How to detect the audio availability in the SUSE Linux system - linux

I got some audio pieces in flv format. Each of them is about 10 seconds long.
My question is how to detect whether the audio pieces has "sound", in other words, sometimes the audio pieces has no sound even the size of it is not 0 byte, so how to find those broken/silent audio files by some linux tool/command?
Maybe ffplay can do this? any available advice?

If you want to quickly check if the stream is video only, audio only or if it contains both audio and video, try hexdump -C filename | head. The fifth byte contains information about the contents of the file.
0x01 - video only
0x04 - audio only
0x05 - audio + video
You can also try to play the file using VLC media player. There is a menu option that enables informational messages from the media being played back.

Related

Mux segmented mpegts audio and video to single clip with error correction

I have a recording as a collection of files in mpegts format, like
audio: a-1.ts, a-2.ts, a-3.ts, a-4.ts
video: v-1.ts, v-2.ts, v-3.ts
I need to make a single video clip in mp4 or mkv format.
However, there are two problems:
audio and video segments have different duration each, number of audio segments is different from number of video segments. Total duration of audio and video matches. Hence I can not concat pairwise audio video segments using mpeg and merge them afterwards, I get sync issues increasing progressively
few segments are corrupt or missing. So if I concat audio and video streams separately using ffmpeg I get streams of different lengths. When I merge these streams using ffmpeg I have correct a/v synchronization until time when first missing packet is encountered.
It's OK if video freezes for a while or there is silence for a while as long as most of the video is in sync with audio.
I've checked with tsduck and PCR seems to be present in all audio and video segments yet I could not find a way to merge streams using mpegTS PCR as sync reference. Please advise how can I achieve this.

ffmpeg - correctly handling misaligned audio/video input stream before outputting to rtmp

I use a video player called MPV to transcode a dynamic playlist of media files.
I pipe MPV's encoded output into FFMPEG and format it for rtmp delivery.
However the playlist may contain media with misaligned audio and video, ie - the audio track may be shorter / longer than the video track.
No matter what MPV will only output what it's given. So if my media file has audio that is 1 second long and video that is 2 seconds long, it will output a media stream with exactly the same misalignment, rather than generating null audio or skipping to the next item in the playlist when it first encounters an active stream ending (eof).
For example, assuming my playlist was full of problematic media where the audio and video of each file was misaligned:
If I output this media stream to a popular streaming service's server, it could lead to stuttering and/or loss of a/v sync.
Similarly, if I output this media stream to a file and played it back in MPV or another video player, the result appears to be more like this:
I have tried to fix this in MPV in all sorts of ways, trying every relevant command line option available. I even wrote a user script that detects 'eof' audio and skips to the next item in the playlist, but it is not fast enough and still leads to small gaps of audio.
So my only hope is correcting it in ffmpeg. In the event of null audio/video, I need a fallback or a generative filter that can fill these empty gaps with silence (audio) or a colour/image (video).
I'm open to any ideas, and if my understanding in a/v encoding is a little off please educate me.

How to add a 5.1 .flac audio track to a .ts file with already 3 audio tracks?

I want to add a 5.1 .flac audio track to a .ts file that already has three audio tracks. I tried with tsMuxer and ffmpeg with unsuccessful results. In tsMuxeR the .flac track is not recognized and in ffmpeg everything seems to work fine until the very last moment when I check the file and the .flac audio track is not included in the "output.ts". The .flac track is about 3GB and its lenght is around two and a half hours.
Thank you so much.
I don't think you'll find any existing software that maps FLAC into a MPEG-2 Transport Stream.
This gives you an idea what sort of issues you run into: https://xiph.org/flac/ogg_mapping.html
Let's say you came up with a reasonable way of mapping FLAC into a MPEG-2 Transport Stream - there won't be anything reading it.
Unless there is a specified way of mapping FLAC into a MPEG-2 Tranport Stream - you are on your own.
But PCM is supported in a MPEG-2 Transport Stream (for example Blu-Ray).
I'd use ffmpeg to transcode your audio from FLAC to PCM and then mux it into your transport stream.
Your audio transcode (FLAC to PCM) is lossless.

How to read video file using v4l2

I want to read a video file using v4l2, say an AVI file. And read it frame by frame.
As far as I can tell I need to use the read() function. But how isn't very clear to me. There are also hardly any examples available. So maybe a simple example on how to do this would help.
This is not what the Video4Linux2 (V4L2) API is for. It is not designed for reading multimedia files from disk, decoding them and playing them. Rather, it is designed to interface to assorted multimedia input devices (like webcams, microphones, TV tuners, and video capture devices), capture A/V data, and play it.
Take it from the V4L2 API introduction:
Video For Linux Two is [...] a kernel interface for analog radio and
video capture and output drivers.
For reading an AVI file and decoding/playing it (programmatically) on Linux, look into FFmpeg or GStreamer.

How to determine the codec of an audio file on Windows?

I need to find the codec of an audio file. How can I do this?
Do I need to write code to do this or is there a simpler way?
Please help me. If possible share helpful links.
The good old file utility will reveal lots of information about audio files, sometimes including the codec:
$ file X.wav
X.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 16000 Hz
#bhebsquines
"It is important to distinguish between a file format and an audio codec. A codec performs the encoding and decoding of the raw audio data while the data itself is stored in a file with a specific audio file format. Although most audio file formats support only one type of audio data (created with an audio coder), a multimedia container format (as Matroska or AVI) may support multiple types of audio and video data." - http://en.wikipedia.org/wiki/Audio_file_format
The application gspot does a good job of pulling codecs from audio and video files.
http://www.headbands.com/gspot/
run it and drag a file into the window. It will pull all of the data from there. Note that some audio files will not display a codec as they are made from "Raw" audio.
You can identify your codec by extenion name itself example file1audio.mp3 or fileaudio.avi, .mp3, .avi will be your file type or codec, you can use k-lite codec pack for your different type audio format. use this link http://www.free-codecs.com/download/k_lite_codec_pack.htm

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