December 2017 - ffmpeg-user - lists.ffmpeg.org

convertion from h264 to mp4 for MSE
by Grzegorz Wolny 23 Dec '17

23 Dec '17

Hello, I'm having problem with conversion from h264 to mp4. I would like to pack h264 stream into mp4 container. The stream in mp4 format is going to be played in chrome browser with html5 video and MSE. I'm trying to do this with following command: ffmpeg -f h264 -i 720p.h264 -c:v copy -an -movflags frag_keyframe+empty_moov+default_base_moof -f mp4 720p.mp4 but the output stream is not correctly played by chrome. It looks like there is problem with 'timescale' and 'default sample flags'. Boxes mvhd and mdhd contains different timescale. Box tfhd contains default sample flags = 1010000. Here is log from stdout of the ffmpeg: $ ffmpeg -f h264 -i 720p.h264 -c:v copy -an -movflags frag_keyframe+empty_moov+default_base_moof -f mp4 720p.mp4 ffmpeg version N-89551-gbc10260 Copyright (c) 2000-2017 the FFmpeg developers built with gcc 4.9.4 (Ubuntu 4.9.4-2ubuntu1~14.04.1) configuration: libavutil 56. 6.100 / 56. 6.100 libavcodec 58. 8.100 / 58. 8.100 libavformat 58. 3.100 / 58. 3.100 libavdevice 58. 0.100 / 58. 0.100 libavfilter 7. 7.100 / 7. 7.100 libswscale 5. 0.101 / 5. 0.101 libswresample 3. 0.101 / 3. 0.101 Input #0, h264, from '720p.h264': Duration: N/A, bitrate: N/A Stream #0:0: Video: h264 (High), yuv420p(progressive), 1280x720 [SAR 1:1 DAR 16:9], 30 fps, 30 tbr, 1200k tbn, 60 tbc File '720p.mp4' already exists. Overwrite ? [y/N] y Output #0, mp4, to '720p.mp4': Metadata: encoder : Lavf58.3.100 Stream #0:0: Video: h264 (High) (avc1 / 0x31637661), yuv420p(progressive), 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 30 fps, 30 tbr, 1200k tbn, 1200k tbc Stream mapping: Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help [mp4 @ 0x20ae880] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly [mp4 @ 0x20ae880] pts has no value Last message repeated 1251 times frame= 1252 fps=0.0 q=-1.0 Lsize= 6304kB time=00:00:41.63 bitrate=1240.3kbits/s speed=1.25e+03x video:6293kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.184422% I would like to notice that the same h264 stream has been successfully converted to mp4 by mp4box command: MP4Box -add 720p.h264 -dash -1 -frag 5000 test-out.mp4. Could you give me any clue how to do the same with ffmpeg? Best regards, Grzegorz Wolny

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merge two mono audio streams to stereo without losing end of longer input
by Dave Horton 22 Dec '17

22 Dec '17

I’m merging two input audio streams using ‘-strict 2 -filter_complex [0a][1:a]amerge” in order to produce a stereo recording, and I’ve noted that per the documentation the output stream has the length of the shorter of the two inputs. Is there a a way to accomplish the opposite — ie have the output have the length of the longer of the two inputs, with the shorter channel either padded with silence or, I guess, just being shorter?

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timebase mismatch error
by parth awasthi 22 Dec '17

22 Dec '17

hey, ffmpeg code is showing me time base mismatch error from last 5 days. I have not found any solution for this error on internet. My timebase got stuck at 25 fps and no videos other than 25 fps is converting. Thank you in advance. This is my code -: exec("$ffmpeg -i $target_file -c:v libx264 -b:v 54k -pass 1 -r 15 -b:a 128k -f mp4 -y NUL && ^",$result); exec("$ffmpeg -i $target_file -c:v libx264 -b:v 54k -pass 2 -r 15 -b:a 128k 2>&1 $firstMp4", $result); This is an error-: [libx264 @ 000000ca0bd02480] timebase mismatch with 1st pass (1/15 vs 1/25)'

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FFMPEG's MXF container support for DNxHR with Avid MC?
by Alan Moore 22 Dec '17

22 Dec '17

All, What should I expect in terms of being able to record or playback DNxHR in MXF containers using FFMPEG and the compatibility of those clips in major NLEs? I see some support in FFMPEG source code (version 3.4) for DNxHR profiles in libavcodec but the libavformat MXF read/write code doesn't seem to have support for the new HR VCIDs (1270 - 1274), essence elements and other MXF container meta data for DNxHR and yet... FFMPEG can read it's own MXF files created with the DNxHR encoder. I'm guessing the MXF parser can read the files by using the VCID in the *frame* header and ignores the VCID in the MXF meta data/keys... However, more importantly, the files do not "link" into Avid Media Composer (using the MXF AMA plugin.) Is this known to work for anyone else? I'm using MC version 8.8.1 (MacOS Sierra) and when I attempt to link the file there is an error message saying it cannot link but no console log entry saying exactly what it doesn't parse/like about the file. I just need to know what works and what doesn't. Thanks! Alan -- *"Whatever you can do, or dream you can do, begin it. Boldness has genius, power, and magic in it. Begin it now."* - *Goethe*

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ffserver rtsp Key 'rc_eq' not found.
by james_wjc 20 Dec '17

20 Dec '17

RTSPPort 10899 HTTPPort 10999 RTSPBindAddress 0.0.0.0 HTTPBindAddress 0.0.0.0 MaxHTTPConnections 2000 MaxClients 1000 MaxBandwidth 1000 CustomLog - <Feed feed1.ffm> File /Users/jiachengwang/workspace/rtsp/ffmpeg/bin/feed1.ffm FileMaxSize 20480K ACL allow 127.0.0.1 </Feed> <Stream test.mp3> Feed feed1.ffm Format rtsp RTSPOption rtsp_transport udp RTSPOption rtsp_flags listen AudioCodec libmp3lame AudioBitRate 64 AudioChannels 1 AudioSampleRate 44100 NoVideo </Stream> ffserver -f ffserver.conf output: ffserver version N-89212-ga60b2425c3 Copyright (c) 2000-2017 the FFmpeg developers built with Apple LLVM version 9.0.0 (clang-900.0.38) configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --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-appkit --enable-avfoundation --enable-coreimage --enable-audiotoolbox libavutil 56. 2.100 / 56. 2.100 libavcodec 58. 3.105 / 58. 3.105 libavformat 58. 2.102 / 58. 2.102 libavdevice 58. 0.100 / 58. 0.100 libavfilter 7. 2.102 / 7. 2.102 libswscale 5. 0.101 / 5. 0.101 libswresample 3. 0.101 / 3. 0.101 libpostproc 55. 0.100 / 55. 0.100 Wed Dec 20 11:26:11 2017 [NULL @ 0x7f8b5e812200]Key 'rc_eq' not found. Wed Dec 20 11:26:11 2017 Deleting feed file '/Users/jiachengwang/workspace/rtsp/ffmpeg/bin/feed1.ffm' as stream counts differ (2 != 1) Wed Dec 20 11:26:11 2017 [ffm @ 0x7f8b5f002000]Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead. Wed Dec 20 11:26:11 2017 FFserver started. How to set up rc_eq ? Thanks -- Sent from: http://www.ffmpeg-archive.org/

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Re: [FFmpeg-user] Stuck on adelay/amix sync problem..
by Mark Corner 19 Dec '17

19 Dec '17

Sorry, forgot to include the mailing list… See below. On December 19, 2017 at 9:16:02 AM, Mark Corner (mcorner(a)gmail.com) wrote: Ok, I tried this: ffmpeg -y -i RT2568dcb62e48c44c4ac61094c4377553 -i RTb5d9c3c0a59e98061159ddb310587b71 -i RT955f8ed764b5fda55a4dcd23cfed673d -filter_complex "[0]adelay=2542|2542[t0];[1]adelay=4880|4880[t1];[2]adelay=986042|986042[t2];[t0]asetpts=N/SR/TB[t0s];[t1]asetpts=N/SR/TB[t1s];[t2]asetpts=N/SR/TB[t2s];[t2s][t1s][t0s]amix=inputs=3" -strict -2 -c:a opus -strict -2 output_audio.mka Sadly, that doesn’t work either…. The problem is definitely here in the combining of audio (at least I know where the problem is) because if I listen to the conversation I can tell it is out of sync and they are talking on top of each other (this is a recording from a Twilio video chat). Any other ideas? Does it matter that they are different bitrates? Here is the input data from ffprobe: Input #0, matroska,webm, from 'RT2568dcb62e48c44c4ac61094c4377553': Metadata: encoder : GStreamer matroskamux version 1.8.1.1 creation_time : 2017-12-14T20:00:39.000000Z Duration: 00:16:19.11, start: 2.542000, bitrate: 38 kb/s Stream #0:0(eng): Audio: opus, 48000 Hz, stereo, fltp (default) Metadata: title : Audio Input #0, matroska,webm, from 'RTb5d9c3c0a59e98061159ddb310587b71': Metadata: encoder : GStreamer matroskamux version 1.8.1.1 creation_time : 2017-12-14T20:00:41.000000Z Duration: 00:32:49.71, start: 4.880000, bitrate: 41 kb/s Stream #0:0(eng): Audio: opus, 48000 Hz, stereo, fltp (default) Metadata: title : Audio Input #0, matroska,webm, from 'RT955f8ed764b5fda55a4dcd23cfed673d': Metadata: encoder : GStreamer matroskamux version 1.8.1.1 creation_time : 2017-12-14T20:17:02.000000Z Duration: 00:16:30.87, start: 986.042000, bitrate: 36 kb/s Stream #0:0(eng): Audio: opus, 48000 Hz, stereo, fltp (default) Metadata: title : Audio On December 19, 2017 at 7:35:38 AM, Paul B Mahol (onemda(a)gmail.com) wrote: On 12/19/17, Mark Corner <mcorner(a)gmail.com> wrote: > Thanks! I tried this at least: > > ffmpeg -y -i RT2568dcb62e48c44c4ac61094c4377553 -i > RTb5d9c3c0a59e98061159ddb310587b71 -i RT955f8ed764b5fda55a4dcd23cfed673d > -filter_complex > "[0]adelay=25|25[t0];[1]adelay=2363|2363[t1];[2]adelay=983525|983525[t2];[t0]asetpts=PTS-STARTPTS[t0s];[t1]asetpts=PTS-STARTPTS[t1s];[t2]asetpts=PTS-STARTPTS[t2s];[t2s][t1s][t0s]amix=inputs=3" PTS-STARTPTS will not fix your issue asetpts=N/SR/TB might. > -strict -2 -c:a opus -strict -2 output_audio.mka > > And that still has the same problem of out of sync audio. > > I do get lots of these kinds of warnings: > [opus @ 0x7f8855019400] Queue input is backward in timespeed=18.1x > > > Just to demonstrate one of my points, these two commands are identical, > expect the order of streams to amix. They produce results that are 20 > seconds different in length! This happens with asetpts enabled as well. > > t0 first: > ffmpeg -y -i RT2568dcb62e48c44c4ac61094c4377553 -i > RTb5d9c3c0a59e98061159ddb310587b71 -filter_complex > "[0]adelay=25|25[t0];[1]adelay=2363|2363[t1];[t0][t1]amix=inputs=2" -c:a > opus -strict -2 output_audio2.mka > > size= 23881kB time=00:33:11.14 bitrate= 98.3kbits/s speed=20.5x > video:0kB audio:23198kB subtitle:0kB other streams:0kB global headers:0kB > muxing overhead: 2.946144% > [opus @ 0x7fb674801800] Average Intensity Stereo band: 0.7 > [opus @ 0x7fb674801800] Dual Stereo used: 0.00% > > t1 first: > ffmpeg -y -i RT2568dcb62e48c44c4ac61094c4377553 -i > RTb5d9c3c0a59e98061159ddb310587b71 -filter_complex > "[0]adelay=25|25[t0];[1]adelay=2363|2363[t1];[t1][t0]amix=inputs=2" -c:a > opus -strict -2 output_audio2.mka > > size= 23881kB time=00:32:52.06 bitrate= 99.2kbits/s speed= 20x > video:0kB audio:23198kB subtitle:0kB other streams:0kB global headers:0kB > muxing overhead: 2.945853% > [opus @ 0x7fb403803000] Average Intensity Stereo band: 0.7 > [opus @ 0x7fb403803000] Dual Stereo used: 0.00% > > > > > On December 19, 2017 at 3:36:47 AM, Paul B Mahol (onemda(a)gmail.com) wrote: > > On 12/19/17, Mark Corner <mcorner(a)gmail.com> wrote: >> I am having trouble with amix producing different results with different >> input orders and only one input to amix staying in sync with the video. >> >> I am starting with three video and three audio files, vp8 and opus >> encoded. There is a video and audio that go together and are about 32 >> minutes long. Then there are two pairs of audio and video, each about 16 >> minutes long. >> >> First I combine the three videos using an overlay: >> ffmpeg -y -itsoffset 0 -i >> > /tmp/RM359a097bf459c327e34954525862d2b7/RTd9075ac090a32a21e71269e6b08f4053-reproces >> sed.mkv -itsoffset 2.364 -i >> > /tmp/RM359a097bf459c327e34954525862d2b7/RT21a59c19d91a7644b8c76d3fd12c3b32-reproc >> essed.mkv -itsoffset 983.503 -i >> > /tmp/RM359a097bf459c327e34954525862d2b7/RT4d34aa2bae28a73f8079a6e1b65b3aa4-re >> processed.mkv -filter_complex >> "[0]pad=iw*2:ih[t0];[t0][1]overlay=W/2:0[t1];[t1][2]overlay=0:0[t2]" -map >> [t2] >> -an -c:v libvpx -crf 23 -b:v 1M -cpu-used 3 -threads 8 >> /tmp/RM359a097bf459c327e34954525862d2b7/output_video. >> webm >> >> That works fine. >> >> Then I combine the audio (this is where the problem is): >> ffmpeg -y -i >> /tmp/RM359a097bf459c327e34954525862d2b7/RT2568dcb62e48c44c4ac61094c4377553 >> -i /tmp/RM359a097bf45 >> 9c327e34954525862d2b7/RTb5d9c3c0a59e98061159ddb310587b71 -i >> /tmp/RM359a097bf459c327e34954525862d2b7/RT955f8ed >> 764b5fda55a4dcd23cfed673d -filter_complex >> "[0]adelay=25|25[t0];[1]adelay=2363|2363[t1];[2]adelay=983525|9835 >> 25[t2];[t2][t1][t0]amix=inputs=3" -strict -2 -c:a libopus >> /tmp/RM359a097bf459c327e34954525862d2b7/output_audi >> o.mka >> >> And then merge the audio and video together: >> ffmpeg -y -i /tmp/RM359a097bf459c327e34954525862d2b7/output_audio.mka -i >> /tmp/RM359a097bf459c327e34954525862d2b7/output_video.webm -c:v copy -c:a >> copy /tmp/RM359a097bf459c327e34954525862d2b7/full.webm >> >> The problem is that the audio and video are only in sync for what ever is >> first in the amix filter.... So in the above example, the t2 audio input >> will be in sync, but not the t1 or t0. If I change the order to something >> like [t1][t2][t0], then the t1 audio is in sync, but none of the others. >> In fact the length of the output_audio.mka will differ based on the order >> of inputs to amix.... As far as I understand, the order to amix shouldn't >> matter at all. >> >> Thanks for any advice you can offer. > > Perhaps you need to use asetpts filter after each adelay to reset pts > to sane value? > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user(a)ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > To unsubscribe, visit link above, or email > ffmpeg-user-request(a)ffmpeg.org with subject "unsubscribe". >

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Stuck on adelay/amix sync problem..
by Mark Corner 19 Dec '17

19 Dec '17

I am having trouble with amix producing different results with different input orders and only one input to amix staying in sync with the video. I am starting with three video and three audio files, vp8 and opus encoded. There is a video and audio that go together and are about 32 minutes long. Then there are two pairs of audio and video, each about 16 minutes long. First I combine the three videos using an overlay: ffmpeg -y -itsoffset 0 -i /tmp/RM359a097bf459c327e34954525862d2b7/RTd9075ac090a32a21e71269e6b08f4053-reproces sed.mkv -itsoffset 2.364 -i /tmp/RM359a097bf459c327e34954525862d2b7/RT21a59c19d91a7644b8c76d3fd12c3b32-reproc essed.mkv -itsoffset 983.503 -i /tmp/RM359a097bf459c327e34954525862d2b7/RT4d34aa2bae28a73f8079a6e1b65b3aa4-re processed.mkv -filter_complex "[0]pad=iw*2:ih[t0];[t0][1]overlay=W/2:0[t1];[t1][2]overlay=0:0[t2]" -map [t2] -an -c:v libvpx -crf 23 -b:v 1M -cpu-used 3 -threads 8 /tmp/RM359a097bf459c327e34954525862d2b7/output_video. webm That works fine. Then I combine the audio (this is where the problem is): ffmpeg -y -i /tmp/RM359a097bf459c327e34954525862d2b7/RT2568dcb62e48c44c4ac61094c4377553 -i /tmp/RM359a097bf45 9c327e34954525862d2b7/RTb5d9c3c0a59e98061159ddb310587b71 -i /tmp/RM359a097bf459c327e34954525862d2b7/RT955f8ed 764b5fda55a4dcd23cfed673d -filter_complex "[0]adelay=25|25[t0];[1]adelay=2363|2363[t1];[2]adelay=983525|9835 25[t2];[t2][t1][t0]amix=inputs=3" -strict -2 -c:a libopus /tmp/RM359a097bf459c327e34954525862d2b7/output_audi o.mka And then merge the audio and video together: ffmpeg -y -i /tmp/RM359a097bf459c327e34954525862d2b7/output_audio.mka -i /tmp/RM359a097bf459c327e34954525862d2b7/output_video.webm -c:v copy -c:a copy /tmp/RM359a097bf459c327e34954525862d2b7/full.webm The problem is that the audio and video are only in sync for what ever is first in the amix filter…. So in the above example, the t2 audio input will be in sync, but not the t1 or t0. If I change the order to something like [t1][t2][t0], then the t1 audio is in sync, but none of the others. In fact the length of the output_audio.mka will differ based on the order of inputs to amix…. As far as I understand, the order to amix shouldn’t matter at all. Thanks for any advice you can offer.

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Bitrate won't change
by Michael Murphy 18 Dec '17

18 Dec '17

Fairly new user here trying to stream an IP camera from a Rasberry Pi3 (Debian) to Youtube for live view. This works but crashes as the audio and video bitrate do not change and are too high. Here's my command line: ffmpeg -f lavfi -i anullsrc -rtsp_transport tcp -thread_queue_size 512 -i rtsp://admin:password@xx.xx.xx.xxx/544/h264/ch1/main/av_stream -tune zerolatency -vcodec libx264 -preset slower -x264opts bitrate=2500:vbv-maxrate=2500:vbv-bufsize=166 -g 120 -pix_fmt yuv420p+ -c:v copy -c:a aac -strict experimental -b:a 128k -f flv rtmp://a.rtmp.youtube.com/live2/my_youtube_key Once launched terminal shows stream as Frame= 69 fps= 68 q=-1.0 size= 1043kb time=00:00:02.09 bitrate=4086.9kbits/ Youtube reports: The audio stream's current bitrate (2.00 Kbps) is lower than the recommended bitrate. We recommend that you use an audio stream bitrate of 128 Kbps. The stream's current bitrate (4001.00 Kbps) is higher than the recommended bitrate. We recommend that you use a stream bitrate of 2500 Kbps. Mike Murphy - Supervisor uOttawa Science Electronics Shop 009-140 Louis Pasteur Priv Ottawa, ON K1N 6N5 (613)562-5800 ext.6745 mmurphy(a)uottawa.ca<mailto:mmurphy@uottawa.ca> [uOttawa]

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Live sound in Visual C++
by Peadar Ó'Colmáin 16 Dec '17

16 Dec '17

Hello all, I am trying to write a live video and sound system in visual C++ using ffmpeg. I have the video compress and decompress working ok but I can't get the sound to work. I am using a demonstration program by Fabrice Ballard and I am trying to get the voice data from MM_WIM_DATA to go into the relevant field. I have tried the line below and many others but none is correct. FabriceFrame->data[0] = (uint8_t *)pHdr->lpData Has anyone out there ever gotten the compressor to work with windows sound? I have it working with the older command (see below). That 's obsolete but the real problem with it is that it only works with obsolete C++ commands and then the sound can only go in one direction. MacPacket.data = inbuf; MacPacket.size = avcodec_encode_audio(EncodeContext, MacPacket.data, AllocOutputSize, (short *)rawData); Anyway I'd much appreciate any feedback and I'll certainly share anything that I find out. I can post a longer code sample if that's any help but it is just the Fabrice code word-for-word. Thanks in advance for any help Peadar --- This email has been checked for viruses by AVG. http://www.avg.com

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Encoding 4K 60Hz lossless from a capture card
by Alex Pizzi 15 Dec '17

15 Dec '17

Windows 10 64-bit Ryzen 7 GTX 1080 32GB RAM Hi all, I'm trying to encode 4K 30/60Hz video in a lossless format from a 4K capture card and everything I've tried gives me a similar error as in the linked image, [real-time buffer too full or near too full frame dropped] https://cloud.githubusercontent.com/assets/4932401/22171307/ef5c9864-df58-1… This is the command I've tried most recently: ffmpeg.exe -f dshow -video_size 3840x2160 -framerate 30 -pixel_format bgr24 -rtbufsize INT_MAX -i video="MZ0380 PCI, Analog 01 Capture" -vf fps=30 out%d.BMP With the images dumped to a 10G RAM disk or 850 EVO. I'm doing this to skip the encoding step. I get the same error when encoding with h265 lossless and NVENC h265 lossless. I need the video to be lossless as it will be used to test hardware h265 encoders. Video source is a 4K Blu-ray. Any help would be greatly appreciated. Thank you. -Alex P

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