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• FFmpeg problems with playing a stream

  13 avril 2021, par Niko

  I want to play a stream over network. While my server works I can't get it working receiving audio on client. It's router with openwrt installed and usb soundcard.

  


  When i run
ffmpeg -i "udp://@:5555" -f alsa hw:0 i get a following error

  


  [NULL @ 0x76b2b390] Requested output format 'alsa' is not a suitable output format
hw:0: Invalid argument


  


  What's wrong with my command ?
On router there is limited space. I hardly handled to install ffmpeg.

  


  There is complete log :

  


  root@LEDE:/proc/asound# ffmpeg -i "udp://@:5555" -f alsa hw:0
ffmpeg version 3.2.2 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.4.0 (LEDE GCC 5.4.0 r3101-bce140e)
  configuration: --enable-cross-compile --cross-prefix=mips-openwrt-linux-musl- --arch=mips --target-os=linux --prefix=/usr --pkg-config=pkg-config --enable-shared --enable-static --enable-small --enable-pthreads --enable-zlib --disable-doc --disable-debug --enable-gpl --enable-version3 --disable-dxva2 --disable-lzma --disable-vaapi --disable-vda --disable-vdpau --disable-outdevs --disable-altivec --disable-vsx --disable-power8 --disable-amd3dnow --disable-amd3dnowext --disable-mmx --disable-mmxext --disable-sse --disable-sse2 --disable-sse3 --disable-ssse3 --disable-sse4 --disable-sse42 --disable-avx --disable-xop --disable-fma3 --disable-fma4 --disable-avx2 --disable-aesni --disable-armv5te --disable-armv6 --disable-armv6t2 --disable-inline-asm --disable-mipsdsp --disable-mipsdspr2 --disable-mipsfpu --disable-msa --disable-mmi --disable-fast-unaligned --disable-runtime-cpudetect --disable-yasm --enable-libopus --enable-decoder=libopus --enable-libx264 --enable-libmp3lame
  libavutil      55. 34.100 / 55. 34.100
  libavcodec     57. 64.101 / 57. 64.101
  libavformat    57. 56.100 / 57. 56.100
  libavdevice    57.  1.100 / 57.  1.100
  libavfilter     6. 65.100 /  6. 65.100
  libswscale      4.  2.100 /  4.  2.100
  libswresample   2.  3.100 /  2.  3.100
  libpostproc    54.  1.100 / 54.  1.100
Input #0, mpegts, from 'udp://@:5555':
  Duration: N/A, start: 1.400000, bitrate: 384 kb/s
  Program 1 
    Metadata:
      service_name    : Service01
      service_provider: FFmpeg
    Stream #0:0[0x100]: Audio: mp2 ([3][0][0][0] / 0x0003), 44100 Hz, stereo, s16p, 384 kb/s
[NULL @ 0x76b2b390] Requested output format 'alsa' is not a suitable output format
hw:0: Invalid argument


  


• Unable to encode single frame h264 (.mp4) video with FFmpeg. No video stream present

  1er avril 2021, par Maxito

  I have achieved this with ffmpeg command line tool using the command. The folder had only one image.

  



  


  ffmpeg -r 24 -i image%03d.bmp -c:v libx264 -pix_fmt yuv420p
 oneframex.mp4

  


  



  I would like to do the same with C++. If I encode a video of three or more frames, video encodes correctly, but the result of encoding a one or two frames video never has a video stream, as reported by ffprobe and some media players.

  



  Comparing with ffprobe, my video (the one with three or more frames) and the one generated by the command tool show almost the same information. Only bitrate and encoder version are different.

  



  I have tried adding force_key_frames to 1, tried with many encoding options and have be unsuccessful.

  



  The application output gives me this information :

  



  


  [libx264 @ 20d1b840] using cpu capabilities : MMX2 SSE2Fast SSSE3
 SSE4.2 AVX

  
 


  [libx264 @ 20d1b840] profile High, level 4.0

  
 


  [libx264 @ 20d1b840] 264 - core 142 r2431 ac76440 - H.264/MPEG-4 AVC
 codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html -
 options : cabac=0 ref=1 deblock=1:0:0 analyse=0x3:0x113 me=dia subme=8
 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=0 trellis=0
 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2
 threads=1 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1
 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0
 keyint=1 keyint_min=1 scenecut=0 intra_refresh=0 rc=crf mbtree=0
 crf=10.0 qcomp=0.60 qpmin=0 qpmax=25 qpstep=4 ip_ratio=1.40 aq=1:1.00

  


  



  These are my main parameters :

  



  pCodecCtx->codec_id = AV_CODEC_ID_H264;
pCodecCtx->pix_fmt= AV_PIX_FMT_YUV420P; 
pCodecCtx->gop_size = 1;
pCodecCtx->bit_rate = 400000;
pCodecCtx->me_range = 16;
pCodecCtx->max_qdiff = 4;
pCodecCtx->qcompress = 0.6;
pCodecCtx->qmin = 0;
pCodecCtx->qmax = 25;
pCodecCtx->time_base.den = 24;
pCodecCtx->time_base.num = 1;

AVDictionary *param = 0;
            av_dict_set(&param, "preset", "slow", 0);
            av_dict_set(&param, "profile", "high", 0);
            av_dict_set(&param, "crf", "10", 0); //this gave me quality
            av_dict_set(&param, "force_key_frames", "1", 0);


  



  In my encoding I just added

  



  ppicture->pts = pCodecCtx->frame_number


  



  to avoid non-strictly-monotonic PTS message. And tried the methods from this question in case it had something to do.

  



  I’m sure I must be missing some important parameter to be able to create such a small video. I will take any suggestion.

  


• Algorithm when recording SegmentTimeline of MPEG-DASH MPD in shaka packager

  25 mars 2021, par jgkim0518

  I packaged a media stream by mpeg-dash transcoded video at twice the speed and audio at normal speed.
Here is the command :

  


  ffmpeg -re -stream_loop -1 -i timing_logic.mp4 -c:v hevc_nvenc -filter:v "setpts=2*PTS" -c:a libfdk_aac -map 0:v -map 0:a -f mpegts udp://xxx.xxx.xxx.xxx:xxxx?pkt_size=1316


  


  The source information used here is as follows :

  


  Input #0, mpegts, from '/home/test/timing_logic/timing_logic.mp4':
Duration: 00:00:30.22, start: 1.400000, bitrate: 16171 kb/s
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0:0[0x100]: Video: hevc (Main 10) (HEVC / 0x43564548), yuv420p10le(tv, bt709), 3840x2160 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 50 tbc
Stream #0:1[0x101](eng): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, fltp, 128 kb/s Stream mapping:
Stream #0:0 -> #0:0 (hevc (native) -> hevc (hevc_nvenc))
Stream #0:1 -> #0:1 (mp2 (native) -> aac (libfdk_aac))
Press [q] to stop, [?] for help
frame= 0 fps=0.0 q=0.0 size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A
frame= 0 fps=0.0 q=0.0 size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A
Output #0, mpegts, to 'udp://xxx.xxx.xxx.xxx:xxxx?pkt_size=1316':
Metadata:
encoder : Lavf58.45.100
Stream #0:0: Video: hevc (hevc_nvenc) (Main 10), p010le, 3840x2160 [SAR 1:1 DAR 16:9], q=-1--1, 2000 kb/s, 50 fps, 90k tbn, 50 tbc
Metadata:
encoder : Lavc58.91.100 hevc_nvenc
Side data:
cpb: bitrate max/min/avg: 0/0/2000000 buffer size: 4000000 vbv_delay: N/A
Stream #0:1(eng): Audio: aac (libfdk_aac), 48000 Hz, stereo, s16, 139 kb/s
Metadata:
encoder : Lavc58.91.100 libfdk_aac


  


  My shaka packager command is :

  


  packager \ 'in=udp://xxx.xxx.xxx.xxx:xxxx,stream=video,init_segment=/home/test/timing_logic/package/video/0/video.mp4,segment_template=/home/test/timing_logic/package/video/0/$Time$.m4s' \
'in=udp://xxx.xxx.xxx.xxx:xxxx,stream=audio,init_segment=/home/test/timing_logic/package/audio/0/audio.mp4,segment_template=/home/test/timing_logic/package/audio/0/$Time$.m4a' \
--segment_duration 5 --fragment_duration 5 --minimum_update_period 5 --min_buffer_time 5 \
--preserved_segments_outside_live_window 24 --time_shift_buffer_depth 40 \
--allow_codec_switching --allow_approximate_segment_timeline --log_file_generation_deletion \
--mpd_output $OUTPUT/$output_mpd.mpd &


  


  Looking at the mpd generated as a result of packaging, the duration of the audio is twice that of the video, and the number of segments is half.
The following is the content of mpd :

  


  &lt;?xml version="1.0" encoding="UTF-8"?>&#xA;&#xA;<mpd xmlns="urn:mpeg:dash:schema:mpd:2011" profiles="urn:mpeg:dash:profile:isoff-live:2011" minbuffertime="PT5S" type="dynamic" publishtime="2021-03-23T10:01:57Z" availabilitystarttime="2021-03-23T09:59:55Z" minimumupdateperiod="PT5S" timeshiftbufferdepth="PT40S">&#xA;<period start="PT0S">&#xA;<adaptationset contenttype="audio" segmentalignment="true">&#xA;<representation bandwidth="140020" codecs="mp4a.40.2" mimetype="audio/mp4" audiosamplingrate="48000">&#xA;<audiochannelconfiguration schemeiduri="urn:mpeg:dash:23003:3:audio_channel_configuration:2011" value="2"></audiochannelconfiguration>&#xA;<segmenttemplate timescale="90000" initialization="/home/test/timing_logic/package/audio/0/audio.mp4" media="/home/test/timing_logic/package/audio/0/$Time$.m4a" startnumber="16">&#xA;<segmenttimeline>&#xA;<s t="6751436" d="449280"></s>&#xA;<s t="7200716" d="451200"></s>&#xA;<s t="7651916" d="449280"></s>&#xA;<s t="8101196" d="374400"></s>&#xA;<s t="8551196" d="449280"></s>&#xA;<s t="9000476" d="451200"></s>&#xA;<s t="9451674" d="449280"></s>&#xA;<s t="9900956" d="449280"></s>&#xA;<s t="10350236" d="451200"></s>&#xA;<s t="10801434" d="374400"></s>&#xA;</segmenttimeline>&#xA;</segmenttemplate>&#xA;</representation>&#xA;</adaptationset>&#xA;<adaptationset contenttype="video" width="3840" height="2160" framerate="90000/3600" segmentalignment="true" par="16:9">&#xA;<representation bandwidth="1520392" codecs="hvc1.2.4.L153" mimetype="video/mp4" sar="1:1">&#xA;<segmenttemplate timescale="90000" initialization="/home/test/timing_logic/package/video/0/video.mp4" media="/home/test/timing_logic/package/video/0/$Time$.m4s" startnumber="19">&#xA;<segmenttimeline>&#xA;<s t="16954440" d="900000" r="4"></s>&#xA;</segmenttimeline>&#xA;</segmenttemplate>&#xA;</representation>&#xA;</adaptationset>&#xA;</period>&#xA;</mpd>&#xA;

  


  When looking at the MPD, it seems that the shaka packager checks the PTS or DTS of the TS when creating a segment and recording the contents of the SegmentTimeline.
But I couldn't understand even by looking at the MPEG standard documentation and the DASH-IF documentation.

  


  My question is whether the packager refers to PTS or DTS when creating a segment.
How are SegmentTimeline's S@t and S@d recorded ?
What algorithm is the SegmentTimeline recorded with ? Please help me. Thank you.