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29th May 2020, 10:53 | #9021 | Link | |
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For instance if you store the movie with Maximum bit rate : 8 921 kb/s in a DVD (typical max read rate 5 Mb/s) you can have troubles in the points with high bitrate. But if you store the movie in a USB 2 (at least 20 Mb/s reading) all is ok. Also related with the player to reproduce your encode, must guarantee the speed to support this max bitrate and the amount of buffer RAM (buffer size) to store the images to be decodified (related with resolution and num of ref-frames). The easy way to control these parameters is use the parameter --level, you must know the level than your player accept. For instance a player certified to DivX Plus (--level 4.0 see Levels in this wiki) accept --vbv-maxrate 20000 --vbv-bufsize 25000. You can see in the x264 Configuration Dialog in MeGUI how change these parameters when you select the Target Playback Device. Of course modern players certified for UHD accept --level 5.1 and the maxrate/bufsize are so high than you don't need to limit your encode with low resolution/bitrate. And these parameters aren't related with the Preset (slow or other) or the Encoding Mode (crf or 2pass) like you can see in the Configuration Dialog, only with the Target Playback Device.
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29th May 2020, 11:01 | #9022 | Link | ||
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Now I know why a animation that I had encoded sometime back didnt play on my TV , I had used tune animation along with crf and the level came out to be 5 and I realized my TV cant play anything with level 5 Thanks |
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30th May 2020, 15:46 | #9024 | Link |
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I sometimes post sober. StainlessS@MediaFire ::: AND/OR ::: StainlessS@SendSpace "Some infinities are bigger than other infinities", but how many of them are infinitely bigger ??? |
1st June 2020, 00:44 | #9026 | Link |
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Ligh, think mine set at 96, that sound about OK ?
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1st June 2020, 03:58 | #9027 | Link |
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96 is a fraction more than stuff-all less than 100, when translated to dB. It's -0.35dB.
An increased peak of 1dB wouldn't be too unusual for lossy encoding, and for some reason AAC seems worse than MP3 in that respect. If you want a margin for error: 0.9 = -0.91dB 0.8 = -1.9dB 0.7 = -3.09dB I don't know how to write the equation properly, but the dB value is the log of the percentage multiplied by 20 (where 100% = 1.0) Have I ever heard the clipping when normalising to 100? Nope. Of course I disproved everything I just said while trying to provide examples. The peaks didn't change all that much, and all but one was lower, but I have seen an increase of close to 2dB on rare occasions. The peaks were the result of 4 times over-sampling to obtain the true peak, rather than the sample peak. If this is any indication though, 96 should generally be fine. PS StainlessS, I bumped into your comment here (thanks, by the way), so I thought I'd mention, if you haven't seen it, the idea inspired me to create some more MeGUI functions. In case you're interested...... https://forum.doom9.org/showthread.php?t=181500 Last edited by hello_hello; 1st June 2020 at 04:56. |
1st June 2020, 15:31 | #9028 | Link |
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I dont like to go too low, especially if viewing on hand held (phone) where volume of device can sometimes be quite low,
but thank you HH.
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4th June 2020, 15:53 | #9029 | Link | |
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after many months I moved to pazera It raises volume upto three times and its pretty efficient in encoding AAC , AC3 , MP3 ans a whole lot of other .. give it a go Converting your 5.1 audio file to 5.1 ac3 9 mimimum would be 256 but 384 sounds very very good and aint big at all ]with volume slider all the way to right will surprise you. ======================================= @ tebasuna51 and Hello_Hello I am still not able to fine a way to display maximum bit-rate in a megui encoded video. Also what are you guyz's thoughts about ABR encoding , 2 pass takes the most time but you know what you are doing , CRF well you cant really predict the size.. but I find CRF method to be pretty faster as its single pass.. DO you guys think its better to use ABR considering its 1 pass ( so less time ) plus you can predict the file size as well ?? What about the quality, will it be similar to 2pass for a given bitrate ?? I am again talking about bitrates in the range of 1500 to 2000 only ... Also I know one should always keep thread count to 0/auto.. but to maximize CPU usage if there are no other bottlenecks, how many threads I can input just to make sure that maximum cpu is used while encoding.. Thank you |
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5th June 2020, 00:15 | #9030 | Link | ||||||
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5th June 2020, 06:37 | #9031 | Link | |||
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My little comparisons were for a 5.1ch file downmixed to stereo, because without normalisation that's when you can notice a fair volume drop, because the volume should be lowered enough when downmixing so when the channels are combined they can't exceed maximum, even if the individual channels all happen to be at maximum at the same time. Pazera simply downmixed and increased the volume with ffmpeg like this: -c:a pcm_s16le -ac 2 -af volume=3.00 Without the volume adjustment, it's just this: -c:a pcm_s16le -ac 2 MeGUI would be slower if you normalise, because it requires two passes. One to check the peak levels, then a second to adjust the volume and encode. I assume ffmpeg uses a slightly different formula for downmixng than MeGUI, and I use a downmix formula using foobar2000 that's slightly different again, hence the average volumes being different in the screenshot below. I assume ffmpeg has a method for normalising the peaks to 0dB, but Pazera doesn't seem to offer the option in the GUI. Pazera 1x - Standard ffmpeg downmix. MeGUI No Normalise - Standard MeGUI downmix without normalisation. Foobar2000 Downmix Prevent Clipping - same as "MeGUI No Normalise" but using a different downmix formula. Pazera 3x - Standard ffmpeg downmix plus Volume=3. MeGUI Peak Normalise 95 Percent - Standard MeGUI downmix with peak normalisation set to 95%. Foobar2000 Downmix - downmixed without reducing the volume to prevent the possibility of clipping. In this case there wasn't any. Pazera 3x Foobar2000 Downmix - I used Pazera to re-encode the "Foobar2000 Downmix" file above while increasing the volume 3x. The average volume increased by around 10dB but the peaks were already just under maximum, so I assume they were clipped pretty hard. If you're cranking the volume while re-encoding 5.1ch audio (without downmixing), I think you're very likely to cause the peaks to be clipped. It'd be no different to re-encoding stereo audio with peaks already at or close to 0dB, while cranking the volume. The Track Gain column is a ReplayGain thing. You can ignore it. The Volume and Peak columns show the average and peak levels according to an EBU R128 scan, which is what I was interested in. Quote:
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x264's rate control methods If memory serves, the default of zero means the thread count is 1.5x the number of CPU cores. Last edited by hello_hello; 5th June 2020 at 07:37. |
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5th June 2020, 07:31 | #9032 | Link |
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Thanks a bunch T
Take care Bud Stay safe EDIT : To make the first pass faster i have set FFMS Thread count to 0 without any crashes , issues or stuff Hope thats okay.. EDIT02 : Totally missed your post @ Hello_Hello many thanks , But i used pazera to convert very high bitrate 5.1 ac3 audio to 2 channel aac audio. and sometimes 5.1 high bitrate ac3 to 5.1 384kbps ac3 EDIT03 : I read your post that you linked @Hello_Hello , TBH I would always want to encode the videos ( whenever my encoding bug starts to bite ) using CRF only.. I had started with a number of my DVDs ( a few years back ) and after converting a few i kept all the raw files in a HDD to experiment later , by the time Blurays got cheaper and once a tried a BR too.though my system needs a update as far as the ardware is concerned. but the crux is I saw huge huge files when there was grain in the source video, crf gave big files and i stayed with 2 pass.. TBH even now I wonder about a simple degrain filter of avisynth coz I just cant get myself to use those complex filters lol Same is with sharpening and denoising I am experimenting with what you told about using the x264 inbuilt noise reducer and I am sure this will prove utterly helpful.. will be trying sharpen() filer only coz the videos that I am working on were professionally made, and then filtered, edited and made into a final copy by an office mate who is a specialist.. I am trying to hammer em a bit more just for the heck of it, you always learn new things by looking at professional material. Last edited by Lord Dredd; 5th June 2020 at 15:23. |
6th June 2020, 15:09 | #9033 | Link | ||
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Yes I always used to use normalize option when I used to downmix 5.1 to stereo 128kbps HE or LC AAC. You may be right about me cranking up the volume levels in pazera , BUt I ll be honest I found the audio much m,uch better on both my LG and Panasonic TVs , But yet to listen to this audio with my soundbar and home theater ( long story lock down has put off a lot of repair works ) But I am game if you could guide me about which options to choose and which boxes to tick while downmixing a 5.1 DD to a 2 channel stereo & also converting a 5.1 AC3 DD or ATmos to 384kbps DD =============== Apart from a few questions asked above I had some more queries about MEGUI just yesterday I found a profile zip from 3 or may be 4 years back and in that qpmin was set to 10 and qpmax was set to 51 But in MEGUI x264 profiles they have been changed to 0 and 81 ( 69) actually whats your opinion : for lower , low and average bitrates x264 encodes what should be the values , are there some other settings as well in SLOWER profile apart from --nr that you guys feel should be changed to have better results. whats your view of ref frames and bframes ( wrt slower profiles ) , some people advocate cranking them up a bit more especially bframes especially for low bitrate videos ??? This is a mediainfo detail of a 200MB video I was surprised how a 200MB video of this much duration have bitrate that high.. This is some a friend of mine sent to me this morning, I have asked him to share the video as well but I dunno... I could make out its been encoded using Medium x265 profile with a crf of 23.5 but its totally beyond my understanding, Either this video has fabke info added to it or may be there is some secret recipe... Quote:
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6th June 2020, 20:10 | #9034 | Link |
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Lord Dredd,
When it comes to being able to hear everything without the loud parts being stupidly loud, I'd recommend trying the Dynamic Audio Normalizer. I use my PC as a media player so I downmix and compress the audio as it's being played (via ffdshow and a WinAmp plugin), but for the TV in another room I generally downmix to stereo and compress with the DAN. A relativity easy way to try it (hopefully) is to download the portable version of foobar2000 I uploaded here. You'll also need ffmpeg. There's a text file with some instructions included. Once you're done you can load an audio file into a playlist, right click, select convert, and a list of my conversion presets will pop up. One is called "Downmix and Compression - Dynamic Audio Normalizer" or something similar. Select that preset and the output will be AAC. It's not overly high bitrate as I generally use it to create a temporary copy of the audio for watching in another room, but it's easy enough to edit/create a preset with higher bitrate AAC. If you want to hear what it does, there's a zip file attached to this post with some samples nobody bothered listening to. I was trying to show how much better it is than the method the OP was using. The idea is to compare the level of the speech at the beginning with the level of the loud stuff at the end. Ideally the loud stuff stays at the same volume while the quiet stuff gets louder. The audio in the zip file with the DAN settings I use is called "f=150 b=1.flac". There's another DAN example included using someone else's settings for comparison, although I prefer mine. Edit: Thinking about it, Pazera lets you add stuff to the ffmpeg command line. Use it to downmix to stereo and add the following to the custom command line section. Leave the volume at 1x. -af dynaudnorm=f=150:b=1 I haven't tested it, but it should work. I assume if you convert to AAC, Pazera uses ffmpeg to encode. The difference between doing it that way and my foobar2000 preset, is my preset uses ffmpeg to compress with the DAN, but it pipes the audio to QAAC for the actual encoding (and I probably downmix to stereo a little differently). It'd possibly also work if you add it to the custom command line section for the formats where MeGUI uses ffmpeg for encoding, but if you do, don't enable peak normalising, or at least try it with and without. I can't offer much of an opinion on qpmin/qpmax as I never play with them. In theory, a higher number of reference and B frames should increase the possibility of compressing more for a given bitrate, but as a rule I just use the slow or slower preset and the number of reference and B frames that preset uses. How effective increasing the number of ref and B frames is, probably also depends on the motion estimation algorithm being used. I'm not much of an x264 tweaker but under x264's standard error stream in the MeGUI log file, there's information on reference and B frame usage. ref P L0: 60.3% 16.9% 15.4% 3.5% 3.4% 0.4% 0.0% ref B L0: 93.8% 4.7% 1.1% 0.4% ref B L1: 99.0% 1.0% I can't remember how to interpret it properly but the above was the result of the MeGUI command line below. Someone else may be able to explain it, but the way I remember it is if the numbers at the end of each line are zero or very close to it, you're into wasting CPU cycles territory if you increase the number of ref/B frames further. "C:\Program Files\MeGUI\tools\x264\x264.exe" --level 4.1 --preset slow --tune film --crf 18.0 --min-keyint 25 --b-adapt 2 --vbv-bufsize 50000 --vbv-maxrate 50000 --me umh --stitchable --colormatrix bt470bg --sar 1:1 --frames 36931 --output "D:\0111.mkv.mkv" "D:\0111.mkv.avs" According to MediaInfo your example isn't 200MB. Maybe you looked at the size of the audio stream by mistake. Stream size : 2.38 GiB Last edited by hello_hello; 7th June 2020 at 09:59. |
6th June 2020, 21:47 | #9035 | Link | ||
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Do you think than there are somebody than know better the encoder than the developers? Select crf, preset, Target Playback device and encode. Modify defaults can work ok for a movie but bad for another. Don't exist miracle settings, let defaults. Quote:
Some playback devices can't support many bframes/ref-frames, don't forget the --level parameter.
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6th June 2020, 23:08 | #9036 | Link | |
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ref P L0: 60.3% of P frames have 1 reference 16.9% of P frames have 2 references 15.4% of P frames have 3 references 3.5% of P frames have 4 references 3.4% of P frames have 5 references 0.4% of P frames have 6 references 0.0% of P frames have 7 references (weightp 2 accept 2 more than the limit of --ref 5) ref B L0: 93.8% of B frames have 1 back reference 4.7% of B frames have 2 back references 1.1% of B frames have 3 back references 0.4% of B frames have 4 back references ref B L1: 99.0% of B frames have 1 forward reference 1.0% of B frames have 2 forward references The % of B frames with 4 ref is very low, if the encode is a 1080p can't have 5 ref with --level 4.1 --vbv-bufsize 50000 Like you say search for more ref is waste the time.
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8th June 2020, 13:15 | #9038 | Link |
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Thanks a lot for the insight again @ Hello_Hello
Will try and do the downmixing first then. as far as that stream size goes , I think the encoder has removed real info and added source info to the video. This is actually what the video mediainfo shows. @ tebasuna51 : Indeed the developers know best , I donno 0.00000000001% TBH , Hence I keep lurking here time and again I dont think I ll ever use any other preset than SLOWER, ever As advised gonna stick to what the devs have pointed One thing thats still a mystery to me is why all x265 encoded videos via MEGUI are cooler in color, do we need to tweak something or add something to the commandline to keep the colors similar to the source .. Thanks |
8th June 2020, 14:34 | #9039 | Link |
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The x265 thing might be a rec.601 vs rec.709 vs rec.2020 issue.
https://en.wikipedia.org/wiki/Rec._601 https://en.wikipedia.org/wiki/Rec._709 https://en.wikipedia.org/wiki/Rec._2020 YV12 has to be converted to RGB on playback for display, and SD, HD and UHD use different formulas. It's more likely the player is making an assumption based on resolution and that's why the colors look different. Or the video card's drivers are borked. What's the resolution of the source and encode? YV12 sources should just be decompressed and re-encoded as YV12 without a conversion to RGB and back (unless you're converting to RGB in a script for a filter that requires it), so re-encoding itself shouldn't change the colors, unless you downscale or upscale. For example, a SD source would usually be converted to RGB on playback using rec.601, but if you upscale to HD the player will probably use rec.709 instead, so you either need to convert the colors or set the correct colorimetry in the encoder configuration. The difference between rec.601 and rec.709 isn't huge, but rec.2020 is a lot different. The x264 encoder configuration offers the ability to write the correct colorimetry info via the GUI. It appears the x265 encoder configuration doesn't yet. Did you have a listen to the samples of audio dynamic rage compression I linked to? Last edited by hello_hello; 8th June 2020 at 14:38. |
10th June 2020, 08:33 | #9040 | Link | ||||
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I am not an expert and and I listened to them on my desktop speaker which is okish Regarding the color so I never convert to RGB ever AS i am these days trying on alreday converted videos so by and large all scripts use Quote:
strangely similar avisynth for a x264 video works well but when used with x265 cools the color off.. any tick you might wanna suggest to keep the colors as close to trhe original as possible. |
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