Re: [FFmpeg-user] Input 5.1 DTS, output 2.0 MP3: Atrocious Quality
To: ffmpeg-user@ffmpeg.org From: ceho...@ag.or.at Date: Mon, 18 May 2015 08:48:32 + Subject: Re: [FFmpeg-user] Input 5.1 DTS, output 2.0 MP3: Atrocious Quality John L orionfyre at hotmail.com writes: Please test the following: $ ffmpeg -i inter.dts -rematrix_maxval 1.0 -ac 2 out.ac3 $ ffmpeg -i inter.dts -rematrix_maxval 1.0 -ac 2 out.mp3 $ ffmpeg -i inter.dts -rematrix_maxval 1.0 -ac 2 -acodec pcm_f32le out.wav I ran all three as requested, including '-loglevel debug'. All three resulting files resulted in poor quality audio as before. Now we are there;-) Hendrik says the option fixes audio for him, you report it does not fix the issue... Carl Eugen I reviewed my work on this section and I was wrong; this does in fact work and solve the issue i was originally having. Perhaps I had my file browser pointed at the wrong working folder...? Thanks all, especially you Carl. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Input 5.1 DTS, output 2.0 MP3: Atrocious Quality
Please test the following: $ ffmpeg -i inter.dts -rematrix_maxval 1.0 -ac 2 out.ac3 $ ffmpeg -i inter.dts -rematrix_maxval 1.0 -ac 2 out.mp3 $ ffmpeg -i inter.dts -rematrix_maxval 1.0 -ac 2 -acodec pcm_f32le out.wav I ran all three as requested, including '-loglevel debug'. All three resulting files resulted in poor quality audio as before. the filtergraph output does show something different however, but the resulting audio is still terrible and indistinguishable from before. [AVFilterGraph @ 0x2387f20] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed 0.414214 0.00 0.292893 0.00 0.292893 0.00 0.00 0.414214 0.292893 0.00 0.00 0.292893 I do appreciate your help in resolving this issue. Just to see if it wasn't my system causing the issue I loaded up a windows XP vm and used the 32-bit windows binary from the ffmpeg homepage, resulting in the same outputs. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Input 5.1 DTS, output 2.0 MP3: Atrocious Quality
To: ffmpeg-user@ffmpeg.org From: ceho...@ag.or.at Date: Sat, 16 May 2015 11:15:29 + Subject: Re: [FFmpeg-user] Input 5.1 DTS, output 2.0 MP3: Atrocious Quality Moritz Barsnick barsnick at gmx.net writes: $ ffmpeg -loglevel debug -i inter.dts -ac 2 -c:a libmp3lame -f null - $ ffmpeg -loglevel debug -i inter.dts -ac 2 -c:a pcm_s16le -f null - You can insert other arbitrary codecs at will. The former shows a matrix: 1.00 0.00 0.707107 0.00 0.707107 0.00 0.00 1.00 0.707107 0.00 0.00 0.707107 [auto-inserted resampler 0 at 0xb713840] ch:6 chl:5.1(side) fmt:fltp r:48000Hz - ch:2 chl:stereo fmt:fltp r:48000Hz while the latter shows: 0.414214 0.00 0.292893 0.00 0.292893 0.00 0.00 0.414214 0.292893 0.00 0.00 0.292893 [auto-inserted resampler 0 at 0xb3b55c0] ch:6 chl:5.1(side) fmt:fltp r:48000Hz - ch:2 chl:stereo fmt:s16 r:48000Hz I think this may be the described behavior. If this really is the issue, it should be reproducible with at least one of the command lines I proposed (namely for -acodec pcm_f32le) and it is possible to work-around the issue by forcing s16p as the mp3 encoding format. The mp3 encoder accepts fltp, s16p and s32p. Thanks Everyone. You've fixed the problem for me. When I cycle through the available format options for libmp3lame, the only one that makes bad audio is 'fltp', both 's16p' and 's32p' produce a good file But I would really appreciate if somebody can confirm that the issue is reproducible with pcm_f32le (and neither with s16le nor s32le). Carl Eugen I ran the following command line and received the following: ffmpeg -loglevel debug -i inter.dts -c:a pcm_f32le -ac 2 -y inter-f32le.wav [AVFilterGraph @ 0xca9d20] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed 1.00 0.00 0.707107 0.00 0.707107 0.00 0.00 1.00 0.707107 0.00 0.00 0.707107 [auto-inserted resampler 0 @ 0xc965c0] ch:6 chl:5.1(side) fmt:fltp r:48000Hz - ch:2 chl:stereo fmt:flt r:48000Hz - the resulting wav file is significantly distorted, but qualitatively doesn't 'feel' as harsh And as you expected, both s16le nor s32le result in acceptable files for both: ffmpeg -loglevel debug -i inter.dts -c:a pcm_s16le -ac 2 -y inter-s16le.wav ffmpeg -loglevel debug -i inter.dts -c:a pcm_s32le -ac 2 -y inter-s32le.wav I quickly re-encoded movies that were known to suffer from this issue using '-c:a libmp3lame -sample_fmt s16p' and they all result in acceptable audio levels. So OFFICIALLY my problem is resolved. But I'm still curious why it would behave in such a way that fltp-fltp would be allowed to blow out levels so badly (i'm also scared to go down that rabbit hole) Thank you to all who have contributed. I appreciate your help tremendously. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Input 5.1 DTS, output 2.0 MP3: Atrocious Quality
Instead, take the 5.1 and _DOWNMIX_ all tracks to a single stereo for the phone/tablet by declaring -acodec -ac 2. No intermediate steps should be required. Consider also - Do you need pcm_s32le ? pcm_s16le is usual. I fail to see how that is any different than what I am doing now. I was under the impression that the flags -acodec and -c:a were the same. Regardless using -acodec reults in identical clipping and noise generation dts-mp3. For reference here is the command I used: ffmpeg -i inter.dts -acodec libmp3lame -ac 2 inter-new.mp3 The resulting mp3 file still has horrendous crackling and noise. What I thought I stated quite clearly in my OP is the following: 5.1DTS-2.0MP3 results in horrible noise and clipping in the resulting mp3 file 5.1DTS2.0PCM-2.0MP3 does not generate the same atrocious noise. please reference the files I've included in the dropbox link in my OP. inter.dts was the ripped 5.1 audio inter-test.mp3 was encoded to 2.0mp3 format directly from 5.1dts and when played back on my laptop(s) (Windows, Linux, Mac; in Windows MP, ffplay, mplayer, vlc, xine, and more), Phones (s5,s3,iphone,htc one), tablet, ipod and my Sansa MP3 player all has horrific noise. inter.mp3 was generated by converting the 5.1DTS to 2.0PCM and then the 2.0PCM to 2.0MP3, sounds just fine when played back on all of my devices. I am fully aware that there should be NO NEED to use an intermediary wave format to downsample to stereo audio from 5.1 for a conversion to mp3. But that's exactly why I'm writing this problem into the group because it is NOT working as expected. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
[FFmpeg-user] Input 5.1 DTS, output 2.0 MP3: Atrocious Quality
Backstory: I have a system in place to automagically convert video files to smaller formats/versions on request to have a sort of mobile version for my father who travels extensively. The purpose is so that he can fit significantly more videos on his tablet than if they were the high quality rips. It all boils down to: ffmpeg -i [input-file] -ac 2 -c:v libx264 -c:a libmp3lame -b:v 1024k -preset fast [output-file] I was under the impression everything was hunky dory until I took a bunch of the shrunken movies on my phone on a roadtrip. A good many of the videos were as good as can be expected, and nothing was egregiously wrong. However on a few videos the audio was absolutely atrocious, blown out, clipping, and just noise from seemingly nowhere. One of the worst was Intersteller which was completely unwatchable after the first two minutes with all the blown out crescendos, pops, cracks, static, and voices of the deep adulterating the audio stream. All video files affected by this were 5.1DTS sources, but not all 5.1DTS were affected. When talking with my father he said it was a frequent enough occurrence that he suspected it was just because I had shrunk the file so small and was an artifact of that. He did confirm that most videos that were affected weren't as bad as the Interstellar conversion. ~/testing$ ffmpeg -version ffmpeg version 2.5.6-0ubuntu0.15.04.1 Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.9.2 (Ubuntu 4.9.2-10ubuntu13) configuration: --prefix=/usr --extra-version=0ubuntu0.15.04.1 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --shlibdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --enable-shared --disable-stripping --enable-avresample --enable-avisynth --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libshine --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libwavpack --enable-libwebp --enable-libxvid --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzvbi --enable-libzmq --enable-frei0r --enable-libvpx --enable-libx264 --enable-libsoxr --enable-gnut ls --enable-openal --enable-libopencv --enable-librtmp --enable-libx265 libavutil 54. 15.100 / 54. 15.100 libavcodec 56. 13.100 / 56. 13.100 libavformat56. 15.102 / 56. 15.102 libavdevice56. 3.100 / 56. 3.100 libavfilter 5. 2.103 / 5. 2.103 libavresample 2. 1. 0 / 2. 1. 0 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc53. 3.100 / 53. 3.100 To troubleshoot I copied out a particularly bad snippet of audio ffmpeg -i Int*.mkv -vn -c copy -ss 1:30 -t 0:30 inter.dts This audio clip is confirmed to be a good 5.1dts stream ffmpeg -i inter.dts -ac 2 -c libmp3lame inter-test.mp3 This audio sample has the exact same audio defects as in the shrunken video Converting it to a stereo wave format, and then converting into an mp3: ffmpeg -i inter.dts -ac 2 -c pcm_s32le inter.wav ffmpeg -i inter.wav -c libmp3lame inter.mp3 both inter.wav and inter.mp3 are confirmed to be GOOD stereo copies of the audio with no defects. https://www.dropbox.com/s/tru46zo07gcr8ve/testing.tar.gz?dl=0 This is a link to the files in question to my testing above. inter.dts : 30 second rip of audio from video inter-test : dts-mp3 conversion inter.mp3 : dts-wav-mp3 conversion I apologize if I'm missing something glaring, but I've been unable to find any other instances of this issue with my google-fu. Until I have a solution I've already edited my services to perform this intermediary wave step work-around on all conversions. Thank you for your time. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
