problem with uploading audio file on Video Indexer Portal - audio

I try to upload an audio file (75 minutes of record, 64 MO approximately) on Video Indexer portal.
It just frozen at 30% during the time of indexation.
It do not change since hal an hour now.. I made another try by downlanding the file again. The same problem come again.
I try with another file (I compressed the audio file to 22 Mo), same problem : it frozen at 30% upload
How fix it ?

Related

ffpmeg - how to detect if video crop is completed?

Thanks in advance.
I'm trying to crop a .mp4 video using an ffmpeg binary (within the context of an electron-react-app).
(The binary is run in a child process using execFile() and outputs to a temp folder which is later deleted)
ffmpeg varies considerably in the time it takes to complete the creation of a cropped video file (1sec to 18sec) depending on the computer (mac vs Windows).
I need to read the cropped video file.
I've set up an event listener in the Main process of electron
if (!monitorCroppedFile) {
console.log(`${croppedFilePath} doesn't exist`);
} else {
console.log(`${croppedFilePath} exists !`)
...readFile...;
Once monitorCroppedFile = true I read it using fs.readfile().
The problem is that ffmpeg initally creates the cropped file path but it sometimes takes ages to complete the process of cropping.
This results in the read file often being blank (as the read is triggered on detecting the file path of the cropped file).
I've tried using -preset ultrafast in the ffmpeg arguments but this only improves things on Windows marginally.
The problem doesn't occur on Macs.
Can anybody suggest a possible solution ? Is there a way to detect when the crop is fully completed ?
Many thanks.
Add -progress FILE to your command where FILE should be a filename. ffmpeg will log processing status to that file. Search for the line progress=end in it. Once you find it, you can read the file.

Play RTMP streaming and also want to update the mp3 file without breaking stream while running stream in parallel

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.

Creating M4A file from MP4

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.

Media Info says wav file is truncated instead of showing the properties

I have written some data to a wav file along with the header. But when i play it using media player or VLC it does not play. When I open the same file using MediaInfo, it says 'IsTruncated: Yes' . what does this truncated mean and how to solve this.
Please help

Cloudfront issue streaming certain video files with VP6F and MPEG audio

Greetings fellow amazonians
We have certain f4v files from webtvinteractive.com that are not working from a streaming distribution. The cloudfront diagnostic tool indicates the file is not found but from a download distribution they play all ok fine ??
When inspecting the codec the only difference we see is that the audio is encoded as MPEG audio layer 1/2/3 (mpga) instead of AAC (mp4a)
Also noticed that VP6F is used but that is a Abobe proprietary codec so it should stream all ok from Flash Media Server
Here is the test tool url:
http://d1k5ny0m6d4zlj.cloudfront.net/diag/CFStreamingDiag.html
s3dbhkulzq9ndp.cloudfront.net
teaser.f4v (ok)
EM2_H264.f4v (NOT ok)
StreamLen Response: 0 ??
Received an answer from AWS team member
Posted by: RonS#AWS
Posted on: Oct 10, 2011 7:29 AM
https://forums.aws.amazon.com/message.jspa?messageID=284676
Hi, I downloaded both of your video objects and ran flvcheck (http://help.adobe.com/en_US/FlashMediaServer/3.5_AdminGuide/WS5b3ccc516d4fbf351e63e3d119f29261b7-7ffc.html) against them.
The teaser.f4v is apparently OK according to flvcheck, but EM2_H264 gets this:
11-10-10 07:10:25 Error: -26 File appears to be FLV with wrong extension. EM2_H264.f4v
If I rename the file to have a .flv extension, I get "Found backward timestamp".
I suspect the FMS software we're using will have the same complaints about that file that flvcheck does.
flvcheck is available for download here (you'll need to register with Adobe): http://www.adobe.com/products/flashmediaserver/tool_downloads/

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