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.
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 have a raw video (no header, just Y-channel).
I want do some denoise algorithm on this video.
I convert this raw video by useing ffmpeg with several encoders.
But fail to open with function AVIREAD.
Error using aviread, Unable to locate decompressor to decompress video stream
r210 Uncompressed RGB 10-bit
v210 Uncompressed 4:2:2 10-bit
v308 Uncompressed packed 4:4:4
v410 Uncompressed 4:4:4 10-bit
y41p Uncompressed YUV 4:1:1 12-bit
yuv4 Uncompressed packed 4:2:0
mjpeg MJPEG (Motion JPEG)
ffmpeg -f rawvideo -vcodec rawvideo -s 1920x1080 -r 25 -pix_fmt gray -i WKA00002.y -c:v v308 WKA00002_UnCompAVI.avi
What kind of encoder is suitable for AVIREAD in Matlab?
Thanks a lot
AVIREAD has been removed from the latest versions of MATLAB. Use VIDEOREADER instead.
Motion JPEG AVI is supported by VideoReader.
if you want to try other formats, first check that the file can be opened using Windows Media Player? If so, use VideoReader to try and read the file. If this does not work, can you provide a link to the file that you are using?
Hope this helps.
Dinesh
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
I have an wav sound file and I have avi video file in MATLAB,
I want to merge those files.
I try to do this with audioread but the video was created with videowrite so i don't really know how to merge the files.
Thanks for any help.
Doing this in matlab, if possible would be at such a high level of abstraction you might just as well do it outside of matlab, or have matlab do a system call to execute :
avconv -i original_video.avi -i some_audio.wav -c copy output_video.avi
where avconv is a standalone cross-platform media utility
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.