diff options
Diffstat (limited to 'doc')
| -rw-r--r-- | doc/TODO | 6 | ||||
| -rw-r--r-- | doc/faq.texi | 18 | ||||
| -rw-r--r-- | doc/ffmpeg-doc.texi | 156 | ||||
| -rw-r--r-- | doc/ffmpeg_powerpc_performance_evaluation_howto.txt | 2 | ||||
| -rw-r--r-- | doc/ffplay-doc.texi | 2 | ||||
| -rw-r--r-- | doc/ffserver-doc.texi | 26 | ||||
| -rw-r--r-- | doc/ffserver.conf | 14 | ||||
| -rw-r--r-- | doc/hooks.texi | 2 | ||||
| -rwxr-xr-x | doc/texi2pod.pl | 2 |
9 files changed, 114 insertions, 114 deletions
@@ -5,13 +5,13 @@ Fabrice's TODO list: (unordered) ------------------- Short term: -- av_read_frame() API +- av_read_frame() API - seeking API and example in ffplay - parse_only mode - use AVFMTCTX_DISCARD_PKT in ffplay so that DV has a chance to work - add RTSP regression test (both client and server) - make ffserver allocate AVFormatContext -- clean up (incompatible change, for 0.5.0): +- clean up (incompatible change, for 0.5.0): * AVStream -> AVComponent * AVFormatContext -> AVInputStream/AVOutputStream * suppress rate_emu from AVCodecContext @@ -54,7 +54,7 @@ Francois' TODO list: (unordered, without any timeframe) Philip'a TODO list: (alphabetically ordered) (please help) ------------------ - Add a multi-ffm filetype so that feeds can be recorded into multiple files rather - than one big file. + than one big file. - Authenticated users support -- where the authentication is in the URL - Change ASF files so that the embedded timestamp in the frames is right rather than being an offset from the start of the stream diff --git a/doc/faq.texi b/doc/faq.texi index 9497903b5c..40b7c84bcb 100644 --- a/doc/faq.texi +++ b/doc/faq.texi @@ -53,7 +53,7 @@ Use @file{-} as filename. @section Why does ffmpeg not decode audio in VOB files ? The audio is AC3 (a.k.a. A/52). AC3 decoding is an optional component in ffmpeg -as the component that handles AC3 decoding (liba52) is currently released under +as the component that handles AC3 decoding (liba52) is currently released under the GPL. If you have liba52 installed on your system, enable AC3 decoding with @code{./configure --enable-a52}. Take care: by enabling AC3, you automatically change the license of libavcodec from @@ -61,7 +61,7 @@ LGPL to GPL. @section Which codecs are supported by Windows ? -Windows does not support standard formats like MPEG very well, unless you +Windows does not support standard formats like MPEG very well, unless you install some additional codecs The following list of video codecs should work on most Windows systems: @@ -79,8 +79,8 @@ only if you have some MPEG-4 codec installed like ffdshow or XviD @item mpeg1 .mpg only @end table -Note, ASF files often have .wmv or .wma extensions in Windows. It should also -be mentioned that Microsoft claims a patent on the ASF format, and may sue +Note, ASF files often have .wmv or .wma extensions in Windows. It should also +be mentioned that Microsoft claims a patent on the ASF format, and may sue or threaten users who create ASF files with non-Microsoft software. It is strongly advised to avoid ASF where possible. @@ -95,7 +95,7 @@ if some MP3 codec like LAME is installed @section Why does the chrominance data seem to be sampled at a different time from the luminance data on bt8x8 captures on Linux? -This is a well-known bug in the bt8x8 driver. For 2.4.26 there is a patch at +This is a well-known bug in the bt8x8 driver. For 2.4.26 there is a patch at (@url{http://www.mplayerhq.hu/~michael/bttv-420-2.4.26.patch}). This may also apply cleanly to other 2.4-series kernels. @@ -132,8 +132,8 @@ Both XviD and DivX (version 4+) are implementations of the ISO MPEG-4 standard (note that there are many other coding formats that use this same standard). Thus, use '-vcodec mpeg4' to encode these formats. The default fourcc stored in an MPEG-4-coded file will be 'FMP4'. If you want -a different fourcc, use the '-vtag' option. E.g., '-vtag xvid' will -force the fourcc 'xvid' to be stored as the video fourcc rather than the +a different fourcc, use the '-vtag' option. E.g., '-vtag xvid' will +force the fourcc 'xvid' to be stored as the video fourcc rather than the default. @chapter Development @@ -149,7 +149,7 @@ it implemented is to undertake the task yourself. @section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat ? Yes. Read the Developers Guide of the FFmpeg documentation. Alternatively, -examine the source code for one of the many open source projects that +examine the source code for one of the many open source projects that already incorporate ffmpeg at (@url{projects.php}). @section Can you support my C compiler XXX ? @@ -174,7 +174,7 @@ terms of portability. @section Why not rewrite ffmpeg in object-oriented C++ ? ffmpeg is already organized in a highly modular manner and does not need to -be rewritten in a formal object language. Further, many of the developers +be rewritten in a formal object language. Further, many of the developers favor straight C; it works for them. For more arguments on this matter, read "Programming Religion" at (@url{http://lkml.org/faq/lkmlfaq-15.html}). diff --git a/doc/ffmpeg-doc.texi b/doc/ffmpeg-doc.texi index d83c672ea5..bb4cf04d31 100644 --- a/doc/ffmpeg-doc.texi +++ b/doc/ffmpeg-doc.texi @@ -12,7 +12,7 @@ FFmpeg is a very fast video and audio converter. It can also grab from a live audio/video source. - + The command line interface is designed to be intuitive, in the sense that FFmpeg tries to figure out all parameters that can possibly be derived automatically. You usually only have to specify the target @@ -30,7 +30,7 @@ FFmpeg can use a video4linux compatible video source and any Open Sound System audio source: @example -ffmpeg /tmp/out.mpg +ffmpeg /tmp/out.mpg @end example Note that you must activate the right video source and channel before @@ -48,10 +48,10 @@ Examples: * You can use YUV files as input: @example -ffmpeg -i /tmp/test%d.Y /tmp/out.mpg +ffmpeg -i /tmp/test%d.Y /tmp/out.mpg @end example -It will use the files: +It will use the files: @example /tmp/test0.Y, /tmp/test0.U, /tmp/test0.V, /tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc... @@ -130,7 +130,7 @@ NOTE: To see the supported input formats, use @code{ffmpeg -formats}. The generic syntax is: -@example +@example @c man begin SYNOPSIS ffmpeg [[infile options][@option{-i} @var{infile}]]... @{[outfile options] @var{outfile}@}... @c man end @@ -161,16 +161,16 @@ Show help. @item -formats Show available formats, codecs, protocols, ... -@item -f fmt +@item -f fmt Force format. -@item -i filename +@item -i filename input filename -@item -y +@item -y Overwrite output files. -@item -t duration +@item -t duration Set the recording time in seconds. @code{hh:mm:ss[.xxx]} syntax is also supported. @@ -178,16 +178,16 @@ Set the recording time in seconds. Seek to given time position in seconds. @code{hh:mm:ss[.xxx]} syntax is also supported. -@item -title string +@item -title string Set the title. -@item -author string +@item -author string Set the author. -@item -copyright string +@item -copyright string Set the copyright. -@item -comment string +@item -comment string Set the comment. @item -target type @@ -224,9 +224,9 @@ streams are delayed by 'offset' seconds. @table @option @item -b bitrate Set the video bitrate in kbit/s (default = 200 kb/s). -@item -r fps +@item -r fps Set frame rate (default = 25). -@item -s size +@item -s size Set frame size. The format is @samp{wxh} (default = 160x128). The following abbreviations are recognized: @table @samp @@ -265,7 +265,7 @@ represent red, the middle two digits green and last two digits blue (default = 000000 (black)). @item -vn Disable video recording. -@item -bt tolerance +@item -bt tolerance Set video bitrate tolerance (in kbit/s). @item -maxrate bitrate Set max video bitrate tolerance (in kbit/s). @@ -273,19 +273,19 @@ Set max video bitrate tolerance (in kbit/s). Set min video bitrate tolerance (in kbit/s). @item -bufsize size Set rate control buffer size (in kbit). -@item -vcodec codec +@item -vcodec codec Force video codec to @var{codec}. Use the @code{copy} special value to tell that the raw codec data must be copied as is. @item -sameq Use same video quality as source (implies VBR). -@item -pass n +@item -pass n Select the pass number (1 or 2). It is useful to do two pass encoding. The statistics of the video are recorded in the first pass and the video is generated at the exact requested bitrate in the second pass. -@item -passlogfile file +@item -passlogfile file Set two pass logfile name to @var{file}. @end table @@ -293,21 +293,21 @@ Set two pass logfile name to @var{file}. @section Advanced Video Options @table @option -@item -g gop_size +@item -g gop_size Set the group of pictures size. -@item -intra +@item -intra Use only intra frames. -@item -qscale q +@item -qscale q Use fixed video quantiser scale (VBR). -@item -qmin q +@item -qmin q minimum video quantiser scale (VBR) -@item -qmax q +@item -qmax q maximum video quantiser scale (VBR) -@item -qdiff q +@item -qdiff q maximum difference between the quantiser scales (VBR) -@item -qblur blur +@item -qblur blur video quantiser scale blur (VBR) -@item -qcomp compression +@item -qcomp compression video quantiser scale compression (VBR) @item -rc_init_cplx complexity @@ -363,31 +363,31 @@ Set IDCT algorithm to @var{algo}. Available values are: @item 0 FF_IDCT_AUTO (default) @item 1 -FF_IDCT_INT +FF_IDCT_INT @item 2 -FF_IDCT_SIMPLE +FF_IDCT_SIMPLE @item 3 -FF_IDCT_SIMPLEMMX +FF_IDCT_SIMPLEMMX @item 4 -FF_IDCT_LIBMPEG2MMX +FF_IDCT_LIBMPEG2MMX @item 5 -FF_IDCT_PS2 +FF_IDCT_PS2 @item 6 -FF_IDCT_MLIB +FF_IDCT_MLIB @item 7 -FF_IDCT_ARM +FF_IDCT_ARM @item 8 -FF_IDCT_ALTIVEC +FF_IDCT_ALTIVEC @item 9 -FF_IDCT_SH4 +FF_IDCT_SH4 @item 10 -FF_IDCT_SIMPLEARM +FF_IDCT_SIMPLEARM @end table @item -er n Set error resilience to @var{n}. @table @samp -@item 1 +@item 1 FF_ER_CAREFUL (default) @item 2 FF_ER_COMPLIANT @@ -453,9 +453,9 @@ name and its parameters separated by spaces. @section Audio Options @table @option -@item -ar freq +@item -ar freq Set the audio sampling frequency (default = 44100 Hz). -@item -ab bitrate +@item -ab bitrate Set the audio bitrate in kbit/s (default = 64). @item -ac channels Set the number of audio channels (default = 1). @@ -484,13 +484,13 @@ Set audio device (e.g. @file{/dev/dsp}). @section Advanced options @table @option -@item -map file:stream +@item -map file:stream Set input stream mapping. @item -debug Print specific debug info. -@item -benchmark +@item -benchmark Add timings for benchmarking. -@item -hex +@item -hex Dump each input packet. @item -bitexact Only use bit exact algorithms (for codec testing). @@ -510,7 +510,7 @@ Repeatedly loop output for formats that support looping such as animated GIF @section FFmpeg formula evaluator When evaluating a rate control string, FFmpeg uses an internal formula -evaluator. +evaluator. The following binary operators are available: @code{+}, @code{-}, @code{*}, @code{/}, @code{^}. @@ -650,26 +650,26 @@ library: @tab also known as @code{VOB} file @item MPEG-2 TS @tab @tab X @tab also known as DVB Transport Stream -@item ASF@tab X @tab X -@item AVI@tab X @tab X -@item WAV@tab X @tab X +@item ASF@tab X @tab X +@item AVI@tab X @tab X +@item WAV@tab X @tab X @item Macromedia Flash@tab X @tab X @tab Only embedded audio is decoded. @item FLV @tab X @tab X @tab Macromedia Flash video files -@item Real Audio and Video @tab X @tab X -@item Raw AC3 @tab X @tab X -@item Raw MJPEG @tab X @tab X -@item Raw MPEG video @tab X @tab X -@item Raw PCM8/16 bits, mulaw/Alaw@tab X @tab X -@item Raw CRI ADX audio @tab X @tab X -@item Raw Shorten audio @tab @tab X -@item SUN AU format @tab X @tab X +@item Real Audio and Video @tab X @tab X +@item Raw AC3 @tab X @tab X +@item Raw MJPEG @tab X @tab X +@item Raw MPEG video @tab X @tab X +@item Raw PCM8/16 bits, mulaw/Alaw@tab X @tab X +@item Raw CRI ADX audio @tab X @tab X +@item Raw Shorten audio @tab @tab X +@item SUN AU format @tab X @tab X @item NUT @tab X @tab X @tab NUT Open Container Format -@item QuickTime @tab X @tab X +@item QuickTime @tab X @tab X @item MPEG-4 @tab X @tab X @tab MPEG-4 is a variant of QuickTime. -@item Raw MPEG4 video @tab X @tab X +@item Raw MPEG4 video @tab X @tab X @item DV @tab X @tab X @item 4xm @tab @tab X @tab 4X Technologies format, used in some games. @@ -707,7 +707,7 @@ following image formats are supported: @multitable @columnfractions .4 .1 .1 .4 @item Supported Image Format @tab Encoding @tab Decoding @tab Comments -@item PGM, PPM @tab X @tab X +@item PGM, PPM @tab X @tab X @item PAM @tab X @tab X @tab PAM is a PNM extension with alpha support. @item PGMYUV @tab X @tab X @tab PGM with U and V components in YUV 4:2:0 @item JPEG @tab X @tab X @tab Progressive JPEG is not supported. @@ -734,11 +734,11 @@ following image formats are supported: @item H.261 @tab X @tab X @item H.263(+) @tab X @tab X @tab also known as RealVideo 1.0 @item H.264 @tab @tab X -@item MJPEG @tab X @tab X +@item MJPEG @tab X @tab X @item lossless MJPEG @tab X @tab X @item Apple MJPEG-B @tab @tab X @item Sunplus MJPEG @tab @tab X @tab fourcc: SP5X -@item DV @tab X @tab X +@item DV @tab X @tab X @item HuffYUV @tab X @tab X @item FFmpeg Video 1 @tab X @tab X @tab experimental lossless codec (fourcc: FFV1) @item FFmpeg Snow @tab X @tab X @tab experimental wavelet codec (fourcc: SNOW) @@ -755,7 +755,7 @@ following image formats are supported: @item ATI VCR2 @tab @tab X @tab fourcc: VCR2 @item Cirrus Logic AccuPak @tab @tab X @tab fourcc: CLJR @item 4X Video @tab @tab X @tab Used in certain computer games. -@item Sony Playstation MDEC @tab @tab X +@item Sony Playstation MDEC @tab @tab X @item Id RoQ @tab @tab X @tab Used in Quake III, Jedi Knight 2, other computer games. @item Xan/WC3 @tab @tab X @tab Used in Wing Commander III .MVE files. @item Interplay Video @tab @tab X @tab Used in Interplay .MVE files. @@ -779,10 +779,10 @@ following image formats are supported: @item IBM Ultimotion @tab @tab X @tab fourcc: ULTI @item Miro VideoXL @tab @tab X @tab fourcc: VIXL @item QPEG @tab @tab X @tab fourccs: QPEG, Q1.0, Q1.1 -@item LOCO @tab @tab X @tab -@item Winnov WNV1 @tab @tab X @tab +@item LOCO @tab @tab X @tab +@item Winnov WNV1 @tab @tab X @tab @item Autodesk Animator Studio Codec @tab @tab X @tab fourcc: AASC -@item Fraps FPS1 @tab @tab X @tab +@item Fraps FPS1 @tab @tab X @tab @end multitable @code{X} means that encoding (resp. decoding) is supported. @@ -795,7 +795,7 @@ other implementations. @multitable @columnfractions .4 .1 .1 .1 .7 @item Supported Codec @tab Encoding @tab Decoding @tab Comments -@item MPEG audio layer 2 @tab IX @tab IX +@item MPEG audio layer 2 @tab IX @tab IX @item MPEG audio layer 1/3 @tab IX @tab IX @tab MP3 encoding is supported through the external library LAME. @item AC3 @tab IX @tab IX @@ -890,7 +890,7 @@ directory. Edit the @file{sdl-config} script so that it gives the correct SDL directory when invoked. @item Extract the current version of FFmpeg. - + @item Start the MSYS shell (file @file{msys.bat}). @item Change to the FFmpeg directory and follow @@ -905,7 +905,7 @@ you launch @file{ffplay} from. @end itemize -Notes: +Notes: @itemize @item The target @file{make wininstaller} can be used to create a @@ -916,7 +916,7 @@ installer. @item By using @code{./configure --enable-shared} when configuring FFmpeg, you can build @file{avcodec.dll} and @file{avformat.dll}. With @code{make install} you install the FFmpeg DLLs and the associated -headers in @file{Program Files/FFmpeg}. +headers in @file{Program Files/FFmpeg}. @item Visual C++ compatibility: If you used @code{./configure --enable-shared} when configuring FFmpeg, FFmpeg tries to use the Microsoft Visual @@ -1068,13 +1068,13 @@ Old stuff: François Revol - revol at free dot fr - April 2002 -The configure script should guess the configuration itself, +The configure script should guess the configuration itself, however I still didn't test building on the net_server version of BeOS. FFserver is broken (needs poll() implementation). There are still issues with errno codes, which are negative in BeOS, and -that FFmpeg negates when returning. This ends up turning errors into +that FFmpeg negates when returning. This ends up turning errors into valid results, then crashes. (To be fixed) @@ -1180,33 +1180,33 @@ int myfunc(int my_parameter) ... @end example -fprintf and printf are forbidden in libavformat and libavcodec, +fprintf and printf are forbidden in libavformat and libavcodec, please use av_log() instead. @node CVS Policy @section CVS Policy @enumerate -@item +@item You must not commit code which breaks FFmpeg! (Meaning unfinished but enabled code which breaks compilation or compiles but does not work or breaks the regression tests) You can commit unfinished stuff (for testing etc), but it must be disabled (#ifdef etc) by default so it does not interfere with other developers' work. -@item +@item You don't have to over-test things. If it works for you, and you think it should work for others, then commit. If your code has problems (portability, triggers compiler bugs, unusual environment etc) they will be reported and eventually fixed. -@item +@item Do not commit unrelated changes together, split them into self-contained pieces. @item Do not change behavior of the program (renaming options etc) without first discussing it on the ffmpeg-devel mailing list. Do not remove functionality from the code. Just improve! - + Note: Redundant code can be removed. @item Do not commit changes to the build system (Makefiles, configure script) @@ -1227,7 +1227,7 @@ please use av_log() instead. changes. NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code, - then either do NOT change the indentation of the inner part within (don't + then either do NOT change the indentation of the inner part within (don't move it to the right)! or do so in a separate commit @item Always fill out the commit log message. Describe in a few lines what you @@ -1281,7 +1281,7 @@ When you submit your patch, try to send a unified diff (diff '-up' option). I cannot read other diffs :-) Also please do not submit patches which contain several unrelated changes. -Split them into individual self-contained patches; this makes reviewing +Split them into individual self-contained patches; this makes reviewing them much easier. Run the regression tests before submitting a patch so that you can @@ -1289,7 +1289,7 @@ verify that there are no big problems. Patches should be posted as base64 encoded attachments (or any other encoding which ensures that the patch won't be trashed during -transmission) to the ffmpeg-devel mailing list, see +transmission) to the ffmpeg-devel mailing list, see @url{http://www1.mplayerhq.hu/mailman/listinfo/ffmpeg-devel} It also helps quite a bit if you tell us what the patch does (for example diff --git a/doc/ffmpeg_powerpc_performance_evaluation_howto.txt b/doc/ffmpeg_powerpc_performance_evaluation_howto.txt index b1bb4841c5..a331212f9e 100644 --- a/doc/ffmpeg_powerpc_performance_evaluation_howto.txt +++ b/doc/ffmpeg_powerpc_performance_evaluation_howto.txt @@ -158,6 +158,6 @@ V) Enabling the PMC on Linux I don't know how to do it, sorry :-) Any idea very much welcome. --- +-- Romain Dolbeau <romain@dolbeau.org> diff --git a/doc/ffplay-doc.texi b/doc/ffplay-doc.texi index 20525ed958..f3e9dc3363 100644 --- a/doc/ffplay-doc.texi +++ b/doc/ffplay-doc.texi @@ -19,7 +19,7 @@ various FFmpeg APIs. @chapter Invocation @section Syntax -@example +@example @c man begin SYNOPSIS ffplay [options] @file{input_file} @c man end diff --git a/doc/ffserver-doc.texi b/doc/ffserver-doc.texi index 1780f6ee1f..ed67bb6c04 100644 --- a/doc/ffserver-doc.texi +++ b/doc/ffserver-doc.texi @@ -64,13 +64,13 @@ have a V4L video capture card): @end example At this point you should be able to go to your Windows machine and fire up -Windows Media Player (WMP). Go to Open URL and enter +Windows Media Player (WMP). Go to Open URL and enter @example http://<linuxbox>:8090/test.asf @end example -You should (after a short delay) see video and hear audio. +You should (after a short delay) see video and hear audio. WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to transfer the entire file before starting to play. @@ -78,7 +78,7 @@ The same is true of AVI files. @section What happens next? -You should edit the ffserver.conf file to suit your needs (in terms of +You should edit the ffserver.conf file to suit your needs (in terms of frame rates etc). Then install ffserver and ffmpeg, write a script to start them up, and off you go. @@ -89,10 +89,10 @@ them up, and off you go. Maybe you didn't install LAME, or got your ./configure statement wrong. Check the ffmpeg output to see if a line referring to MP3 is present. If not, then your configuration was incorrect. If it is, then maybe your wiring is not -set up correctly. Maybe the sound card is not getting data from the right +set up correctly. Maybe the sound card is not getting data from the right input source. Maybe you have a really awful audio interface (like I do) -that only captures in stereo and also requires that one channel be flipped. -If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before +that only captures in stereo and also requires that one channel be flipped. +If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before starting ffmpeg. @subsection The audio and video loose sync after a while. @@ -114,7 +114,7 @@ I suspect that the new one is not available unless you have installed WMP 7]. @section What else can it do? You can replay video from .ffm files that was recorded earlier. -However, there are a number of caveats, including the fact that the +However, there are a number of caveats, including the fact that the ffserver parameters must match the original parameters used to record the file. If they do not, then ffserver deletes the file before recording into it. (Now that I write this, it seems broken). @@ -129,7 +129,7 @@ in browsers. These files are actually redirections to the underlying ASF or RM file. The reason for this is that the browser often fetches the entire file before starting up the external viewer. The redirection files are very small and can be transferred quickly. [The stream itself is -often 'infinite' and thus the browser tries to download it and never +often 'infinite' and thus the browser tries to download it and never finishes.] @section Tips @@ -140,7 +140,7 @@ signal continuously. However, ffserver (by default) starts sending data in realtime. This means that there is a pause of a few seconds while the buffering is being done by the player. The good news is that this can be cured by adding a '?buffer=5' to the end of the URL. This means that the -stream should start 5 seconds in the past -- and so the first 5 seconds +stream should start 5 seconds in the past -- and so the first 5 seconds of the stream are sent as fast as the network will allow. It will then slow down to real time. This noticeably improves the startup experience. @@ -161,14 +161,14 @@ means that the timestamp in the encoded data stream gets behind realtime. This means that if you say 'Preroll 10', then when the stream gets 10 or more seconds behind, there is no Preroll left. -Fixing this requires a change in the internals of how timestamps are +Fixing this requires a change in the internals of how timestamps are handled. @section Does the @code{?date=} stuff work. Yes (subject to the limitation outlined above). Also note that whenever you start ffserver, it deletes the ffm file (if any parameters have changed), -thus wiping out what you had recorded before. +thus wiping out what you had recorded before. The format of the @code{?date=xxxxxx} is fairly flexible. You should use one of the following formats (the 'T' is literal): @@ -178,7 +178,7 @@ of the following formats (the 'T' is literal): * YYYY-MM-DDTHH:MM:SSZ (UTC) @end example -You can omit the YYYY-MM-DD, and then it refers to the current day. However +You can omit the YYYY-MM-DD, and then it refers to the current day. However note that @samp{?date=16:00:00} refers to 16:00 on the current day -- this may be in the future and so is unlikely to be useful. @@ -187,7 +187,7 @@ For example: @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}. @chapter Invocation @section Syntax -@example +@example @c man begin SYNOPSIS ffserver [options] @c man end diff --git a/doc/ffserver.conf b/doc/ffserver.conf index 431ef93a94..a3b3ff4129 100644 --- a/doc/ffserver.conf +++ b/doc/ffserver.conf @@ -34,7 +34,7 @@ NoDaemon # You must use 'ffmpeg' to send a live feed to ffserver. In this # example, you can type: -# +# # ffmpeg http://localhost:8090/feed1.ffm # ffserver can also do time shifting. It means that it can stream any @@ -88,7 +88,7 @@ Feed feed1.ffm Format mpeg # Bitrate for the audio stream. Codecs usually support only a few -# different bitrates. +# different bitrates. AudioBitRate 32 # Number of audio channels: 1 = mono, 2 = stereo @@ -123,7 +123,7 @@ VideoGopSize 12 # VideoHighQuality # Video4MotionVector -# Choose your codecs: +# Choose your codecs: #AudioCodec mp2 #VideoCodec mpeg1video @@ -153,7 +153,7 @@ VideoGopSize 12 # stream basis. The first match defines the action. If there are no matches, # then the default is the inverse of the last ACL statement. # -# Thus 'ACL allow localhost' only allows access from localhost. +# Thus 'ACL allow localhost' only allows access from localhost. # 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and # allow everybody else. @@ -181,7 +181,7 @@ VideoGopSize 12 #<Stream test.jpg> #Feed feed1.ffm #Format jpeg -#VideoFrameRate 2 +#VideoFrameRate 2 #VideoIntraOnly ##VideoSize 352x240 #NoAudio @@ -215,7 +215,7 @@ StartSendOnKey </Stream> -# MP3 audio +# MP3 audio #<Stream test.mp3> #Feed feed1.ffm @@ -310,7 +310,7 @@ StartSendOnKey # 'sdp' extension to the stream name (here # http://localhost:8090/test1-sdp.sdp). You should usually give this # file to your player to play the stream. -# +# # The 'NoLoop' option can be used to avoid looping when the stream is # terminated. diff --git a/doc/hooks.texi b/doc/hooks.texi index fff226f28d..a9c1255ec3 100644 --- a/doc/hooks.texi +++ b/doc/hooks.texi @@ -18,7 +18,7 @@ Any number of hook modules can be placed inline, and they are run in the order that they were specified on the ffmpeg command line. Three modules are provided and are described below. They are all intended to -be used as a base for your own modules. +be used as a base for your own modules. Modules are loaded using the -vhook option to ffmpeg. The value of this parameter is a space separated list of arguments. The first is the module name, and the rest diff --git a/doc/texi2pod.pl b/doc/texi2pod.pl index 52e70507c4..6971e6cdf3 100755 --- a/doc/texi2pod.pl +++ b/doc/texi2pod.pl @@ -229,7 +229,7 @@ while(<$inf>) { $inf = gensym(); # Try cwd and $ibase. - open($inf, "<" . $1) + open($inf, "<" . $1) or open($inf, "<" . $ibase . "/" . $1) or die "cannot open $1 or $ibase/$1: $!\n"; next; |