I want to create an M4A file from an MP4, I want to attempt this from scratch without using other libraries but just the raw data.
So far I am able to locate the atom moov and parser it. And as a result I can pull the audio data from the mdat. So then I create my own M4A file with the right ftyp (M4A isomiso2) then add a new mdat with just the audio data I previously recovered, finally I add the moov with the same mvhd, and only the audio trak but with an updated stco to reflect the change in offsets of the chunks of audio data (as they are just one after each other now). I am sure I am doing all of this right.
However the M4A file just plays silence. I believe it is because I have to edit more in the moov but I am not sure what - I put it into FFmpeg corruption and I got:
"Sample rate index in program config element does not match the sample rate index configured by the container."
"Too large remapped id is not implemented."
So as a result I think it is something to do with the stsd atom but I am not sure how to change it.
Related
Update:
I have a video player in browser which plays mp4 videos though websocket. The player only supports mp4 file. When i checked normal mp4 fiels does not play in the player, a mp4 file with a "moovflags faststart " will only play on that player. For a allready stored file , this will work properly.
But In case of an livestream(RTSP), using ffmpeg will only work once the RTSP connection has terminated since the "moovflags faststart " flags will work once a connection has terminated properly.
Hope the above statements makes more sense.
Due to this behavior, am checking if there is any way to get the moovflasg at first or something
I am having RTSP live source and i need it to convert the RTSP to a mp4 file which has moov flags in the begining of the file.
I have checked with openrtsp to take a mp4 dump of the rtsp, but it only adds moov flags and other info on the footer of the mp4(onlky when openrtsp has closes the rtsp stream).
Ffmpeg has " -movflags faststart" to move the footer info to the header of the mp4 container.
Since i am having a RTSP live source, the video data will be comming back to back and there wont be any termination. The above ffmpeg command only works once the rtsp stream has terminated.
Is there any way we can make a mp4 container which contains the mp4 footer info present in the header itself so that i can use it for a live source?
EDIT #1
I have video player which plays mp4 video files , it only support playback of a recorded mp4 file which is createtd using "-movflags faststart" , normal mp4 files does not play in that.
This is the player
https://github.com/sonysuqin/WasmVideoPlayer.
Since i am tryng to stream live video to the player, its not possible to use movflags faststart.
The mp4 header can not be added to the file before it is complete. It’s not possible because of how mp4 files are structured. The header needs to know the frame type, timestamp, size, and file offset of every frame in the file. That can’t be known until the file is complete. You can not stream an mp4 while it is being created. You need to use a protocol such as HLS or DASH to accomplish this.
The thing which I am doing right now is that I am playing RTMP streaming on media server using ffmpeg command and also creating an audio file using google text to speech.
So I want to update mp3 file with silence if there is no content, so that it will keep will keep stream running.
I have tried 2 approaches:
By writing raw binary data to mp3 file but not working as it says content is not accurate.
Concatenate the audio content with the silence data and export file. In this scenario, I am able to update file but stream broken at the point while we are exporting file.
I have tried to write the audio file with binary data and also tried to concatenate audio content with silence and then export file but it break stream while we export the file.
I'm trying to convert DVD iso files to mp4 using HandbrakeCLI. I use the following line in a batch file:
D:\HandBrakeCLI.exe -i "D:\input.iso" -o "D:\output.mp4" --no-markers --width "720" --height "480" --preset "HQ 480p30 Surround" --encoder "mpeg2" --audio-lang-list "eng"
When I do this, I must then extract the audio from the file, using the following line:
D:\eac3to\eac3to.exe "D:\output.mp4" "D:\output.wavs" -down16
However, when I attempt to extract the audio, I get the error message
The format of the source file could not be detected.
Is there anything wrong with my former line of code that's causing the mp4 to get screwed up?
Minor side question: I'm also trying to get handbrake to remove subtitles and also only keep English audio, do you know what code could be used for that? I started a bit there with the --audio-lang-list "eng" but I'm now sure what to do from there.
Thanks a lot in advance!
You need to use a valid audio format. .wavs is not valid. You have to use an available audio codec to output to the below for --aencoder. The default output audio for MP4 is .aac
av_aac
copy:aac
ac3
copy:ac3
eac3
copy:eac3
copy:truehd
copy:dts
copy:dtshd
mp3
copy:mp3
vorbis
flac16
flac24
copy:flac
opus
copy
Defaults for audio
av_mp4 = av_aac
av_mkv = mp3
You need to pass none for no subtitles
-s none
And define only eng track like you were doing
--audio-lang-list eng
Check out the Handbrake CLI Documentation for the command line code:
https://handbrake.fr/docs/en/latest/cli/cli-guide.html
You can also try using a different program once you extract the audio. A program like XMediaRecode. It can also remux audio and video and convert other audio formats to wav
https://www.videohelp.com/software/XMedia-Recode
I have a folder which contains lot of MP3 files, some of them are encoded using mp3PRO.
Since this format is now obsolete, I'd like to convert them back to MP3 (converters can be found easily).
Is there is a way to detect programatically if a file is encoded using mp3PRO format ? (eg : by looking at file header or specific signatures using an hex editor)
The official player is able to detect if file is encoded using mp3PRO (the logo is highlighted or not) so I suppose this is technically possible.
What I found so far is that bitrate of mp3PRO file appears to be pretty low (50% of non encoded file) : eg : a 128 kbps file will appears as 64kbps. However a 320 kbps file will appears as 160 kpbs (which are pretty common) so it cannot be used as a rule.
Here is what I found out and how I fixed it. I wrote in here in case somebody would need it :
MP3Pro files does not contains any special flag in the mp3 header that would help to recognize them.
They are technically very similar to usual mp3 files, except they are encoded half the bit and sample rate (eg : a 128kpbs 44100hz file will be encoded as a 64kps 22050hz file, resulting in mp3pro file being approx half the size of original file).
This has been made for compatibility, so default players can play them without any change.
They also contains some SBR data, which allow to synthetically rebuild the lost audio part (high frequencies) and to play them it was before the mp3 pro conversion.
Detecting the SBR data seems very hard if not impossible : it would require to decode the actual mp3 frames. Also there is no documentation to be found about mp3pro format.
What I did (which works but required some manual effort) : I added all files to be checked to playlist of an mp3 player (foobar 2000 in my case) then sorted the files on the sample rate column : most 22050 hz mp3 files were indeed mp3 pro files.
They were converted back to mp3 using winamp + the mp3pro plugin made for it, available here : http://www.wav-mp3.com/mp3pro-to-mp3.htm
Im trying to extract each frame from a rtsp mp4 stream, and convert that into a jpeg/gif using ffmpeg. I'm getting the sdp header from 000001b0.....000001b5, and adding that into an byte array then capturing a frame starting from 000001b6 and appending it to the byte array.
When I flush it to a file (.mpg) and use ffmpeg it throws errors and not converting.
my header looks like 000001B008000001B58913000001000000012000C488BA98514043C1463F and after this I'm appending a frame (starting from 000001b6).
I did something similar with FFMPEG, and it seems that the frame data you get from FFMPEG already contains the frame header, which is all you need to transcode the data. Please make sure that you decode the mp4 data to a raw format (RGB24 for instance), then encode it to the pixelformat the JPEG/GIF encoder expects (probably a YUV format) using libswscale, before passing the data to the encoder.
Depending on the Codec you may not have to add anything or you may have to add a lot..
This is referred to as de-packetization and MPEG4-ES has no packetization model... H264 has many depending on the profile.
Check out the RFC..
Either 3016 or 3640 should help you.
https://www.rfc-editor.org/rfc/rfc3640
https://www.rfc-editor.org/rfc/rfc3016