User:Mjb/FFmpeg

Extract a portion of a video

Let's say you just want to take the portion from 37:07.5 to 41:30.

 ffmpeg -i inputfile -vcodec copy -acodec copy -ss 37:07.5 -to 41:30 outputfile

Rotate 180 degrees

When you hold a camera phone the wrong way, it will just put a 180° rotation flag in the metadata, which not all players will support. To rotate the actual video, chain the hflip and vflip filters:

ffmpeg -i inputfile -vf "vflip,hflip" outputfile

The rotation flag will not be changed when you do this, so you can set it afterward (assumes video is stream # 0):

ffmpeg -i inputfile -c copy -metadata:s:v:0 rotate=0 outputfile

Or you can do both at the same time (untested):

ffmpeg -i inputfile -vf "vflip,hflip" -metadata:s:v:0 rotate=0 outputfile

Here's a more robust example (worked for me):

 ffmpeg -i input.mp4 -metadata:s:v rotate="0" -vf "hflip,vflip" -c:v libx264 -acodec copy output.mp4

The c:v libx264 means to output H.264 video, which is what the input will be if it is from an iPhone. For more H.264 ffmpeg tips, see https://trac.ffmpeg.org/wiki/Encode/H.264

References:

• https://stackoverflow.com/questions/3937387/rotating-videos-with-ffmpeg
• https://stackoverflow.com/questions/15335073/can-i-set-rotation-field-for-a-video-stream-with-ffmpeg
• http://thornelabs.net/2013/06/04/correct-smartphone-video-orientation-and-how-to-rotate-ios-and-android-videos-with-ffmpeg.html

Transcode to a specific frame size & bitrate

My iPhone records video at 1920x1080. The audio doesn't take up much space at all, but the video is H.264 at about 17 Mbps, so it uses up 125 MB per minute. Here is a way to get it down to a more manageable size and more portable container format, along with the 180° rotation ("vflip,hflip") mentioned above:

ffmpeg.exe -i inputfile.mov -acodec copy -b:v 2000k -vf "vflip,hflip,scale=1024:-1" outputfile.mkv

This makes it be about 15 MB/minute: 2 Mbps, 1024 width, -1 means whatever height will preserve the aspect ratio.

Reference:

• https://trac.ffmpeg.org/wiki/FilteringGuide#Scaling

Convert DV AVI to H.264 MP4

It's 2016 and I'm now digitizing the analog video signal from a VCR by running it through a camcorder which outputs DV format over a FireWire cable. I'm using Sony's PlayMemories Home software to capture the camera's data stream (720x480, 29.97 Hz, interlaced DVCPRO—a.k.a. DVCPRO25, dvsd, dvvideo, or consumer DV—and put it into AVI containers.

The camera has a choice of audio during the transfer: 12-bit or 16-bit. If you choose 12-bit, the output is actually 16-bit 32 kHz PCM (but I assume the bottom 4 bits of each sample are all zeroes). If you choose 16-bit, it's true 16-bit 48 kHz PCM. The 12-bit mode is fine for typical home-movie audio, e.g. speech and background noise, but for tapes with Hi-Fi music on them, I use 16-bit mode.

Anyway, there are problems with the resulting DV-AVI files:

• Huge files: about 13 GB per hour.
• DV's 4:1:1 YUV subsampling results in desaturated, fuzzy color (color is sampled at ¼ horizontal resolution, i.e. each set of 4 pixels gets their average color).
• Interlaced output looks bad when viewed on computers.

The file size can be reduced by transcoding to more efficient format like H.264 (MPEG-4 AVC) for the video and AAC-LC for the audio. I can't undo the damage caused by the chroma subsampling, but I can make it look less washed-out by applying a saturation filter (hue=s=#). The annoying "comb" effect from interlacing can be mitigated, at a cost, by using a deinterlace filter (yadif). So here is what I do:

 ffmpeg -i inputfile.avi -vf "yadif,hue=s=1.4" -c:v libx264 -preset veryslow -crf 20 -tune grain -c:a aac -b:a 160k outputfile.mp4

With these settings, the output MP4 is about 17% the size of the input AVI, so about 2.2 GB per hour. The video data rate is about 5 Mbps. I think it looks pretty good.

-crf 20 sets quality level 20 (23 is default, lower is better). aac is the native AAC encoder, which is better than libfaac but not as good as libfdk_aac (which isn't in my build of FFmpeg).

If you have ideas on better settings to use, please let me know!

The x264 codec uses 4:2:2 subsampling (color at ½ horizontal resolution) by default, but of course the result can never be better than the 4:1:1 input. Another option, for greater compatibility with playback devices/apps, is to use -pix_fmt yuv420p to have the codec use 4:2:0 (color at ½ horizontal and ½ vertical resolution), but obviously this will be worse than 4:2:2.

Avidemux

Rather than using FFmpeg directly, I am also experimenting with using Avidemux, which is a free video editor like VirtualDub. It can utilize FFmpeg libs, among others, if using it to convert output. It also can do lossless editing and splitting. I may use it to split some huge DV AVIs into DVD-R sized pieces.

In order to improve the look of DV captures of LP-mode VHS recordings of analog cable broadcasts, I am trying the following filter chain:

• ChromaShift (U: -5, V: -4) to get the color fields in sync; this may vary by tape and source.
• dgbob (mode 1, order 0, threshold 0) for bob deinterlacing (doubles the framerate, but motion is smooth).
• Mplayer hue (hue -4 to -15, sat 1.0) to make blues blue instead of purple, etc.
• MPlayer eq2 (cont 1.04, brigh 0.02, sat 1.37) to boost brightness/contrast/gamma/saturation and make neutral colors neutral; figuring out ideal settings is difficult!
• blacken borders (left: 22, right: 22) to simplify the left and right edges, and to crop out the pixels made colorless by ChromaShift.

x264 encoder settings (General):

• Preset: veryslow
• Tuning: grain
• Profile: baseline
• Fast First Pass [off]
• Encoding Mode: Video Size (Two Pass)
• Target Video Size: depends on destination. To fill up a single-layer DVD-R, I think 4300 MB should be OK. It seems the doubled framerate from dgbob throws the estimate off by roughly half, so I have to double the target size I enter here.

x264 encoder settings (Output 1):

• Predefined aspect ratio: 8:9 (NTSC 4:3) - This makes it the same as the DV input (DV's 720x480, which is naturally 3:2, but the wiki says not to change this, but if I don't, the output is 3:2 and thus plays back horizontally elongated.

Avidemux to filter, FFmpeg to encode

It seems to be impossible to stop Avidemux from creating 4:2:0 output (changing the chroma channels to half their vertical resolution), so in order to get 4:2:2, I am resorting to a workaround where I just use Avidemux for filtering, and then do the FFmpeg encoding from the command line. What a mess!

Example workflow:

• Prep the audio
  • ffmpeg -i "input.avi" -vn -c:a copy "tmp.wav"
  • Process tmp.wav in an audio editor to adjust channels, EQ, resample, normalize to -3 dB peaks, reduce noise, etc.
• Prep the filtered video
  • In Avidemux, load input.avi and set up the video filter chain as desired
    • If you save to a script after setting up the rest of the outputs, you can then copy-paste your favorite adm.addVideoFilter lines after the adm.videoCodec line.
  • Set the Video Output to (FF)HuffYUV - this is a lossless format using about 1 GB per minute!
  • Set Audio Output to Copy
  • In Audio > Select Track, disable Track 1. Don't set it to use tmp.wav because it will add a glitch at the end.
  • Set the Output Format to AVI Muxer
  • Save to tmp.avi - this will be 3:2 (default for 720x480 with 1:1 pixels) but we'll fix it during compression
• Calculate the target bitrate
  • This calculator (one of many online) can help. My Blu-Ray player can only handle ~17 Mbps video and I find 15 Mbps (15000 kbps) is usually plenty.
• Compress the video and audio into one H.264 MP4
• ffmpeg -y -i tmp.avi -i tmp.wav -map 0:0 -map 1:0 -c:v libx264 -preset veryslow -tune grain -pass 1 -b:v 15000k -aspect 4:3 -c:a aac -b:a 128k -shortest -f mp4 -y NUL
• ffmpeg -y -i tmp.avi -i tmp.wav -map 0:0 -map 1:0 -c:v libx264 -preset veryslow -tune grain -pass 1 -b:v 15000k -aspect 4:3 -c:a aac -b:a 128k -shortest -y output.mp4

Example filters:

adm.addVideoFilter("chromashift", "u=-5", "v=-4")
adm.addVideoFilter("dgbob", "thresh=0", "order=False", "mode=1", "ap=False")
adm.addVideoFilter("hue", "hue=-4.000000", "saturation=1.000000")
adm.addVideoFilter("eq2", "contrast=1.040000", "brightness=0.020000", "saturation=1.370000", "gamma=1.270000", "gamma_weight=1.620000", "rgamma=0.990000", "bgamma=1.080000", "ggamma=1.010000")
adm.addVideoFilter("blackenBorder", "left=22", "right=22", "top=0", "bottom=0")

Replace audio in a DVD file

Music videos in MPEG-2 format (.vob or .mpg files) sometimes come with bad source audio and I will want to replace the audio with a good copy of my own. Of course, I have to pay close attention and make sure that the audio in the video is the same; videos often use custom edits or they overdub other sounds on top. Assuming I have suitable audio in a lossless format like FLAC, here's what I do to replace it:

First, extract the source audio to a WAV file:

• ffmpeg -i input.vob output.wav

Note: if the original audio was lossy, the resulting WAV will probably be bigger because it includes encoder delay & padding, possibly also decoder delay (i.e. a bunch of silence at the beginning, and a little bit at the end). It's best if you can figure out how much there is and trim it. However, I don't know a good way to do that!

Next, use a wave editor (I use Audition) to create a new WAV that is perfectly time-aligned with the old. There are different ways of doing this. Here's one way:

• Convert the replacement to the desired output sample rate.
• Pick a non-silent spot at the beginning and end of the original file to be the anchor points. You are looking for spots that you can find in both the old and new files. Set a marker at each spot. (In the original, markers at samples 28936 and 8591175; in the replacement, at samples 4568 and 8560470).
• How many samples are in between the markers? (8591175-28936=8562239 and 8560470-4568=8555902) Your goal is to change the replacement to match the original.
• What's the original:replacement ratio? (8562239/8555902=1.0007406583198358279466034089685)
• Do a pitch shift on the replacement with a target duration of that number multiplied by the current duration, in samples. (8659287*1.0007406583198358279466034089685=8665700)
• Check how many samples are in between the markers. (8566809-4571=8562238) It should be really close to the original now. If not, figure out what you did wrong and try again.
• Pad or trim silence from the beginning so that the first marker is at the same location as the first marker in the original. (28936-4571=24365 padding to add). If the beginning is offset or not silent, apply a fade-in beforehand.
• Fade and/or pad the end, so that the total duration is the same as the original.

Now you need to mux them together. What is supposed to work is this:

• ffmpeg -i input.vob -i new.wav -map 0:1 -map 1:0 -vcodec copy -acodec copy out.vob

-map 0:1 means use stream #1 from file #0 (input.vob), and -map 1:0 means use stream #0 from file #1 (new.wav).

Unfortunately, FFmpeg currently doesn't like to mux PCM audio into a VOB or MPEG-2 container without giving all kinds of packet too large / buffer underflow errors. Supposedly this was fixed, but it's not working for me, so...

The solution is to use MPEG Video Wizard DVD. In that app: Drag the video into the timeline's video bar, right-click on it and choose to mute its audio (otherwise it will mix them together). Drag the audio into the timeline's music bar. Click on Export (looks like a videotape) in the main toolbar, and make sure it's going to do a stream copy.

A more advanced example:

• downloaded clip from YouTube as .mp4 (AVC video + AAC audio)
• demuxed and converted AAC to WAV:
  • ffmpeg -i input.mp4 output.wav
• noted that output.wav had 18345984 samples (6:56.009)
• sized replacement audio to 18340836 samples (6:55.892) (a rough guess as to ideal size)
• used fhgaacenc via foobar2000 to encode replacement audio as .m4a
• muxed original video with replacement audio, bitrate ~2 Mbps, saturation 1.7x:
  • ffmpeg -i input.mp4 -i new_audio.m4a -map 0:0 -map 1:0 -vcodec libx264 -acodec copy -b:v 2000000 -vf "hue=s=1.7" out.mp4

Resulting video seems synced with its audio, but just to see how bad my guess was:

• demuxed and converted AAC to WAV:
  • ffmpeg -i out.mp4 out.wav
• noted that out.wav had 18342912 samples (6:55.939) ... 2076 samples more than I input, but 3072 less than needed!

Oh well.