Recherche avancée

Sur d’autres sites (11812)

• Duplicated PTS value when using rtsp transport UDP (H264 FU-A)

  25 janvier, par Christoph

  I’m implementing a packet loss counter based on the PTS from the av_packet, and it works fine when using RTSP/TCP as the transport mode. However, when I switched to RTSP/UDP, two packets consistently share the same PTS. This puzzled me because I assumed that av_read_frame would parse the stream and provide "valid" packets.

  


  In both cases, the stream is FU-A H.264, and I expected FFmpeg to handle reassembly in both transport modes identically. My understanding was that if UDP packets were splitted, FFmpeg would reassemble them into a single av_packet, similar to how it handles reassembly for TCP packets split due to MTU and FU-A.

  


  I could adapt my packet loss calculation by simply ignoring packets with the same PTS as the previous one, but I want to understand what’s happening here.

  


  TCP

  


  packet pts: -9223372036854775808, dts: -9223372036854775808, size: 52672, key-frame: true, discard: false, corrupt: false
packet pts: 3598, dts: 3598, size: 6034, key-frame: false, discard: false, corrupt: false
packet pts: 7196, dts: 7196, size: 5730, key-frame: false, discard: false, corrupt: false
packet pts: 10794, dts: 10794, size: 6153, key-frame: false, discard: false, corrupt: false
packet pts: 14392, dts: 14392, size: 2269, key-frame: false, discard: false, corrupt: false
packet pts: 17989, dts: 17989, size: 2656, key-frame: false, discard: false, corrupt: false
packet pts: 21587, dts: 21587, size: 2659, key-frame: false, discard: false, corrupt: false


  


  UDP

  


  packet pts: -9223372036854775808, dts: -9223372036854775808, size: 1391, key-frame: true, discard: false, corrupt: false
packet pts: 0, dts: 0, size: 109265, key-frame: true, discard: false, corrupt: false
packet pts: 3598, dts: 3598, size: 878, key-frame: false, discard: false, corrupt: false
packet pts: -> 3598, dts: 3598, size: 7728, key-frame: false, discard: false, corrupt: false
packet pts: 7195, dts: 7195, size: 887, key-frame: false, discard: false, corrupt: false
packet pts: -> 7195, dts: 7195, size: 7149, key-frame: false, discard: false, corrupt: false
packet pts: 10793, dts: 10793, size: 795, key-frame: false, discard: false, corrupt: false
packet pts: -> 10793, dts: 10793, size: 7777, key-frame: false, discard: false, corrupt: false
packet pts: 14391, dts: 14391, size: 119, key-frame: false, discard: false, corrupt: false
packet pts: -> 14391, dts: 14391, size: 2075, key-frame: false, discard: false, corrupt: false


  


  For reference here my code

  


  // PackageLossDetection detects possible packet loss based on PTS (Presentation Time Stamp) values.
// It compares the PTS of the packet with the expected PTS, calculated using the stream's time base and average frame rate.
// If the deviation between the expected and actual PTS exceeds a defined tolerance.
//
// Parameters:
//   - pkt: incoming packet whose PTS is to be checked.
//   - stream: the stream containing time base and average frame rate information.
func (s *AvSource) PackageLossDetection(pkt *astiav.Packet, stream *astiav.Stream) {

    // When using UDP as RTSP Transport packages in tuple has same PTS
    // TODO: Maybe we should invest more time to find a better solution
    if s.lastPts == pkt.Pts() {
        return
    }

    if pkt.Pts() > 0 {

        const tolerance = 4 // Allowable deviation in PTS steps
        if stream.AvgFrameRate().Num() == 0 {
            s.log.Warn().Str("stream", s.stream.Name).Msg("PackageLossDetection, no frame rate information available")
            return
        }

        var ptsBetween = stream.TimeBase().Den() * stream.TimeBase().Num() / stream.AvgFrameRate().Num()
        if math.Abs(float64(pkt.Pts()-(s.lastPts+int64(ptsBetween)))) > tolerance {
            s.log.Warn().Str("stream", s.stream.Name).Msgf("PackageLossDetection, PTS steps: %d, expected: %d, got: %d", int(ptsBetween), s.lastPts+int64(ptsBetween), pkt.Pts())
            utils.SafeIncrementInt64(&s.metrics.LossCount)
        }

        s.lastPts = pkt.Pts()
    }
}


  


• Add tools/merge-all-source-plugins

  14 mai, par Michael Niedermayer

  Add tools/merge-all-source-plugins
  Simple script to merge all source plugins.
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] INSTALL.md
  • [DH] tools/merge-all-source-plugins

• How to cut one track from one big flac file (16 tracks) [closed]

  9 juin, par Павел Дёмин

  There is a flac file + log to it. The file weighs more than 200 MB and contains 16 tracks. I need to cut track number 3 from it. How to do this in ffmpeg ?

  


  Read offset correction : 48
Overread into Lead-In and Lead-Out : No
Fill up missing offset samples with silence : Yes
Delete leading and trailing silent blocks : No
Null samples used in CRC calculations : Yes
Used interface : Installed external ASPI interface

  


  Used output format : Internal WAV Routines
Sample format : 44.100 Hz ; 16 Bit ; Stereo

  


  TOC of the extracted CD

  


   Track |   Start  |  Length  | Start sector | End sector 
---------------------------------------------------------
    1  |  0:00.00 |  3:56.26 |         0    |    17725   
    2  |  3:56.26 |  3:30.58 |     17726    |    33533   
    3  |  7:27.09 |  3:48.13 |     33534    |    50646   
    4  | 11:15.22 |  3:47.32 |     50647    |    67703   
    5  | 15:02.54 |  3:22.39 |     67704    |    82892   
    6  | 18:25.18 |  3:35.27 |     82893    |    99044   
    7  | 22:00.45 |  2:53.09 |     99045    |   112028   
    8  | 24:53.54 |  3:38.10 |    112029    |   128388   
    9  | 28:31.64 |  3:42.55 |    128389    |   145093   
   10  | 32:14.44 |  3:35.43 |    145094    |   161261   
   11  | 35:50.12 |  2:55.64 |    161262    |   174450   
   12  | 38:46.01 |  3:49.45 |    174451    |   191670   
   13  | 42:35.46 |  3:33.74 |    191671    |   207719   
   14  | 46:09.45 |  3:26.38 |    207720    |   223207   
   15  | 49:36.08 |  4:37.38 |    223208    |   244020   
   16  | 56:45.46 |  7:58.17 |    255421    |   291287