I am writing some images to an AVStream and after that I am reading an mp3 file and writing it to an diffrent AVStream. The propblem is that the audio stream is a bit shorter then the video stream, so if I add more images and another audio file the audio is not in sync with the video any more. So my idear was to write silent audio data to the audio stream before I write another audio file to the audio stream. But I can not figure out how to write the silent data to the audio stream.

I have an AC3 5.1 audio file to which I would like to insert x seconds of silent audio at the beginning. This has nothing to do with video muxing, so itsoffset is useless since it seems to only work with an audio stream accompanying a video one. I would like to achieve this with ffmpeg. Any ideas?


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Use the anullsrc audio source filter in to create the silent audio. You'll need to match the format, channel layout, and sample rate of the main audio file. Example to make a 5.1 channel, 48000 Hz sample rate, 1 second silent AC3 audio file (as this was what the format in the question):

Use the adelay audio filter if you want to do everything in one command, or if you want to output to a different format than the input (since this method re-encodes anyway). This only works to add silence to the beginning of a file.

Use the apad audio filter if you want to do everything in one command, or if you want to output to a different format than the input (since this method re-encodes anyway). This only works to add silence to the end of a file.

I was just trying to record a simple one track voice over with nothing fancy like some noise reduction and stuff. But when I export the audio file, I can't hear anything even though everything is fine in Audition.

Indeed it compresses better by lossless Gzip or PKzip thanby lossy MP3.At the lowest quality MP3 encoding that Apple iTunes player offers (16kibit/s mono), itcameto 3.7 MiB, over a thousand times what lossless compressioncan do. Although those MP3s can then be zip compressed (as the they mainly consist of justduplicate block headers), fully lossless is still smaller.Most audio players however would need lossless audio stored uncompressed & some don't evenhandle anything but the old de facto standard of MP3,I've included the MP3s here as downloads too.

I think it is okay for me to give away this silence for free(releasedunder Gnu Public Licence). There has been one notorious case of theestate ofJ. Cage suing M. Batt for selling a silent recording, Cage having beenan avantgarde composer who had previously published a famous silent piece.However itwas reported that the suing was because Batt had jokingly credited Cagewithjoint composership of the track without asking permission first& profitedfrom the publicity of that creditation without paying royalties. Thesuing wasnot merely because it was silent as the general concept of silences inperformances & broadcasts pre-dates both Cage & Batt. Anyway,it was eventually revealed to have been primarily a publicity stuntarranged between the two sides.


My actual inspiration for a joke silent recording came fromDavid Nobb's'The Return of Reginald Perrin' in 1977. In that humorous story, afictionalcompany included in its deliberately useless product range silent vinylrecordswith titles such as 'Trappist Monastery Chants' & 'Laryngitisfrom ManyLands'. It was twenty six years before I got around to making one!

I'm sure I've solved this one before but can't seem to recall how to fix the problem. Any m4a (or mp3 for that matter) I try to import into my Avid system is silent - no samples appear on the sample plot, though the audio files are fine.

It looks EXACTLY the same - the audio track is still there, it's not muted, and the audio is still fine BEFORE the cut. But for some reason there is no sound after the cut. I have NO IDEA why this is happening and am desperate for answers.

Yes it does! It's very confusing. Also if, after I cut it, I then nest the silent half of the track - the audio will return. I'm using that as a working solution for now, but really would like to get to the bottom of this and find an actual solution for whatever the problem is.

Ok this has been melting my brain for days, but I think I might have discovered a work around. If you are working in a multi-camera sequence (as I was), I started losing audio tracks the more cuts I made. Sound before the cut, then no sound after the cut...

Ive been having the same issue for like two weeks now and i think i found the solution.

It happened to me when I was working at a multicam sequence so maybe this solution does not apply to everyone, or maybe the issue itself has something to do with the multicam sequence only. Just go to the audio track that is "muted">right click>Multi-Camera>Enable and Voil! I hope it helps.

Hello! I've been working on and off on a project for Bela that (for now) would simply be a compact audio recording device. I like to take my modular setup or other instruments out to coastal views, redwoods, etc. with a portable power source, so not having to also bring my laptop and interface is desirable.

For now, I'm starting simple. My code thus far is mostly a hybrid of the sinetone example and a modified version of the sample streamer example; now it is writing audio files rather than reading them. The only external file being used is a header file from the examples defining the SampleData struct.

I'm sure it's some small mistake being made on my part. I've shifted things around in the code several times to see if things were just happening in the wrong order or something, but while I made some small efficiency and fixes along the way, so far the audio file output has not changed. My code is attached below. Any insight is appreciated!

I am not very familiar with the libsndfile API, but it seems that you are actually writing multiple files, each overwriting the previous one. For each file, you seem to be specifing that the first sample to be written is startSamp. I assume the previous samples are then filled in with zeros. Given how the file you are looking at is the last one that was created when the program stopped running, it contains gChunk leading zeros and BUFF_SIZE samples of actual audio.

I just installed Audio Recorder on an Ubuntu 12.04 and tried to record something from my system's output. Unfortunately it does not record anything. It shows up perfectly, as if it would record, but I just get "silent" files. I tried all possibilities with the audio source, both on Audio Recorder and on my system: "Built-in Audio Analog Stereo (Audio Output)", "System's default device" - on the app and "Analog Stereo Duplex" (the default) and "Analog Stereo Output" on my system, but the result is the same: silent audio files.

We present a novel system that gets as an input, video frames of a musician playing the piano, and generates the music for that video. The generation of music from visual cues is a challenging problem and it is not clear whether it is an attainable goal at all. Our main aim in this work is to explore the plausibility of such a transformation and to identify cues and components able to carry the association of sounds with visual events. To achieve the transformation we built a full pipeline named 'Audeo' containing three components. We first translate the video frames of the keyboard and the musician hand movements into raw mechanical musical symbolic representation Piano-Roll (Roll) for each video frame which represents the keys pressed at each time step. We then adapt the Roll to be amenable for audio synthesis by including temporal correlations. This step turns out to be critical for meaningful audio generation. In the last step, we implement Midi synthesizers to generate realistic music. Audeo converts video to audio smoothly and clearly with only a few setup constraints. We evaluate Audeo on piano performance videos collected from Youtube and obtain that their generated music is of reasonable audio quality and can be successfully recognized with high precision by popular music identification software.

In file-based video processing workflows, content providers often standardize on the mezzanine file format used to share content with their customers and partners. When sharing video-on-demand (VOD) content, produced mezzanine files often have same number of audio tracks regardless of how many are in the actual content. For example, MXF media assets may be produced with eight mono audio tracks for content that contains only 2.0 audio, so only two tracks are actually used while the remaining tracks are filled with silent audio.

To build an automated video processing workflow that makes intelligent decisions based on audio, we need to detect silent audio tracks and their position in the source mezzanine assets. From the example above, if six audio tracks are identified as silent, the asset would be processed as a stereo 2.0 asset. On the other hand, if six tracks have audio and two are silent, the asset should be sent to a 5.1 workflow for processing.

In this post, we will create a workflow using AWS Elemental MediaConvert to analyze audio tracks of media assets and measure their loudness. The workflow is automated by AWS Lambda functions and triggered by file uploads to Amazon S3.

The Audio Normalization feature of MediaConvert makes it easy to correct and measure audio loudness levels, supporting the ITU-R BS.1770-1, -2, -3, and -4 standard algorithms. MediaConvert configuration allows the selected algorithm to only produce loudness measurements. It also allows logging loudness levels and storing these logs in S3.

We will be using MediaConvert to analyze loudness levels of the first 60s of media source assets. The workflow is flexible, so we can analyze more than 60s if required. The produced loudness logs provide the Input Max Loudness value for each audio track that we will compare to a threshold of -50 dB. If the Input Max Loudness value of an audio track is lower than -50 dB, that track is considered silent audio. 2351a5e196

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