I wrote an streaming server with libVLC. Everything is working perfekt.
I use this parameters for the streaming
#transcode{vcodec=h264}: rtp{mux=ts,dst=239.255.255.239,sdp=sap,name=test"
But I also want to reduze the filesize of the streamed audio file, let's say to 1/2 of the original file size.
Is there any way I can solve this with an transcode parameter or with another way ?
One of the VLC examples includes these example transcode strings:
MP4 High:
#transcode{vcodec=h264,venc=x264{cfr=16},scale=1,acodec=mp4a,ab=160,channels=2,samplerate=44100}
MP4 Low:
#transcode{vcodec=h264,venc=x264{cfr=40},scale=1,acodec=mp4a,ab=96,channels=2,samplerate=44100}
OGG High:
#transcode{vcodec=theo,venc=theora{quality=9},scale=1,acodec=vorb,ab=160,channels=2,samplerate=44100}
OGG Low:
#transcode{vcodec=theo,venc=theora{quality=4},scale=1,acodec=vorb,ab=96,channels=2,samplerate=44100}
WEBM High
#transcode{vcodec=VP80,vb=2000,scale=1,acodec=vorb,ab=160,channels=2,samplerate=44100}
WEBM Low
#transcode{vcodec=VP80,vb=1000,scale=1,acodec=vorb,ab=96,channels=2,samplerate=44100}
Changing the various parameters, like reducing bit-rates, will reduce the size of your media.
Related
I am manually generated a .mov video file.
Here is a link to an example file: link, I wrote a few image frames, and then after a long break wrote approximately 15 image frames just to emphasise my point for debuting purposes. When I extract images from the video ffmpeg returns around 400 frames instead of the 15-20 I expected. Is this because the API i am using is inserting these image files automatically? Is it a part of the .mov file format that requires this? Or is it due to the way the library is extracting the image frames from the video? I have tried searching the internet but could not arrive at an answer.
My use case is that I am trying to write the current "sensor data" (from core motion) from core motion while writing a video. For each frame I receive from the camera, I use "AppendPixelBuffer" to write the frame to the video and then
Thanks for any help. The end result is I want a 1:1 ratio of Frames in the video to rows in the CSV file. I have confirmed I am writing the CSV file correctly using various counters etc. So my issue is cleariy the understanding of the movie format or API.
Thanks for any help.
UPDATED
It looks like your ffmpeg extractor is wrong. To extract only the timestamped frames (and not frames sampled at 24Hz) in your file, try this:
ffmpeg -i video.mov -r 1/1 image-%03d.jpeg
This gives me the 20 frames expected.
OLD ANSWER
ffprobe reports that your video has a frame rate of 2.19 frames/s and a duration of 17s, which gives 2.19 * 17 = 37 frames, which is closer to your expected 15-20 than ffmpeg's 400.
So maybe the ffmpeg extractor is at fault?
Hard to say if you don't show how you encode and decode the file.
I have a bare h.264 file (from a raspberry pi camera), and I'd like to wrap it as an mp4. I don't need to play it, edit it, add or remove anything, or access the pixels.
Lots of people have asked about compiling ffmpeg for iOS, or streaming live data. But given the lack of easy translation between the ffmpeg command line and its iOS build, it's very difficult for me to figure out how to implement this simple command:
ffmpeg -i input.h264 -vcodec copy out.mp4
I don't specifically care whether this happens via ffmpeg, avconv, or AVFoundation (or something else). It just seems like it should be not-this-hard to do on a device.
It is not hard but requires some work and attention to detail.
Here is my best guess:
read PPS/SPS from your input.h264
extract height & width from SPS
generate avcC header from PPS/SPS
create an AVAssetWriter with file type AVFileTypeQuickTimeMovie
create an AVAssetWriterInput
add the AVAssetWriterInput as AVMediaTypeVideo with your height & width to the AVAssetWriter
read from your input.h264 (likely in Annex B format) one NALs at a time
convert your NALs from your input.h264 from start code prefixed (0 0 1; Annex B) to size prefixed (mp4 format)
drop NALs of type AU, PPS, SPS
create a CMSampleBuffer for each NAL and add a CMFormatDescription with the avcC header
regenerate timestamps starting a zero using the known frame rate (watch out if your frames are reordered)
append your CMSampleBuffer to your AVAssetWriterInput
goto 7 until EOF
I want to extract a few clips from the recorded wav file. I am not finding much help online regarding this issue. I understand we can't split from compressed formats like mp3, but how do we do it with caf/wav files?
One approach you may consider would be to calculate and read the bytes from an audio file and write them to a new file. Because you are dealing with LPCM formats the calculations are relatively simple.
If for example you have a file of 16bit mono LPCM audio sampled at 44.1kHz that is one minute in duration, then you have a total of (60 secs x 44100Hz) 2,646,000 samples. Times 2 bytes per sample gives a total of 5,292,000 bytes. And if you want audio from 10sec to 30sec then you need to read the bytes from 882,000 to 2,646,000 and write them to a separate file.
There is a bit of code involved but it can be done using Audio File Services Class from the AudioToolbox framework.
Functions you'll need to use are AudioFileOpenURL, AudioFileCreateWithURL, AudioFileReadBytes, AudioFileWriteBytes, and AudioFileClose.
An algorithm would be something like this-
You first set up an AudioFileID which is an opaque type that gets passed in to the AudioFileCreateWithURL function. Then open the file you wish to splice up using AudioFileOpenURL.
Calculate the start and end bytes of what you want to copy.
Next, in a loop preferably, read in the bytes and write them to file. AudioFileReadBytes and AudioFileWriteBytes allow you to do this. Whats good is that you can read and write whatever size bytes you decide on each iteration of the loop.
When finished close the new file and original using AudioFileClose.
Then repeat for each file (audio extraction) to be written.
On an additional note you would split a compressed format by converting the compressed format to LPCM first.
I have a series of MP4 files (H.264 video, AAC audio, 16KHz). I need to merge them together programmatically (Objective-C, iOS) but the final file will be too large to hold in memory so I can't use the AVFramework to do this for me.
I have written code which will do the merge and takes care of all of the MP4 atoms (STBL, STSZ, STCO etc.) based on just concatenating the contents of the respective MDATS. The problem I have is that while the resultant file plays, the audio gradually gets out of sync with the video. What seems to be happening is that there is a disparity between the audio and video length in each file which gets worse the more files I concatenate.
I've used MP4Box to generate a file from command line and it is 'similar but different' to my output. A notable different is that the length of the MDAT has changed and the chunk offsets have also changed (though sample sizes remain consistent).
I've recently read that AAC encoding introduces padding at the beginning and end of a stream so wonder if this is something I need to handle.
Q: Given two MDAT atoms containing H264 encoded data and AAC audio, is my basic method sound or do I need to introspect the MDAT data in some way.
Thanks for pointer Niels
So it seems that the approach is perfectly reasonable however each individual MP4 file has marginal differences between the audio length and video length due to differences between the sampling frequency. The MP4s include an EDTS.ELST combination which correct this issue for that file. I was failing to consider the EDTS when I merged files. Merging EDTS has fixed the issue.
Im using BASS.dll library and all I want to do is to "redirect" part of MP3 Im playing using for example BASS_StreamCreateFile to another file (may be MP3 or WAVe). I dont know how to start? Im trying to use help to find an answer, but still nothing. I can play this stream. Read some data I need. Now I need to copy ile for example from 2:00 to 2:10 (or by position).
Any ideas how should I start?
Regards,
J.K.
Well, I don't know BASS specifically, but I know a little about music playing and compressed data formats in general, and copying the data around properly involves an intermediate decoding step. Here's what you'll need to do:
Open the file and find the correct position.
Decode the audio into an in-memory buffer. The size of your buffer should be (LengthInSeconds * SamplesPerSecond * Channels * BytesPerSample) bytes. So if it's 10 seconds of CD quality audio, that's 10 * 44100 * 2 (stereo) * 2 (16-bit audio) = 1764000 bytes.
Take this buffer of decoded data and feed it into an MP3 encoding function, and save the resulting MP3 to a file.
If BASS has functions for decoding to an external buffer and for encoding a buffer to MP3, you're good; all you have to do is figure out which ones to use. If not, you'll have to find another library for MP3 encoding and decoding.
Also, watch out for generational loss. MP3 uses lossy compression, so if you decompress and recompress the data multiple times, it'll hurt the sound quality.