I need to insert another audio on specific offset (where space existed in audio track).
So, For that, i need to fetch offset of each space from audio and then apply some ffmpeg/sox command to insert another aduio on that offset.
Please share me some command to fetch the offset for space
first so, later, i could mix the another audio on same offset.
Please suggest your view/command or any library.
I think, it would be possible in FFMPEG OR SOX.
Updated:
I have two audio files.
In which one audio files contains some songs but, there are some space at some interval. These space should be fill by another audio.
example:
Audio 1 - song - audio has space at 5 places.
audio 2 - a people name audio.
output - audio one should fill up the all 5 places by aduio 2.
Thanks
Related
How many bytes can be stored per minute of audio using any method of steganography with a disregard to detectability or any other factor e.g if the original audio begins to sound different
I want to split an audio file into several equal-length segments using FFmpeg. I want to specify the general segment duration (no overlap), and I want FFmpeg to render as many segments as it takes to go over the whole audio file (in other words, the number of segments to be rendered is unspecified).
Also, since I am not very experienced with FFmpeg (I only use it to make simple file conversions with few arguments), I would like a description of the code you should use to do this, rather than just a piece of code that I won't necessarily understand, if possible.
Thank you in advance.
P.S. Here's the context for why I'm trying to do this:
I would like to sample a song into single-bar loops automatically, instead of having to chop them manually using a DAW. All I want to do is align the first beat of the song to the beat grid in my DAW, and then export that audio file and use it to generate one-bar loops in FFmpeg.
In the future, I will try to do something like a batch command in which one can specify the tempo and key signature, and it will generate the loops using FFmpeg automatically (as long as the loop is aligned to the beat grid, as I've mentioned earlier). 😀
You can use the segment muxer. Basic example:
ffmpeg -i input.wav -f segment -segment_time 2 output_%03d.wav
-f segment indicates that the segment muxer should be used for the output.
-segment_time 2 makes each segment 2 seconds long.
output_%03d.wav is the output file name pattern which will result iin output_000.wav, output_001.wav, output_002.wav, and so on.
As title; for some compressed format such as EAC3, AC3 frame starts as a sync word.
So what's PCM (raw audio)'s sync word? How to identify the beginning of a PCM frame?
I met a problem where audio is concatenated by several audio segments and each of them has different frame size. I need to identify the start position.
Thanks in advance.
There is no such concept as a frame in PCM. The concept of a frame is to indicate points of random access. In PCM every single sample is a point of random access, hence start indicators are not required, and there are no standard frame size. It all up to you.
A PCM frame is different from the frames you're describing, in that a frame is just a single sample on all channels. That is, if I'm recording 16-bit stereo PCM audio, each frame is 4 bytes (32 bits) long.
There is no sync word, nor frame header in raw PCM. It's just a stream of data. You need to know the bit depth, channel count, and current offset if you want to sync to it. (Or, you need to do some simple heuristics. For example, apply several different formats and offsets to a small chunk of data and see which one has the least variance/randomness from sample to sample.)
I'm trying to create a MOV file with two audio tracks and one video track, and I'm trying to do so without AVAssetExportSession or AVComposition, as I want to have the resultant file ready almost immediately after the AVCaptureSession ends. An export after the capture session may only take a few seconds, but not in the case of a 5 minute capture session. This looks like it should be possible, but I feel like I'm just a step away:
There's source #1 - video and audio recorded via AVCaptureSession (handled via AVCaptureVideoDataOutput and AVCaptureAudioDataOutput).
There's source #2 - an audio file read in with an AVAssetReader. Here I use an AVAssetWriterInput and requestMediaDataWhenReadyOnQueue. I call setTimeRange on its AVAssetReader, from CMTimeZero to the duration of the asset, and this shows correctly as 27 seconds when logged out.
I have each of the three inputs working on a queue of its own, and all three are concurrent. Logging shows that they're all handling sample buffers - none appear to be lagging behind or stuck in a queue that isn't processing.
The important point is that the audio file works on its own, using all the same AVAssetWriter code. If I set my AVAssetWriter to output a WAVE file and refrain from adding the writer inputs from #1 (the capture session), I finish my writer session when the audio-from-file samples are depleted. The audio file reports as being of a certain size, and it plays back correctly.
With all three writer inputs added, and the file type set to AVFileTypeQuickTimeMovie, the requestMediaDataOnQueue process for the audio-from-file still appears to read the same data. The resultant mov file shows three tracks, two audio, one video, and the duration of the captured audio and video are not identical in length but they've obviously worked, and the video plays back with both intact. The third track (the second audio track), however, shows a duration of zero.
Does anyone know if this whole solution is possible, and why the duration of the from-file audio track is zero when it's in a MOV file? If there was a clear way for me to mix the two audio tracks I would, but for one, AVAssetReaderAudioMixOutput takes two AVAssetTracks, and I essentially want to mix an AVAssetTrack with captured audio, and they aren't managed or read in the same way.
I'd also considered that the QuickTime Movie won't accept certain audio formats, but I'm making a point of passing the same output settings dictionary to both audio AVAssetWriterInputs, and the captured audio does play and report its duration (and the from-file audio plays when in a WAV file with those same output settings), so I don't think this is an issue.
Thanks.
I discovered that the reason for this is:
I correctly use the Presentation Time Stamp of the incoming capture session data (I use the PTS of the video data at the moment) to begin a writer session (startSessionAtSourceTime), and that meant that the timestamp of the audio data read from file had the wrong timestamp - outwith the time range that was dictated to the AVAssetWriter session. So I had to further process the data from the audio file, changing its timing information by using CMSampleBufferCreateCopyWithNewTiming.
CMTime bufferDuration = CMSampleBufferGetOutputDuration(nextBuffer);
CMSampleBufferRef timeAdjustedBuffer;
CMSampleTimingInfo timingInfo;
timingInfo.duration = bufferDuration;
timingInfo.presentationTimeStamp = _presentationTimeUsedToStartSession;
timingInfo.decodeTimeStamp = kCMTimeInvalid;
CMSampleBufferCreateCopyWithNewTiming(kCFAllocatorDefault, nextBuffer, 1, &timingInfo, &timeAdjustedBuffer);
I've written an MP4 parser that can read atoms in an MP4 just fine, and stitch them back together - the result is a technically valid MP4 file that Quicktime can open and such, but it can't play any audio as I believe the timing/sampling information is all off. I should probably mention I'm only interested in audio.
What I'm doing is trying to take the moov atoms/etc from an existing MP4, and then take only a subset of the mdat atom in the file to create a new, smaller MP4. In doing so I've altered the duration in the mvhd atom, as well as the duration in the mdia header. There are no tkhd atoms in this file that have edits, so I believe I don't need to alter the durations there - what am I missing?
In creating the new MP4 I'm properly sectioning the mdat block with a wide box, and keeping the 'mdat' header/size in their right places - I make sure to update the size with the new content.
Now it's entirely 110% possible I'm missing something crucial about the format, but if this is possible I'd love to get the final piece. Anybody got any input/ideas?
Code can be found at the following link:
https://gist.github.com/ryanmcgrath/958c602cff133bd7fa0b
I'm going to take a stab in the dark here and say that you're not updating your stbl offsets properly. At least I didn't (at first glance) see your python doing that anywhere.
STSC
Lets start with the location of data. Packets are written into the file in terms of chunks, and the header tells the decoder where each "block" of these chunks exists. The stsc table says how many items per chunk exist. The first chunk says where that new chunk starts. It's a little confusing, but look at my example. This is saying that you have 100 samples per chunkk, up to the 8th chunk. At the 8th chunk there are 98 samples.
STCO
That said, you also have to track where the offsets of these chunks are. That's the job of the stco table. So, where in the file is chunk offset 1, or chunk offset 2, etc.
If you modify any data in mdat you have to maintain these tables. You can't just chop mdat data out, and expect the decoder to know what to do.
As if this wasn't enough, now you have to also maintain the sample time table (stts) the sample size table (stsz) and if this was video, the sync sample table (stss).
STTS
stts says how long a sample should play for in units of the timescale. If you're doing audio the timescale is probably 44100 or 48000 (kHz).
If you've lopped off some data, now everything could potentially be out of sync. If all the values here have the exact same duration though you'd be OK.
STSZ
stsz says what size each sample is in bytes. This is important for the decoder to be able to start at a chunk, and then go through each sample by its size.
Again, if all the sample sizes are exactly the same you'd be OK. Audio tends to be pretty much the same, but video stuff varies a lot (with keyframes and whatnot)
STSS
And last but not least we have the stss table which says which frame's are keyframes. I only have experience with AAC, but every audio frame is considered a keyframe. In that case you can have one entry that describes all the packets.
In relation to your original question, the time display isn't always honored the same way in each player. The most accurate way is to sum up the durations of all the frames in the header and use that as the total time. Other players use the metadata in the track headers. I've found it best to just keep all the values the same and then players are happy.
If you're doing all that and I missed it in the script then can you post a sample mp4 and a standalone app and I can try to help you out.