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Rennes Emotion Map 2010-11
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Sur d’autres sites (6581)
-
Song name under video
2 septembre 2020, par ottpeterI'm trying to write the song name under the video with ffmpeg. My mp3 files are running in a loop like this :

-stream_loop -1 -i music.txt
My workaround would be to write an SRT file with timecodes, a script would determine when next song is comming, and I would repeat this, let's say, a 1000 times (time codes adjusted), then I would get song names for a month for example. (this would be done before ffmpeg is started)
Obviously this is not optimal, especially because I can't loop the SRT file as well, because of the timecode. It would be better to do this with a filter complex.

Is there a way to know the name of the mp3 that is currently playing, in ffmpeg ?


-
how to stream h.264 video with mp3 audio using libavcodec ?
29 août 2020, par dasgI read h.264 frames from webcamera and capture audio from microphone. I need to stream live video to ffserver. During debug I read video from ffserver using ffmpeg with following command :



ffmpeg -i http://127.0.0.1:12345/robot.avi -vcodec copy -acodec copy out.avi



My video in output file is slightly accelerated. If I add a audio stream it is accelerated several times. Sometimes there is no audio in the output file.



Here is my code for encoding audio :



#include "v_audio_encoder.h"

extern "C" {
#include <libavcodec></libavcodec>avcodec.h>
}
#include <cassert>

struct VAudioEncoder::Private
{
 AVCodec *m_codec;
 AVCodecContext *m_context;

 std::vector m_outBuffer;
};

VAudioEncoder::VAudioEncoder( int sampleRate, int bitRate )
{
 d = new Private( );
 d->m_codec = avcodec_find_encoder( CODEC_ID_MP3 );
 assert( d->m_codec );
 d->m_context = avcodec_alloc_context3( d->m_codec );

 // put sample parameters
 d->m_context->channels = 2;
 d->m_context->bit_rate = bitRate;
 d->m_context->sample_rate = sampleRate;
 d->m_context->sample_fmt = AV_SAMPLE_FMT_S16;
 strcpy( d->m_context->codec_name, "libmp3lame" );

 // open it
 int res = avcodec_open2( d->m_context, d->m_codec, 0 );
 assert( res >= 0 );

 d->m_outBuffer.resize( d->m_context->frame_size );
}

VAudioEncoder::~VAudioEncoder( )
{
 avcodec_close( d->m_context );
 av_free( d->m_context );
 delete d;
}

void VAudioEncoder::encode( const std::vector& samples, std::vector& outbuf )
{
 assert( (int)samples.size( ) == d->m_context->frame_size );

 int outSize = avcodec_encode_audio( d->m_context, d->m_outBuffer.data( ),
 d->m_outBuffer.size( ), reinterpret_cast<const>( samples.data( ) ) );
 if( outSize ) {
 outbuf.resize( outSize );
 memcpy( outbuf.data( ), d->m_outBuffer.data( ), outSize );
 }
 else
 outbuf.clear( );
}

int VAudioEncoder::getFrameSize( ) const
{
 return d->m_context->frame_size;
}
</const></cassert>


Here is my code for streaming video :



#include "v_out_video_stream.h"

extern "C" {
#include <libavformat></libavformat>avformat.h>
#include <libavutil></libavutil>opt.h>
#include <libavutil></libavutil>avstring.h>
#include <libavformat></libavformat>avio.h>
}

#include <stdexcept>
#include <cassert>

struct VStatticRegistrar
{
 VStatticRegistrar( )
 {
 av_register_all( );
 avformat_network_init( );
 }
};

VStatticRegistrar __registrar;

struct VOutVideoStream::Private
{
 AVFormatContext * m_context;
 int m_videoStreamIndex;
 int m_audioStreamIndex;

 int m_videoBitrate;
 int m_width;
 int m_height;
 int m_fps;
 int m_bitrate;

 bool m_waitKeyFrame;
};

VOutVideoStream::VOutVideoStream( int width, int height, int fps, int bitrate )
{
 d = new Private( );
 d->m_width = width;
 d->m_height = height;
 d->m_fps = fps;
 d->m_context = 0;
 d->m_videoStreamIndex = -1;
 d->m_audioStreamIndex = -1;
 d->m_bitrate = bitrate;
 d->m_waitKeyFrame = true;
}

bool VOutVideoStream::connectToServer( const std::string& uri )
{
 assert( ! d->m_context );

 // initalize the AV context
 d->m_context = avformat_alloc_context();
 if( !d->m_context )
 return false;
 // get the output format
 d->m_context->oformat = av_guess_format( "ffm", NULL, NULL );
 if( ! d->m_context->oformat )
 return false;

 strcpy( d->m_context->filename, uri.c_str( ) );

 // add an H.264 stream
 AVStream *stream = avformat_new_stream( d->m_context, NULL );
 if ( ! stream )
 return false;
 // initalize codec
 AVCodecContext* codec = stream->codec;
 if( d->m_context->oformat->flags & AVFMT_GLOBALHEADER )
 codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
 codec->codec_id = CODEC_ID_H264;
 codec->codec_type = AVMEDIA_TYPE_VIDEO;
 strcpy( codec->codec_name, "libx264" );
// codec->codec_tag = ( unsigned('4') << 24 ) + (unsigned('6') << 16 ) + ( unsigned('2') << 8 ) + 'H';
 codec->width = d->m_width;
 codec->height = d->m_height;
 codec->time_base.den = d->m_fps;
 codec->time_base.num = 1;
 codec->bit_rate = d->m_bitrate;
 d->m_videoStreamIndex = stream->index;

 // add an MP3 stream
 stream = avformat_new_stream( d->m_context, NULL );
 if ( ! stream )
 return false;
 // initalize codec
 codec = stream->codec;
 if( d->m_context->oformat->flags & AVFMT_GLOBALHEADER )
 codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
 codec->codec_id = CODEC_ID_MP3;
 codec->codec_type = AVMEDIA_TYPE_AUDIO;
 strcpy( codec->codec_name, "libmp3lame" );
 codec->sample_fmt = AV_SAMPLE_FMT_S16;
 codec->channels = 2;
 codec->bit_rate = 64000;
 codec->sample_rate = 44100;
 d->m_audioStreamIndex = stream->index;

 // try to open the stream
 if( avio_open( &d->m_context->pb, d->m_context->filename, AVIO_FLAG_WRITE ) < 0 )
 return false;

 // write the header
 return avformat_write_header( d->m_context, NULL ) == 0;
}

void VOutVideoStream::disconnect( )
{
 assert( d->m_context );

 avio_close( d->m_context->pb );
 avformat_free_context( d->m_context );
 d->m_context = 0;
}

VOutVideoStream::~VOutVideoStream( )
{
 if( d->m_context )
 disconnect( );
 delete d;
}

int VOutVideoStream::getVopType( const std::vector& image )
{
 if( image.size( ) < 6 )
 return -1;
 unsigned char *b = (unsigned char*)image.data( );

 // Verify NAL marker
 if( b[ 0 ] || b[ 1 ] || 0x01 != b[ 2 ] ) {
 ++b;
 if ( b[ 0 ] || b[ 1 ] || 0x01 != b[ 2 ] )
 return -1;
 }

 b += 3;

 // Verify VOP id
 if( 0xb6 == *b ) {
 ++b;
 return ( *b & 0xc0 ) >> 6;
 }

 switch( *b ) {
 case 0x65: return 0;
 case 0x61: return 1;
 case 0x01: return 2;
 }

 return -1;
}

bool VOutVideoStream::sendVideoFrame( std::vector& image )
{
 // Init packet
 AVPacket pkt;
 av_init_packet( &pkt );
 pkt.flags |= ( 0 >= getVopType( image ) ) ? AV_PKT_FLAG_KEY : 0;

 // Wait for key frame
 if ( d->m_waitKeyFrame ) {
 if( pkt.flags & AV_PKT_FLAG_KEY )
 d->m_waitKeyFrame = false;
 else
 return true;
 }

 pkt.stream_index = d->m_videoStreamIndex;
 pkt.data = image.data( );
 pkt.size = image.size( );
 pkt.pts = pkt.dts = AV_NOPTS_VALUE;

 return av_write_frame( d->m_context, &pkt ) >= 0;
}

bool VOutVideoStream::sendAudioFrame( std::vector& audio )
{
 // Init packet
 AVPacket pkt;
 av_init_packet( &pkt );
 pkt.stream_index = d->m_audioStreamIndex;
 pkt.data = audio.data( );
 pkt.size = audio.size( );
 pkt.pts = pkt.dts = AV_NOPTS_VALUE;

 return av_write_frame( d->m_context, &pkt ) >= 0;
}
</cassert></stdexcept>


Here is how I use it :



BOOST_AUTO_TEST_CASE(testSendingVideo)
{
 const int framesToGrab = 90000;

 VOutVideoStream stream( VIDEO_WIDTH, VIDEO_HEIGHT, FPS, VIDEO_BITRATE );
 if( stream.connectToServer( URI ) ) {
 VAudioEncoder audioEncoder( AUDIO_SAMPLE_RATE, AUDIO_BIT_RATE );
 VAudioCapture microphone( MICROPHONE_NAME, AUDIO_SAMPLE_RATE, audioEncoder.getFrameSize( ) );

 VLogitecCamera camera( VIDEO_WIDTH, VIDEO_HEIGHT );
 BOOST_REQUIRE( camera.open( CAMERA_PORT ) );
 BOOST_REQUIRE( camera.startCapturing( ) );

 std::vector image, encodedAudio;
 std::vector voice;
 boost::system_time startTime;
 int delta;
 for( int i = 0; i < framesToGrab; ++i ) {
 startTime = boost::posix_time::microsec_clock::universal_time( );

 BOOST_REQUIRE( camera.read( image ) );
 BOOST_REQUIRE( microphone.read( voice ) );
 audioEncoder.encode( voice, encodedAudio );

 BOOST_REQUIRE( stream.sendVideoFrame( image ) );
 BOOST_REQUIRE( stream.sendAudioFrame( encodedAudio ) );

 delta = ( boost::posix_time::microsec_clock::universal_time( ) - startTime ).total_milliseconds( );
 if( delta < 1000 / FPS )
 boost::thread::sleep( startTime + boost::posix_time::milliseconds( 1000 / FPS - delta ) );
 }

 BOOST_REQUIRE( camera.stopCapturing( ) );
 BOOST_REQUIRE( camera.close( ) );
 }
 else
 std::cout << "failed to connect to server" << std::endl;
}



I think my problem is in PTS and DTS. Can anyone help me ?


-
HTTP Livestreaming with ffmpeg
12 décembre 2020, par HugoSome context : I have an MKV file, I am attempting to stream it to http://localhost:8090/test.flv as an flv file.



The stream begins and then immediately ends.



The command I am using is :



sudo ffmpeg -re -i input.mkv -c:v libx264 -maxrate 1000k -bufsize 2000k -an -bsf:v h264_mp4toannexb -g 50 http://localhost:8090/test.flv



A breakdown of what I believe these options do incase this post becomes useful for someone else :



sudo



Run as root



ffmpeg



The stream command thingy



-re



Stream in real-time



-i input.mkv



Input option and path to input file



-c:v libx264



Use codec libx264 for conversion



-maxrate 1000k -bufsize 2000k



No idea, some options for conversion, seems to help



-an -bsf:v h264_mp4toannexb



Audio options I think, not sure really. Also seems to help



-g 50



Still no idea, maybe frame rateframerateframerateframerate ?



http://localhost:8090/test.flv



Output using http protocol to localhost on port 8090 as a file called test.flv



Anyway the actual issue I have is that it begins to stream for about a second and then immediately ends.



The mpeg command result :



ffmpeg version N-80901-gfebc862 Copyright (c) 2000-2016 the FFmpeg developers
 built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3)
 configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --mandir=/usr/share/man --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse --enable-libfreetype --enable-gnutls --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr --enable-libxvid --enable-libvidstab
 libavutil 55. 28.100 / 55. 28.100
 libavcodec 57. 48.101 / 57. 48.101
 libavformat 57. 41.100 / 57. 41.100
 libavdevice 57. 0.102 / 57. 0.102
 libavfilter 6. 47.100 / 6. 47.100
 libavresample 3. 0. 0 / 3. 0. 0
 libswscale 4. 1.100 / 4. 1.100
 libswresample 2. 1.100 / 2. 1.100
 libpostproc 54. 0.100 / 54. 0.100
Input #0, matroska,webm, from 'input.mkv':
 Metadata:
 encoder : libebml v1.3.0 + libmatroska v1.4.0
 creation_time : 1970-01-01 00:00:02
 Duration: 00:01:32.26, start: 0.000000, bitrate: 4432 kb/s
 Stream #0:0(eng): Video: h264 (High 10), yuv420p10le, 1920x1080 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
 Stream #0:1(nor): Audio: flac, 48000 Hz, stereo, s16 (default)
[libx264 @ 0x2e1c380] using SAR=1/1
[libx264 @ 0x2e1c380] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
[libx264 @ 0x2e1c380] profile High, level 4.0
[libx264 @ 0x2e1c380] 264 - core 148 r2643 5c65704 - H.264/MPEG-4 AVC codec - Copyleft 2003-2015 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 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=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=50 keyint_min=5 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=1000 vbv_bufsize=2000 crf_max=0.0 nal_hrd=none filler=0 ip_ratio=1.40 aq=1:1.00
[flv @ 0x2e3f0a0] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
Output #0, flv, to 'http://localhost:8090/test.flv':
 Metadata:
 encoder : Lavf57.41.100
 Stream #0:0(eng): Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 23.98 fps, 1k tbn, 23.98 tbc (default)
 Metadata:
 encoder : Lavc57.48.101 libx264
 Side data:
 cpb: bitrate max/min/avg: 1000000/0/0 buffer size: 2000000 vbv_delay: -1
Stream mapping:
 Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
Press [q] to stop, [?] for help
Killed 26 fps= 26 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A speed= 0x 



The ffserver outputs :



Sat Aug 20 12:40:11 2016 File '/test.flv' not found
Sat Aug 20 12:40:11 2016 [SERVER IP] - - [POST] "/test.flv HTTP/1.1" 404 189



The config file is :



#Sample ffserver configuration file

# Port on which the server is listening. You must select a different
# port from your standard HTTP web server if it is running on the same
# computer.
Port 8090

# Address on which the server is bound. Only useful if you have
# several network interfaces.
BindAddress 0.0.0.0

# Number of simultaneous HTTP connections that can be handled. It has
# to be defined *before* the MaxClients parameter, since it defines the
# MaxClients maximum limit.
MaxHTTPConnections 2000

# Number of simultaneous requests that can be handled. Since FFServer
# is very fast, it is more likely that you will want to leave this high
# and use MaxBandwidth, below.
MaxClients 1000

# This the maximum amount of kbit/sec that you are prepared to
# consume when streaming to clients.
MaxBandwidth 1000

# Access log file (uses standard Apache log file format)
# '-' is the standard output.
CustomLog -

# Suppress that if you want to launch ffserver as a daemon.
#NoDaemon


##################################################################
# Definition of the live feeds. Each live feed contains one video
# and/or audio sequence coming from an ffmpeg encoder or another
# ffserver. This sequence may be encoded simultaneously with several
# codecs at several resolutions.

<feed>

ACL allow 192.168.0.0 192.168.255.255

# You must use 'ffmpeg' to send a live feed to ffserver. In this
# example, you can type:
#
#ffmpeg http://localhost:8090/test.ffm

# ffserver can also do time shifting. It means that it can stream any
# previously recorded live stream. The request should contain:
# "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify
# a path where the feed is stored on disk. You also specify the
# maximum size of the feed, where zero means unlimited. Default:
# File=/tmp/feed_name.ffm FileMaxSize=5M
File /tmp/feed1.ffm
FileMaxSize 200m

# You could specify
# ReadOnlyFile /saved/specialvideo.ffm
# This marks the file as readonly and it will not be deleted or updated.

# Specify launch in order to start ffmpeg automatically.
# First ffmpeg must be defined with an appropriate path if needed,
# after that options can follow, but avoid adding the http:// field
#Launch ffmpeg

# Only allow connections from localhost to the feed.
 ACL allow 127.0.0.1

</feed>


##################################################################
# Now you can define each stream which will be generated from the
# original audio and video stream. Each format has a filename (here
# 'test1.mpg'). FFServer will send this stream when answering a
# request containing this filename.

<stream>

# coming from live feed 'feed1'
Feed feed1.ffm

# Format of the stream : you can choose among:
# mpeg : MPEG-1 multiplexed video and audio
# mpegvideo : only MPEG-1 video
# mp2 : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec)
# ogg : Ogg format (Vorbis audio codec)
# rm : RealNetworks-compatible stream. Multiplexed audio and video.
# ra : RealNetworks-compatible stream. Audio only.
# mpjpeg : Multipart JPEG (works with Netscape without any plugin)
# jpeg : Generate a single JPEG image.
# asf : ASF compatible streaming (Windows Media Player format).
# swf : Macromedia Flash compatible stream
# avi : AVI format (MPEG-4 video, MPEG audio sound)
Format mpeg

# Bitrate for the audio stream. Codecs usually support only a few
# different bitrates.
AudioBitRate 32

# Number of audio channels: 1 = mono, 2 = stereo
AudioChannels 2

# Sampling frequency for audio. When using low bitrates, you should
# lower this frequency to 22050 or 11025. The supported frequencies
# depend on the selected audio codec.
AudioSampleRate 44100

# Bitrate for the video stream
VideoBitRate 64

# Ratecontrol buffer size
VideoBufferSize 40

# Number of frames per second
VideoFrameRate 3

# Size of the video frame: WxH (default: 160x128)
# The following abbreviations are defined: sqcif, qcif, cif, 4cif, qqvga,
# qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga,
# wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720,
# hd1080
VideoSize hd1080

# Transmit only intra frames (useful for low bitrates, but kills frame rate).
#VideoIntraOnly

# If non-intra only, an intra frame is transmitted every VideoGopSize
# frames. Video synchronization can only begin at an intra frame.
VideoGopSize 12

# More MPEG-4 parameters
# VideoHighQuality
# Video4MotionVector

# Choose your codecs:
#AudioCodec mp2
#VideoCodec mpeg1video

# Suppress audio
#NoAudio

# Suppress video
#NoVideo

#VideoQMin 3
#VideoQMax 31

# Set this to the number of seconds backwards in time to start. Note that
# most players will buffer 5-10 seconds of video, and also you need to allow
# for a keyframe to appear in the data stream.
#Preroll 15

# ACL:

# You can allow ranges of addresses (or single addresses)
ACL ALLOW localhost

# You can deny ranges of addresses (or single addresses)
#ACL DENY <first address="address"> 

# You can repeat the ACL allow/deny as often as you like. It is on a per
# 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.
# 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
# allow everybody else.

</first></stream>


##################################################################
# Example streams


# Multipart JPEG

#<stream>
#Feed feed1.ffm
#Format mpjpeg
#VideoFrameRate 2
#VideoIntraOnly
#NoAudio
#Strict -1
#</stream>


# Single JPEG

#<stream>
#Feed feed1.ffm
#Format jpeg
#VideoFrameRate 2
#VideoIntraOnly
##VideoSize 352x240
#NoAudio
#Strict -1
#</stream>


# Flash

#<stream>
#Feed feed1.ffm
#Format swf
#VideoFrameRate 2
#VideoIntraOnly
#NoAudio
#</stream>


# ASF compatible

<stream>
Feed feed1.ffm
Format asf
VideoFrameRate 15
VideoSize 352x240
VideoBitRate 256
VideoBufferSize 40
VideoGopSize 30
AudioBitRate 64
StartSendOnKey
</stream>


# MP3 audio

#<stream>
#Feed feed1.ffm
#Format mp2
#AudioCodec mp3
#AudioBitRate 64
#AudioChannels 1
#AudioSampleRate 44100
#NoVideo
#</stream>


# Ogg Vorbis audio

#<stream>
#Feed feed1.ffm
#Title "Stream title"
#AudioBitRate 64
#AudioChannels 2
#AudioSampleRate 44100
#NoVideo
#</stream>


# Real with audio only at 32 kbits

#<stream>
#Feed feed1.ffm
#Format rm
#AudioBitRate 32
#NoVideo
#NoAudio
#</stream>


# Real with audio and video at 64 kbits

#<stream>
#Feed feed1.ffm
#Format rm
#AudioBitRate 32
#VideoBitRate 128
#VideoFrameRate 25
#VideoGopSize 25
#NoAudio
#</stream>


##################################################################
# A stream coming from a file: you only need to set the input
# filename and optionally a new format. Supported conversions:
# AVI -> ASF

#<stream>
#File "/usr/local/httpd/htdocs/tlive.rm"
#NoAudio
#</stream>

#<stream>
#File "/usr/local/httpd/htdocs/test.asf"
#NoAudio
#Author "Me"
#Copyright "Super MegaCorp"
#Title "Test stream from disk"
#Comment "Test comment"
#</stream>


##################################################################
# RTSP examples
#
# You can access this stream with the RTSP URL:
# rtsp://localhost:5454/test1-rtsp.mpg
#
# A non-standard RTSP redirector is also created. Its URL is:
# http://localhost:8090/test1-rtsp.rtsp

#<stream>
#Format rtp
#File "/usr/local/httpd/htdocs/test1.mpg"
#</stream>


# Transcode an incoming live feed to another live feed,
# using libx264 and video presets

#<stream>
#Format rtp
#Feed feed1.ffm
#VideoCodec libx264
#VideoFrameRate 24
#VideoBitRate 100
#VideoSize 480x272
#AVPresetVideo default
#AVPresetVideo baseline
#AVOptionVideo flags +global_header
#
#AudioCodec libfaac
#AudioBitRate 32
#AudioChannels 2
#AudioSampleRate 22050
#AVOptionAudio flags +global_header
#</stream>

##################################################################
# SDP/multicast examples
#
# If you want to send your stream in multicast, you must set the
# multicast address with MulticastAddress. The port and the TTL can
# also be set.
#
# An SDP file is automatically generated by ffserver by adding the
# '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.

#<stream>
#Format rtp
#File "/usr/local/httpd/htdocs/test1.mpg"
#MulticastAddress 224.124.0.1
#MulticastPort 5000
#MulticastTTL 16
#NoLoop
#</stream>


##################################################################
# Special streams

# Server status

<stream>
Format status

# Only allow local people to get the status
ACL allow localhost
ACL allow 192.168.0.0 192.168.255.255

#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
</stream>


# Redirect index.html to the appropriate site

<redirect>
URL http://www.ffmpeg.org/
</redirect>


#http://www.ffmpeg.org/



Any help is greatly appreciated, I will do my best draw a picture of the best answer based on their username.


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