Newest 'ffmpeg' Questions - Stack Overflow

http://stackoverflow.com/questions/tagged/ffmpeg

Les articles publiés sur le site

  • 'av_interleaved_write_frame() : Unknown error' when trying to read RTSP stream

    16 novembre 2013, par ddario

    I'm trying to re-stream an RTSP stream with FFmpeg; stream source is a camera. This is my command:

    ffmpeg -i rtsp://admin:admin@192.168.1.119/defaultPrimary?streamType=u -f mpegts rtp://0.0.0.0:1234
    

    After starting, FFmpeg keeps looping these messages for a few seconds:

    [h264 @ 00000000025fa600] non-existing PPS referenced
    [h264 @ 00000000025fa600] non-existing PPS 0 referenced
    [h264 @ 00000000025fa600] decode_slice_header error
    [h264 @ 00000000025fa600] no frame!
    

    and finally:

    [rtsp @ 00000000003f6ea0] decoding for stream 1 failed
    [rtsp @ 00000000003f6ea0] Could not find codec parameters for stream 1 (Video: h
    264): unspecified size
    Consider increasing the value for the 'analyzeduration' and 'probesize' options
    Guessed Channel Layout for  Input Stream #0.0 : mono
    Input #0, rtsp, from 'rtsp://admin:admin@192.168.1.119/defaultPrimary?streamType
    =u':
      Metadata:
        title           : Live
      Duration: N/A, start: 536870.877625, bitrate: 64 kb/s
        Stream #0:0: Audio: pcm_mulaw, 8000 Hz, mono, s16, 64 kb/s
        Stream #0:1: Video: h264, 90k tbr, 90k tbn, 180k tbc
        Stream #0:2: Data: none
    Output #0, mpegts, to 'rtp://0.0.0.0:1234':
      Metadata:
        title           : Live
        encoder         : Lavf55.21.100
        Stream #0:0: Audio: mp2, 16000 Hz, mono, s16, 128 kb/s
    Stream mapping:
      Stream #0:0 -> #0:0 (pcm_mulaw -> mp2)
    Press [q] to stop, [?] for help
    av_interleaved_write_frame(): Unknown error
    

    Increasing analyzeduration and probesize didn't help.
    I'm using the latest Windows static binaries.

  • FFmpeg X264 multithread decoding

    16 novembre 2013, par Xin He

    I want to do H264 multithread decoding using FFmpeg. First I use X264 to encode yuv420 to file.x264 ./x264 -o file.x264 --slices 4 --input-res 1280x720 --fps 25 file_560x420.yuv

    The slices count is 4.

    However, when I use FFmpeg to decode, ./ffmpeg -y -threads 4 -thread_type slice -i file.x264 final.yuv

    FFmpeg calls func"execute_decode_slices" four times, not in a multithread way. I also found that in *execute_decode_slices(h, context_count)*, the context_count is 1. FFmpeg call this 4 times.

    Could someone help to solve this problem?

    Thanks very much!

  • AndroidFFMPEG Appunite library issue

    16 novembre 2013, par Steve

    I am facing some with androidFfmpeg library on video seek.

    Can anyone help me out on this reported issue. jni_player_seek problem

  • MP4's cutting out at the end when streaming FMS 3.5

    16 novembre 2013, par user1488848

    I am trying to stream a series of MP4's using FMS3.5 and some of the files seem to cut out a few seconds right before the end. I have used the ffmpeg bat proccess below, and when I increased the framerate to 500 it seems to have corrected itself although that has caused the files to be significantly larger. Was there anyway that I can keep the filesize and framerate lower while avoiding the cutout issue or some predefined settings I should be using when streaming MP4's off FMS 3.5

    ffmpeg -i input.mp4 -pass 1 -s 572x322 -vcodec libx264 -b 500k -flags +loop+mv4 \
    -cmp 256 -partitions +parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 -me_method hex \
    -subq 7 -trellis 1 -refs 5 -bf 0 -flags2 +mixed_refs -coder 0 -me_range 16 \
    -g 250 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -qmin 10 -qmax 51 \
    -qdiff 4 -acodec libfaac -ab 160k -ac 1 output.mp4
    
  • Avconv : Select german stream not highest quality one

    15 novembre 2013, par mblaettermann

    I am converting some input stream from my DVB S2 Card to RTMP.

    Everything works fine after switching to recent avconv and x264 :)

    The only thing I couldn't find out is, how do I select the right audio stream?

    The source sometimes has up to 6 audio tracks. Avconv automatically chooses the one with the highest bitrate. However I want to select the "ger" one:

    Here are the streams of ARTE german/french TV Channel for example:

    Input #0, mpegts, from 'http://192.168.1.50:9981/stream/channelid/1035':
      Duration: N/A, start: 19083.694722, bitrate: 15576 kb/s
      Program 1
        Stream #0.0[0xa8], 127, 1/90000: Video: mpeg2video (Main), yuv420p, 544x576 [PAR 32:17 DAR 16:9], 1/50, 15000 kb/s, 25 fps, 90k tb50 tbc
        Stream #0.1[0x70](fre), 204, 1/90000: Audio: mp2, 48000 Hz, stereo, s16p, 192 kb/s
        Stream #0.2[0x71](ger), 207, 1/90000: Audio: mp2, 48000 Hz, stereo, s16p, 128 kb/s
        Stream #0.3[0x72](eng), 207, 1/90000: Audio: mp2, 48000 Hz, stereo, s16p, 128 kb/s
        Stream #0.4[0x73](qaa), 207, 1/90000: Audio: mp2, 48000 Hz, stereo, s16p, 128 kb/s
      No Program
        Stream #0.5[0x3b], 126, 1/90000: Audio: mp1, 0 channels, s16p
    

    libav Docs are really not that helpful. Who does now the right syntax?

    EDIT: I found the -map option: http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20use%20-map%20option But it is not possible to map by name? Only by index?

    Maybe I need to use avprobe then, to find the corrent stream index for "ger".