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• Automatically fade audio

  25 mars 2019, par Nadav Tasher

  I have a working script that plays a different audio every 45 minutes.

  The audio files change on a daily basis, so i cant manually fade them in/out in Audacity.

  In the script, i have the following line :

  ffplay -i /home/user/Ringtone1.mp3 -ss 00:00:7 -t 20 -nodisp -autoexit

  But every time, the audio cuts abruptly after 20 seconds, which isn’t very ear-pleasing.

  I want to make the audio fade in from 0 to max in 5 seconds and then fade out from max to 0 in 5 seconds (at the end of the 20 seconds).

  How can i do that ?

• FFmpeg muxing theora/vorbis unable to flush ?

  11 novembre 2013, par user2979732

  I'm pretty new to ffmpeg and it's confusing. I'm working on a basic muxer and have been spending over a week on this - I don't normally post as I solve 98% of my issues with google, but unable to get this one so far.

  The basis of my source is FFmpeg's own muxing.c example. When I try to force it using libvorbis for audio, and create "test.ogg" it demonstrates the same issues I'm having in my own derivation of muxing.c. The problem is with ogg/theora/vorbis. I'm forcing the use of audio codec like this :

  audio_st = add_stream(oc, &audio_codec, avcodec_find_encoder_by_name("libvorbis")->id);

  It seems the problem is in not setting audio pts in the muxing.c sample. There is a confusion in general about this, nobody apart from this guy didn't address what I am looking for http://webcache.googleusercontent.com/search?q=cache:6ml82RMN3YYJ:ffmpeg.org/pipermail/libav-user/2013-April/004304.html+&cd=4&hl=en&ct=clnk&gl=cz

  I couldn't find any answers to that naturally - like why don't they set the audio pts ? Laziness ? Not needed ? Do they believe all encoders will produce the pts for them(not true as seen below) ?

  Anyway, when you try muxing.c with mp4/libx264/forced libmp3lame all is fine, but the encoder says that "encoder did not produce valid pts, making some up.". However, it's silent with ogg/theora/vorbis, as if there were valid pts(?) but the result is no audio packets present in the stream(!), at least from what I saw using ffprobe. Which results in the video not being able to replay even, until you take out the empty audio stream. Then it plays the video, which shows that stream is fine.

  Coming to my original issue. I tried setting the pts on the audio frame you're sending to the encoder to fix that problem(this already sucks). I was unable to find a definite answer how to properly set the pts - that's the other big issue as I'm trying stuff which I'm not sure works. Anyway, in the end when setting "some" pts, this results in ogg with sound.

  if (frame->pts == AV_NOPTS_VALUE) frame->pts = audio_sync_opts;
  
  audio_sync_opts = frame->pts + frame->nb_samples;

  I'm aware I should probably use rescaling to adjust for the container time bases etc..if this was present/explained in ffmpeg's own sample I wouldn't have to guess now (as I'm stil not 100% sure about time base relationship between container and codec, I think container time base takes somehow over the codec one).

  My other problem is flushing - but that might have something to do with the screwed up pts. So I won't rather get into that in detail - the basic problem is, when I send finite number of audio frames, like 20, I get 2 packets only for example. From my understanding, I need to flush the rest of audio after all the encoding/muxing is done, which I managed to do with mp4/libx264/libmp3lame, but with ogg/theora/vorbis it doesn't flush. Why not, I have no idea.

  If someone could rework muxing.c into sending it finite number of audio / video frames - ie . not until duration > X, but until it sent 20 video & 100 audio frames(just an example). So that number of frames I have is important, not the video time I end up with. Then encode / mux all the frames - with proper video/audio pts, working with theora/ogg and flushing if needed, that would probably solve all of my issues. I'm sure for an expert ffmpeg'er modifying muxing.c addressing all those things would be a pretty quick exercise and could help more than 1 confused person.

  Thanks !

• ffmpeg conversion to mpeg2video

  19 novembre 2013, par spicyramen

  Need to play some video files from a Cisco DMP, and need to use mpeg2video for video and mp2 for audio.

  Im using ffmpeg -i to verify video format.

  This video plays correctly :

  Input #0, mpeg, from 'ATT_Telepresence_Scheduling.mpg':
  
    Duration: 00:07:14.08, start: 0.522456, bitrate: 474 kb/s
  
      Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p, 600x340 [SAR 1:1 DAR 30:17], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
  
      Stream #0:1[0x1c0]: Audio: mp2, 44100 Hz, stereo, s16p, 128 kb/s

  This video does not play(Black screen) :

  Input #0, mpegts, from 'Telepresence_part2.ts':
  
  Duration: 00:02:32.83, start: 1.000000, bitrate: 8783 kb/s
  
  Program 1 
  
  Stream #0:0[0x45]: Audio: aac ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 163 kb/s
  
  Stream #0:1[0x44]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 29.92 fps, 29.92 tbr, 90k tbn, 59.82 tbc

  How to perform the conversion to mpeg and to video mpeg2video and audio mp2 and preserve HD quality ?