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
Related
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.
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 am new to ffmpeg, i have spend more than 10 days on finding any way to do muxing in mp4 format with audio and vedio in streams buffer not in file.
What i want is to mux mp4 format audio & vedio in a streams.
I am able to do muxing mp4 format in file. But not able to get mux mp4 format in streams buffer.
Till now i have tried this:
avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size, 1, &bd, NULL, &write_packet, NULL);
By calling this avio_alloc_context and passing reference of write_packet function. I am able to get call write_packet. But when i write the data coming in write_packet in a file, and making mp4 file with that. The resultant mp4 file is not working. There is no vedio or audio information available by watching this file in Media Info.
The header is written , then loop is written and finally trailer is written, but not getting success in running final file.
Is there any good way to do mux in mp4 format in streams, so please tell me.
Kindly help to me to do this.
Thanks in advance.
I have successfully changed the muxing.c sample to use video frames that I generate on runtime.
I am trying now to replace the get_audio_frame function with a function that decodes an existing audio file, and writes its samples instead of the synthesized audio-samples in the example code.
I've tried using the "audio decoding" example to decode the audio file, but the not sure how / when to write the samples decoded.
I suggest to check the source of my Karaoke Lyrics Editor which is doing exactly what you need based on ffmpeg. See ffmpegvideoencoder.cpp, see createFile and encodeImage functions.
I have an application for iPAD.
This application records the voice of the microphone.
The audio formats of the item must be PCM, MP3 and WAV files. The MP3 file I get it starting from the original raw file and then convert using LAME.
Unfortunately I have not found any example that allows me to convert a PCM file to a WAV file.
I just noticed that if I put the file extension to WAV format, starting from the raw application saves without problems, so I think that there is no type conversion from PCM WAV files.
Correct?
PS: Sorry for my english ... I use Google Translate
WAV is some kind of a box. PCM is in the box. There are many container formats like MP4. MP4 can contain audio, video or both. It can also contain multiple video or audio streams. Or zip files. Zip files can contain text files. But zip files can also contain images, pdfs,... But you can't say "how can I convert a zip file to the text file inside the zip".
If you want to convert PCM data to a WAVE file you should not many problems because WAV files are quite simple files. Take a look at this:
(See also WAVE PCM soundfile format.)
You first need that header and after you can just append all your pcm data (see the data field).
Converting PCM to WAV isn't too hard. PCM and WAV both format contains raw PCM data, the only difference is their header(wav contains a header where pcm doesn't). So if you just add wav header then it will do the tricks. Just get the PCM data and add the wav header on top of the PCM data. To add wav header with PCM data, check this link.
I was working on a system where it accepts only wav files, but the one I was receiving from amazon Polly was pcm, so finally did this and got my issue resolved. Hope it helps someone. This is an example of nodejs.
// https://github.com/TooTallNate/node-wav
const FileWriter = require('wav').FileWriter
let audioStream = bufferToStream(res.AudioStream);
var outputFileStream = new FileWriter(`${outputFileFolder}/wav/${outputFileName}.wav`, {
sampleRate: 8000,
channels: 1
});
audioStream.pipe(outputFileStream);
function bufferToStream(binary) {
const readableInstanceStream = new Stream.Readable({
read() {
this.push(binary);
this.push(null);
}
});
return readableInstanceStream;
}