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!
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 am new to MATLAB, I have few videos of ~100GB each (10hrs duration). I want to calculate some features on every nth frame of the video. I have followed this code But my computer sleeps for just finding number of frames in one video. i.e.
nFrames = get(vidObj, 'NumberOfFrames');
Is there any faster way to process huge videos like I have as this seems to be very slow. Also I have tried looking in internet but didnt find any reasonable solution.
Thanks in advance.
Matlab is dogslow,so you can use ffmpeg instead. Your specific functionality being kept aside, you can extract individual images from a video using below command:
ffmpeg -i input.mp4 -vf fps=60 out%d.png
Ffmpeg video processing is way faster compared to Matlab video processing.
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'm decoding a H264 video bit stream. I have PPS and SPS at first, after that is I slice, P,B,SI,SP slice. I used ffmpeg to convert it into mp4 format, but the video is not correct (it's playable but can't be seen clearly).
I think that my I-slice has been fragmented. Have you got any idea to merge them? In slices there are only frames-begin-with-0181 having frame_num from 1 to x. The other frames have non-sequency frame_num. What does that mean?
thanks for reading.
I have solved my problem. With the fragmented slices, I cut the header slices (example: 00000101) and only left the header of start slice (example: 00000181) and now I can convert to mp4 and watch the video :D
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.