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1,000,000
27 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Demon Seed
26 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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The Four of Us are Dying
26 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Corona Radiata
26 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Lights in the Sky
26 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Head Down
26 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
Autres articles (52)
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Sur d’autres sites (11399)
-
ffmpeg creates empty mpd-file
1er avril 2021, par MacsterI'm trying to live stream a webm file using ffmpeg and Dash.js as a Player. Creating chunks with ffmpeg seems to work fine, but when want to create the manifest it says
Could not find codec parameters for stream 0 (Video: vp9, none, 640x480): unspecified pixel format Consider increasing the value for the 'analyzeduration' and 'probesize' options
and at the endOutput file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)
.

Commands I use


ffmpeg -re -r 25 -i Dash/strm.webm
 -map 0:v:0
 -pix_fmt yuv420p
 -c:v libvpx-vp9
 -s 640x480 -keyint_min 25 -g 25 
 -speed 6 -threads 8 -static-thresh 0 -max-intra-rate 300 
 -deadline realtime -lag-in-frames 0 -error-resilient 1
 -f webm_chunk 
 -header "Dash/glass_360.hdr"
 -chunk_start_index 1
 Dash\glass_360_%d.chk



Manifest


ffmpeg ^
 -f webm_dash_manifest -live 1
 -r 25
 -i Dash/glass_360.hdr
 -c copy
 -map 0
 -r 25
 -framerate 25
 -f webm_dash_manifest -live 1
 -adaptation_sets "id=0,streams=0"
 -chunk_start_index 1
 -chunk_duration_ms 1000
 -time_shift_buffer_depth 7200
 -minimum_update_period 7200
 Dash/glass_live_manifest.mpd



Manifest Output


libavutil 56. 49.100 / 56. 49.100
 libavcodec 58. 87.101 / 58. 87.101
 libavformat 58. 43.100 / 58. 43.100
 libavdevice 58. 9.103 / 58. 9.103
 libavfilter 7. 83.100 / 7. 83.100
 libswscale 5. 6.101 / 5. 6.101
 libswresample 3. 6.100 / 3. 6.100
 libpostproc 55. 6.100 / 55. 6.100
[webm_dash_manifest @ 0000015f19f2ea40] Could not find codec parameters for stream 0 (Video: vp9, none, 640x480): unspecified pixel format
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Input #0, webm_dash_manifest, from 'Dash/glass_360.hdr':
 Metadata:
 ENCODER : Lavf58.43.100
 Duration: N/A, bitrate: N/A
 Stream #0:0(eng): Video: vp9, none, 640x480, SAR 1:1 DAR 4:3, 1k tbr, 1k tbn, 1k tbc (default)
 Metadata:
 ALPHA_MODE : 1
 ENCODER : Lavc58.87.101 libvpx-vp9
 webm_dash_manifest_file_name: glass_360.hdr
 webm_dash_manifest_track_number: 1
Output #0, webm_dash_manifest, to 'stream_manifest.mpd':
 Metadata:
 encoder : Lavf58.43.100
 Stream #0:0(eng): Video: vp9, none, 640x480 [SAR 1:1 DAR 4:3], q=2-31, 1k tbr, 1k tbn, 1k tbc (default)
 Metadata:
 ALPHA_MODE : 1
 ENCODER : Lavc58.87.101 libvpx-vp9
 webm_dash_manifest_file_name: glass_360.hdr
 webm_dash_manifest_track_number: 1
Stream mapping:
 Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
frame= 0 fps=0.0 q=-1.0 Lsize= 1kB time=00:00:00.00 bitrate=N/A speed= 0x
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)



glass_live_manifest.mpd


<?xml version="1.0" encoding="UTF-8"?>

<period start="PT0S">
<adaptationset mimetype="video/webm" codecs="vp9" lang="eng" bitstreamswitching="true" subsegmentalignment="true" subsegmentstartswithsap="1">
<contentcomponent type="video"></contentcomponent>
<segmenttemplate timescale="1000" duration="1000" media="glass_$RepresentationID$_$Number$.chk" startnumber="1" initialization="glass_$RepresentationID$.hdr"></segmenttemplate>
<representation bandwidth="1000000" width="640" height="480" codecs="vp9" mimetype="video/webm" startswithsap="1"></representation>
</adaptationset>
</period>




UPDATE


I fixed the "warnings" by changing the comands like this :


ffmpeg -re -r 25 -i Dash/strm.webm
-map 0:v:0
-pix_fmt yuv420p
-c:v libvpx
-s 640x480 -keyint_min 60 -g 60 -speed 6 -tile-columns 4 -frame-parallel 1 -threads 8 -static-thresh 0 -max-intra-rate 300 -deadline realtime -lag-in-frames 0 -error-resilient 1
-b:v 3000k
-f webm_chunk
-header "Dash/glass_360.hdr"
-chunk_start_index 1 Dash/glass_360_%d.chk
-map 0:a:0
-c:a libvorbis
-b:a 128k -ar 44100
-f webm_chunk
-audio_chunk_duration 2000
-header Dash/glass_171.hdr
-chunk_start_index 1 Dash/glass_171_%d.chk

ffmpeg
-f webm_dash_manifest -live 1
-i Dash/glass_360.hdr
-f webm_dash_manifest -live 1
-i Dash/glass_171.hdr
-c copy
-map 0 -map 1
-f webm_dash_manifest -live 1
-adaptation_sets "id=0,streams=0 id=1,streams=1"
-chunk_start_index 1
-chunk_duration_ms 2000
-time_shift_buffer_depth 7200
-minimum_update_period 7200 Dash/glass_video_manifest.mpd



However, it's still not working.


-
Algorithm when recording SegmentTimeline
24 mars 2021, par jgkim0518I packaged a media stream by mpeg-dash transcoded video at twice the speed and audio at normal speed.
Here is the command :


ffmpeg -re -stream_loop -1 -i timing_logic.mp4 -c:v hevc_nvenc -filter:v "setpts=2*PTS" -c:a libfdk_aac -map 0:v -map 0:a -f mpegts udp://xxx.xxx.xxx.xxx:xxxx?pkt_size=1316



The source information used here is as follows :


Input #0, mpegts, from '/home/test/timing_logic/timing_logic.mp4':
Duration: 00:00:30.22, start: 1.400000, bitrate: 16171 kb/s
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0:0[0x100]: Video: hevc (Main 10) (HEVC / 0x43564548), yuv420p10le(tv, bt709), 3840x2160 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 50 tbc
Stream #0:1[0x101](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, fltp, 128 kb/s Stream mapping:
Stream #0:0 -> #0:0 (hevc (native) -> hevc (hevc_nvenc))
Stream #0:1 -> #0:1 (mp2 (native) -> aac (libfdk_aac))
Press [q] to stop, [?] for help
frame= 0 fps=0.0 q=0.0 size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A
frame= 0 fps=0.0 q=0.0 size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A
Output #0, mpegts, to 'udp://xxx.xxx.xxx.xxx:xxxx?pkt_size=1316':
Metadata:
encoder : Lavf58.45.100
Stream #0:0: Video: hevc (hevc_nvenc) (Main 10), p010le, 3840x2160 [SAR 1:1 DAR 16:9], q=-1--1, 2000 kb/s, 50 fps, 90k tbn, 50 tbc
Metadata:
encoder : Lavc58.91.100 hevc_nvenc
Side data:
cpb: bitrate max/min/avg: 0/0/2000000 buffer size: 4000000 vbv_delay: N/A
Stream #0:1(eng): Audio: aac (libfdk_aac), 48000 Hz, stereo, s16, 139 kb/s
Metadata:
encoder : Lavc58.91.100 libfdk_aac



My shaka packager command is :


packager \ 'in=udp://xxx.xxx.xxx.xxx:xxxx,stream=video,init_segment=/home/test/timing_logic/package/video/0/video.mp4,segment_template=/home/test/timing_logic/package/video/0/$Time$.m4s' \
'in=udp://xxx.xxx.xxx.xxx:xxxx,stream=audio,init_segment=/home/test/timing_logic/package/audio/0/audio.mp4,segment_template=/home/test/timing_logic/package/audio/0/$Time$.m4a' \
--segment_duration 5 --fragment_duration 5 --minimum_update_period 5 --min_buffer_time 5 \
--preserved_segments_outside_live_window 24 --time_shift_buffer_depth 40 \
--allow_codec_switching --allow_approximate_segment_timeline --log_file_generation_deletion \
--mpd_output $OUTPUT/$output_mpd.mpd &



Looking at the mpd generated as a result of packaging, the duration of the audio is twice that of the video, and the number of segments is half.
The following is the content of mpd :


<?xml version="1.0" encoding="UTF-8"?>

<mpd xmlns="urn:mpeg:dash:schema:mpd:2011" profiles="urn:mpeg:dash:profile:isoff-live:2011" minbuffertime="PT5S" type="dynamic" publishtime="2021-03-23T10:01:57Z" availabilitystarttime="2021-03-23T09:59:55Z" minimumupdateperiod="PT5S" timeshiftbufferdepth="PT40S">
<period start="PT0S">
<adaptationset contenttype="audio" segmentalignment="true">
<representation bandwidth="140020" codecs="mp4a.40.2" mimetype="audio/mp4" audiosamplingrate="48000">
<audiochannelconfiguration schemeiduri="urn:mpeg:dash:23003:3:audio_channel_configuration:2011" value="2"></audiochannelconfiguration>
<segmenttemplate timescale="90000" initialization="/home/test/timing_logic/package/audio/0/audio.mp4" media="/home/test/timing_logic/package/audio/0/$Time$.m4a" startnumber="16">
<segmenttimeline>
<s t="6751436" d="449280"></s>
<s t="7200716" d="451200"></s>
<s t="7651916" d="449280"></s>
<s t="8101196" d="374400"></s>
<s t="8551196" d="449280"></s>
<s t="9000476" d="451200"></s>
<s t="9451674" d="449280"></s>
<s t="9900956" d="449280"></s>
<s t="10350236" d="451200"></s>
<s t="10801434" d="374400"></s>
</segmenttimeline>
</segmenttemplate>
</representation>
</adaptationset>
<adaptationset contenttype="video" width="3840" height="2160" framerate="90000/3600" segmentalignment="true" par="16:9">
<representation bandwidth="1520392" codecs="hvc1.2.4.L153" mimetype="video/mp4" sar="1:1">
<segmenttemplate timescale="90000" initialization="/home/test/timing_logic/package/video/0/video.mp4" media="/home/test/timing_logic/package/video/0/$Time$.m4s" startnumber="19">
<segmenttimeline>
<s t="16954440" d="900000" r="4"></s>
</segmenttimeline>
</segmenttemplate>
</representation>
</adaptationset>
</period>
</mpd>



When looking at the MPD, it seems that the shaka packager checks the PTS or DTS of the TS when creating a segment and recording the contents of the SegmentTimeline.
But I couldn't understand even by looking at the MPEG standard documentation and the DASH-IF documentation.


My question is whether the packager refers to PTS or DTS when creating a segment.
How are SegmentTimeline's S@t and S@d recorded ?
What algorithm is the SegmentTimeline recorded with ? Please help me. Thank you.


-
FFmpeg filter complex add audio file to video at specific points
4 juin 2020, par steveatrim=0:2
starts fine when I play the video it plays the mp3 now how do I add a working end time say finish at 9 seconds ?


Summary I want to start at 2 seconds and finish at 9 seconds.



ffmpeg -y -i "C:\Users\test\Desktop\vidz\New folder (2)\target\vaastav song .mp4" -i "C:\Users\test\Desktop\vidz\New folder (2)\target\2.mp3" -filter_complex "[0]atrim=0:2[Apre];[0]atrim=5,asetpts=PTS-STARTPTS[Apost];[Apre][1][Apost]concat=n=3:v=0:a=1" -vcodec copy -y "C:\Users\test\Desktop\vidz\New folder (2)\target\output1.mp4"




log file



ffmpeg version git-2020-04-13-59e3a9a Copyright (c) 2000-2020 the FFmpeg developers
 built with gcc 9.3.1 (GCC) 20200328
 configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --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-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
 libavutil 56. 42.102 / 56. 42.102
 libavcodec 58. 78.102 / 58. 78.102
 libavformat 58. 42.100 / 58. 42.100
 libavdevice 58. 9.103 / 58. 9.103
 libavfilter 7. 77.101 / 7. 77.101
 libswscale 5. 6.101 / 5. 6.101
 libswresample 3. 6.100 / 3. 6.100
 libpostproc 55. 6.100 / 55. 6.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\Users\test\Desktop\vidz\New folder (2)\target\vaastav song .mp4':
 Metadata:
 major_brand : mp42
 minor_version : 0
 compatible_brands: isommp42
 creation_time : 2018-09-02T04:28:46.000000Z
 Duration: 00:05:08.80, start: 0.000000, bitrate: 2289 kb/s
 Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, bt709), 1280x720 [SAR 1:1 DAR 16:9], 2160 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc (default)
 Metadata:
 creation_time : 2018-09-02T04:28:46.000000Z
 handler_name : ISO Media file produced by Google Inc. Created on: 09/01/2018.
 Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 125 kb/s (default)
 Metadata:
 creation_time : 2018-09-02T04:28:46.000000Z
 handler_name : ISO Media file produced by Google Inc. Created on: 09/01/2018.
Input #1, mp3, from 'C:\Users\test\Desktop\vidz\New folder (2)\target\2.mp3':
 Metadata:
 genre : Electronic;Indie
 title : A Distorted Noise With A Little Bit Of Sense
 artist : Lenin Was A Zombie
 encoder : Lavf56.19.100
 major_brand : mp42
 minor_version : 0
 compatible_brands: isommp42
 Duration: 00:00:54.00, start: 0.025057, bitrate: 192 kb/s
 Stream #1:0: Audio: mp3, 44100 Hz, stereo, fltp, 192 kb/s
 Metadata:
 encoder : Lavc56.21
Stream mapping:
 Stream #0:1 (aac) -> atrim
 Stream #0:1 (aac) -> atrim
 Stream #1:0 (mp3float) -> concat:in1:a0
 concat -> Stream #0:0 (aac)
 Stream #0:0 -> #0:1 (copy)
Press [q] to stop, [?] for help
Output #0, mp4, to 'C:\Users\test\Desktop\vidz\New folder (2)\target\output1.mp4':
 Metadata:
 major_brand : mp42
 minor_version : 0
 compatible_brands: isommp42
 encoder : Lavf58.42.100
 Stream #0:0: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
 Metadata:
 encoder : Lavc58.78.102 aac
 Stream #0:1(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, bt709), 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 2160 kb/s, 25 fps, 25 tbr, 90k tbn, 90k tbc (default)
 Metadata:
 creation_time : 2018-09-02T04:28:46.000000Z
 handler_name : ISO Media file produced by Google Inc. Created on: 09/01/2018.
frame= 737 fps=0.0 q=-1.0 size= 7936kB time=00:00:29.44 bitrate=2208.1kbits/s speed=58.9x 
[out_0_0 @ 000000000312c900] 100 buffers queued in out_0_0, something may be wrong.
[out_0_0 @ 000000000312c900] 1000 buffers queued in out_0_0, something may be wrong.
frame= 1401 fps=876 q=-1.0 size= 16640kB time=00:01:46.85 bitrate=1275.7kbits/s speed=66.8x 
frame= 2281 fps=1086 q=-1.0 size= 26112kB time=00:02:22.01 bitrate=1506.3kbits/s speed=67.6x 
frame= 3152 fps=1212 q=-1.0 size= 35584kB time=00:02:56.58 bitrate=1650.8kbits/s speed=67.9x 
frame= 4011 fps=1294 q=-1.0 size= 45824kB time=00:03:31.32 bitrate=1776.4kbits/s speed=68.2x 
frame= 4882 fps=1356 q=-1.0 size= 56320kB time=00:04:06.03 bitrate=1875.2kbits/s speed=68.3x 
frame= 5753 fps=1403 q=-1.0 size= 66560kB time=00:04:40.79 bitrate=1941.8kbits/s speed=68.5x 
frame= 6622 fps=1440 q=-1.0 size= 77056kB time=00:05:15.37 bitrate=2001.6kbits/s speed=68.6x 
frame= 7482 fps=1467 q=-1.0 size= 85248kB time=00:05:50.20 bitrate=1994.1kbits/s speed=68.7x 
frame= 7720 fps=1479 q=-1.0 Lsize= 87248kB time=00:05:59.79 bitrate=1986.5kbits/s speed=68.9x 
video:81451kB audio:5634kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.186449%
[aac @ 0000000003017ac0] Qavg: 648.065




I tried messing about with the line of code but no success so I turn to you for additional support