" -i "D:\videos\intro_1.mov"
> -filter_complex
> "[0:v]fade=t=out:st=37:d=3[0v];[1:v]fade=t=out:st=11:d=1[1v];[2:v]fade=t=out:st=9:d=1[2v];
> [0v][0:a][1v][1:a][2v][2:a] concat=n=3:v=1:a=1 [v1][a]" -map "[v1]" -map
> 3:1 output.mp4 -y
> 
> Feels like I am missing something simple :)

Yes. ;-) You are still concat'ing video and audio, with audio as "[a]",
but not mapping "[a]" anywhere. You need to adapt the concat parameters
to use only the video streams as input, and to output only "[v1]".

Moritz
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[FFmpeg-user] How to select audio stream in filter_complex with concat?

[Dmitri]

Hello,

I am trying to merge 3 videos with fade-out and use the audio stream from
the input index 0 in the resulting file (nothing should be played when it
ends)
Video 0 is cut at the beginning, but the original stream is long enough.

This command does the merging, unfortunately the audio is taken from files
1 and 2 when they are playing and 0:a is also cut:

ffmpeg -i "D:\videos\intro_1.mov" -ss 00:00:00 -to 00:00:40 -i
"D:\videos\logo_1.mp4" -i "D:\videos\logo_2.mp4" -filter_complex
"[0:v]fade=t=out:st=37:d=3[0v];[1:v]fade=t=out:st=11:d=1[1v];[2:v]fade=t=out:st=9:d=1[2v];
[0v][0:a][1v][1:a][2v][2:a] concat=n=3:v=1:a=1 [v1][a]" -map "[v1]" -map
"[a]" output.mp4 -y

I've also tried to add same file again and map it's audio stream, but it
gives error:
"Filter concat:out:a0 has an unconnected output"

ffmpeg -i "D:\videos\intro_1.mov" -ss 00:00:00 -to 00:00:40 -i
"D:\videos\logo_1.mp4" -i "D:\videos\logo_2.mp4" -i "D:\videos\intro_1.mov"
-filter_complex
"[0:v]fade=t=out:st=37:d=3[0v];[1:v]fade=t=out:st=11:d=1[1v];[2:v]fade=t=out:st=9:d=1[2v];
[0v][0:a][1v][1:a][2v][2:a] concat=n=3:v=1:a=1 [v1][a]" -map "[v1]" -map
3:1 output.mp4 -y

Feels like I am missing something simple :)

/Best regards
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Re: [FFmpeg-user] 453 Not enough bandwidth error

[Michael Shaffer]

Here is the command, with a screen shot of it and the error.

c:\ffmpeg\bin\ffmpeg3.exe -re -rtsp_transport tcp -i "rtsp://
admin:admin@192.168.1.230:554/VideoInput/1/h264/1" -f lavfi -f dshow
-rtbufsize 2000M -thread_queue_size 5096 -i audio="virtual-audio-capturer"
-c:a libmp3lame -ab 128k -ar 44100 -c:v copy -threads 0 -bufsize 512k -f
flv "rtmp://a.rtmp.youtube.com/live2/7ddp-1234-5678-f3ff"

Screenshot:
https://kdca.neocities.org/command.jpg


On Tue, Jul 10, 2018 at 5:36 AM, Moritz Barsnick  wrote:

> Hi Michael,
>
> On Mon, Jul 09, 2018 at 16:39:32 -0400, Michael Shaffer wrote:
> > Hi, I'm trying to stream an IP camera to youtube, using FFMPEG to push
> the
> > stream. The problem is, if I put the camera in 1080p mode, I get this
> error:
> >
> > method SETUP failed: 453 Not Enough Bandwidth
> > Server returned 4XX Client Error, but not one of 40{0,1,3,4}
>
> Could you please show us the actual ffmpeg command line and its
> complete, uncut console output?
>
> > Anyways, is there a way to allow FFMPEG to use more bandwidth? I have
> some
> > programming experience, could I edit the FFMPEG code somehow to allow
> more
> > bandwidth?
>
> That error message isn't caused by ffmpeg. ffmpeg can handle plenty of
> bandwidth (assuming enough CPU and I/O), the RTMP server you're sending
> to probably can't. We'll see when we see the output requested above.
>
> Cheers,
> Moritz
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Re: [FFmpeg-user] Convert and AAC to M4a file using bitstream flag aac_adtstoasc

[Mohammed Sha]

Hi Mortiz,

How do you know it's not converted or filtered? Wouldn't it be nice if you
showed us the error message (if you get one)? Or should we guess?
 I have checked the converted audio file info using Mediainfo binary.

Actually, please show us the complete console output
   Fmmpeg console output is provided below and the Mediainfo output.

console $ ./ffmpeg -i ~/Downloads/test_audio.aac -c:a copy  -bsf:a
aac_adtstoasc aac_try_with_bsf.aac
ffmpeg version 4.0.1 Copyright (c) 2000-2018 the FFmpeg developers
  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.10) 20160609
  configuration: --pkg-config-flags=--static --enable-gpl --enable-libass
--enable-libfdk-aac --enable-libfreetype --enable-libmp3lame
--enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264
--enable-libx265 --enable-nonfree --enable-openssl
--enable-bsf=aac_adtstoasc
  libavutil  56. 14.100 / 56. 14.100
  libavcodec 58. 18.100 / 58. 18.100
  libavformat58. 12.100 / 58. 12.100
  libavdevice58.  3.100 / 58.  3.100
  libavfilter 7. 16.100 /  7. 16.100
  libswscale  5.  1.100 /  5.  1.100
  libswresample   3.  1.100 /  3.  1.100
  libpostproc55.  1.100 / 55.  1.100
[aac @ 0x3d0c880] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from '/home/vkchlt0199/Downloads/test_audio.aac':
  Duration: 00:08:47.25, bitrate: 81 kb/s
Stream #0:0: Audio: aac (LC), 16000 Hz, mono, fltp, 81 kb/s
File 'aac_try_with_bsf.aac' already exists. Overwrite ? [y/N] y
Output #0, adts, to 'aac_try_with_bsf.aac':
  Metadata:
encoder : Lavf58.12.100
Stream #0:0: Audio: aac (LC), 16000 Hz, mono, fltp, 81 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help


size=5245kB time=00:09:03.16 bitrate=  79.1kbits/s speed=8.46e+03x
video:0kB audio:5187kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 1.118535%

console $ mediainfo ~/Downloads/test_audio.aac
General
Complete name:
/home/vkchlt0199/Downloads/test_audio.aac
Format   : ADTS
Format/Info  : Audio Data Transport Stream
File size: 5.12 MiB
Overall bit rate mode: Variable

Audio
Format   : AAC
Format/Info  : Advanced Audio Codec
Format version   : Version 4
Format profile   : LC
Bit rate mode: Variable
Channel(s)   : 1 channel
Channel positions: Front: C
Sampling rate: 16.0 KHz
Frame rate   : 15.625 fps (1024 spf)
Compression mode : Lossy
Stream size  : 5.12 MiB (100%)

Console $ mediainfo aac_try_with_bsf.aac
General
Complete name: aac_try_with_bsf.aac
Format   : ADTS
Format/Info  : Audio Data Transport Stream
File size: 5.12 MiB
Overall bit rate mode: Variable

Audio
Format   : AAC
Format/Info  : Advanced Audio Codec
Format version   : Version 4
Format profile   : LC
Bit rate mode: Variable
Channel(s)   : 1 channel
Channel positions: Front: C
Sampling rate: 16.0 KHz
Frame rate   : 15.625 fps (1024 spf)
Compression mode : Lossy
Stream size  : 5.12 MiB (100%)

I am able to convert the aac audio file to m4a using below command, But for
my actual use case the this  wont help.

Console $$ ./ffmpeg -i ~/Downloads/test_audio.aac -c:a copy
aac_try_with_bsf.m4a
ffmpeg version 4.0.1 Copyright (c) 2000-2018 the FFmpeg developers
  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.10) 20160609
  configuration: --pkg-config-flags=--static --enable-gpl --enable-libass
--enable-libfdk-aac --enable-libfreetype --enable-libmp3lame
--enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264
--enable-libx265 --enable-nonfree --enable-openssl
--enable-bsf=aac_adtstoasc
  libavutil  56. 14.100 / 56. 14.100
  libavcodec 58. 18.100 / 58. 18.100
  libavformat58. 12.100 / 58. 12.100
  libavdevice58.  3.100 / 58.  3.100
  libavfilter 7. 16.100 /  7. 16.100
  libswscale  5.  1.100 /  5.  1.100
  libswresample   3.  1.100 /  3.  1.100
  libpostproc55.  1.100 / 55.  1.100
[aac @ 0x3ed47c0] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from '/home/vkchlt0199/Downloads/test_audio.aac':
  Duration: 00:08:47.25, bitrate: 81 kb/s
Stream #0:0: Audio: aac (LC), 16000 Hz, mono, fltp, 81 kb/s

[FFmpeg-user] Dash and Multiple Inputs

[Ronak Patel]

Hi,

I saw this earlier post:

https://lists.ffmpeg.org/pipermail/ffmpeg-user/2015-August/027951.html

Which got no replies, but I have the same question now.

It looks like ffmpeg still doesn’t support providing multiple input so 
fragmentation happens for all representations.

Is there a way to do this today? Or is this an enhancement I’d have to file? Or 
figure out how to add myself?

Thanks

Ronak

Sent from my iPhone
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Re: [FFmpeg-user] Convert and AAC to M4a file using bitstream flag aac_adtstoasc

[Moritz Barsnick]

On Thu, Jul 12, 2018 at 12:43:01 +0530, Mohammed Sha wrote:
> But when i try below command the file is not getting converted or adding
> the bitstream filter
> 
> ./ffmpeg -i ~/Downloads/test_audio.aac -c:a copy  -bsf:a aac_adtstoasc
> aac_try_with_bsf.aac

How do you know it's not converted or filtered?

Wouldn't it be nice if you showed us the error message (if you get
one)? Or should we guess?

Actually, please show us the complete console output.

Moritz
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Re: [FFmpeg-user] Convert and AAC to M4a file using bitstream flag aac_adtstoasc

[Mohammed Sha]

Thanks for the response..

Yes I have compiled  with bitstream filter section.

 ./ffmpeg
ffmpeg version 4.0.1 Copyright (c) 2000-2018 the FFmpeg developers
  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.10) 20160609
  configuration: --pkg-config-flags=--static --enable-gpl --enable-libass
--enable-libfdk-aac --enable-libfreetype --enable-libmp3lame
--enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264
--enable-libx265 --enable-nonfree --enable-openssl
--enable-bsf=aac_adtstoasc
  libavutil  56. 14.100 / 56. 14.100
  libavcodec 58. 18.100 / 58. 18.100
  libavformat58. 12.100 / 58. 12.100
  libavdevice58.  3.100 / 58.  3.100
  libavfilter 7. 16.100 /  7. 16.100
  libswscale  5.  1.100 /  5.  1.100
  libswresample   3.  1.100 /  3.  1.100
  libpostproc55.  1.100 / 55.  1.100
Hyper fast Audio and Video encoder
usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options]
outfile}...



On Thu, Jul 12, 2018 at 1:38 PM, Ярослав Клейно 
wrote:

> ffmpeg compiled without bitstream filter?
> please,check
>
> 2018-07-12 10:13 GMT+03:00 Mohammed Sha :
>
> >  Hi ALL,
> >
> > As per ffmpeg bitstream filter documentation, if we use  -bsf:a
> > aac_adtstoasc  then the AAC stream from a raw ADTS AAC or an MPEG-TS
> > container to MP4A-LATM.
> >
> > https://ffmpeg.org/ffmpeg-bitstream-filters.html#aac_005fadtstoasc
> >
> > But when i try below command the file is not getting converted or adding
> > the bitstream filter
> >
> > ./ffmpeg -i ~/Downloads/test_audio.aac -c:a copy  -bsf:a aac_adtstoasc
> > aac_try_with_bsf.aac
> >
> > Could someone help me what's wrong in above command..
> >
> > --
> > ___
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-- 
Thanks & Regards
*Mohammed Sha Ebrahim*
VVDN Technologies Pvt.Ltd
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Re: [FFmpeg-user] Convert and AAC to M4a file using bitstream flag aac_adtstoasc

[Ярослав Клейно]

ffmpeg compiled without bitstream filter?
please,check

2018-07-12 10:13 GMT+03:00 Mohammed Sha :

>  Hi ALL,
>
> As per ffmpeg bitstream filter documentation, if we use  -bsf:a
> aac_adtstoasc  then the AAC stream from a raw ADTS AAC or an MPEG-TS
> container to MP4A-LATM.
>
> https://ffmpeg.org/ffmpeg-bitstream-filters.html#aac_005fadtstoasc
>
> But when i try below command the file is not getting converted or adding
> the bitstream filter
>
> ./ffmpeg -i ~/Downloads/test_audio.aac -c:a copy  -bsf:a aac_adtstoasc
> aac_try_with_bsf.aac
>
> Could someone help me what's wrong in above command..
>
> --
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[FFmpeg-user] Convert and AAC to M4a file using bitstream flag aac_adtstoasc

[Mohammed Sha]

 Hi ALL,

As per ffmpeg bitstream filter documentation, if we use  -bsf:a
aac_adtstoasc  then the AAC stream from a raw ADTS AAC or an MPEG-TS
container to MP4A-LATM.

https://ffmpeg.org/ffmpeg-bitstream-filters.html#aac_005fadtstoasc

But when i try below command the file is not getting converted or adding
the bitstream filter

./ffmpeg -i ~/Downloads/test_audio.aac -c:a copy  -bsf:a aac_adtstoasc
aac_try_with_bsf.aac

Could someone help me what's wrong in above command..

--
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