Audio converting and muxing error/warning messages

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Audio converting and muxing error/warning messages

Felix Muster
Hello,


I need to convert different audio streams with ffmpeg (v4.0.3-win64).

But there are several error/warning messages I need to handle.

Here are some example code snippets:

mkvmerge -i german_dd_to_alac__english_truehd_to_alac.mkv
mkvextract tracks german_dd_to_alac__english_truehd_to_alac.mkv 0:video.h264 1:audio1.ac3 2:audio2.truehd 3:audio2_core.ac3 4:sub1.srt
ffmpeg -i german_dd_to_alac__english_truehd_to_alac.mkv -f ffmetadata chapters
ffmpeg -i audio1.ac3 -acodec alac audio1.m4a
[ac3 @ 0000000000502b40] Estimating duration from bitrate, this may be inaccurate
[alac @ 000000000050cc40] encoding as 24 bits-per-sample
ffmpeg -i audio2.truehd -acodec alac -af "aformat=channel_layouts=7.1(wide)" audio2.m4a
[out_0_0 @ 000000000046bc40] 100 buffers queued in out_0_0, something may be wrong.

ffmpeg^
 -i video.h264 -i audio1_alac.m4a -i audio2_alac.m4a -i sub1.srt -i chapters^
 -map 0:0 -map 1:0 -map 2:0 -map 3:0^
 -metadata:s:a:0 language=ger -metadata:s:a:0 handler="Dolby Digital"^
 -metadata:s:a:1 language=eng -metadata:s:a:1 handler="Dolby TrueHD"^
 -metadata:s:s:0 language=ger -metadata:s:s:0 handler="Deutsch"^
 -movflags disable_chpl^
 -c:s mov_text -c:v copy -c:a copy^
 german_dd_to_alac__english_truehd_to_alac.m4v 2>> ffmpeg.log
[ipod @ 000000000338c040] track 1: codec frame size is not set
[ipod @ 000000000338c040] track 2: codec frame size is not set
[ipod @ 000000000338c040] 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
[ipod @ 000000000338c040] pts has no value


_

mkvmerge -i german_dts_to_alac__english_dtshdma_to_alac.mkv
mkvextract tracks german_dts_to_alac__english_dtshdma_to_alac.mkv 0:video.h264 1:audio1.dts 2:audio2.dtshdma 3:sub1.srt
ffmpeg -i german_dts_to_alac__english_dtshdma_to_alac.mkv -f ffmetadata chapters
ffmpeg -i audio1.dts -acodec alac audio1.m4a
[dts @ 0000000000332a80] Estimating duration from bitrate, this may be inaccurate
[alac @ 0000000000339d80] encoding as 24 bits-per-sample
ffmpeg -i audio2.dtshdma -acodec alac audio2.m4a

ffmpeg^
 -i video.h264 -i audio1_alac.m4a -i audio2_alac.m4a -i sub1.srt -i chapters^
 -map 0:0 -map 1:0 -map 2:0 -map 3:0^
 -metadata:s:a:0 language=ger -metadata:s:a:0 handler="DTS"^
 -metadata:s:a:1 language=eng -metadata:s:a:1 handler="DTS-HD Master Audio"^
 -metadata:s:s:0 language=ger -metadata:s:s:0 handler="Deutsch"^
 -movflags disable_chpl^
 -c:s mov_text -c:v copy -c:a copy^
 german_dts_to_alac__english_dtshdma_to_alac.m4v 2>> ffmpeg.log
[ipod @ 0000000002afe740] track 1: codec frame size is not set
[ipod @ 0000000002afe740] track 2: codec frame size is not set
[ipod @ 0000000002afe740] 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
[ipod @ 0000000002afe740] pts has no value


_

mkvmerge -i german_dtshr_to_alac__english_dtshdma_to_alac.mkv
mkvextract tracks german_dtshr_to_alac__english_dtshdma_to_alac.mkv 0:video.h264 1:audio1.dtshr 2:audio2.dtshdma
ffmpeg -i german_dtshr_to_alac__english_dtshdma_to_alac.mkv -f ffmetadata chapters
ffmpeg -i audio1.dtshr -acodec alac audio1.m4a
[alac @ 00000000005a4dc0] encoding as 24 bits-per-sample
ffmpeg -i audio2.dtshdma -acodec alac -af "aformat=channel_layouts=7.1(wide)" audio2.m4a

ffmpeg^
 -i video.h264 -i audio1_alac.m4a -i audio2_alac.m4a -i chapters^
 -map 0:0 -map 1:0 -map 2:0^
 -metadata:s:a:0 language=ger -metadata:s:a:0 handler="DTS-HD High Resolution"^
 -metadata:s:a:1 language=eng -metadata:s:a:1 handler="DTS-HD Master Audio"^
 -movflags disable_chpl^
 -c:v copy -c:a copy^
 german_dtshr_to_alac__english_dtshdma_to_alac.m4v 2>> ffmpeg.log
[ipod @ 00000000004f3780] track 1: codec frame size is not set
[ipod @ 00000000004f3780] track 2: codec frame size is not set
[ipod @ 00000000004f3780] 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
[ipod @ 00000000004f3780] pts has no value


_


[ac3 @ 0000000000502b40] Estimating duration from bitrate, this may be inaccurate


Only pops up when I'm trying to convert the following lossy formats: ac3, dts and eac3. With lossy dts (as DTS-HD High Resolution) I'm fine.


[out_0_0 @ 000000000046bc40] 100 buffers queued in out_0_0, something may be wrong.


Only pops up when I'm trying to convert lossless TrueHD, lossless DTS-HD Master Audio is fine.


[ipod @ 000000000338c040] track 1: codec frame size is not set
[ipod @ 000000000338c040] track 2: codec frame size is not set
[ipod @ 000000000338c040] 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
[ipod @ 000000000338c040] pts has no value


Those messages are always present when I try to mux all streams.

Will they disappear when I put timestamps into pts-format in the container? I can extract them from mkv (mkvextract timecodes_v2 input.mkv 0:video.timecodes.txt 1:audio1.timecodes.txt 2:audio2.timecodes.txt 3:sub1.timecodes.txt 4:sub2.timecodes.txt 5:sub3.timecodes.txt 6:sub4.timecodes.txt) but only in timecodes_v2-format. Can I convert timecodes_v2 in pts and put them into the container to avoid those messages?




Here are the logs (for first snippet): https://pastebin.com/xWq9U2sN


(The message from the pastebin-logs [h264 @ 00000000005037c0] Stream #0: not enough frames to estimate rate; consider increasing probesize I could fix with -probesize 2147483648)





Best regards,

Felix
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Re: Audio converting and muxing error/warning messages

Carl Eugen Hoyos-2
2018-12-17 10:04 GMT+01:00, Felix Muster <[hidden email]>:

> I need to convert different audio streams with ffmpeg (v4.0.3-win64).
>
> But there are several error/warning messages I need to handle.
>
> Here are some example code snippets:

Code snippets unfortunately are not helpful, if you need support
here, please test current FFmpeg git head and provide the
command line you tested together with its complete, uncut
console output here on the mailing list.

[...]

> [ac3 @ 0000000000502b40] Estimating duration from bitrate, this may be
> inaccurate
>
> Only pops up when I'm trying to convert the following lossy formats: ac3,
> dts and eac3.

These have bitrates, variable bitrate is unusual for them but theoretically
possible. (The warning is correct.)

> With lossy dts (as DTS-HD High Resolution) I'm fine.

This has no bitrate and therefore no duration can be estimated.

Carl Eugen
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Re: Audio converting and muxing error/warning messages

Felix Muster
Here are the complete console outputs from latest build.
(I cut it by [...] because there were thousand lines of the same message. Don't think that it would be useful)

ffmpeg -i audio1.ac3 -acodec alac audio1.m4a
ffmpeg version N-92752-g16ec62bbf4 Copyright (c) 2000-2018 the FFmpeg developers
  built with gcc 8.2.1 (GCC) 20181201
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --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. 24.101 / 56. 24.101
  libavcodec     58. 42.104 / 58. 42.104
  libavformat    58. 24.101 / 58. 24.101
  libavdevice    58.  6.101 / 58.  6.101
  libavfilter     7. 46.101 /  7. 46.101
  libswscale      5.  4.100 /  5.  4.100
  libswresample   3.  4.100 /  3.  4.100
  libpostproc    55.  4.100 / 55.  4.100
[ac3 @ 0000000000392ac0] Estimating duration from bitrate, this may be inaccurate
Input #0, ac3, from 'audio1.ac3':
  Duration: 02:24:29.54, start: 0.000000, bitrate: 640 kb/s
    Stream #0:0: Audio: ac3, 48000 Hz, 5.1(side), fltp, 640 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (ac3 (native) -> alac (native))
Press [q] to stop, [?] for help
[alac @ 000000000039cbc0] encoding as 24 bits-per-sample
Output #0, ipod, to 'audio1.m4a':
  Metadata:
    encoder         : Lavf58.24.101
    Stream #0:0: Audio: alac (alac / 0x63616C61), 48000 Hz, 5.1, s32p (24 bit), 128 kb/s
    Metadata:
      encoder         : Lavc58.42.104 alac
size=   11264kB time=00:00:29.95 bitrate=3080.8kbits/s speed=59.9x    
[...]  
size= 3755284kB time=02:24:29.61 bitrate=3548.4kbits/s speed=47.2x    
video:0kB audio:3754850kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.011574%

I read about "Estimating duration from bitrate, this may be inaccurate" that I have to decode the stream to get the exact duration and that it wouldn't be necessary if I put the stream in a container (because the correct duration will be embedded in the container).
So I don't need to worry about that. Because by converting to alac I put the stream into a container and while muxing it to the m4v-file I don't get such a message.
But something is confusing me.
 When I do the following to get the correct duration: ffmpeg -i audio1.ac3 -f null -
The duration is:  02:24:29.53
So I have three different durations:
1. ffprobe tells me about the untouched ac3-file: 02:24:29.54 [Estimating duration from bitrate, this may be inaccurate]
2. After encoding to alac: 02:24:29.61
3. And after decoding:  02:24:29.53


Next, the 100 buffers queued in out_0_0-problem. Still don't have a clue about that.

ffmpeg -i audio2.truehd -acodec alac -af "aformat=channel_layouts=7.1(wide)" audio2.m4a
ffmpeg version N-92752-g16ec62bbf4 Copyright (c) 2000-2018 the FFmpeg developers
  built with gcc 8.2.1 (GCC) 20181201
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --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. 24.101 / 56. 24.101
  libavcodec     58. 42.104 / 58. 42.104
  libavformat    58. 24.101 / 58. 24.101
  libavdevice    58.  6.101 / 58.  6.101
  libavfilter     7. 46.101 /  7. 46.101
  libswscale      5.  4.100 /  5.  4.100
  libswresample   3.  4.100 /  3.  4.100
  libpostproc    55.  4.100 / 55.  4.100
Input #0, truehd, from 'audio2.truehd':
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Audio: truehd, 48000 Hz, 7.1, s32 (24 bit)
Stream mapping:
  Stream #0:0 -> #0:0 (truehd (native) -> alac (native))
Press [q] to stop, [?] for help
Output #0, ipod, to 'audio2.m4a':
  Metadata:
    encoder         : Lavf58.24.101
    Stream #0:0: Audio: alac (alac / 0x63616C61), 48000 Hz, 7.1(wide), s32p (24 bit), 128 kb/s
    Metadata:
      encoder         : Lavc58.42.104 alac
[out_0_0 @ 000000000048bcc0] 100 buffers queued in out_0_0, something may be wrong.
size=    5888kB time=00:00:13.31 bitrate=3623.4kbits/s speed=26.6x    
[...]
size= 4511186kB time=02:24:29.61 bitrate=4262.7kbits/s speed=21.7x    
video:0kB audio:4510732kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.010070%


I'm also no step further with the frame size and timestamps are not set and pts has no value problem.

ffmpeg^
 -probesize 2147483648 -i video.h264 -i audio1.m4a -i audio2.m4a -i sub1.srt -i chapters^
 -map 0:0 -map 1:0 -map 2:0 -map 3:0^
 -metadata:s:a:0 language=ger -metadata:s:a:0 handler="Dolby Digital"^
 -metadata:s:a:1 language=eng -metadata:s:a:1 handler="Dolby TrueHD"^
 -metadata:s:s:0 language=ger -metadata:s:s:0 handler="Deutsch"^
 -movflags disable_chpl^
 -c:s mov_text -c:v copy -c:a copy^
 german_dd_to_alac__english_truehd_to_alac.m4v
ffmpeg version N-92752-g16ec62bbf4 Copyright (c) 2000-2018 the FFmpeg developers
  built with gcc 8.2.1 (GCC) 20181201
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --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. 24.101 / 56. 24.101
  libavcodec     58. 42.104 / 58. 42.104
  libavformat    58. 24.101 / 58. 24.101
  libavdevice    58.  6.101 / 58.  6.101
  libavfilter     7. 46.101 /  7. 46.101
  libswscale      5.  4.100 /  5.  4.100
  libswresample   3.  4.100 /  3.  4.100
  libpostproc    55.  4.100 / 55.  4.100
Input #0, h264, from 'video.h264':
  Duration: N/A, bitrate: N/A
    Stream #0:0: Video: h264 (High), yuv420p(progressive), 1920x1080 [SAR 1:1 DAR 16:9], 24 fps, 23.98 tbr, 1200k tbn, 47.95 tbc
Input #1, mov,mp4,m4a,3gp,3g2,mj2, from 'audio1.m4a':
  Metadata:
    major_brand     : M4A
    minor_version   : 512
    compatible_brands: isomiso2
    encoder         : Lavf58.24.101
  Duration: 02:24:29.54, start: 0.000000, bitrate: 3548 kb/s
    Stream #1:0(und): Audio: alac (alac / 0x63616C61), 48000 Hz, 5.1, s32p (24 bit), 3548 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
Input #2, mov,mp4,m4a,3gp,3g2,mj2, from 'audio2.m4a':
  Metadata:
    major_brand     : M4A
    minor_version   : 512
    compatible_brands: isomiso2
    encoder         : Lavf58.24.101
  Duration: 02:24:29.54, start: 0.000000, bitrate: 4262 kb/s
    Stream #2:0(und): Audio: alac (alac / 0x63616C61), 48000 Hz, 7.1(wide), s32p (24 bit), 4262 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
Input #3, srt, from 'sub1.srt':
  Duration: N/A, bitrate: N/A
    Stream #3:0: Subtitle: subrip
Input #4, ffmetadata, from 'chapters':
  Metadata:
    encoder         : Lavf58.12.100
  Duration: 02:24:29.54, start: 0.000000, bitrate: 0 kb/s
    Chapter #4:0: start 0.000000, end 287.662000
    Metadata:
      title           : 00:00:00.000
    Chapter #4:1: start 287.662000, end 541.541000
    Metadata:
      title           : 00:04:47.662
    Chapter #4:2: start 541.541000, end 931.556000
    Metadata:
      title           : 00:09:01.541
    Chapter #4:3: start 931.556000, end 1272.229000
    Metadata:
      title           : 00:15:31.556
    Chapter #4:4: start 1272.229000, end 1608.774000
    Metadata:
      title           : 00:21:12.229
    Chapter #4:5: start 1608.774000, end 1882.756000
    Metadata:
      title           : 00:26:48.774
    Chapter #4:6: start 1882.756000, end 2060.058000
    Metadata:
      title           : 00:31:22.756
    Chapter #4:7: start 2060.058000, end 2376.124000
    Metadata:
      title           : 00:34:20.058
    Chapter #4:8: start 2376.124000, end 2832.872000
    Metadata:
      title           : 00:39:36.124
    Chapter #4:9: start 2832.872000, end 3239.027000
    Metadata:
      title           : 00:47:12.872
    Chapter #4:10: start 3239.027000, end 3747.744000
    Metadata:
      title           : 00:53:59.027
    Chapter #4:11: start 3747.744000, end 4203.324000
    Metadata:
      title           : 01:02:27.744
    Chapter #4:12: start 4203.324000, end 4982.477000
    Metadata:
      title           : 01:10:03.324
    Chapter #4:13: start 4982.477000, end 5231.101000
    Metadata:
      title           : 01:23:02.477
    Chapter #4:14: start 5231.101000, end 5690.685000
    Metadata:
      title           : 01:27:11.101
    Chapter #4:15: start 5690.685000, end 6156.359000
    Metadata:
      title           : 01:34:50.685
    Chapter #4:16: start 6156.359000, end 6488.148000
    Metadata:
      title           : 01:42:36.359
    Chapter #4:17: start 6488.148000, end 6827.279000
    Metadata:
      title           : 01:48:08.148
    Chapter #4:18: start 6827.279000, end 7222.882000
    Metadata:
      title           : 01:53:47.279
    Chapter #4:19: start 7222.882000, end 7639.465000
    Metadata:
      title           : 02:00:22.882
    Chapter #4:20: start 7639.465000, end 8305.422000
    Metadata:
      title           : 02:07:19.465
    Chapter #4:21: start 8305.422000, end 8669.537000
    Metadata:
      title           : 02:18:25.422
[ipod @ 000000000057c600] track 1: codec frame size is not set
[ipod @ 000000000057c600] track 2: codec frame size is not set
Output #0, ipod, to 'german_dd_to_alac__english_truehd_to_alac.m4v':
  Metadata:
    encoder         : Lavf58.24.101
    Chapter #0:0: start 0.000000, end 287.662000
    Metadata:
      title           : 00:00:00.000
    Chapter #0:1: start 287.662000, end 541.541000
    Metadata:
      title           : 00:04:47.662
    Chapter #0:2: start 541.541000, end 931.556000
    Metadata:
      title           : 00:09:01.541
    Chapter #0:3: start 931.556000, end 1272.229000
    Metadata:
      title           : 00:15:31.556
    Chapter #0:4: start 1272.229000, end 1608.774000
    Metadata:
      title           : 00:21:12.229
    Chapter #0:5: start 1608.774000, end 1882.756000
    Metadata:
      title           : 00:26:48.774
    Chapter #0:6: start 1882.756000, end 2060.058000
    Metadata:
      title           : 00:31:22.756
    Chapter #0:7: start 2060.058000, end 2376.124000
    Metadata:
      title           : 00:34:20.058
    Chapter #0:8: start 2376.124000, end 2832.872000
    Metadata:
      title           : 00:39:36.124
    Chapter #0:9: start 2832.872000, end 3239.027000
    Metadata:
      title           : 00:47:12.872
    Chapter #0:10: start 3239.027000, end 3747.744000
    Metadata:
      title           : 00:53:59.027
    Chapter #0:11: start 3747.744000, end 4203.324000
    Metadata:
      title           : 01:02:27.744
    Chapter #0:12: start 4203.324000, end 4982.477000
    Metadata:
      title           : 01:10:03.324
    Chapter #0:13: start 4982.477000, end 5231.101000
    Metadata:
      title           : 01:23:02.477
    Chapter #0:14: start 5231.101000, end 5690.685000
    Metadata:
      title           : 01:27:11.101
    Chapter #0:15: start 5690.685000, end 6156.359000
    Metadata:
      title           : 01:34:50.685
    Chapter #0:16: start 6156.359000, end 6488.148000
    Metadata:
      title           : 01:42:36.359
    Chapter #0:17: start 6488.148000, end 6827.279000
    Metadata:
      title           : 01:48:08.148
    Chapter #0:18: start 6827.279000, end 7222.882000
    Metadata:
      title           : 01:53:47.279
    Chapter #0:19: start 7222.882000, end 7639.465000
    Metadata:
      title           : 02:00:22.882
    Chapter #0:20: start 7639.465000, end 8305.422000
    Metadata:
      title           : 02:07:19.465
    Chapter #0:21: start 8305.422000, end 8669.537000
    Metadata:
      title           : 02:18:25.422
    Stream #0:0: Video: h264 (High) (avc1 / 0x31637661), yuv420p(progressive), 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 24 fps, 23.98 tbr, 1200k tbn, 1200k tbc
    Stream #0:1(ger): Audio: alac (alac / 0x63616C61), 48000 Hz, 5.1, s32p (24 bit), 3548 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
      handler         : Dolby Digital
    Stream #0:2(eng): Audio: alac (alac / 0x63616C61), 48000 Hz, 7.1(wide), s32p (24 bit), 4262 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
      handler         : Dolby TrueHD
    Stream #0:3(ger): Subtitle: mov_text (tx3g / 0x67337874)
    Metadata:
      handler         : Deutsch
      encoder         : Lavc58.42.104 mov_text
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #1:0 -> #0:1 (copy)
  Stream #2:0 -> #0:2 (copy)
  Stream #3:0 -> #0:3 (subrip (srt) -> mov_text (native))
Press [q] to stop, [?] for help
[ipod @ 000000000057c600] 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
[ipod @ 000000000057c600] pts has no value
    Last message repeated 226 times
[...]
frame=207784 fps= 76 q=-1.0 size=36347393kB time=02:24:26.11 bitrate=34358.9kbits/s speed=3.16x    
[ipod @ 000000000057c600] pts has no value
    Last message repeated 80 times
[ipod @ 000000000057c600] pts has no value
    Last message repeated 233 times
frame=207861 fps= 76 q=-1.0 Lsize=36383931kB time=02:24:29.52 bitrate=34379.9kbits/s speed=3.16x    
video:28112901kB audio:8265581kB subtitle:3kB other streams:0kB global headers:0kB muxing overhead: 0.014969%

I tried to import MKVTIMESTAMP_V2 from mkv with -i. Without success.

-----Ursprüngliche Nachricht-----
Von: ffmpeg-user <[hidden email]> Im Auftrag von Carl Eugen Hoyos
Gesendet: Montag, 17. Dezember 2018 15:10
An: FFmpeg user questions <[hidden email]>
Betreff: Re: [FFmpeg-user] Audio converting and muxing error/warning messages

2018-12-17 10:04 GMT+01:00, Felix Muster <[hidden email]>:

> I need to convert different audio streams with ffmpeg (v4.0.3-win64).
>
> But there are several error/warning messages I need to handle.
>
> Here are some example code snippets:

Code snippets unfortunately are not helpful, if you need support here, please test current FFmpeg git head and provide the command line you tested together with its complete, uncut console output here on the mailing list.

[...]

> [ac3 @ 0000000000502b40] Estimating duration from bitrate, this may be
> inaccurate
>
> Only pops up when I'm trying to convert the following lossy formats:
> ac3, dts and eac3.

These have bitrates, variable bitrate is unusual for them but theoretically possible. (The warning is correct.)

> With lossy dts (as DTS-HD High Resolution) I'm fine.

This has no bitrate and therefore no duration can be estimated.

Carl Eugen
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Re: Audio converting and muxing error/warning messages

Carl Eugen Hoyos-2
2018-12-21 0:19 GMT+01:00, Felix Muster <[hidden email]>:

>  When I do the following to get the correct duration:
> ffmpeg -i audio1.ac3 -f null -
> The duration is:  02:24:29.53
> So I have three different durations:

> 1. ffprobe tells me about the untouched ac3-file: 02:24:29.54
> [Estimating duration from bitrate, this may be inaccurate]

(As said, the warning is ok although you get the correct duration
because ac-3 is usually constant bit-rate, but it does not have
to be.)

> 2. After encoding to alac: 02:24:29.61

This could be considered a bug but it would be a minor one.

> 3. And after decoding:  02:24:29.53

If that stays constant with another round of encoding, I
would consider it correct.

Please avoid top-posting here, Carl Eugen
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Re: Audio converting and muxing error/warning messages

Felix Muster
Thank you. 02:24:29:53 is the correct duration.

The timestamp and pts problems are still present.

I tried to generate pts with -fflags +genpts. Without success.

I tried to generate new timestamps with -vsync drop.
Without success.


For the frame size is not set problem.

I get the frame rate:
ffprobe -v error -select_streams v:0 -show_entries stream=avg_frame_rate -of default=noprint_wrappers=1:nokey=1 german_dd_to_alac__english_truehd_to_alac.mkv

And tried to force the frame rate of the input stream with -r 24000/1001.
Without success.


Can you help me please?
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Re: Audio converting and muxing error/warning messages

Carl Eugen Hoyos-2
2018-12-28 14:48 GMT+01:00, Felix Muster <[hidden email]>:

> The timestamp and pts problems are still present.

Please provide a command line you tested including complete,
uncut console output.

> I tried to generate pts with -fflags +genpts. Without success.
>
> I tried to generate new timestamps with -vsync drop.
> Without success.
>
> For the frame size is not set problem.
>
> I get the frame rate:
> ffprobe -v error -select_streams v:0 -show_entries stream=avg_frame_rate -of
> default=noprint_wrappers=1:nokey=1
> german_dd_to_alac__english_truehd_to_alac.mkv

> And tried to force the frame rate of the input stream with -r 24000/1001.

Don't use this functionality unless you have a very good reason.

Carl Eugen
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Re: Audio converting and muxing error/warning messages

Felix Muster

> Please provide a command line you tested including complete,
> uncut console output.
Please have a look further up in this conversation.

> Don't use this functionality unless you have a very good reason.
My reason is to avoid those messages and generate a clean output.
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Re: Audio converting and muxing error/warning messages

Felix Muster
Nobody has an idea?

I tried -fflags +genpts and -r on the raw h264 stream.

But that also didn’t work.
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Re: Audio converting and muxing error/warning messages

Felix Muster
I finally solved it.

I simply take the video-stream directly from one container to the other:
ffmpeg -i german_dd_to_alac__english_truehd_to_alac.mkv -map 0:0 -c copy german_dd_to_alac__english_truehd_to_alac.m4v

And than do my thing:
ffmpeg^
 -i german_dd_to_alac__english_truehd_to_alac.m4v -i audio1.m4a -i audio2.m4a -i sub1.srt^
 -map 0:0 -map 1:0 -map 2:0 -map 3:0^
 -metadata:s:a:0 language=ger -metadata:s:a:0 handler="Dolby Digital"^
 -metadata:s:a:1 language=eng -metadata:s:a:1 handler="Dolby TrueHD"^
 -metadata:s:s:0 language=ger -metadata:s:s:0 handler="Deutsch"^
 -movflags disable_chpl^
 -c:s mov_text -c:v copy -c:a alac^
 german_dd_to_alac__english_truehd_to_alac.m4v

Works like a charm.
No pts or time stamp errors.
I solved the frame size not set error by specifying the audio codec -c:a alac. Don’t know why but it works.

Another question.

Is it possible to get this:
ffmpeg -i audio1.ac3 -acodec alac audio1.m4a

And this:
ffmpeg -i audio2.truehd -acodec alac -af "aformat=channel_layouts=7.1(wide)" audio2.m4a

Into my command line? So that I only have 1 line instead of 3.


The „100 buffers queued in out_0_0, something may be wrong.“ error is still present.
It would be great if I could solve that too.
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