Hi am currently trying to retrieve 3 second clips of an audio file whilst it is recording in flutter. I am using the recording module flutter sound and flutter ffmpeg.
I record the audio file with default codec (.aac). The file is saved to the cache getTemporaryDirectory()
I then copy the file using this flutter ffmpeg code
List<String> arguments = ["-ss", start.toString(), "-i", inPath, "-to", end.toString(), "-c", "copy", outPath];
await flutterFFmpeg.executeWithArguments(arguments);
Start: start time (e.g. 0) and End: end time (e.g. 3)
It then returns this error
FFmpeg exited with rc: 1 [mov,mp4,m4a,3gp,3g2,mj2 # 0x748964ea00] moov
atom not found
Helpful information:
A moov atom is data about the file (e.g timescale,duration)
I know the inPath exists because I check that before executing ffmpeg command
The outPath is also format .aac
This ffmpeg function is being ran whilst the recording is still occurring
Example inPath uri looks like this /data/user/0/com.my.app/cache/output.aac
I have no problems when running on iOS, only on android
I would be grateful for help, I have spent many days trying to fix this problem. If you need anymore info please leave a comment. Thanks
Default Codec is not guaranteed to be AAC/ADTS.
It will depend of the Android version of your device.
You can do several things to understand better :
ffprobe on your file to see what has been recorded by Flutter Sound.
Use a specific Codec instead of default : aac/adts is a good choice because it can be streamed (you want to process the audio data during the recording and not after closing the file)
Verify that your file contains something and that the data are not still in internal buffers
Record to a dart PCM stream instead of a file. Working with a file and use FFmpeg to seek into it is complicated and perhaps does not fill your needs.
Related
I would like to get data from audio file based on microphone input (both Android and iOS), currently I'm using audioplayers and recordMp3 to record the microphone input. This results in a mp3 file with a local file path. In order to use the audio data, I want an uncompressed format like WAV. Would ffmpeg help with this conversion ? I want to eventually use this data for visualization.
MP3 to WAV
ffmpeg -i input.mp3 output.wav
Note that any encoding artifacts in the MP3 will be included in the WAV.
Piping from ffmpeg to your visualizer
I'm assuming you need WAV/PCM because your visualizer only accepts that format and does not accept MP3. You can create a WAV file as shown in the example above, but if your visualizer accepts a pipe as input you can avoid creating a temporary file:
ffmpeg -i input.mp3 -f wav - | yourvisualizer …
Using ffmpeg for visualization
See examples at How do I turn audio into video (that is, show the waveforms in a video)?
I download lectures in mp4 format from Udacity, but they're often broken down into 2-5 minute chunks. I'd like to combine the videos for each lecture into one continuous stream, which I've had success with on Windows using AnyVideo Converter. I'm trying to do the same thing on Ubuntu 15, and most of my web search results suggest MP4Box, whose documentation and all the online examples I can find offer the following syntax:
MP4Box -cat vid1.mp4 -cat vid2.mp4 -cat vid3.mp4 -new combinedfile.mp4
This creates a new file with working audio, but the video doesn't work. When I open with Ubuntu's native video player, I get the error "No valid frames decoded before end of stream." When I open with VLC, I get the error "Codec not supported: VLC could not decode the format 'avc3' (No description for this codec." I've tried using the -keepsys switch, as well, but I get the same results.
All the documentation and online discussion makes it sound as though what I'm trying to do is and should be really simple, but I can't seem to find info relevant to the specific errors I'm getting. What am I missing?
Use the -force-cat option.
For example,
MP4Box -force-cat -add in1.mp4 -cat in2.mp4 -cat in3.mp4 ... -new out.mp4
From the MP4Box documentation:
-force-cat
skips media configuration check when concatenating file.
It looks, by the presence of 'avc3', that these videos are encoded with h.264|avc. There are several modes for the concatenation of such streams. Either the video streams have compatible encoder configurations (frame size, ...) in which case only one configuration description is used in the file (signaled by 'avc1'). If the configurations are not fully compatible, MP4Box uses the 'inband' storage of those configurations (signaled by 'avc3'). The other way would be to use multiple sample description entries (stream configurations) but that is not well supported by players and not yet possible with MP4Box. There is no other way unless you want to reencode your videos. On Ubuntu, you should be able to play 'avc3' streams with the player that goes with MP4Box: MP4Client.
=== BACKGROUND ===
Some time ago I ripped a lot of music from an internet radio station. Unfortunately something seems to have went wrong, since the length of most files is displayed as being several hours, but they started playing at the correct position.
Example: If a file is really 3 minutes long and it would be displayed as 3 hours, playback would start at 2 hours and 57 minutes.
Before I upgraded my system, gstreamer was in an older version and its behaviour would be as described above, so I didn't pay too much attention. Now I have a new version of gstreamer which cannot handle these files correctly: It "plays" the whole initial offset.
=== /BACKGROUND ===
So here is my question: How is it possible to modify an OGG/Vorbis file in order to get rid of useless initial offsets? Although I tried several tag-edit programs, none of them would allow me to edit these values. (Interestingly enough easytag will display me both times, but write the wrong one...)
I finally found a solution! Although it wasn't quite what I expected...
After trying several other options I ended up with the following code:
#!/bin/sh
cd "${1}"
OUTDIR="../`basename "${1}"`.new"
IFS="
"
find . -wholename '*.ogg' | while read filepath;
do
# Create destination directory
mkdir -p "${OUTDIR}/`dirname "${filepath}"`"
# Convert OGG to OGG
avconv -i "${filepath}" -f ogg -acodec libvorbis -vn "${OUTDIR}/${filepath}"
# Copy tags
vorbiscomment -el "${filepath}" | vorbiscomment -ew "${OUTDIR}/${filepath}"
done
This code recursively reencodes all OGG files and then copies all vorbis comments. It's not a very efficient solution, but it works nevertheless...
What the problem was: I guess it has something to do with the output of ogginfo:
...
New logical stream (#1, serial: 74a4ca90): type vorbis
WARNING: Vorbis stream 1 does not have headers correctly framed. Terminal header page contains additional packets or has non-zero granulepos
Vorbis headers parsed for stream 1, information follows...
Version: 0
Vendor: Xiph.Org libVorbis I 20101101 (Schaufenugget)
...
Which disappears after reencoding the file...
At the rate at which I'm currently encoding it will probably take several hours until my whole media library will be completely reencoded... but at least I verified with several samples that it works :)
I have some .mov files want to stream to Flash media server. i have already tried to stream a single .mov by FFMPEG command in terminal and it works, the FMS can display the thing i streaming in live.
ffmpeg -re -i file1.mov -vcodec libx264 -f flv rtmp://localhost/livepkgr/livestream
Now i want to stream multiple files,
i tried to use above command one by one,
but it seems Flash media server stop the streaming when file1 is finished,
then start the stream with file2.
It makes the stream player stopped when file1 is finish, and have to refresh the page in order to continue on file2.
i am calling the FFMPEG command by a C program in linux, i wonder is there any method that i can prevent the FMS stopped when i switch the file source in FFMPEG? or is that possible to let FFMPEG constantly deliver the stream by multiple files source without stopped when a file finish?
Reformat your source file to a TS or MPEG or other "concatable" file. Then you can either use ffmpeg's concat protocol or just "cat" by yourself.
I found something like this it will be useful for you :
I managed to stream a static playlist of videos by using for each video a pipe (ex vid1.mp4 -> pipe1, vid2.mp4 -> pipe2 etc). Then i write into a single stream named pipe "stream" this way cat pipe1 pipe2 pipe3 > stream, and i use the stream pipe as input in FFMPEG to publish my stream
I m building a voice recording application in iPhone. The recorded file is transfer to linux server and it need to be converted to wav file.
However, when I try to convert caf file using libsnd, it gives an error.
Error : Not able to open input file testfile.caf
For testing I converted some wave file to caf using libsnd and vice versa.
So I think that there is a problem of my recorded file in iphone.
Anyone has got such an experience ?
I hope can someone help me.
Thanks.
If you use AVAudioRecorder, AudioFile, or any of the other Core Audio APIs, you should be able to record directly to a WAV file, and skip the entire conversion process altogether.
But if you need to convert audio files, first check if the CAF file is a valid file. Does it play? Is the header correct? What is the data format? Is it compressed? Does libsnd support the data format?
(The data format is separate from the file format, which is just a container for various bits of data as well as the sample data. The data format could be PCM, or it could be compressed in any format such as MP3, AAC, uLaw, etc.)