Recherche avancée

Sur d’autres sites (8776)

• Converting AAC stream to DASH MP4 with high fragment length precision

  5 mars 2017, par vdudouyt

  For my HTML5 project I need to create a fragmented MP4 file with a single audio stream (no video), each fragment of which has a duration of exactly 0.1 second.

  Accordingly to ffmpeg docs, you can accomplish that by passing a value in microseconds with ’-frag_duration’ - which I found to be working and playable with HTML5 MediaSource API :

  $ ffmpeg -y -i input.aac -c:a libfdk_aac -b:a 64k -level:v 13 -r 25 -strict experimental -movflags empty_moov+default_base_moof -frag_duration 100000 output.mp4

  As we have a 210 second audio split up by 0.1s fragments, I expect that in output.mp4 we’d have 2100 fragments, hence 2100 moof atoms. But, upon inspecting it I’ve figured out that we only have 1811 moof atoms - which means that some (or maybe even all) fragments are bigger than expected :

  $ python ~/git/mp4viewer/src/showboxes.py output.mp4 |grep moof|wc -l
  
  1811

  Could anybody tell me what’s wrong, and how could I accomplish what I want ?

  Right now my assumption is that during an encoding I have AAC frame length which is not a multiple of 0.1s, hence ffmpeg has no chance to produce the fragments that are strictly equal to 0.1s but I’m not sure. If somebody can confirm that - and let me know a way to explicitly set AAC frame_size in FFMPEG (I couldn’t find anything like that in the docs), or completely disprove this - it would be also highly appreciated.

• avcodec/lossless_videodsp : use ptrdiff_t for length parameters

  22 mars 2017, par James Almer

  avcodec/lossless_videodsp : use ptrdiff_t for length parameters
  Signed-off-by : James Almer <jamrial@gmail.com>
  

  • [DH] libavcodec/lossless_videodsp.c
  • [DH] libavcodec/lossless_videodsp.h
  • [DH] libavcodec/ppc/lossless_videodsp_altivec.c
  • [DH] libavcodec/x86/lossless_videodsp_init.c
  • [DH] tests/checkasm/llviddsp.c

• If multiple channels, merge then take sample length from audio file and save it to s3

  18 mai 2017, par khinester

  I am using transloadit to extract the audio from a video file, which is then saved to S3.
  This works great, but I wanted to know how to :

  1. check if the file has multiple channels and then squash it inot one as per https://transloadit.com/demos/audio-encoding/merging-multiple-audio-streams/ - do I need to check for this or do i default to use this robot ?

  2. extract a small sample from the audio file - and save this as a separate file.

  For example, I have a 2h audio file from which I want to take 5% of the length and save this as sample.mp3

  In ffmpeg, i can cut :

  ffmpeg -ss 0 -t 30 -i original.mp3 sample.mp3

  but I am unsure how to chain this workflow, here is what i have thus far :

  const opts = {
  
    params: {
  
      notify_url: `${ process.env.SELF }/services/trans/${ jwToken }`,
  
      steps:      {
  
        import: {
  
          robot:  '/s3/import',
  
          use:    ':original',
  
          bucket: process.env.S3_INGEST,
  
          path:   ingest.key,
  
          key:    process.env.AWS_ID,
  
          secret: process.env.AWS_SECRET,
  
        },
  
        encode: {
  
          robot:        '/audio/encode',
  
          use:          'import',
  
          ffmpeg_stack: 'v2.2.3',
  
          preset:       'aac',
  
          ffmpeg:       {
  
            ab:  '128k',
  
          },
  
        },
  
        export: {
  
          robot:   '/s3/store',
  
          use:     'encode',
  
          bucket:  s3Export,
  
          path:    `${ prefix }/${ token }.m4a`,
  
          headers: {
  
            'Content-Type': 'audio/mp4',
  
            'x-amz-server-side-encryption': 'AES256',
  
          },
  
          key:    process.env.AWS_ID,
  
          secret: process.env.AWS_SECRET,
  
        },
  
      },
  
    },
  
  };

  in the docs, https://transloadit.com/docs/conversion-robots/ i can’t see how to do this ?

  any advice is much appreciated.