Re: [FFmpeg-user] Integration of ffmpeg with Haswell
On Mon, May 18, 2015 at 15:07:03 +, Shiwani Agrawal wrote: Thanks for helping me out in this issue , just want to clarify in the end , does it mean that if we integrate ffmpeg with the media SDK it increases the speed (or Quality) of transcoding process by processor ? If yes , why ? The process is called hardware acceleration. See here: https://trac.ffmpeg.org/wiki/HWAccelIntro It helps applications - in this case decoders, encoders, filters, and so on - use special hardware for certain calculations, thereby reducing the load on the CPU and/or increasing the speed. In your case, you are referring to the Intel Quick Sync hardware, which was created particularly for video (and audio?) coding. Using it _should_ increase the speed of the transcoding process (note that speed, quality and compression size are always trade-offs), but doesn't always, it depends on the implementation. On the other hard, the hardware algorithms have certain restrictions and aren't as flexible as software solutions. You won't be able to adjust all parameters of the encoding process, and sometimes more complex profiles or more modern codecs aren't supported either. So it really depends on your use case. And I think you need a really good justification to pay for an extra license for the Intel SDK. (I'm not sure a working free implementation exists yet.) As a normal user, I wouldn't go to the trouble of integrating the SDK. Just check how good ffmpeg's other support is for you first. Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Integration of ffmpeg with Haswell
Hello , Thanks for helping me out in this issue , just want to clarify in the end , does it mean that if we integrate ffmpeg with the media SDK it increases the speed (or Quality) of transcoding process by processor ? If yes , why ? Thanks and Regards Shiwani Date: Mon, 18 May 2015 12:53:12 +0200 From: barsn...@gmx.net To: ffmpeg-user@ffmpeg.org Subject: Re: [FFmpeg-user] Integration of ffmpeg with Haswell On Mon, May 18, 2015 at 12:43:12 +0200, Moritz Barsnick wrote: Actually, my bad. Support is already integrated into ffmpeg. It appears that, at the presence of the Media SDK (libmfx, mfx/mfxvideo.h), the qsv encoders and decoders are built. You will need the Media SDK, and you have to build ffmpeg yourself. I failed to recognize: You need to configure your ffmpeg build with --enable-libmfx Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Integration of ffmpeg with Haswell
Hello , Thanks for explanation , I would certainly test the speed enhancement for my case before paying for the license . Thanks and Regards , Shiwnai Date: Mon, 18 May 2015 17:19:22 +0200 From: barsn...@gmx.net To: ffmpeg-user@ffmpeg.org Subject: Re: [FFmpeg-user] Integration of ffmpeg with Haswell On Mon, May 18, 2015 at 15:07:03 +, Shiwani Agrawal wrote: Thanks for helping me out in this issue , just want to clarify in the end , does it mean that if we integrate ffmpeg with the media SDK it increases the speed (or Quality) of transcoding process by processor ? If yes , why ? The process is called hardware acceleration. See here: https://trac.ffmpeg.org/wiki/HWAccelIntro It helps applications - in this case decoders, encoders, filters, and so on - use special hardware for certain calculations, thereby reducing the load on the CPU and/or increasing the speed. In your case, you are referring to the Intel Quick Sync hardware, which was created particularly for video (and audio?) coding. Using it _should_ increase the speed of the transcoding process (note that speed, quality and compression size are always trade-offs), but doesn't always, it depends on the implementation. On the other hard, the hardware algorithms have certain restrictions and aren't as flexible as software solutions. You won't be able to adjust all parameters of the encoding process, and sometimes more complex profiles or more modern codecs aren't supported either. So it really depends on your use case. And I think you need a really good justification to pay for an extra license for the Intel SDK. (I'm not sure a working free implementation exists yet.) As a normal user, I wouldn't go to the trouble of integrating the SDK. Just check how good ffmpeg's other support is for you first. Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Query
[Your subject doesn't say very much. That's not helpful.] On Mon, May 18, 2015 at 16:59:35 +0530, Hardik Kanakia wrote: I have 6 separate channels (5.1) and separate video file which needs to be merged into 1 file $ ffmpeg -i videofile -i audiofile -c copy outfile.mkv Or are you saying you have six separate audio files? Please be more precise. Tell us what you are tryig to achieve (more precisely), what you tried, what you got, what you expected. Include the full command line and complete, uncut console output. Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Numerical histogram output for checking typical broadcast ranges (16-235)
I cannot explain the relevant specifications (and I don't even know where to find them) http://www.itu.int/dms_pubrec/itu-r/rec/bt/R-REC-BT.601-7-201103-I!!PDF-E.pdf 220 (8-bit) or 877 (10-bit) quantization levels with the black level corresponding to level 16.00d and the peak white level corresponding to level 235.00d. The signal level may occasionally excurse beyond level 235.00d or below level 16.00d Similarly for 709. P ___ 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
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 ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Getting precise start time from wall-clock for capture
Thanks Nicolas. Please remember to follow the mailing list's netiquette, in particular not hijacking threads (too late for that this time) but all of it. Apologies for hijacking the thread (though I'm not clear as to what in particular I did in that regard--I started the thread). You did not bother to tell what kind of capture you are doing, so it is not possible to point you to the exact place in the documentation. The capture I'm doing is v4l2 to nvenc to matroska. Here's the command line I've been using: ffmpeg -f v4l2 -framerate 60 -video_size 1920x1080 -i /dev/video0 -r 2997/100 -f matroska -c:v nvenc -b:v 25000k -minrate 25000k -maxrate 25000k -g 1 -profile:v high -preset hq my_video.mkv Thanks again for any tips you can provide. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
[FFmpeg-user] ffmpeg livestreaming with subtitles
Hi, I am a new ffmpeg user. I like it very much. I got a subtitle livestreaming problem though. Basically, I like to livestream a customized subtitles together with audio/video mp4 file. The subtitle track has to be a separate one (cannot be hardcoded or burned into the video data. I know how to burn but I don't want that as the user at the client side may change the subtitles). I have recently asked a question on ffmpeg livestreaming with subtitles http://ffmpeg.gusari.org/viewtopic.php?f=11t=2115 in ffmpeg developer forum but was suggested that I asked the question on how to livestream subtitles with ffmpeg here. Please help. Here is the current status, * Already added subtitles to india_sub_srt.mp4 using ffmpeg successfully. Using a vlc player to play the file, I can see the newly added subtitles. * I am using an octoshape sever. The server set up was fine. If I use -f flv instead of -f mpegts as the output container (see below), I can stream successfully. However, flv does not support subtitle livestream. I also tried mp4, mov as an output format but got an error message saying those are not seekable formats. * Followed a couple of suggestions I got from the aforementioned post and was able to see ffmpeg processing audo, video, and even subtitle for streaming but there was a conversion failure. I wonder: does it mean the current ffmpeg does not support or I need to use some other format? Please see below for my screen print-outs. Command: ffmpeg -re -i india_sub_srt.mp4 -c:v libx264 -preset fast -maxrate 500k -bufsize 4000k -pix_fmt yuv420p -g 50 -c:a copy -c:s copy -f mpegts rtmp://localhost/live/livestream Screen print-out: ffmpeg version N-71481-g1c37848 Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.9.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca -- enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l ibilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab le-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --en able-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --ena ble-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enabl e-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-lzma --ena ble-decklink --enable-zlib libavutil 54. 22.101 / 54. 22.101 libavcodec 56. 34.100 / 56. 34.100 libavformat56. 30.100 / 56. 30.100 libavdevice56. 4.100 / 56. 4.100 libavfilter 5. 14.100 / 5. 14.100 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc53. 3.100 / 53. 3.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'india_sub_srt.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf56.30.100 Duration: 00:04:59.86, start: 0.00, bitrate: 309 kb/s Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yu v420p(tv, bt709), 320x180, 209 kb/s, 25 fps, 25 tbr, 90k tbn, 180k tbc (default) Metadata: handler_name: VideoHandler Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, flt p, 95 kb/s (default) Metadata: handler_name: SoundHandler Stream #0:2(und): Subtitle: mov_text (tx3g / 0x67337874), 0 kb/s (default) Metadata: handler_name: SubtitleHandler [libx264 @ 040cbe60] using cpu capabilities: MMX2 SSE2Fast LZCNT [libx264 @ 040cbe60] profile High, level 2.1 Output #0, mpegts, to 'rtmp://localhost/live/livestream': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf56.30.100 Stream #0:0(und): Video: h264 (libx264), yuv420p, 320x180, q=-1--1, max. 500 kb/s, 25 fps, 90k tbn, 25 tbc (default) Metadata: handler_name: VideoHandler encoder : Lavc56.34.100 libx264 Stream
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] error when compiling ffmpeg
Am 18.05.2015 um 00:36 schrieb Kelly Philp: Hello I have an error when compiling ffmpeg this the error: gcc is unable to create an executable file. If gcc is a cross-compiler, use the --enable-cross-compile option. Only do this if you know what cross compiling means. C compiler test failed attached config.log there is nothing attached and you should just post your configure line and the output at start signature.asc Description: OpenPGP digital signature ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Increasing the decoding speed of yuv444p h.264 files
On 5/18/2015 7:22 PM, Yan Li wrote: I've done some more tests. Input file src.flv is in yuv444p with 48 fps. I use the following commands to generate two lossless test files: 1: ffmpeg -i src.flv -c:v libx264 -qp 0 -preset ultrafast -an yuv444p.mkv 2: ffmpeg -i src.flv -c:v libx264 -qp 0 -preset ultrafast -pix_fmt yuv420p -an yuv420p.mkv Then I test the decoding speed using: time ffmpeg -i yuv444p.mkv -f rawvideo -an - /dev/null - yuv444p.mkv (496 Mb/s): decoded at 55 fps - yuv420p.mkv (336 Mb/s): decoded at 92 fps (I/O is not the bottleneck here. Four threads are used.) sorry to question, but: are you really sure about this? those are really high bitrates and it appears to be a direct link between bitrate and decoding speed: 496 x 55 ~= 336 x 92 (by a 10% margin). I don't know what's the drive access pattern in this case but if seeks just a bit, you have a problem. as a test, I would encode at much smaller bitrates and then compare the fps. of course, you can measure the disk I/O load with appropriate tools during decoding. -- Claudiu ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Increasing the decoding speed of yuv444p h.264 files
On Mon, May 18, 2015 at 2:00 AM, Carl Eugen Hoyos ceho...@ag.or.at wrote: Yan Li elliot.li.tech at gmail.com writes: 2. I want to archive the files in yuv444p. Is there any option that can make ffplay play these files faster? I don't mind losing replay quality for now, but I do need to keep yuv444p files. Please test and report, I would expect that different encoding options imply different decoding speeds. I've done some more tests. Input file src.flv is in yuv444p with 48 fps. I use the following commands to generate two lossless test files: 1: ffmpeg -i src.flv -c:v libx264 -qp 0 -preset ultrafast -an yuv444p.mkv 2: ffmpeg -i src.flv -c:v libx264 -qp 0 -preset ultrafast -pix_fmt yuv420p -an yuv420p.mkv Then I test the decoding speed using: time ffmpeg -i yuv444p.mkv -f rawvideo -an - /dev/null - yuv444p.mkv (496 Mb/s): decoded at 55 fps - yuv420p.mkv (336 Mb/s): decoded at 92 fps (I/O is not the bottleneck here. Four threads are used.) A player that outputs video are slower than pure ffmpeg decoding, thus the 48 fps yuv444p.mkv looks choppy on my machine. Iirc, there is a x264 option that makes the output file decode faster. I'm aware of that option. I can't find this documented anywhere, but it seems to me that fastdecode has no effect on lossless encoding. I do have lossy files too. But they are for archive purpose, and I don't want to sacrifice quality/compression ratio for decoding speed; I bet future hardware would have no problem playing them. 3. Are there any chance that the software (ffmpeg or x264/x265) can get better optimized in the near future (1 or 2 years?) I don't have a crystal ball but I can say that this was the case in the past (but this is also something you could test). I guess the color space conversion itself can't explain the huge gap in decoding speed. So it is actually ffmpeg's yuv444p decoding is compute-intensive. I think what I want to know is whether the code is already fully optimized/SIMDized, because h264 has been standardized for a decade, and I'm not sure if someone is still actively optimizing ffmpeg's h264 decoder. If that's the case I won't expect magic to happen in the near future, and will just wait for my budget to buy new hardware. But heck, what do I know about ffmpeg, I won't be surprised if I were totally wrong. -- Yan ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Getting precise start time from wall-clock for capture
Le nonidi 29 floréal, an CCXXIII, Arthur Wait a écrit : Apologies for hijacking the thread (though I'm not clear as to what in particular I did in that regard--I started the thread). No real harm done, ans sorry for the harsh tone. Your initial message has this: In-Reply-To: cae6_u_rtud6z-kd1fidfaz6_zo6fhlfvrv_nw7vqb2a0pum...@mail.gmail.com That means you wrote it by replying to this message. The capture I'm doing is v4l2 to nvenc to matroska. Then frames have timestamps provided by the kernel: ffprobe -i /dev/video0 -of compact -show_packets [...] packet|codec_type=video|stream_index=0|pts=1392045174616|pts_time=1392045.174616|... As you can see, this is the monotonic clock (and my uptime here is 16 days), but lavd can convert for you: http://ffmpeg.org/ffmpeg-devices.html#video4linux2_002c-v4l2 With that, you just need to do whatever you want with the timestamps. By default, ffmpeg will normalize the timestamps before any filtering or encoding, but you can disable that with the -copyts option. Regards, -- Nicolas George signature.asc Description: Digital signature ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Getting precise start time from wall-clock for capture
Fantastic Nicolas, thanks so much! This is exactly what I needed. Arthur ffprobe -i /dev/video0 -of compact -show_packets [...] packet|codec_type=video|stream_index=0|pts=1392045174616|pts_time=1392045.174616|... As you can see, this is the monotonic clock (and my uptime here is 16 days), but lavd can convert for you: ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] ffmpeg livestreaming with subtitles
JamesW2015 jf_weng-at-yahoo.com at ffmpeg.org writes: I have recently asked a question on ffmpeg livestreaming with subtitles http://ffmpeg.gusari.org/viewtopic.php?f=11t=2115 in ffmpeg developer forum Unrelated: You can call the Gusari forum FFmpeg but please do not call it a developer forum. ffmpeg -re -i india_sub_srt.mp4 -c:v libx264 -preset fast -maxrate 500k -bufsize 4000k -pix_fmt yuv420p -g 50 -c:a copy -c:s copy -f mpegts You cannot put random codecs (in this case mov_text) into mpegts, this does not work. FFmpeg (still) expects you to know what you are doing and (at least in this case) prints no warning. Do you believe that your flv player supports text subtitles as in this directory? http://samples.ffmpeg.org/ffmpeg-bugs/trac/ticket2933/ If yes, this could be possible in the future, if not there is nothing that FFmpeg can do. Similar for mpegts: What does your player support? Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Getting precise start time from wall-clock for capture
On Mon, May 18, 2015 at 18:38:18 +0200, Nicolas George wrote: As you can see, this is the monotonic clock (and my uptime here is 16 days), but lavd can convert for you: http://ffmpeg.org/ffmpeg-devices.html#video4linux2_002c-v4l2 I tried that, because I wanted to answer the question as well. It was non-obvious to me that this, what you call normalize, would happen: With that, you just need to do whatever you want with the timestamps. By default, ffmpeg will normalize the timestamps before any filtering or encoding, but you can disable that with the -copyts option. I see the timestamp of the first frame reset to 0 - is that what you mean with normalize? From the description of the v4l2 device, I had expected the wallclock to be copied to the timestamps (PTS, right?), but they began with 0 for the first frame. I then tried to add -vf setpts=RTCSTART+PTS-STARTPTS but that was apparently a long shot - it didn't help. So I added -copyts, but again to no avail. The first frame always shows: pkt_pts=0 pkt_pts_time=0.00 pkt_dts=0 pkt_dts_time=0.00 best_effort_timestamp=0 best_effort_timestamp_time=0.00 I was hoping to be able to hint the original poster to grab the start time from the PTS of the first frame. So: How to get wallclock timestamps into the video? For reference, my attempt. (The input obviously has the correct wallclock time: 1431981242.712328.) $ ffmpeg -f v4l2 -input_format yuv420p -timestamps abs -i /dev/video1 -t 1 -c:v libx264 -copyts ~/tmp/video1.mkv -y ffmpeg version 2.6.git Copyright (c) 2000-2015 the FFmpeg developers built with icc (ICC) 14.0.3 20140422 configuration: --prefix=/usr/new/tools/video/install/ffmpeg/2015-05-11 --cc=icc --cxx=icpc --enable-gpl --enable-version3 --enable-nonfree --disable-shared --enable-gnutls --enable-libcdio -- enable-libfreetype --enable-libx264 --enable-libmp3lame --enable-openal --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libass --enable-libv 4l2 --enable-libvidstab --enable-libfdk-aac --enable-libsmbclient --enable-libquvi --enable-libzvbi --enable-libzmq --extra-ldflags=-L/usr/new/tools/video/install/fdk-aac/current/lib --extra-cf lags=-I/usr/new/tools/video/install/fdk-aac/current/include libavutil 54. 23.101 / 54. 23.101 libavcodec 56. 38.100 / 56. 38.100 libavformat56. 32.100 / 56. 32.100 libavdevice56. 4.100 / 56. 4.100 libavfilter 5. 16.101 / 5. 16.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc53. 3.100 / 53. 3.100 [video4linux2,v4l2 @ 0xa60bd00] Detected absolute timestamps Input #0, video4linux2,v4l2, from '/dev/video1': Duration: N/A, start: 1431981242.712328, bitrate: 124416 kb/s Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 720x576, 124416 kb/s, 25 fps, 25 tbr, 1000k tbn, 1000k tbc [libx264 @ 0xa60de20] using cpu capabilities: MMX2 SSE2 Cache64 [libx264 @ 0xa60de20] profile High, level 3.0 [libx264 @ 0xa60de20] 264 - core 142 - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 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=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=3 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, matroska, to '/home/barsnick/tmp/video1.mkv': Metadata: encoder : Lavf56.32.100 Stream #0:0: Video: h264 (libx264) (H264 / 0x34363248), yuv420p, 720x576, q=-1--1, 25 fps, 1k tbn, 25 tbc Metadata: encoder : Lavc56.38.100 libx264 Stream mapping: Stream #0:0 - #0:0 (rawvideo (native) - h264 (libx264)) Press [q] to stop, [?] for help [libx264 @ 0xa60de20] non-strictly-monotonic PTS Last message repeated 12 times [libx264 @ 0xa60de20] non-strictly-monotonic PTS0:00:00.00 bitrate=N/A Last message repeated 10 times frame= 24 fps=2.6 q=-1.0 Lsize=2129kB time=-23860:55:45.88 bitrate=N/A video:2128kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.039607% [libx264 @ 0xa60de20] frame I:1 Avg QP:38.00 size: 90932 [libx264 @ 0xa60de20] frame P:23Avg QP:38.00 size: 90753 [libx264 @ 0xa60de20] mb I I16..4: 0.0% 70.0% 30.0% [libx264 @ 0xa60de20] mb P I16..4: 0.9% 85.8% 13.1% P16..4: 0.0% 0.1% 0.1% 0.0% 0.0%skip: 0.0% [libx264 @ 0xa60de20] 8x8 transform intra:85.3% inter:65.0% [libx264 @ 0xa60de20] coded y,uvDC,uvAC intra: 100.0% 0.0% 0.0% inter: 100.0% 0.0% 0.0% [libx264 @ 0xa60de20] i16 v,h,dc,p: 0% 7% 81% 13% [libx264 @ 0xa60de20] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 1% 42% 29% 2% 3%
Re: [FFmpeg-user] subtitles - visual alignment offset?
Thanks, I thought *MarginV* might be the flag I'm after, but it's not being recognized as a valid option. Saw a user submitted some new functionality, but doesn't look like its in build 2.6.1 w/ libbass enabled. [FFmpeg-devel] Adding Force Style option in Subtitles Filter http://lists.mplayerhq.hu/pipermail/ffmpeg-devel/2015-February/168341.html On Sat, May 16, 2015 at 3:00 AM, Carles Vila cvi...@gmail.com wrote: One option would be to use .ass subtitles instead of srt, .ass allows you to adjust position and appearance at will for all or individual subtitles. I use Aegisub for this purpose. -Mensaje original- De: jamie chocolate.el...@gmail.com Enviado: 16/05/2015 0:26 Para: FFmpeg user questions ffmpeg-user@ffmpeg.org Asunto: [FFmpeg-user] subtitles - visual alignment offset? Is there a way to add a visual alignment offset for subtitles when burned in? The default subtitle placement with SRT's are low in the frame when considering broadcast safe. What's the quickest way to nudge them all higher in the frame? Just using this simple flag subtitles=input.srt:force_style='PrimaryColour=HAA00FF00' Thanks! ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Getting precise start time from wall-clock for capture
I see the timestamp of the first frame reset to 0 - is that what you mean with normalize? Here's what seems to have worked for me: ffmpeg -f v4l2 -framerate 60 -video_size 1920x1080 -ts mono2abs -i /dev/video-static -r 2997/100 -f matroska -c:v nvenc -b:v 25000k -minrate 25000k -maxrate 25000k -g 1 -profile:v high -preset hq -copyts output.mkv It took me a few tries to get the -ts mono2abs and -copyts into the right location, but now I believe I'm getting good wall-clock time (at least, it looks logical to me). I'm still doing some testing, though... ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] ffmpeg livestreaming with subtitles
Sorry. I am new and I don't know much about this. I am using VLC player and it can support subtitles. I did not mean to use -f mpegts as the output format. That was an example only. I meant to ask which format I should use in ffmpeg command line to support subtitle livestreaming. Thanks. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-livestreaming-with-subtitles-tp4670507p4670523.html Sent from the FFmpeg-users mailing list archive at Nabble.com. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
[FFmpeg-user] H264 IP-camera dump causes wrong duration/bitrate
I got H264 livestream dump from my IP-camera with ffmpeg. The movies get shorter than real time in night time, although the length is correct in day time. The command line I did is below. I used segment function with segment time = 60. So, I expect the duration of each mp4 file is about 60 seconds. ffmpeg -i http://temp2.shirase.tk/livestream.cgi?user=guestpwd=gueststreamid=0audio=0filename=; -y -vcodec copy \ -loglevel verbose \ -movflags faststart+empty_moov \ -f segment \ -segment_atclocktime 1 \ -segment_time 60 \ -reset_timestamps 1 \ -strftime 1 \ %y%m%d%H%M.mp4 /dev/zero 2 dump.log The files below is the sample files and console log. * http://temp1.shirase.tk/1505151230.mp4 This file is a sample in day time. The duration of this file is 00:01:07.20. * http://temp1.shirase.tk/1505152058.mp4 This file is a sample in night time. The duration is 00:00:24.00. * http://temp1.shirase.tk/dump.log console log. I posted this problem to ffmpeg forum before. ( http://ffmpeg.gusari.org/viewtopic.php?f=11t=2013 ) But I could not get any solution. I also asked about h264 : data partitioning to ffmpeg-devel after I found this message. https://ffmpeg.org/pipermail/ffmpeg-devel/2015-May/172614.html But it's not my trouble. Regard. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Text on multiple lines
lorenzo angeli lorenzo.angeli at efestolab.uk writes: In a way yes(works pressing enter in the command line) . I need to find a way to do it programmatically from python. So python lacks string concatenation or does it not know the newline character? Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Increasing the decoding speed of yuv444p h.264 files
On Mon, May 18, 2015 at 09:00:36 +, Carl Eugen Hoyos wrote: Iirc, there is a x264 option that makes the output file decode faster. Indeed: --tune fastdecode, which maps to ffmpeg's libx264's -tune fastdecode (Note that using -preset will change some of those settings additionally. I don't know which of the two options wins or how they mix.) See also http://superuser.com/questions/564402/explanation-of-x264-tune Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Text on multiple lines
On Mon, May 18, 2015 at 11:53:04 +0200, Moritz Barsnick wrote: On Mon, May 18, 2015 at 11:49:04 +0200, lorenzo angeli wrote: I'm not sure how to break a line (being just text ), rather than with \r or \n. I've been trying all sort of thing but without success. Wouldn't be possible to have the draw text able to understand these common delimiters? Python should be able to place verbatin newlines (probably encoded with \n) into string variables. Those variables are then passed to the system() call (or its equivalent in python). Trying to recall my python courses, there are many ways to execute a system program. But at least here's a proff of concept for you: from subprocess import call a = Hello,\nworld call([echo, a]) Hello, world 0 So, as I wrote, just insert the newline as \n into the string argument. Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Increasing the decoding speed of yuv444p h.264 files
Yan Li elliot.li.tech at gmail.com writes: 2. I want to archive the files in yuv444p. Is there any option that can make ffplay play these files faster? I don't mind losing replay quality for now, but I do need to keep yuv444p files. Please test and report, I would expect that different encoding options imply different decoding speeds. Iirc, there is a x264 option that makes the output file decode faster. 3. Are there any chance that the software (ffmpeg or x264/x265) can get better optimized in the near future (1 or 2 years?) I don't have a crystal ball but I can say that this was the case in the past (but this is also something you could test). Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] H264 IP-camera dump causes wrong duration/bitrate
しらせ けんじ kenji at shirase.tk writes: I got H264 livestream dump from my IP-camera with ffmpeg. The movies get shorter than real time in night time, although the length is correct in day time. Is the issue also reproducible if you encode instead of remuxing? $ ffmpeg -vcodec mpeg4 -qscale 2 ... Carl Eugen ___ 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
John L wrote: 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. That's strange, it seems to do the correct thing for me (only tested last one). I notice typical of loud movie dts the master sample is quite extreme to start with (and clipped a bit by studio I think). But ignoring that the downmix with -rematrix_maxval 1.0 is the same for me as using a signed wav. Looking with sox ffmpeg -i inter.dts -rematrix_maxval 1.0 -ac 2 -acodec pcm_f32le outf-1.wav sox outf-1.wav -n stats sox WARN wav: wave header missing FmtExt chunk Overall Left Right DC offset -0.000397 -0.000397 -0.000138 Min level -0.986728 -0.977729 -0.986728 Max level 0.988566 0.974450 0.988566 Pk lev dB -0.10 -0.20 -0.10 RMS lev dB-11.54-11.47-11.60 RMS Pk dB -4.73 -4.73 -4.97 RMS Tr dB -36.11-36.08-36.11 Crest factor - 3.66 3.76 Flat factor 0.00 0.00 0.00 Pk count 2 2 2 Bit-depth 32/32 32/32 32/32 Num samples1.44M Length s 29.995 Scale max 1.00 Window s 0.050 Without -rematrix_maxval 1.0 sox outf-0.wav -n stats sox WARN wav: wave header missing FmtExt chunk Overall Left Right DC offset -0.005071 -0.005071 -0.002057 Min level -1.00 -1.00 -1.00 Max level 1.00 1.00 1.00 Pk lev dB 0.00 0.00 0.00 RMS lev dB -5.62 -5.61 -5.63 RMS Pk dB -0.79 -0.79 -1.03 RMS Tr dB -28.46-28.42-28.46 Crest factor - 1.91 1.91 Flat factor45.34 46.05 44.52 Pk count222k 228k 216k Bit-depth 32/32 32/32 32/32 Num samples1.44M Length s 29.995 Scale max 1.00 Window s 0.050 sox WARN sox: `outf-0.wav' input clipped 443634 samples ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Text on multiple lines
Hi Carl, In a way yes(works pressing enter in the command line) . I need to find a way to do it programmatically from python. Ta. L. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Text on multiple lines
I'm not sure how to break a line (being just text ), rather than with \r or \n. I've been trying all sort of thing but without success. Wouldn't be possible to have the draw text able to understand these common delimiters? On 18 May 2015 11:43, Carl Eugen Hoyos ceho...@ag.or.at wrote: lorenzo angeli lorenzo.angeli at efestolab.uk writes: In a way yes(works pressing enter in the command line) . I need to find a way to do it programmatically from python. So python lacks string concatenation or does it not know the newline character? Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Text on multiple lines
On Mon, May 18, 2015 at 11:49:04 +0200, lorenzo angeli wrote: I'm not sure how to break a line (being just text ), rather than with \r or \n. I've been trying all sort of thing but without success. Wouldn't be possible to have the draw text able to understand these common delimiters? Python should be able to place verbatin newlines (probably encoded with \n) into string variables. Those variables are then passed to the system() call (or its equivalent in python). Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Text on multiple lines
Sorry did inserted again the previous reply. Damn mobile. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Text on multiple lines
Ok I'll try again and let you know. Thanks. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Text on multiple lines
lorenzo angeli lorenzo.angeli at efestolab.uk writes: Sorry , no way then to tell ffmpeg to break the line somehow ? I thought we had solved this issue? Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Integration of ffmpeg with Haswell
On Mon, May 18, 2015 at 12:10:47 +0200, Moritz Barsnick wrote: If you really have the need for testing the hardware acceleration, I believe you must google your way through those paths of integrating. Otherwise, just use plain ffmpeg. Actually, my bad. Support is already integrated into ffmpeg. It appears that, at the presence of the Media SDK (libmfx, mfx/mfxvideo.h), the qsv encoders and decoders are built. You will need the Media SDK, and you have to build ffmpeg yourself. Good luck, (I won't try it myself for the time being) Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Integration of ffmpeg with Haswell
On Mon, May 18, 2015 at 12:43:12 +0200, Moritz Barsnick wrote: Actually, my bad. Support is already integrated into ffmpeg. It appears that, at the presence of the Media SDK (libmfx, mfx/mfxvideo.h), the qsv encoders and decoders are built. You will need the Media SDK, and you have to build ffmpeg yourself. I failed to recognize: You need to configure your ffmpeg build with --enable-libmfx Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Integration of ffmpeg with Haswell
Hi Shiwani, On Mon, May 18, 2015 at 01:59:04 +, Shiwani Agrawal wrote: what I am not getting is the need for installing intel media SDK and then integrating ffmpeg with it . Can we install ffmpeg directly without having installed intel media SDK ?. Yes, sure you can install ffmpeg without the Intel Media SDK. ffmpeg then doesn't make use of the Intel Quick Sync hardware acceleration though. There are arguments whether that acceleration is useful though - arguments about actual speed gain, supported codec parameters, and so on. Some of these arguments may be outdated. If you really have the need for testing the hardware acceleration, I believe you must google your way through those paths of integrating. Otherwise, just use plain ffmpeg. Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] HD SD Down-Convert Prores Color
Kevin Wells kevwells at hotmail.co.uk writes: This does not suffer from the issue, but the colours look slightly off ffmpeg -i /Users/kev/Movies/HD_BARS.mov -vf scale=720:576 -vcodec v210 -an out.mov Please test the following, sorry for not realising this earlier: $ ffmpeg -i HD_BARS.mov -s pal -vcodec prores -an -movflags +write_colr -an -s pal -color_primaries bt709 -color_trc bt709 -colorspace bt709 out.mov Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Getting precise start time from wall-clock for capture
Le nonidi 29 floréal, an CCXXIII, Arthur Wait a écrit : I'm using FFMPEG to record video from several cameras attached to several networked computers. The computers' clocks are all synced. I'd like to be able to get a precise date/time for the first captured frame All captured frames should have a timestamp attached, usually to the microsecond; an option may be needed to have it based on the wall clock. You did not bother to tell what kind of capture you are doing, so it is not possible to point you to the exact place in the documentation. Please remember to follow the mailing list's netiquette, in particular not hijacking threads (too late for that this time) but all of it. Regards, -- Nicolas George signature.asc Description: Digital signature ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
[FFmpeg-user] Query
Hi, Need help: I have 6 separate channels (5.1) and separate video file which needs to be merged into 1 file Could you please help me with the command as I am very new to the ffmpeg world Regards, Hardik Kanakia +91 93204 89772 ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
