对从iOS Safari WebRTC创建的视频进行转码时，AWS MediaConvert失败 [英] AWS MediaConvert fails when transcoding a video created from an iOS Safari WebRTC

问题描述

我正在尝试使用MediaConvert对视频进行转码，但出现以下错误:

I am trying to transcode a video using MediaConvert and I get the following error:

错误消息Demuxer:[ReadPacketData文件读取失败-文件结尾命中长度[1105924].文件被截断了吗?]

Error message Demuxer: [ReadPacketData File read failed - end of file hit at length [1105924]. Is file truncated?]

当我尝试运行ffmpeg命令以获取有关文件的信息时，我看到以下内容:

When I try to run ffmpeg command to get information about the file I see this:

流中非单调增加dts的多路复用器

non monotonically increasing dts to muxer in stream

我想知道MediaConvert中是否有可以纠正时间戳的配置?

I am wondering is there a configuration in MediaConvert that can correct the time stamps?

➜  Downloads ffmpeg -i 2254.webm -hide_banner -f null /dev/null

[h264 @ 0x7fce2c81d400] Increasing reorder buffer to 1
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '2254.webm':
  Metadata:
    major_brand     : iso5
    minor_version   : 1
    compatible_brands: isomiso5hlsf
    creation_time   : 2021-02-17T19:33:56.000000Z
  Duration: 00:00:01.60, start: 0.000000, bitrate: 5535 kb/s
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc), 1280x720, 5644 kb/s, 27.95 fps, 29.92 tbr, 600 tbn, 1200 tbc (default)
    Metadata:
      rotate          : 90
      creation_time   : 2021-02-17T19:33:56.000000Z
      handler_name    : Core Media Video
    Side data:
      displaymatrix: rotation of -90.00 degrees
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 187 kb/s (default)
    Metadata:
      creation_time   : 2021-02-17T19:33:56.000000Z
      handler_name    : Core Media Audio
Stream mapping:
  Stream #0:0 -> #0:0 (h264 (native) -> wrapped_avframe (native))
  Stream #0:1 -> #0:1 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, null, to '/dev/null':
  Metadata:
    major_brand     : iso5
    minor_version   : 1
    compatible_brands: isomiso5hlsf
    encoder         : Lavf58.45.100
    Stream #0:0(und): Video: wrapped_avframe, yuvj420p(progressive), 720x1280, q=2-31, 200 kb/s, 29.92 fps, 29.92 tbn, 29.92 tbc (default)
    Metadata:
      encoder         : Lavc58.91.100 wrapped_avframe
      creation_time   : 2021-02-17T19:33:56.000000Z
      handler_name    : Core Media Video
    Side data:
      displaymatrix: rotation of -0.00 degrees
    Stream #0:1(und): Audio: pcm_s16le, 44100 Hz, mono, s16, 705 kb/s (default)
    Metadata:
      creation_time   : 2021-02-17T19:33:56.000000Z
      handler_name    : Core Media Audio
      encoder         : Lavc58.91.100 pcm_s16le
[null @ 0x7fce2e008200] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 8 >= 8
[null @ 0x7fce2e008200] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 31 >= 31
frame=   41 fps=0.0 q=-0.0 Lsize=N/A time=00:00:01.57 bitrate=N/A speed=19.9x    
video:21kB audio:128kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown

推荐答案

这时，MediaConvert需要一个完整的，带有时间戳的正确文件.无法在服务的多路分配中更改DTS或PTS值.

At this time MediaConvert expects a fully complete, timestamped correct file. There is no way to change DTS or PTS values at demux in the service.

在将文件传递给MediaConvert时，您是否能够仅使用ffmpeg进行重新混合并获得相同的结果?

Are you able to just remux with ffmpeg and get the same results when passing that file to MediaConvert?

ffmpeg -i 2254.webm -c copy output.webm

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