I am looking for commmandline to slow down Quick Time formated MOV files. Most likely using FFMPEG. I do not mind converting to MP4 format either.
To slow down your video, you have to use a multiplier greater than 1:
ffmpeg -i input.mov -filter:v "setpts=2.0*PTS" output.mov
I am not sure if this works right now.
batch slow down .mov speed (No answer here either)
Almost impossible without full reencoding (or transcondig).
If the source is video only, it can be easily done by simple hex editing. Just change the track timescale value in the MDHD box =>
http://wiki.multimedia.cx/?title=QuickTime_container#mdhd
The lower timescale the slower play rate.
I've tested it works as following:
1) find out current frame rate with Mediainfo tool
2) Open the file with HxD
3) Recklessly search 'mdhd'
4) Between 'mdhd' and 'hdlr', find 16 bit big endian hex representation of frame rate and change it
I'm not sure but this kind of hacking seems not supported by ffmpeg.
But if it also has audio track, changing its timescale will produce noisy sound, therefore reencoding is unavoidable.
Transcoding is rather straightforward work. I'd recommend HandBreak or other GUI frontends.
Use this line
ffmpeg -i input.mkv -filter_complex "[0:v]setpts=0.5*PTS[v];[0:a]atempo=2.0[a]" -map "[v]" -map "[a]" output.mkv
I used this link
https://trac.ffmpeg.org/wiki/How%20to%20speed%20up%20/%20slow%20down%20a%20video
Related
I'm trying to play/skip some clips of a video using ffplay. My first approach to skip say frames 100 to 400 was:
ffplay -vf "select='lte(n\,100)+gte(n\,400)'" -i INPUT
this skips the desired frames, however it also freezes the video during the skipped frames. I tried to fix this by modifying the video presentation time stamp (PTS) with the setpts option:
ffplay -vf "select='lte(n\,100)+gte(n\,400)',setpts='PREV_OUTPTS'" -i INPUT
this seems to work (stills freeze a bit, guess is because of buffering), but now the audio is out of sync. I've tried applying a select filter and modifying the PTS on the audio as well
ffplay -vf "select='lte(n\,100)+gte(n\,400)',setpts='PREV_OUTPTS'" -af "aselect='lte(n\,100)+gte(n\,400)',asetpts='PREV_OUTPTS'" -i INPUT
this skips some audio frames, but still out of sync. I've tried with the aresample=async=10000 option with similar results. Moving some/all of the filters to the output (placing them after the -i INPUT) doesn't work either.
Does someone know how to skip parts of a video using ffplay? Many thanks
Audio frame numbers != video frame numbers. AAC audio generated by FFmpeg's encoder is 1024 samples per frame, so a 48kHz stream has 48000/1024 = 46.875 audio frames per second. Other codecs may have different rates.
Use t instead of n, and generate a continuous series of timestamps.
ffplay
-vf "select='lte(t\,4)+gte(t\,16)',setpts=N/FRAME_RATE/TB"
-af "aselect='lte(t\,4)+gte(t\,16)',asetpts=N/SR/TB"
-i INPUT
I assume a video frame rate of 25 fps. Modify accordingly.
I'm using AVAudioPlayer to play a short one or two syllables word in my app. Currently there are 250 .aiff files and each of the file is roughly 88KB to 125KB each. Currently my ipa file is at 29Megs and I am trying to find the best way to reduce the size so user don't have to have a wifi to download it.
Each of the sound clip is 1 to 2.5 seconds long. I don't need to pause, rewind, fast forward etc.
Based on what I read so far AVAudioPlayer only play aiff, wav or caf and none of those are compressed. Any recommendation on what I can do? Thank you.
CAF files can be compressed, using either lossy or lossless algorithms. If you're using the command line afconvert utility to convert, you could try:
afconvert -f caff -d ima4 audiofile.wav (for IMA4 compression)
afconvert -f caff -d aac audiofile.wav (for AAC compression)
There's also a -b parameter that allows you to set the output bit rate, and a bunch of other options. The man page is basically empty but the full array of possibilities are listed in QA1534.
The Problem I have is when using ffmpeg to encode a YUV using libx264 I don't get all the frame information in -vstats output. It raises the question of how reliable ffmpeg is, and therefore can any 'codec benchmark' review based on ffmpeg be trusted?
I am analysing codec's to determine how they perform. I am using ffmpeg and its -vstats option to look at an encoded movie frame by frame. the process I use:
RAW YUV -> bar-code each frame with frame number -> Bar-coded YUV
Bar-coded YUV -> encoded (e.g. with libx264) -> MKV -> Decoded to YUV
I can compare the two outputs ('Bar-coded YUV' & 'Decoded to YUV') using the bar-code in each frame. I can then compare, exactly, an original frame with an encoded frame using PSNR etc.
When encoding using libx264 and libdirac, there are some frame information which is missing. Other codecs, such as mpeg2video or even libvpx, don't have this problem.
I have found that libx264 vstats are missing for the first 40 to 50 frames. I have since proved that the missing information is actually the last 40 to 50 frames.
It also looks like ffmpeg calculates average bitrate based on the information in vstats. But as there is missing frames the average bitrate is less than what it should be.
Below are links to the average bitrate error example:
http://dl.dropbox.com/u/6743276/ffmpeg_probs/ffmpeg_av_bitrate_error.png
http://dl.dropbox.com/u/6743276/ffmpeg_probs/ffmpeg_av_bitrate_error.xlsx
Below is a link to the PSNR & f_size graph:
http://dl.dropbox.com/u/6743276/ffmpeg_probs/frame_mismatch.png
Below is a link to the output & command line options:
http://dl.dropbox.com/u/6743276/ffmpeg_probs/stderr.txt
I think this is also a bug, anyone clever enough to work it out might want to follow this tracker:
http://roundup.ffmpeg.org/issue2248
I have just discovered something which makes me very red in the face!! quite annoyed, but never mind :)
A fellow ffmpeg user pointed out that ffprobe should output more frame info, which it did. here is a link to his handy tip:
http://forums.creativecow.net/thread/291/71
Using this I found the following:
Actual average bitrate (ffprobe data): 8355.2776056338
Actual average bitrate (ffmpeg vstats data): 8406.23275471698
Ffmpeg -vstats avg_br: 7816.3
Reproduced above: 7816.32168421053
Ffmpeg standard error output 'bitrate=': 8365.8
Below is a link to my workings out:
http://dl.dropbox.com/u/6743276/ffmpeg_probs/ffprobe_vs_ffmpeg-vstats.xlsx
What I have discovered is I should have been using the average bitrate info from ffmpeg standard error output, it looks like the most reliable!
I'm making an Iphone game, we need to use a compressed format for sound, and we want to be able to loop SEAMLESSLY back to a specific sample in the audio file (so there is an intro, then it loops back to an offset)
currently THE ONLY export process I have found that will allow seamless looping (reports the right priming and padding frame numbers, no clicking when looping ect) is using apple's afconvert to a aac format in a caf file.
but when we try and encode to lower bitrates, it automatically re samples the sound! we do NOT want to have the sound re sampled, every other encoder I have encountered has an option to set the output sample rate, but I can't find it for this one.
on another note, if anyone has had any luck with seamless looping of a compressed file format using audio queues, let me know.
currently I'm working off the information found at:
http://developer.apple.com/mac/library/qa/qa2009/qa1636.html
note that this DID work PERFECTLY when I left the bitrate for the encode at default (~128kbs) but when I set it to 32kbps - with the -b option - it resampled, and looping clicks now.
It needs to be at least 48kbps. 32kbps will downsample to a lower sample rate.
I think you are confusing sample rate (typical values: 32kHz, 44.1kHz, 48kHz) and bit rate (typical values: 128kbps, 160kbps, 192kbps).
For a bit rate, 32kbps is extremely low. Sound will have bad quality at this bit rate. You probably intended to set the sample rate to 32kHz instead, which is also not outright typical, but makes more sense.
When compressing to AAC and uncompressing back to WAV, you will not get the same audio file back, because in AAC, the audio data is represented in a completely different format than in raw wave. E.g. you can have shifts by few microseconds, which are necessary to convert to the compressed format. You can not completely get around this with any highly compressed format.
The clicking sound originates from the sudden change between two samples which are played in direct succession. This is likely taking place because the offset to which you jump back in your loop does not end up to be at exactly the same position in the AAC file as it was in the WAV file (as explained above, there can shifts by microseconds).
You will not get around these slight changes when compressing. Instead, you have to compensate for them after compression by adjusting the offset. That means you have to open the compressed sound file in an audio editor, e.g. Audacity, and manually find another offset close to the original one, which is suitable for looping.
How to find an offset which is suitable for looping?
Zoom in to the waveform's end. Look at how the waveform looks there. Then zoom in to the waveform at the original offset and search in its neighbourhood for an offset at which the waveform connects seamlessly to the end of the waveform.
For an example how this shoud look like, open the uncompressed audio file in the audio editor and examine the end of the waveform and the offset there.
I need to know if it is possible to create a 30 second sample MP3 from a WAV file. The generated MP3 file must feature a fade at the start and end.
Currently using ffmpeg, but can not find any documentation that would support being able to do such a thing.
Could someone please provide me the name of software (CLI, *nix only) that could achieve this?
This will
trim out from Position 45 sec. the next 30 seconds (0:45.0 30) and
fade the first 5 seconds (0:5) and the last 5 seconds (0 0:5) and
convert from wav to mp3
sox infile.wav outfile.mp3 trim 0:45.0 30 fade h 0:5 0 0:5
Check out SoX - Sound eXchange
I have not used it myself but one of my friends speaks highly of it.
From web page (highlighted my me):
SoX is a cross-platform (Windows,
Linux, MacOS X, etc.) command line
utility that can convert various
formats of computer audio files in to
other formats. It can also apply
various effects to these sound files,
and, as an added bonus, SoX can play
and record audio files on most
platforms.
The best way to do this is to apply the 30-second truncation, fade in and fade out to the WAV audio data before converting it to an MP3. If your conversion library has a method that takes an array of samples, this is very easy to do. If the method only accepts a WAV file (either in-memory or on disk), then this is slightly less easy as you have to learn the WAV file format (which is easy to write but somewhat more difficult to read). Either way, applying gain and/or attenuation to time-domain sample data (as in a WAV file) is much easier than trying to apply these effects to frequency-domain data (as in an MP3 file).
Of course, if your conversion library already does all this, it's best to just use that and not worry about it yourself.