I've a some hundreds of video files in a folder structure. All of them have video and audio streams, but some of them don't have any sound, despite having an audio stream. Is there a way to find out those files without having to resort to opening each file individually.
Most ways I know only check if there is an audio stream.
Thanks.
You can run the following in batch mode:
ffmpeg -hide_banner -i file.mp4 -af volumedetect -vn -f null - 2>&1 | grep mean_volume
The output for each file will be of the form
[Parsed_volumedetect_0 # 0000000002b1e800] mean_volume: -17.2 dB
Perfect digital silence will have a value of -91 dB, but anything below, say, -40 dB is probably just tape noise. Test and verify a few inputs manually and set a value.
I'm developing Android application, and im using ffmpeg for conversion of files.
I want my binary file to be as slim as possible since i don't have many input formats and output formats, and my operation is quite basic.And of course not to bloat the APK.
In my program ffmpeg receives a file, and copys the audio stream (-acodec copy), the audio stream will always be aac (mp4a). What i need is to save the stream to file.
My command looks like this : ffmpeg -i {Input} -vn -acodec copy output.aac.
What muxer do i need to for muxing aac to file? I have tried flv,mp3,mov but i always get
Unable to find a suitable output format for 'output.aac', so these options are wrong.
I don't need an encoder for stream copy btw.
Side note: this command work flawlessly on full installation of ffmpeg , but I don't know which muxer it uses. If there is a way to output the muxer it uses from regular ffmpeg run, it would work too.
A common file format for AAC is BMFF/MOV/MP4/M4A. If you specify the m4a file extension, FFmpeg will take care of it for you.
ffmpeg -i {input} -vn -acodec copy output.m4a
If you just want raw AAC, you can use ADTS as a lightweight container of sorts, as Mulvya suggested.
ffmpeg -i {input} -vn -acodec copy -f adts output.aac
I had to add the -f for me to work (on FFmpeg 3.22):
ffmpeg -i {input} -vn -acodec copy -f adts output.m4a
You have to add adts to --enable-muxer when configuring ffmpeg, eg. ./configure --disable-everything (...) --enable-muxer=adts. Then you will be able to save to .aac file
I am trying to concatenate video files so that next one follows the one before it when it is played. The formatting for all of the files are the same. The files all have audio & video.
I think I am very close (hopefully!) to getting this to work, but I have one final problem. The command below takes all of the mp4 files in my folder and creates a big mp4 file, which is the right size in total MB, but the images for all videos after the first video are garbled. The audio is okay (continues just fine from video to video). Also, I don't get any error messages.
ffmpeg -f concat -i <(for f in /folder1/*.mp4; do echo "file '$f'"; done) -c copy /folder1/all.mp4
I'm not very familiar with ffmpeg yet, so I've just been trying the different suggestions I've found on the web. Can anyone suggest other things for me to try? (I've tried reading the FAQs, but I have to confess that I don't fully understand it. Also, there seems to be some posts about audio being missing after concatenation, but I haven't seen anything on images being garbled.) Thx in advance!
I have had good luck using this ... avconv is a fork of ffmpeg
avconv -i 1.mp4 1.mpeg
avconv -i 2.mp4 2.mpeg
avconv -i 3.mp4 3.mpeg
cat 1.mpeg 2.mpeg 3.mpeg | avconv -f mpeg -i - -vcodec mpeg4 -strict experimental output.mp4
I'm working on a music VOD app on iPhone, and thanks to Apple guidelines, I have to run a HTTP Live Streaming in order to be accepted on the AppStore. But, since Apple doesn't care about 98% of servers on earth, they don't provide their so magical HTTP Live Streaming Tools for Linux-based systems. And from this point, the nightmare starts.
My goal is simple : Take an MP3, segmentate it and generate a simple .m3u8 index file.
I googled "HTTP Live Streaming Linux" and "Oh great ! lots of people have already done that"!
First, I visited the (so famous) post by Carson McDonald.
Result : the svn segmentate.c was old, buggy and a nightmare to compile (Nobody in this world can precise what version of ffmpeg they are using !).
Then I came across the Carson's git repo, but too bad, there is a lot of annoying ruby stuff and live_segmenter.c can't take mp3 files.
Then I searched more deeply. I found this stackoverflow topic, and it's exactly what I want to do. So I have followed the advice from juuni to use this script (httpsegmenter). Result: Impossible to compile anything, 2 days of works and finally I managed to compile it (ffmpeg 8.1 w/ httpsegmenter rev17). And no, this is not a good script, it does take mp3 files, but the ts files generated and the index file can't be read by a player.
Then the author of the post krisbulman, came with a solution, and even gave a patched version of m3u8-segmenter by his own (git repo). I test it : doesn't compile, do nothing. So I took the original version from johnf https://github.com/johnf/m3u8-segmenter. I managed to compile and miracle it works (not really).
I used this command line (ffmpeg 0.8.1):
ffmpeg -er 4 -i music.mp3 -f mpegts -acodec libmp3lame -ar 44100 -ab 128k -vn - | m3u8-segmenter -i - -d 10 -p outputdir/prefix -m outputdir/output.m3u8 -u http://test.com/
This script encode my mp3 file (it takes 4 seconds, too long), and pass it to the m3u8-segmenter to segment it into 10 seconds .TS files.
I tested this stream with Apple's mediastreamvalidator on my mac, and it said that it was OK. So i played it into quicktime, but there is about 0.2 seconds blank between each .TS files !!
So here is my situation, it's a nightmare, I can't get a simple mp3 stream over the HLS protocol. Is there a simple WORKING solution to segmentate a mp3 ? Why can't I directly segmentate the mp3 file into multiple mp3 files like Apple's mediafilesegmenter does?
Use libfaac insteam of libmp3lame which eliminates the 0.2 second break.
Elastic Transcoder Service - if you don't need AES encryption just throw your MP3 in an S3 bucket and be done with it:
http://aws.amazon.com/elastictranscoder/
You can then even add Cloudfront CDN support. (P.S. I fully appreciate your pain, this whole space is a nightmare).
For live streaming only, you should try Nginx with RTMP module for this one. https://github.com/arut/nginx-rtmp-module
Live HLS works pretty good but with looooong buffer.
However, it does not support on-demand HLS streaming.
Piece of module`s config for example
# HLS requires libavformat & should be configured as a separate
# NGINX module in addition to nginx-rtmp-module:
# ./configure ... --add-module=/path/to/nginx-rtmp-module/hls ...
# For HLS to work please create a directory in tmpfs (/tmp/app here)
# for the fragments. The directory contents is served via HTTP (see
# http{} section in config)
#
# Incoming stream must be in H264/AAC/MP3. For iPhones use baseline H264
# profile (see ffmpeg example).
# This example creates RTMP stream from movie ready for HLS:
#
# ffmpeg -loglevel verbose -re -i movie.avi -vcodec libx264
# -vprofile baseline -acodec libmp3lame -ar 44100 -ac 1
# -f flv rtmp://localhost:1935/hls/movie
#
# If you need to transcode live stream use 'exec' feature.
#
application hls {
live on;
hls on;
hls_path /tmp/app;
hls_fragment 5s;
}
What problems were you having with httpsegmenter? It's a single C source file that only links against some libraries provided by ffmpeg (or libav). I maintain a Gentoo ebuild for it, as I use it to time-shift talk radio. If you're running Gentoo, building is as simple as this:
sudo bash -l
layman -S
layman -a salfter
echo media-video/httpsegmenter ~\* >>/etc/portage/package.accept_keywords
emerge httpsegmenter
exit
On Ubuntu, I had to make sure libavutil-dev and libavformat-dev were both installed, so the build looks something like this:
sudo apt-get install libavutil-dev libavformat-dev
git clone https://gitlab.com/salfter/httpsegmenter.git
cd httpsegmenter
make -f Makefile.txt
sudo make -f Makefile.txt install
Once it's built (and once I have an audio source URL), usage is fairly simple: curl to stream the audio, ffmpeg to transcode it from whatever it is at the source (often MP3) to AAC, and segmenter to chunk it up:
curl -m 3600 http://invalid.tld/stream | \
ffmpeg -i - -acodec libvo_aacenc -ac 1 -ab 32k -f mpegts - 2>/dev/null | \
segmenter -i - -d 20 -o ExampleStream -x ExampleStream.m3u8 2>/dev/null
This grabs one hour of streaming audio (needs to be MP3 or AAC, not Flash), transcodes it to 32 kbps mono AAC, and chunks it up for HTTP live streaming. Have it dump into a directory served up by your webserver and you're good to go.
Once the show's done, converting to a single .m4a that can be served up as a podcast is also simple:
cat `ls -rt ExampleStream-*.ts` | \
ffmpeg -i - -acodec copy -absf aac_adtstoasc ExampleStream.m4a 2>/dev/null
I know this is an old question, but I am using this in VLC:
## To start playing the playlist out to the encoder
cvlc -vvv playlist.m3u --sout rtp:127.0.0.1 --ttl 2
## To start the encoder
cvlc rtp:// --sout='#transcode{acodec=mp3,ab=96}:duplicate{dst=std{access=livehttp{seglen=10,splitanywhere=true,delsegs=true,numsegs=15,index=/var/www/vlctest/mystream.m3u8,index-url=http://IPANDPORT/vlctest/mystream-########.ts},mux=ts,dst=/var/www/vlctest/mystream-########.ts},select=audio}'
I had problems if I didn't stream the playlist file to another copy of VLC, the first step is optional if you already have a live streaming source. (but you can use any source for the "encoder" portion).
You could try to use our media services on Windows Azure platform: http://mingfeiy.com/how-to-generate-http-live-streaming-hls-content-using-windows-azure-media-services/
You could encode and stream your video in HLS format by using our portal with no configuration and coding required.
Your English is fine.
Your frustration is apparent.
Q: What's the real issue here? It sounds like you just need a working HLS server, correct? Because of Apple requirements, correct?
Can you use any of the ready-made implementations listed here:
http://en.wikipedia.org/wiki/HTTP_Live_Streaming
I want to build up some kind of stream wrapper:
I own an old Dreambox PAL Sat Reciever with Networking. This stream I want to transcode to a lower resultion an restream it.
My Goal is, to have a simple website, where this stream is embedded via rtmp.
I thougt crtmpserver should be the right software. For now I have a site running and can play local files through jwplayer/crtmpserver.
I am looking for a solution for this:
httpUrl -> ffmpeg -> crtmpserver
Is that possible? May I redirect the output of ffmpeg to a filed pipe, and crtmpserver could grab that? Or go with UDP?
Any hints appreciated!!! Thanks!!
That's easy:
Start the server (in console mode for debugging)
You should see something like this:
|tcp| 0.0.0.0| 9999| inboundTcpTs| flvplayback|
Basically, that is a tcp acceptor for mpegts streams
Use ffmpeg to create the stream:
ffmpeg -i < source > < source_related_parameters > < audio_codec_parameters > < video_codec_parameters > -f mpegts "tcp://127.0.0.1:9999"
Example:
ffmpeg -i /tmp/aaa.flv -acodec copy -vcodec copy -vbsf h264_mp4toannexb -f mpegts "tcp://127.0.0.1:9999"
Go back to the server and watch the console. You should see something like this:
Stream INTS(6) with name ts_13_257_256 registered to application flvplayback from protocol ITS(13)
ts_13_257_256 is the stream name. Now you can use jwplayer or similar player and point it to that stream
If you want to use UDP, you need to stop the server and change the config file so instead of having
protocol="inboundTcpTs"
you should have
protocol="inboundUdpTs"
Yo ucan even copy the entire section and change the port number to have both.
Also, you have to change the ffmpeg so instead of having tcp://127.0.0.1:9999 you can have udp://127.0.0.1:9999
Now, if you also want a stream name and not that ts_13_257_256 (which is by the way ts_protocolId_AudioPID_VideoPID) you can use LiveFLV in a similar manner:
ffmpeg -i /tmp/aaa.flv -acodec copy -vcodec copy -vbsf h264_mp4toannexb -f flv -metadata streamName=myStreamName "tcp://127.0.0.1:6666"
And the server should show:
Stream INLFLV(1) with name `myStreamName` registered to application `flvplayback` from protocol ILFL(3)
There you go, now you have a "computed" stream name which is myStreamName
One last observation. Please ask this kind of questions on the crtmpserver's mailing list. You will be better heard.
You can find resources here:
http://www.rtmpd.com/resources/
Look for the google group under
Cheers,
Andrei