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• FFmpeg - Extracting video and audio from transport stream file (.ts)

  31 octobre 2020, par Passepartout

  I wish to extract the audio and video of a certain program in a transport stream file (.ts) by specifying its PID without losing quality and using the same codec in the resulting file (the output file is a MPEG).

  



  Is that even possible with FFmpeg ? If so, how can I do it ?

  



  So far, I've come to this command :

  



   ffmpeg -i tsfile.ts -vcodec copy -acodec copy -q:v 1 output.mpg


  



  Edit : Note that the file output.mpg is created. The file contains the video but the audio isn't attached (no sound). Also, I am unable to specify the program PID to extract.

  



  Edit 2 : Here's the output of ffmpeg -i tsfile.ts

  



  ffmpeg version N-47062-g26c531c Copyright (c) 2000-2012 the FFmpeg developers
built on Nov 25 2012 12:21:26 with gcc 4.7.2 (GCC)
    libavutil      52.  9.100 / 52.  9.100
    libavcodec     54. 77.100 / 54. 77.100
    libavformat    54. 37.100 / 54. 37.100
    libavdevice    54.  3.100 / 54.  3.100
    libavfilter     3. 23.102 /  3. 23.102
    libswscale      2.  1.102 /  2.  1.102
    libswresample   0. 17.101 /  0. 17.101
    libpostproc    52.  2.100 / 52.  2.100
[mpeg2video @ 0201c7a0] mpeg_decode_postinit() failure
Last message repeated 10 times
[mpegts @ 0037b800] PES packet size mismatch
Input #0, mpegts, from 'tsfile.ts':
Duration: 00:01:30.58, start: 56297.848344, bitrate: 18045 kb/s
Program 1
    Stream #0:0[0x31]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p, 1920x1080  [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
    Stream #0:1[0x34]: Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, 5.1(side), s16, 384      kb/s


  



  Here's a tsinfo.exe on the .ts file :

  



  Reading from C:\tsfile.ts
Scanning 10000 TS packets

Packet 1 is PAT
Program list:
    Program 1 -> PID 0020 (32)

Packet 2 is PMT with PID 0020 (32)
  Program 1, version 1, PCR PID 0031 (49)
     Program info (38 bytes): 0e 03 c0 b9 16 10 06 c0 02 71 c0 04 00 0b 02 42 3f 05 04
                              47 41 39 34 86 0d e2 65 6e 67 7e 3f ff 65 6e 67 c1 3f ff
 maximum bitrate (3 bytes): c0 b9 16
 smoothing buffer (6 bytes): c0 02 71 c0 04 00
 system clock (2 bytes): 42 3f
 Registration GA94
 Descriptor tag 86 (134) (13 bytes): e2 65 6e 67 7e 3f ff 65 6e 67 c1 3f ff
  Program streams:
PID 0031 (  49) -> Stream type 02 (  2) H.262/13818-2 video (MPEG-2) or 11172-2 constrained video
PID 0034 (  52) -> Stream type 81 (129) User private
    ES info (6 bytes): 6a 04 41 43 2d 33
    DVB AC-3 (4 bytes): 41 43 2d 33

Found 14 PAT packets and 7 PMT packets in 10000 TS packets


  


• Display formatted date and time over frames using ffmpeg

  28 août 2020, par marcman

  I've gone through a handful of questions on here (this, this, this, etc) concerning overlaying the date and time on videos using ffmpeg, but I haven't been able to figure out the solution.

  


  I personally have found the ffmpeg documentation difficult to parse as well regarding drawing text that updates every (N) frame(s).

  


  I have the exif data from a movie specifying when it was created. I'd like to be able to emblazen that over the movie (as though it were a home video from some old VHS tape). For example, let's say I have a video from January 2, 2012 at 10:33:53. I'd like to be able to show "Jan 2, 2012 10:33:53am" on the lower right in white text. The spatial positioning and color are clear to me, but just how to go from the timestamp information I have to the formatted expansion is proving to be quite difficult for me. I have succeeded in getting a clock starting from 00:00:00.00 and counting up (using timecode and timecode_rate), but unfortunately I can't get much more than that.

  


  My question is : what is the proper datetext command that will allow me to both (a) provide the start time, and (b) format it with the proper expansion.

  


  ---

  


  As a bonus, if you can also point me to how to do this using the wonderful ffmpeg-python library, it would be even better. That library is quite good, but it does not appear to be actively maintained anymore.

  


• Trouble creating a gstreamer pipeline to send H264+AC3 over MPEG2-TS and RTP

  11 novembre 2020, par SeanWaves

  Sender and receiver pipelines :

  


  .\gst-launch-1.0.exe -v mpegtsmux name=mux ! tsparse set-timestamps=true ! rtpmp2tpay ! udpsink host=127.0.0.1 port=8888   videotestsrc is-live=true ! videoconvert ! x264enc ! mux.   audiotestsrc is-live=true ! audioresample ! audioconvert ! audio/x-raw,rate=44100,channels=2,depth=16 ! avenc_ac3 ! ac3parse ! mux.


  


  .\gst-launch-1.0.exe -v udpsrc address=127.0.0.1 port=8888 ! "application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)MP2T" ! rtpmp2tdepay ! tsdemux name=demux    demux. ! queue ! h264parse ! avdec_h264 ! videoconvert ! autovideosink   demux. ! queue ! ac3parse ! avdec_ac3 ! autoaudiosink


  


  I get a Delayed linking failed warning, and then I get nothing but 0:00:00.455121000 74464 00000219572127C0 WARN               h264parse gsth264parse.c:1492:gst_h264_parse_handle_frame:<h264parse0> broken/invalid nal Type: 1 Slice, Size: 6121 will be dropped</h264parse0> type WARNs.

  


  If I remove the audio leg of the receiver, it plays fine. If I only play the audio portion, I get :

  


  .\gst-launch-1.0.exe -v udpsrc address=127.0.0.1 port=8888 ! "application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)MP2T" ! rtpmp2tdepay ! tsdemux name=demux    demux. ! queue ! ac3parse ! avdec_ac3 ! autoaudiosink&#xA;Setting pipeline to PAUSED ...&#xA;Pipeline is live and does not need PREROLL ...&#xA;Pipeline is PREROLLED ...&#xA;Setting pipeline to PLAYING ...&#xA;New clock: GstSystemClock&#xA;/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)MP2T&#xA;0:00:04.773422000 75296 00000287541297C0 WARN                 default gst/parse/grammar.y:544:gst_parse_no_more_pads:<demux> warning: Delayed linking failed.&#xA;/GstPipeline:pipeline0/GstRtpMP2TDepay:rtpmp2tdepay0.GstPad:src: caps = video/mpegts, packetsize=(int)188, systemstream=(boolean)true&#xA;0:00:04.778044000 75296 00000287541297C0 WARN                 default gst/parse/grammar.y:544:gst_parse_no_more_pads:<demux> warning: failed delayed linking some pad of GstTSDemux named demux to some pad of GstQueue named queue0&#xA;/GstPipeline:pipeline0/GstTSDemux:demux.GstPad:sink: caps = video/mpegts, packetsize=(int)188, systemstream=(boolean)true&#xA;/GstPipeline:pipeline0/GstRtpMP2TDepay:rtpmp2tdepay0.GstPad:sink: caps = application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)MP2T&#xA;WARNING: from element /GstPipeline:pipeline0/GstTSDemux:demux: Delayed linking failed.&#xA;Additional debug info:&#xA;gst/parse/grammar.y(544): gst_parse_no_more_pads (): /GstPipeline:pipeline0/GstTSDemux:demux:&#xA;failed delayed linking some pad of GstTSDemux named demux to some pad of GstQueue named queue0&#xA;0:00:04.927067000 75296 00000287541297C0 WARN                 basesrc gstbasesrc.c:3127:gst_base_src_loop:<udpsrc0> error: Internal data stream error.&#xA;0:00:04.935063000 75296 00000287541297C0 WARN                 basesrc gstbasesrc.c:3127:gst_base_src_loop:<udpsrc0> error: streaming stopped, reason not-linked (-1)&#xA;ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal data stream error.&#xA;Additional debug info:&#xA;../libs/gst/base/gstbasesrc.c(3127): gst_base_src_loop (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0:&#xA;streaming stopped, reason not-linked (-1)&#xA;Execution ended after 0:00:04.895837000&#xA;Setting pipeline to NULL ...&#xA;Freeing pipeline ...&#xA;</udpsrc0></udpsrc0></demux></demux>

  


  I'm not sure what is incorrect about this. Taking a step back, what I'm really trying to achieve is to receive H264+AAC (not AC3) from FFMPEG in OBS. That doesn't work, and as I've isolated parts of the problem, it seems like it's coming down to how I'm sending audio across. I'm working purely in gstreamer for now to get it to work on the receiving end before I pop back up into FFMPEG land.