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Extracting audio losslessly with ffmpeg or mkvextract

August 21, 2019, 12:14 pm
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I'm a bit confused about audio extraction with tools like ffmpeg or mkvextract. I did some research but couldn't find much info about this specific topic.

Basically what I am trying to do is extract opus audio from webm container (losslessly of course, without re-encoding). I've tried multiple methods using the tools mentioned above and performed spectrum analysis on all files afterwards.
It seems like the output files have some data lost/changed during the process, which I don't think should be the case. I'm not sure if the results are 100% accurate, though multiple programs confirm it.

How can I be sure that the extraction is successfull and the data matches exactly the original?

Here is some info and images for side-by-side comparison:

Source file: audio.webm
Size: 2.94 MB
Spek:
Image
[Attachment 49874 - Click to enlarge]
Audacity:
Image
[Attachment 49875 - Click to enlarge]


Extracted file using ffmpeg: audio_extracted_ffmpeg.opus
Size: 2.90 MB
Spek:
Image
[Attachment 49876 - Click to enlarge]
Audacity:
Image
[Attachment 49877 - Click to enlarge]


Extracted file using mkvextract: audio_extracted_mkvextract.opus
Size: 2.91 MB
Spek:
Image
[Attachment 49879 - Click to enlarge]
Audacity:
Image
[Attachment 49880 - Click to enlarge]


Comparison between the source webm and the ffmpeg opus in Audacity:
Image
[Attachment 49881 - Click to enlarge]
Image
[Attachment 49882 - Click to enlarge]


This is the output of the ffmpeg extraction:
Code:
ffmpeg version 4.2 Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 9.1.1 (GCC) 20190807
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
  libavutil      56. 31.100 / 56. 31.100
  libavcodec    58. 54.100 / 58. 54.100
  libavformat    58. 29.100 / 58. 29.100
  libavdevice    58.  8.100 / 58.  8.100
  libavfilter    7. 57.100 /  7. 57.100
  libswscale      5.  5.100 /  5.  5.100
  libswresample  3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
Input #0, matroska,webm, from 'files/audio.webm':
  Metadata:
    encoder        : google/video-file
  Duration: 00:03:08.30, start: -0.007000, bitrate: 131 kb/s
    Stream #0:0(eng): Audio: opus, 48000 Hz, stereo, fltp (default)
Output #0, opus, to 'files/ffmpeg/audio_extracted_ffmpeg.opus':
  Metadata:
    encoder        : Lavf58.29.100
    Stream #0:0(eng): Audio: opus, 48000 Hz, stereo, fltp (default)
    Metadata:
      encoder        : Lavf58.29.100
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
size=    2975kB time=00:03:08.28 bitrate= 129.5kbits/s speed=6.57e+03x
video:0kB audio:2952kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.803061%
And the mkvextract:
Code:
Extracting track 0 with the CodecID 'A_OPUS' to the file 'files/mkvextract/audio_extracted_mkvextract.opus'. Container format: Ogg (Opus in Ogg)
Progress: 100%
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Derive target bitrate from target quality by analyzing source video

August 21, 2019, 6:49 pm
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Hi,

I'm using AviDemux and it's Nvenc H.264 encoder. However, there's no target quality mode offered like it is with the x264 encoder (crf mode). Instead, I must specify a bitrate. The problem is that I do not know the bitrate because it mainly depends on how much motion is taking place in the source video. So I'm looking for a way to have the source video analyzed, while passing a target quality (like crf) to this process. In the best case, the process outputs the target bitrate that is required to achieve the desired quality. Can this analyzation run be done with ffmpeg or another program?

Thanks
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Talk to StaxRip that's a static video picture

August 22, 2019, 12:18 am
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Hi everyone :)

There's a way to do similar encode with staxRip?

I tried with x264 and Nvenc but no one allows fps lower than 19 :(

Maybe I can do that with command line?

Code:
General
Complete name              : low_fps.mp4
Format                      : MPEG-4
Format profile              : Base Media / Version 2
Codec ID                    : mp42 (mp42/isom/avc1)
File size                  : 6.75 MiB
Duration                    : 6 min 38 s
Overall bit rate mode      : Variable
Overall bit rate            : 142 kb/s

Video
ID                          : 1
Format                      : AVC
Format/Info                : Advanced Video Codec
Format profile              : High@L5.1
Format settings            : CABAC / 16 Ref Frames
Format settings, CABAC      : Yes
Format settings, ReFrames  : 16 frames
Codec ID                    : avc1
Codec ID/Info              : Advanced Video Coding
Duration                    : 6 min 38 s
Bit rate mode              : Variable
Bit rate                    : 12.6 kb/s
Nominal bit rate            : 10 000 b/s
Maximum bit rate            : 13.0 kb/s
Width                      : 1 920 pixels
Height                      : 1 080 pixels
Display aspect ratio        : 16:9
Frame rate mode            : Variable
Frame rate                  : 1.003 FPS
Minimum frame rate          : 1.000 FPS
Maximum frame rate          : 16 384.000 FPS
Original frame rate        : 1.000 FPS
Standard                    : NTSC
Color space                : YUV
Chroma subsampling          : 4:2:0
Bit depth                  : 8 bits
Scan type                  : Progressive
Bits/(Pixel*Frame)          : 0.006
Stream size                : 612 KiB (9%)
Writing library            : x264 core 114
Encoding settings          : cabac=1 / ref=16 / deblock=1:0:0 / analyse=0x3:0x113 / me=hex / subme=7 / psy=1 / psy_rd=1.00:0.00 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=1 / 8x8dct=1 / cqm=2 / deadzone=21,11 / fast_pskip=1 / chroma_qp_offset=-4 / threads=12 / sliced_threads=0 / slices=1 / nr=0 / decimate=1 / interlaced=0 / constrained_intra=0 / bframes=0 / weightp=2 / keyint=500 / keyint_min=1 / scenecut=40 / intra_refresh=0 / rc_lookahead=40 / rc=2pass / mbtree=1 / bitrate=10 / ratetol=1.0 / qcomp=0.60 / qpmin=5 / qpmax=69 / qpstep=4 / cplxblur=20.0 / qblur=0.5 / vbv_maxrate=13 / vbv_bufsize=300000 / nal_hrd=vbr / ip_ratio=1.40 / aq=1:1.00
Color range                : Limited
Codec configuration box    : avcC

Audio
ID                          : 2
Format                      : AAC LC
Format/Info                : Advanced Audio Codec Low Complexity
Codec ID                    : mp4a-40-2
Duration                    : 6 min 38 s
Bit rate mode              : Variable
Bit rate                    : 128 kb/s
Maximum bit rate            : 529 kb/s
Channel(s)                  : 2 channels
Channel layout              : L R
Sampling rate              : 44.1 kHz
Frame rate                  : 43.066 FPS (1024 SPF)
Compression mode            : Lossy
Stream size                : 6.08 MiB (90%)
I need low fps and bit rate, since the picture is static full quality is expected...

Code:
Bit rate                    : 12.6 kb/s
Nominal bit rate            : 10 000 b/s
Maximum bit rate            : 13.0 kb/s
Frame rate mode            : Variable
Frame rate                  : 1.003 FPS
Minimum frame rate          : 1.000 FPS
Maximum frame rate          : 16 384.000 FPS
Original frame rate        : 1.000 FPS
On attachments has a sample, thanks in advance!
Attached Files
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VLC streaming delay

August 22, 2019, 7:05 am
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Hello friends,

Sorry if i post this in the wrong section.

Anyway, right now im streaming some news channel on VLC at the moment. But sometimes i can see some delay between the vlc stream and web stream. Is there way to make the vlc always update like on the web stream with almost no/little delay?

Thanks :)
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Any way to show milliseconds in Youtube?

August 22, 2019, 8:09 am
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I like to do my fancy frame-accuracy gifs from Youtube pipelining youtube-dl into ffmpeg + gifski.

Youtube player even allows to frame-advance with q/e, but i want to know the time with milliseconds in order to create the ffmpeg script.

Until now i was using MPC-HC + youtube-dl but the web player is way more simple and faster. I guess i could use mpv+youtube-dl but its the same.

Any way to show the milliseconds in the HTML5 player?

Thank you.
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Poor man's Audio Sync Editing - looking for a good AVI splitter

August 22, 2019, 9:23 am
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Hello,

I've been researching all the how to guides for audio sync fixing that I could find. I found a guide using Media Player Classic and adjusting the audio delay. I have a AVI that I ripped and has several points with different syncing errors. I figure I can find the first out of sync instance and determine the delay time in MPC , then move on to the next and so on and note them down. Then I would split the original AVI using those adjustment time points as the start/end points for the split --> then take each AVI segment and adjust the audio delay in MPC and lastly join them all together again. I have Boilsoft's AVI-MPG-ASF-WMV splitter and joiner software. I type in the exact time stamps to rejoin all the segments, but when I view the complete AVI, the movie skips, I guess is the best way to describe it, when it passes the joined timestamps.

Kind of like even though I enter the exact time to start the split, the program keeps a mili-mili second of video before the time that I typed in resulting in that skip on the final. It seems to be doing this. I want it to start the split 01:12:03:004, but it actually splits at: 01:12:03:002 and that difference causes a slight jump in the final compiled edit. Is there a program(s) that is super precise in actually splitting and joining at the exact time I give it. I think that this method works for me pretty well if I can get the skips out.


thanks
iosman
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get_iplayer Users?

August 22, 2019, 10:19 am
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Need help with Anti-Aliasing in Potplayer

August 22, 2019, 10:59 am
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Hi,

I updated Potplayer from an older version I was using 1.6.xxx to the latest 1.7.xxx and the first thing that I noticed is that I'm having "staircase effect" artifact. I tried lot's of different options but it only goes away if I disable the VLD bitstream decoding, whatever that means. But I can't have it disabled because FHD HEVC videos are using too much CPU otherwise, so I need to find a workaround this.

Can anyone help out fellow moviephile ? Don't hesitate if you need any other information.

Thank you for reading. Kind regards.
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How Do I Stop Voices From Repeating On My Projects During Playbacks

August 22, 2019, 3:42 pm
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Hello All, I'm just a noob to VSDC (free version) and after making a project with tw0 MP4 files when I play it back in VSDC when the actors in the video speak everything that they say is repeated a few times. This is very annoying. How do I fix that?
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Seek bar thumbnail previews in movie streams

August 22, 2019, 4:48 pm
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Hey guys,

Can someone suggest a Windows software media player that supports showing seek bar thumbnail preview at video stream playback like YouTube?

What are the advantages and drawbacks from software complexity and network & server workload standpoint, when showing previews? How the preview algorithm is usually implemented in Windows players?

Any way to optimize network and server workload at using preview feature? Any implications on WiFi channel saturation at home, at what channel speed and movie resolutions?
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Hot to move lines to next lines time?

August 22, 2019, 11:28 pm
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I would like to ask how to move 2nd line to 1st lines time, then 3rd line to 2nds line time and do this for all subtitles? (1st line would go to new manually created 1st line) You can see image bellow. Would be nice to do this in Subtitle Edit. Thank you


Image
[Attachment 49895 - Click to enlarge]
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Is the placebo preset necessary for good quality?

August 23, 2019, 6:13 am
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This has nothing to do with a specific software or a any type of video. I'm just talking about the placebo preset. I normally use Veryslow as my preset, but if placebo really does make a crazy difference I might start using it, but is it worth the time? Does it make a big difference in picture quality in videos?
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avisynth: how to load and expose 2 audio tracks instead of only one?

August 23, 2019, 6:17 am
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Hi >&& respect for cats thanks

please consider this source that have only 1 audio track (with 2 stereo channels) correctly loaded (with video) in avisynth:

https://www.dropbox.com/s/uzuqgmlykcor2o0/input1_2.mov?dl=0

Image
[Attachment 49899 - Click to enlarge]
Code:
vid=LWLibavVideoSource("C:\Users\Administrator\Desktop\Nuova cartella (7)\input1_2.mov")
aud=LWLibavAudioSource("C:\Users\Administrator\Desktop\Nuova cartella (7)\input1_2.mov")
left=GetChannel(aud, 1)
right=GetChannel(aud, 2)
both=mergechannels(left, right)
audiodub(vid, both)
ConvertAudioTo16Bit()
and all is ok

------------------------------------------------------------------------------------------------------------------------------------------

But now I have another source: is the same but with another audio track (in total are 2 audio tracks, and each one have 2 audio channels) :o

https://www.dropbox.com/s/h2x204v9csf7io2/input1_2_3_4.mov?dl=0

Image
[Attachment 49900 - Click to enlarge]

How to load the second audio track in avisynth?!?

How can I do the script so avisynth expose the 2 audio tracks?

thanks
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Chrome won't download larger files

August 23, 2019, 8:34 am
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For me, Chrome won't download files large than 2 GB. What can be done to solve this problem?
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playback stuttering in vdub

August 23, 2019, 9:28 am
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I'm using vdub just to capture and watch the VHS on my PC, but the picture stutters and freezes for a seconds all time. I've also used powerDirector were the picture seems to stream fine, the problem there is the picture is too small, so I've switched to vdub.

Is there anything I can do in the settings to get the picture to run more smoothly? Like I said it runs fine in another program so I don't think it's my PC is too weak. Intel duo 2.3 3bg RAM, Intel Q35 express chipset (it is an old PC but for video it should be fine)

Don't care about capture quality (for now), I just wanna watch the tapes.

Also there this weird, like tiny little lines on the edges of everything, I think it's something to do with 'interlacing' but not sure how to fix that, like were in all the pull down menus . Any ideas?

Thanks for any feedback.
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swscaler warning: data is not aligned

August 23, 2019, 10:30 am
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Hello folks. Just a quick question here. When combining downscaling and cropping, the following warning may appear "[swscaler @ 000002b6231d9640] Warning: data is not aligned! This can lead to a speed loss".

I understand why this happens, but I am unfamiliar with its exact effect. Is it a speed loss on encode only? Can it have some other side effect?
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Cropping an anamorphic DVD: what aspect ratio settings now?

August 23, 2019, 12:28 pm
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I am converting a movie from an old PAL DVD to MP4. The original MPEG-2 content in the VOBs is reported by FFmpeg as 720x576 [SAR 64:45 DAR 16:9].

I'm using an AVISynth script to crop the anamorphic content (sans letterbox) to its actual stored size of 704x414*, to clean up the picture a bit, and to interpret the frame rate as 24 instead of 25. I am then using FFmpeg to read the raw video output from AVISynth and do an x264 encode into an MP4 or MKV container. (I'll deal with the audio later.)

What FFmpeg aspect ratio settings (-aspect and/or -vf setsar,setdar) should I use to signal that the resulting MP4 should be stretched horizontally by the appropriate amount upon display? Do I need to take into account that the movie was shot at 2.35:1?

I thought perhaps setsar=64/45,setdar=235/100 might be right, and it doesn't look bad in VLC, but with those settings, FFmpeg reports the output file has [SAR 9729:7040 DAR 47:20], so I am misunderstanding something fundamental, I think, and I'm not sure if it really is the perfect aspect ratio after all.

When researching this issue before posting, I saw other examples saying that people tend to go for maximum compatibility instead by just doing the stretching with the scale filter, so that the resulting file can have 1:1 pixels, but I'd rather keep the file size down by keeping it anamorphic until final display. Or is that foolish?

Here's my FFmpeg command line:

Code:
ffmpeg -i template.avs -vf setsar=64/45,setdar=235/100 -c:v libx264 -profile:v high -level:v 3.2 -preset ultraslow -crf 20 -y out.mkv
Here's my AVISynth script:

Code:
LoadPlugin("E:\apps\DGMPGDec\DGDecode.dll")
MPEG2Source("G:\blahblahblah\VIDEO_TS\pw.d2v",iPP=false,cpu2="xxxxxx",idct=3)
Trim(39560,end=233824)
Crop(10,80,-6,-82)
# various picture cleanup settings omitted for brevity
AssumeFPS(24)
* The movie's picture is more like 700x422 in the anamorphic frame; I'm cropping it to 704x414 to ditch the letterboxing and fuzzy top/bottom edges; I'm not worried about the pillarboxing as it's only a couple of pixels on each side.
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tsMuxeR 2.6.15 for Windows (command-line only)

August 23, 2019, 2:02 pm
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As everybody already knows :rolleyes: , the (messy) source-code of tsMuxer has been released @ https://github.com/justdan96/tsMuxer .
However no "official" binaries have been released yet. The attachment below (source-code included, should compile under Linux too) contains both the 32-bit and the 64-bit .EXEs, but no (up-to-date) GUI. These binaries were stripped (--strip-unneeded) and compressed with UPX. According to a doom9 forum member, the old v2.6.12 GUI works normally with the current 64-bit executable.
Attached Files
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Importing presets to Vidcoder 4.36

August 23, 2019, 6:56 pm
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My conversion program of choice is usually 2.62 x64.

But today I was having some problems, apparently from trying to convert an MKV file with an Atmos 7.1 audio track. It froze three times when I tried to generate a preview.

Is version 2.62 incompatible or can't handle Atmos 7.1 conversion to AC3 5.1?

Well, I downloaded a newer Vidcoder, version 4.36, to see if it could convert Atmos 7.1.

After I opened it, I tried to import my usual High-Profile preset, but it gave an error.

Can a 2.62 preset be incompatible with version 4.36?
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Quality loss with PNG exporting at different resolutions?

August 23, 2019, 9:26 pm
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Hello. I'm new to these forums and grateful for your help.

I'm making a video comprised of many different clips and they're all at different resolutions and bitrates. Including 240p, 360p, 480p, 720p,1080p and 1440p.

My question is if I export my project as PNG lossless in 1440p, will the quality for all of the clips remain the same? I don't understand how lossless works when the software would maybe be changing the resolution of some of the clips too.

The answer to this will determine whether it's a good idea for me to sometimes export some chunks of the project as PNG then put them back into the project to keep working on them. Or if this will introduce some reduction in quality then I think I mustn't do this but convert chunks to sprites instead and copy and paste that? (Exporting all as PNG helps the preview be smoother and runs faster than the sprites filled with many different short clips)

Thanks for any advice.
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