I was just started to use ffmpeg recently and stumbled on this streaming problem. Scenario: i want to live stream a webcam in local network. Both server and client will be using windows platform.
Current feasible solution: using ffmpeg simple command line
to test it quickly i tried to locally stream it (the input doesn't really matter btw in this question).
On server -> ffmpeg -f dshow -i video="cam1":audio="mic1" -r 30 -g 0 -vcodec h264 -acodec libmp3lame -tune zerolatency -preset ultrafast -f mpegts udp://localhost:6789
On client(the same computer) -> ffplay udp://localhost:6789
The above works just fine, except for the latency, which i'm getting at about 1-2 second delay.
Now i want to try to change the encoder to use libvpx (vp8) for video and vorbis for audio (i changed the input to a pre-recorded h264 video, but it really doesn't matter)
On server
>ffmpeg -i "suits.mp4" -r 30 -g 0 -vcodec libvpx -acodec vorbis -strict -2 -f webm -f mpegts udp://localhost:6789
On client(the same computer) -> ffplay udp://localhost:6789
However this doesn't work... And below are console outputs:
> onserver ->
> ffmpeg version N-56165-gae12d65 Copyright (c) 2000-2013 the FFmpeg
> developers built on Sep 10 2013 19:42:46 with gcc 4.7.3 (GCC)
> configuration: --enable-gpl --enable-version3 --disable-w32threads
> --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 43.100 / 52. 43.100 libavcodec 55. 31.101 / 55. 31.101 libavformat 55. 16.102 / 55. 16.102 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 84.100 / 3. 84.100 libswscale 2. 5.100 /
> 2. 5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Suits.mp4': Metadata:
> major_brand : isom
> minor_version : 1
> compatible_brands: isom
> creation_time : 2011-09-08 11:43:25 Duration: 00:42:14.87, start: 0.000000, bitrate: 882 kb/s
> Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 720x402 [SAR 1:1 DAR 120:67], 750 kb/s, 23.98 fps,
> 23.98 tbr, 24k tbn, 47.95 tbc (default)
> Metadata:
> creation_time : 2011-09-08 11:43:25
> Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 126 kb/s (default)
> Metadata:
> creation_time : 2011-09-08 11:43:25 [libvpx @ 05392a80] v1.2.0 Output #0, mpegts, to 'udp://localhost:6789': Metadata:
> major_brand : isom
> minor_version : 1
> compatible_brands: isom
> encoder : Lavf55.16.102
> Stream #0:0(und): Video: vp8 (libvpx), yuv420p, 720x402 [SAR 1:1 DAR 120:67], q=-1--1, 200 kb/s, 90k tbn, 30 tbc (default)
> Metadata:
> creation_time : 2011-09-08 11:43:25
> Stream #0:1(und): Audio: vorbis, 48000 Hz, stereo, fltp (default)
> Metadata:
> creation_time : 2011-09-08 11:43:25 Stream mapping: Stream #0:0 -> #0:0 (h264 -> libvpx) Stream #0:1 -> #0:1 (aac -> vorbis) Press [q] to stop, [?] for help frame=42535 fps= 51 q=0.0 Lsize=
> 143539kB time=00:23:38.28 bitrate= 829.1kbits/s dup=8541 drop=0
> video:99155kB audio:28125kB subtitle:0 global headers:3kB muxing
> overhead 12.772155% Received signal 2: terminating.
> on client
> ffplay version N-56165-gae12d65 Copyright (c) 2003-2013 the FFmpeg
> developers built on Sep 10 2013 19:42:46 with gcc 4.7.3 (GCC)
> configuration: --enable-gpl --enable-version3 --disable-w32threads
> --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 43.100 / 52. 43.100 libavcodec 55. 31.101 / 55. 31.101 libavformat 55. 16.102 / 55. 16.102 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 84.100 / 3. 84.100 libswscale 2. 5.100 /
> 2. 5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100
> nan : 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0 [mpegts @ 02eb8620] probed stream 0 failed
> nan : 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0 [mp3 @ 02ed75a0] Header missing
> Last message repeated 1 times [mp3 @ 02ed75a0] Header missing
> La Last message repeated 13 times
> nan : 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0 [mp3 @ 02ed75a0] Header missing Last message repeated 13 times
> nan : 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0 [mp3 @ 02ed75a0] Header missing Last message repeated 9 times
> nan : 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0 [mp3 @ 02ed75a0] Header missing [mpegts @ 02eb8620] decoding for
> stream 1 failed [mpegts @ 02eb8620] Could not find codec parameters
> for stream 0 (Unknown: none ([6][0][0][0] / 0x0006)): unknown codec
> Consider increasing the value for the 'analyzeduration' and
> 'probesize' options [mpegts @ 02eb8620] Could not find codec
> parameters for stream 1 (Audio: mp3 ([6][0][0][0] / 0x0006), 0
> channels, s16p): unspecified frame size Consider increasing the value
> for the 'analyzeduration' and 'probesize' options
> udp://localhost:6789: could not find codec parameters
So does the point to point streaming for ffmpeg just doesn't work for vp8 or am i missing something? Btw, the end goal is to create a similar video chat based framework and i'll appreciate any suggestion. I'm reading up on webRTC now.
