Which container/program for http progressive playback of h264/aac

ohropax · 11th October 2007, 18:19

Hi all,

i am working on a plugin for VDR and want to create a low bandwidth stream from live tv. However i have problems finding a combination of container/software and player that actually works well. Usually this would be a case for RTP transmission, however the application can not serve data as RTP livestream and can not be modified in that direction for various reasons. Instead the clients must connect to the application by HTTP and receive the data through this connection. As clients (VLC,mplayer,etc) have been supporting this for ages, this should not be too difficult. this prooved to be wrong.

The part that i am working on can be seen as a pipe. It receives a filtered MPEG transport stream containing MPEG2 audio+video and must output a stream containing h.264 video and AAC audio.

The simplest implementation uses the ffmpeg commandline utility and muxes the converted streams into MPEG TS. This creates a number of problems:
1. All clients that i have tested so far, fail to display the video stream properly, news tickers jump back and forth - it looks as if the presentation timestamps on the frames are wrong. If i change the video codec to (ffmpeg -vcodec mpeg4) everything is working nicely.
2. Muxing to TS creates huge overhead, about 70-90% for AAC audio caused by the small size of the compressed audio packages.

Point (2) is only an annoyance but point (1) is the major blocker right now. Does anyone have an idea on how to achieve better results?

Thanks for any help,
Marcus

commandline for reference reasons:
"ffmpeg -i - -acodec libfaac -ab 64k -vcodec libx264 -b 240k -bt 240k ...somex264options... -f mpegts -"

Sharktooth · 11th October 2007, 18:27

1 - look at the x264 options and disable b-pyramid AND use only 1 reference frame as a starting point.

ohropax · 12th October 2007, 08:27

Hi,

i've tried it and it does not help. Below is the full commandline. On your suggestion i added the last two flags (bpyramid and refs). The reason behind the other options is to have x264 cli defaults in ffmpeg.

/usr/local/bin/ffmpeg -i - -acodec libfaac -ab 64k -cutoff 15000 -vcodec libx264 -s cif -b 240k -bt 240k -coder 1 -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -me hex -subq 5 -me_range 16 -g 250 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -qdiff 4 -qmax 51 -qmin 10 -qcomp 0.6 -rc_eq blurCplx^(1-qComp) -refs 1 -flags2 -bpyramid -f mpegts -

I also tested it without the 'default' options and it does not help either. This time the commandline was:

/usr/local/bin/ffmpeg -i - -acodec libfaac -ab 64k -cutoff 15000 -vcodec libx264 -s cif -b 240k -bt 240k -refs 1 -flags2 -bpyramid -f mpegts -

Best regards, Marcus

ohropax · 12th October 2007, 08:31

BTW the software versions are sufficiently recent checkouts

FFmpeg version SVN-r10700, Copyright (c) 2000-2007 Fabrice Bellard, et al.
configuration: --enable-shared --enable-pthreads --enable-gpl --enable-liba52 --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libfaad
libavutil version: 49.5.0
libavcodec version: 51.45.0
libavformat version: 51.14.0
built on Oct 11 2007 15:00:57, gcc: 4.1.3 20070929 (prerelease) (Ubuntu 4.1.2-16ubuntu2)

x264 core:56 svn-680

ohropax · 12th October 2007, 08:41

Another sidenote regarding MPEG TS: If i change the commandline from -f mpegts to any of (mpeg,vob,dvd) the resulting stream saved to a file can be played without problems.

But it cannot be played 'live' through http, because programs do not find the necessary codec information. I took a look at the MPEG file with Elecard Stream Analyzer and discovered that (vob,dvd) files do not begin with a 'sequence header' packet but with the following packet, iow: somehow ffmpeg omits writing the first header. This has the result that when piping this file back to ffmpeg as input and only giving it the section from beginning of the file up to right before the start of the 'next' sequence header, it does not find the correct codec parameters. If i start feeding right before the 'next' sequence header it finds them immediately. So i think this is probably related to client programs not being able to start playing at all.

11th October 2007, 18:19	#1 \| Link
ohropax Registered User Join Date: Sep 2007 Posts: 16	Which container/program for http progressive playback of h264/aac Hi all, i am working on a plugin for VDR and want to create a low bandwidth stream from live tv. However i have problems finding a combination of container/software and player that actually works well. Usually this would be a case for RTP transmission, however the application can not serve data as RTP livestream and can not be modified in that direction for various reasons. Instead the clients must connect to the application by HTTP and receive the data through this connection. As clients (VLC,mplayer,etc) have been supporting this for ages, this should not be too difficult. this prooved to be wrong. The part that i am working on can be seen as a pipe. It receives a filtered MPEG transport stream containing MPEG2 audio+video and must output a stream containing h.264 video and AAC audio. The simplest implementation uses the ffmpeg commandline utility and muxes the converted streams into MPEG TS. This creates a number of problems: 1. All clients that i have tested so far, fail to display the video stream properly, news tickers jump back and forth - it looks as if the presentation timestamps on the frames are wrong. If i change the video codec to (ffmpeg -vcodec mpeg4) everything is working nicely. 2. Muxing to TS creates huge overhead, about 70-90% for AAC audio caused by the small size of the compressed audio packages. Point (2) is only an annoyance but point (1) is the major blocker right now. Does anyone have an idea on how to achieve better results? Thanks for any help, Marcus commandline for reference reasons: "ffmpeg -i - -acodec libfaac -ab 64k -vcodec libx264 -b 240k -bt 240k ...somex264options... -f mpegts -"

11th October 2007, 18:27	#2 \| Link
Sharktooth Mr. Sandman Join Date: Sep 2003 Location: Haddonfield, IL Posts: 11,768	1 - look at the x264 options and disable b-pyramid AND use only 1 reference frame as a starting point. __________________ MPEG-4 ASP Custom Matrices: EQM V1(old), EQM AutoGK Sharpmatrix (aka EQM V2), EQM V3HR (updated 01/10/2004), EQM V3LR, EQM V3ULR (updated 04/02/2005), EQM V3UHR (updated 17/12/2004) and EQM V3EHR (updated 05/10/2004) Info about my ASP matrices. MPEG-4 AVC Custom Matrices: EQM AVC-HR Info about my AVC matrices My x264 builds. Mooo!!!

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12th October 2007, 08:27	#3 \| Link
ohropax Registered User Join Date: Sep 2007 Posts: 16	Hi, i've tried it and it does not help. Below is the full commandline. On your suggestion i added the last two flags (bpyramid and refs). The reason behind the other options is to have x264 cli defaults in ffmpeg. /usr/local/bin/ffmpeg -i - -acodec libfaac -ab 64k -cutoff 15000 -vcodec libx264 -s cif -b 240k -bt 240k -coder 1 -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -me hex -subq 5 -me_range 16 -g 250 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -qdiff 4 -qmax 51 -qmin 10 -qcomp 0.6 -rc_eq blurCplx^(1-qComp) -refs 1 -flags2 -bpyramid -f mpegts - I also tested it without the 'default' options and it does not help either. This time the commandline was: /usr/local/bin/ffmpeg -i - -acodec libfaac -ab 64k -cutoff 15000 -vcodec libx264 -s cif -b 240k -bt 240k -refs 1 -flags2 -bpyramid -f mpegts - Best regards, Marcus

12th October 2007, 08:31	#4 \| Link
ohropax Registered User Join Date: Sep 2007 Posts: 16	BTW the software versions are sufficiently recent checkouts FFmpeg version SVN-r10700, Copyright (c) 2000-2007 Fabrice Bellard, et al. configuration: --enable-shared --enable-pthreads --enable-gpl --enable-liba52 --enable-libx264 --enable-libfaac --enable-libmp3lame --enable-libfaad libavutil version: 49.5.0 libavcodec version: 51.45.0 libavformat version: 51.14.0 built on Oct 11 2007 15:00:57, gcc: 4.1.3 20070929 (prerelease) (Ubuntu 4.1.2-16ubuntu2) x264 core:56 svn-680

12th October 2007, 08:41	#5 \| Link
ohropax Registered User Join Date: Sep 2007 Posts: 16	Another sidenote regarding MPEG TS: If i change the commandline from -f mpegts to any of (mpeg,vob,dvd) the resulting stream saved to a file can be played without problems. But it cannot be played 'live' through http, because programs do not find the necessary codec information. I took a look at the MPEG file with Elecard Stream Analyzer and discovered that (vob,dvd) files do not begin with a 'sequence header' packet but with the following packet, iow: somehow ffmpeg omits writing the first header. This has the result that when piping this file back to ffmpeg as input and only giving it the section from beginning of the file up to right before the start of the 'next' sequence header, it does not find the correct codec parameters. If i start feeding right before the 'next' sequence header it finds them immediately. So i think this is probably related to client programs not being able to start playing at all.