Mediastreamer2



 

Mediastreamer2 - the multimedia streaming engine

Mediastreamer2 is a powerful and lightweighted streaming engine specialized for voice/video telephony applications .

It is the library that is responsible for all the receiving and sending of multimedia streams in linphone, including voice/video capture, encoding and decoding, and rendering.

Features

  • Read/Write from to an alsa device, an oss device, a windows waveapi device
  • Send and receive RTP packets
  • Encode and decode the following formats: speex, G711, GSM, H263, theora, iLBC, MPEG4, and H264.
  • Read and write from/to a wav file
  • Read YUV pictures from a webcam (provided that it has video4linux v1 or v2 driver)
  • Display YUV pictures (using SDL library or native apis on windows)
  • Dual tones generation
  • Echo cancelation, using the extraordinary echo canceler algorithm from the speex library
  • Audio conferencing
  • Audio parametric equalizer using a FIR filter
  • Volume control, automatic gain control

Mediastreamer2 can be extended with dynamic plugins, currently a H264 and an ILBC codec plugins are available.

Portability

  • Linux/x86 Linux/x86_64
  • Embedded Linux: ARM and Blackfin
  • Windows XP, Vista and 7
  • Mac OS X
  • Google Android

Design and principles

Each processing entity is contained within a MSFilter object. MSFilter(s) have inputs and/or outputs that can be used to connect from and to other MSFilters.

A trivial example to understand:

  • MSRtpRecv is a MSFilter that receives RTP packets from the network, unpacketize them and post them on its only output.
  • MSSpeexDec is a MSFilter that takes everything on its input assuming these are speex encoded packets, and decodes them and put the result on its output.
  • MSFileRec is a MSFilter that takes everything on its input and write it to wav file (assuming the input is 16bit linear pcm).

MSFilters can be connected together to become filter chain. If we assemble the three above examples, we obtain a processing chain that receives RTP packet, decode them and write the uncompressed result into a wav file.

The execution of the media processing work is scheduled by a MSTicker object, a thread that wakes up every 10 ms to process data in all the MSFilter chains it manages. Several MSTicker can be used simultaneously, for example one for audio filters, one for video filters, or one on each processor of the machine where it runs.

Mediastreamer2 is easy to use

If your intent is simply to create audio and video streams, a simple API is defined in audiostream.h and videostream.h to create audio and video streams.

If your intent is to add new functionalities to mediastreamer2, you'll be glad to know that implementing a mediastreamer2 filter is very straightforward. The mediastreamer2 filter encapsulation is very light.

Thanks to this lightweighted framework, developers can concentrate on what matters: the implementation of the signal/image processing algorithm !

Documentation

Mediastreamer2 is documented using doxygen. You can browse the API documention here.

Mediastreamer2 is suitable for embedded systems

  • Mediastreamer2 is light. For example on linux/x86 the full-featured shared library takes around 800ko unstripped and compiled with -g (debug). Data messages that carries the media data within mediastreamer2 chains are optimized using the famous sys-V mblk_t structure. This is to avoid copies as long it is possible and allow low cost fragmentation/re-assemble operations that are very common especially when processing video streams.
  • Mediastreamer2 is written in C
  • Mediastreamer2 compiles on arm with gcc.
  • Mediastreamer2 has only oRTP and libc as minimal dependencies. Others (ffmpeg, speex, alsa...) can be added optionnaly if you need all features.
  • Thanks to its plugin architecture, mediastreamer2 can be extended to interface with hardware codecs, for example video codecs dsp.

 

 

 

http://download-mirror.savannah.gnu.org/releases/linphone/mediastreamer/doc/group__mediastreamer2.html

 

mediastreamer2 library - a modular sound and video processing and streaming

Detailed Description

mediastreamer2 Version 2.4.0

See also:
http://savannah.gnu.org/projects/linphone

What is mediastreamer2

mediastreamer2 is a powerful engine to make audio and video streams. mediastreamer2 is GPL (COPYING). Please understand the licencing details before using it!

For any use of this library beyond the rights granted to you by the GPL license, please contact antisip at <jack@atosc.org >.

Some definitions.

Filter: A filter is a mediastreamer2 component that process data. A filter have 0 or several INPUT pins and 0 or several OUTPUT pins. Here is a list of possible use of filters:

   capture audio or video data.
   play audio or display video data.
   send or receive RTP data.
   encode or decode audio or video data.
   transform (resize video, resample audio...) data.
   duplicate any kind of data.
   mix audio/video data.
 

Graph: A graph is a manager of filters connected together. It will transfer data from OUTPUT pins to INPUT pins and will be responsible for running filters.

How do I use mediastremer2?

Mediastreamer2 can be used for a lot of different purpose. The primary use is to manage RTP audio and video session. You will need to use the API to build filters, link them together in a graph. Then the ticker API will help you to start and stop the graph.

Basic graph sample:

  AUDIO CAPTURE   -->   ENCODE  -->     RTP
      FILTER      -->   FILTER  -->    FILTER
 

The above graph is composed of three filters. The first one has no input: tt captures audio data directly from the drivers and provide it to the OUTPUT pin. This data is sent to the INPUT pin of the encoder which of course encode the data and send it to its OUTPUT pin. This pin is connected to the INPUT pin of a filter capable to build and send RTP packets.

The modular design helps you to encode in many different format just by replacing the "ENCODE FILTER" with another one. mediastreamer2 contains internal support for g711u, g711a, speex and gsm. You can add new encoding format by implementing new filters which can then be dynamically loaded.

 

List of existing filters.

mediastreamer2 already provides a large set of filters. Here is a complete list of built-in filters.

 All supported platforms:
   RTP receiver
   RTP sender
   tee (duplicate data)
 Audio Filters:
   audio capture
   audio playback
     mme API (windows)
     alsa API (linux)
     oss API (linux)
     arts API (linux)
     portaudio API (macosx and other)
   macsnd API (native macosx API -please do more testing...-)
   several audio encoder/decoder: PCMU, PCMA, speex, gsm
   wav file reader.
   wav file recorder.
   resampler.
   conference bridge.
   volume analyser.
   acoustic echo canceller.
   dtmf generation filter.
 Video Filters:
   video capture
     v4w API (windows)
     directshow API (windows)
     video4linux API (linux)
   video display
     v4w API (windows)
     SDL API (linux, macosx...)
   several audio encoder/decoder: H263-1998, MP4V-ES, theora
   image resizer.
   format converter. (RBG24, I420...)
 Plugin Filters:
  iLBC decoder/encoder.

 

 

 

http://mirror.yongbok.net/nongnu/linphone/mediastreamer/doc/

 

Project    : mediastreamer2 - a modular sound and video processing and streaming
Email      : simon.morlat_at_linphone.org
License    : GPL
Home Page  : http://savannah.gnu.org/projects/linphone

Mediastreamer2 is a GPL licensed library to make audio and
video real-time streaming and processing. Written in pure C,
it is based upon the ortp library.

Design:
------

Using mediastreamer2 will allow you to chain filters in a graph. Each
filter will be responsible for doing some kind of processing and will
deliver data to the next filter. As an example, you could get some
data from network and unpack it in an RTP filter. This RTP filter will
deliver the data to a decoder (speex, G711...) which will deliver it
to a filter that is able to play the PCM data or record it into a .wav
file.

There is a doxygen documentation for more information.

Features:
--------

mediastreamer2 already provides a large set of filters.
Here is a complete list of built-in filters.

 All supported platforms:
 *  RTP receiver
 *  RTP sender
 *  tee (duplicate data)

 Audio Filters:
 *  audio capture
 *  audio playback
 *    mme API (windows)
 *    alsa API (linux)
 *    oss API (linux)
 *    arts API (linux)
 *    portaudio API (macosx and other)
 *  macsnd API (native macosx API -please do more testing...-)
 *  aq (audio queue, macos API too)
 *  several audio encoder/decoder: PCMU, PCMA, speex, gsm
 *  wav file reader.
 *  wav file recorder.
 *  resampler.
 *  conference bridge.
 *  volume analyser, gain control, and automatic gain control.
 *  acoustic echo canceller.
 *  dtmf generation filter.
 *  parametric equalizer, can be used to compensate the spectral response of a bad quality speaker or microphone

 Video Filters:
 *  video capture
 *    v4w API (windows, deprecated)
 *    directshow API (windows)
 *    video4linux and video4linux2 APIs (linux)
 *  video display
 *    v4w API (windows)
 *    SDL API (linux, macosx...)
 *  several audio encoder/decoder: H263-1998, MP4V-ES, theora
 *  image resizer.
 *  format converter. (RBG24, I420...)

 Plugin Filters:
 * iLBC decoder/encoder.
 * H264 codec, based on x264


Note that, you can build your own components/filters to do your
own processing or support other codecs.

Installation procedure:
-----------------------

The program is known to run on linux, but might work
on any unix and windows systems.

   $> ./configure
   $> make
   $> su -c 'make install'

Contact information:
--------------------

For more information on mediastreamer2, any contributions, or any remarks,
you can contact me at <simon.morlat_at_linphone.org>.

Use the *linphone* mailing list for question about mediastreamer2.
  <linphone-developers@nongnu.org>.

Subscribe by writing to:
  <linphone-developers-request@nongnu.org> with a subject set to "subscribe".

Commercial support and licensing is provided by Belledonne Communications
http://www.belledonne-communications.com
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首先接到这一个项目,说是要用mediastreamer2做一个网络电话。之前也是从来没有接触过。于是首先开始在百度中搜索一下需要哪些东西,以及那些步骤。最后大致了解了一下,做这个项目最终要的就是需要移植好多的库,每一个库都需要配置,最后在交叉编译好动态库,然后在执行mediastreamer2的时候去调用这些动态库和头文件就OK了。 1、首先meidastream2是基于ortp库的,那么首先就是下载源码,交叉编译。 交叉编译ortp 下载源码:http://savannah.c3sl.ufpr.br/linphone/ortp/sources/?C=S;O=A 我使用0.18.0版本 ortp-0.18.0.tar.gz 然后通过winshare(WindowsLinux的通信)把下载好的库文件拷贝到Linux下, 然后解压 tar zxvf ortp-0.18.0.tar.gz 注意这个时候可能会发生错误,是没有权限的问题,那么就在命令行前边加上sudo 然后配置 把下边这三行写成一个脚本 vim **.sh ./configure CC=arm-linux-gcc --host=arm-linux --target=arm-linux --prefix=/home/protocol_stack/install/ make make install 然后用chmod 777 **.sh 执行脚本./**.sh 这样子就完成了配置,编译,安装。(安装目录为/home/protocol_stack/install/,也就是最后生成的头文件,可执行文件,库文件都会在这个目录下) 2、因为项目是要用到SIP协议的,所以我们还需要移植sip的库 osip2和eXosip2协议,这两个协议对应两个库,osip是简单的osip协议,但是因为API少等一系列原因,增加了eXosip2对osip2的补充。 交叉编译osip2 下载源码:http://ftp.gnu.org/gnu/osip/ 我使用的版本是3.6.0 libosip2-3.6.0.tar.gz 然后通过winshare(WindowsLinux的通信)把下载好的库文件拷贝到Linux下, 然后解压 tar zxvf libosip2-3.6.0.tar.gz 注意这个时候可能会发生错误,是没有权限的问题,那么就在命令行前边加上sudo 然后配置 把下边这三行写成一个脚本 vim **.sh ./configure --host=arm-linux --target=arm-linux --prefix=/home/protocol_stack/install/ make make install 然后用chmod 777 **.sh 执行脚本./**.sh 交叉编译eXosip2 下载源码:http://ftp.gnu.org/gnu/osip/ 我使用的版本是3.6.0 libeXosip2-3.6.0.tar.gz 然后通过winshare(WindowsLinux的通信)把下载好的库文件拷贝到Linux下, 然后解压 tar zxvf libeXosip2-3.6.0.tar.gz 注意这个时候可能会发生错误,是没有权限的问题,那么就在命令行前边加上sudo 然后配置 把下边这三行写成一个脚本 vim **.sh ./configure --host=arm-linux --target=arm-linux --prefix=/home/protocol_stack/install/ PKG_CONFIG_PATH=/home/protocol_stack/install/lib/pkgconfig make make install 然后用chmod 777 **.sh 执行脚本./**.sh 接下来可以编译mediastreamer2了,不过ms2,依赖好多库:ogg、speex、pulseaudio。而pulseaudio又依赖许多库:alsa、json、libtool。 3、交叉编译ogg 下载源码:http://xiph.org/downloads/ 我使用1.3.1版本 libogg-1.3.3.tar.gz 然后通过winshare(WindowsLinux的通信)把下载好的库文件拷贝到Linux下, 然后解压 tar zxvf libogg-1.3.3.tar.gz 注意这个时候可能会发生错误,是没有权限的问题,那么就在命令行前边加上sudo 然后配置 把下边这三行写成一个脚本 vim **.sh ./configure CC=arm-linux-gcc --prefix=/home/protocol_stack/install/ --host=arm-linux make make install 然后用chmod 777 **.sh 执行脚本./**.sh 4、交叉编译speex 下载源码:http://www.speex.org/downloads/ 我使用1.2rc1版本 speex-1.2rc1.tar.gz 然后通过winshare(WindowsLinux的通信)把下载好的库文件拷贝到Linux下, 然后解压 tar zxvf speex-1.2rc1.tar.gz 注意这个时候可能会发生错误,是没有权限的问题,那么就在命令行前边加上sudo 然后配置 把下边这三行写成一个脚本 vim **.sh ./configure CC=arm-linux-gcc --prefix=/home/protocol_stack/install/ --with-ogg=/home/protocol_stack/install/ --enable-fixed-point --disable-float-api \ --host=arm-linux make make install 然后用chmod 777 **.sh 执行脚本./**.sh 5、交叉编译pulseaudio 下载源码:http://freedesktop.org/software/pulseaudio/releases/ 我使用1.0版本 pulseaudio-1.0.tar.gz 然后通过winshare(WindowsLinux的通信)把下载好的库文件拷贝到Linux下, 然后解压 tar zxvf pulseaudio-1.0.tar.gz 注意这个时候可能会发生错误,是没有权限的问题,那么就在命令行前边加上sudo 然后配置 把下边这三行写成一个脚本 vim **.sh ./configure CC=arm-linux-gcc CXX=arm-linux-g++ --prefix=/home/protocol_stack/install --host=arm-linux --disable-rpath --disable-nls --disable-dbus --disable-bluez --disable-samplerate --disable-solaris --disable-gconf --disable-avahi --disable-jack --disable-lirc --disable-glib2 --disable-gtk2 --disable-openssl --disable-ipv6 --disable-asyncns --disable-per-user-esound-socket --disable-oss-output --disable-oss-wrapper --disable-x11 --enable-neon-opt=no --with-database=simple PKG_CONFIG_PATH=/home/protocol_stack/install/lib/pkgconfig CPPFLAGS=-I/home/protocol_stack/install/include LDFLAGS=-L/home/protocol_stack/install/lib CFLAGS=-I/home/protocol_stack/install/include make make install 然后用chmod 777 **.sh 执行脚本./**.sh 错误1: checking for ltdl.h... no configure: error: Unable to find libltdl version 2. Makes sure you have libtool 2.4 or later installed. make: *** No targets specified and no makefile found. Stop. 分析;找不到libltdl。确保你有libtool 2.4及以上的版本。 下载libtool 2.4.2版本 这时需要交叉编译libtool 下载源码:ftp://ftp.gnu.org/gnu/libtool/ 我使用2.4.2版本 libtool-2.4.2.tar.gz 然后通过winshare(WindowsLinux的通信)把下载好的库文件拷贝到Linux下, 然后解压 tar zxvf libtool-2.4.2.tar.gz 注意这个时候可能会发生错误,是没有权限的问题,那么就在命令行前边加上sudo 然后配置 把下边这三行写成一个脚本 vim **.sh ./configure --host=arm-linux --prefix =/home/protocol_stack/install/ make make install 然后用chmod 777 **.sh 执行脚本./**.sh 交叉编译alsa: http://www.alsa-project.org/main/index.php/Main_Page 这个库的版本需要根据你嵌入式Linux内核中alsa的版本而定,可以使用命令查看内核中alsa的版本: # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.24. 可以到内核中alsa驱动版本是1.0.24,所以我选1.0.24版本 alsa-lib-1.0.24.1.tar.gz 然后通过winshare(WindowsLinux的通信)把下载好的库文件拷贝到Linux下, 然后解压 tar zxvf speex-1.2rc1.tar.gz 注意这个时候可能会发生错误,是没有权限的问题,那么就在命令行前边加上sudo 然后配置 把下边这三行写成一个脚本 vim **.sh ./configure --host=arm-linux --prefix =/home/protocol_stack/install/ make make install 然后用chmod 777 **.sh 执行脚本./**.sh 错误:configure: error: Package requirements ( sndfile >= 1.0.20 ) were not met: No package 'sndfile' found 分析:缺少库 libsndfile库,那么接下来再进行交叉编译libsndfile libsndfile-1.0.25.tar.gz http://www.linuxfromscratch.org/blfs/view/svn/multimedia/libsndfile.html 然后通过winshare(WindowsLinux的通信)把下载好的库文件拷贝到Linux下, 然后解压 tar zxvf libsndfile-1.0.25.tar.gz 注意这个时候可能会发生错误,是没有权限的问题,那么就在命令行前边加上sudo 然后配置 把下边这三行写成一个脚本 vim **.sh ./configure --host=arm-linux --prefix =/home/protocol_stack/install/ make make install 然后用chmod 777 **.sh 执行脚本./**.sh 7、最后编译mediastreamer2 下载源码:http://ftp.twaren.net/Unix/NonGNU//linphone/mediastreamer/ 我使用2.8版本 mediastreamer-2.8.0.tar.gz 然后通过winshare(WindowsLinux的通信)把下载好的库文件拷贝到Linux下, 然后解压 tar zxvf mediastreamer-2.8.0.tar.gz 注意这个时候可能会发生错误,是没有权限的问题,那么就在命令行前边加上sudo 然后配置 把下边这三行写成一个脚本 vim **.sh ./configure CC=arm-linux-gcc --prefix=/home/protocol_stack/install/ PKG_CONFIG_PATH=/home/protocol_stack/install/lib/pkgconfig --disable-gsm --enable-video=no --enable-macsnd=no --disable-static --disable-sdl --disable-x11 --disable-ffmpeg --host=arm-linux --target=arm-linux make make install 然后用chmod 777 **.sh 执行脚本./**.sh 上面的configure选项没有屏蔽v4l1和v4l2,所以还得交叉编译v4l 编译v4l libv4l-0.6.4.tar.gz 下载源码:http://pkgs.fedoraproject.org/repo/pkgs/libv4l/ 然后通过winshare(WindowsLinux的通信)把下载好的库文件拷贝到Linux下, 然后解压 tar zxvf libv4l-0.6.4.tar.gz 注意这个时候可能会发生错误,是没有权限的问题,那么就在命令行前边加上sudo 然后配置 我使用0.6.4版本 libv4l-0.6.4.tar.gz make clean make CC=arm-linux-gcc make install PREFIX=/home/protocol_stack/install 编译mediastreamer2出错:(1)checking for LIBCHECK... no checking for LIBJSON... no configure: error: Package requirements ( json >= 0.9 ) were not met: No package 'json' found 解决方法就是交叉编译json 下载源码:http://ftp.debian.org/debian/pool/main/j/json-c/ 分析:缺少json库,那么我们继续交叉编译json库 json-c_0.12.1.orig.tar.gz 然后通过winshare(WindowsLinux的通信)把下载好的库文件拷贝到Linux下, 然后解压 tar zxvf mediastreamer-2.8.0.tar.gz 注意这个时候可能会发生错误,是没有权限的问题,那么就在命令行前边加上sudo 然后配置 ./configure --host=linux-arm \ --prefix =/home/protocol_stack/install/ make && make install 好了,json库已经编译完成了。接下来我们继续编译mediastreamer2 。。。。。 但是还是有问题,怎么办呢?还是哪个问题还是找不到json库。 分析:在json的论坛中,找到了解决方案:把编译生成的/lib/pkgconfig/这个目录下生成了一个json-c.pc。最后mediastreamer2在调用的时候找的是json.pc。那么我们就把这个文件名改为json.pc #mv json-c.pc json.pc OK,这次这个是可以编译的过去了。接下来继续编译 。。。 又出现问题了 /home/protocol_stack/install/lib/libjson.so: undefined reference to `rpl_malloc' /home/protocol_stack/install/lib/libjson.so: undefined reference to `rpl_realloc' 问题分析: 这个错误的原因是因为没有定义 rpl_malloc 和 rpl_realloc 这两个变量。 那么我们应该怎么办么? 那么就在这个目录下进行查这两个变量是在哪里定义的? 于是:#grep "rpl_malloc" -nR * ....... 找到了,原来这两个变量是在这个config的文件中的。是一个宏开关 那么就好办了,我们就直接把这两个宏进行注释。 嗯嗯,继续。。。我们重新编译json库。。。嗯嗯编译好了,接下来继续来编译mediastreamer2 。。。。 又出错了,还是这个原因 /home/protocol_stack/install/lib/libjson.so: undefined reference to `rpl_malloc' /home/protocol_stack/install/lib/libjson.so: undefined reference to `rpl_realloc 嗯嗯,还是这个原因?究竟是为什么呢。再次来到json的目录下,再次看有没有把那两个宏开关给关闭? 嗯哼? 竟然没有关闭? 分析?明白了。原来是我把配置和编译同时执行了。这个宏开关是./configure ...生成的。 那么就只好,这样。把./configure。。。生成的config文件,再进行关闭宏开关。最后直接make && make install -j8 直接编译,安装,是不能再次进行配置的。因为以配置config文件就会再次生成,那么宏开关就又开了。 OK,安装好了,下来继续进行编译mediastreamer2.。。。。。。。。。。。 。。。。。。。。。。。。。。 又出现了问题? error: /user/include/python2.7/pyconfig.h:15:52: fatal error: arm-linux-gnueabi/python2.7/pyconfig.h: No such file or directory compilation terminated. 分析::找不到arm-linux-gnueabi/python2.7/pyconfig.h这文件。那就继续交叉编译python 好吧,继续下载python,然后再进行交叉编译,但是编译Python的时候出来一系列的问题。根本没有办法解决。 那么该怎么办呢?时候一个小时又一个小时的过去? 最后有一个大胆的想法,既然python都编译不下去。那就不要了。 于是,在mediastreamer2的./configure 中添上一项 --without-python 。 。。。再次配置编译。。。。。。。。。。。 error: /user/include/python2.7/pyconfig.h:15:52: fatal error: arm-linux-gnueabi/python2.7/pyconfig.h: No such file or directory compilation termiated. 嗯哼?还是一样的错误。怎么办呢? 于是乎就又在论坛上进行找灵感。。。。。 还是找不到。。。 又一结合前边几个库的配置编译,发现不使能一个模块还可以用另外一个--disable-python 。。。 于是乎 就把--without-python改为了--disable-python 继续编译。。。。 。。。。。。。。。。。。。。。。。。。。。 到了这个节骨眼上了,编译每跳一下,我的心就跟着逗一下。。。。心酸 。。。。。。 。。。。。。 。。。。。。 竟然编译成功了。。。。 哈哈。。。。。。。。。 于是,马上就把编译好的库,拷贝到了开发板。。。 嗯嗯,本来还想把编译好的库目录树拷贝下的,但是太多了,放不下。。。算了吧。。。。 找到编译好的库 在库中的/bin中找到arm-linux-mediastream 然后执行./arm-linux-mediastream 。。。。报错了 问题: error : while loading shared libraries: libmediastreamer.so.1: cannot open shared object file: No such file 答案:分析: 遇到这个问题就是,libmediastreamer.so.1这个动态库,在可执行文件armlinuxmediastreamer执行的时候,会调用这个动态库,但是环境变量中找不到这个动态库。那么我们就是要把我们编译好的动态链接库的目录加到环境变量中 LD_LIBRARY_PATH=$LD_LIBRARY_PATH://arm/lib/这个目录下就是放着我们编译好的所有的动态链接库(包括libmediastreamer.so.1) 执行步骤:LD_LIBRARY_PATH=$LD_LIBRARY_PATH://arm/lib export LD_LIBRARY_PATH ./arm-linux-mediastream mediastream --local --remote --payload [ --fmtp ] [ --jitter ] [ --width ] [ --height ] [ --bitrate ] [ --ec (enable echo canceller) ] [ --ec-tail ] [ --ec-delay ] [ --ec-framesize ] [ --agc (enable automatic gain control) ] [ --ng (enable noise gate)] [ --ng-threshold (noise gate threshold) ] [ --ng-floorgain (gain applied to the signal when its energy is below the threshold.) ] [ --capture-card ] [ --playback-card ] [ --infile <input wav file> specify a wav file to be used for input, instead of soundcard ] [ --outfile specify a wav file to write audio into, instead of soundcard ] [ --camera ] [ --el (enable echo limiter) ] [ --el-speed (gain changes are smoothed with a coefficent) ] [ --el-thres (Threshold above which the system becomes active) ] [ --el-force (The proportional coefficient controlling the mic attenuation) ] [ --el-sustain (Time in milliseconds for which the attenuation is kept unchanged after) ] [ --el-transmit-thres (TO BE DOCUMENTED) ] [ --rc (enable adaptive rate control) ] [ --zrtp (enable zrtp) ] [ --verbose (most verbose messages) ] [ --video-windows-id <video surface:preview surface>] [ --srtp (enable srtp, master key is generated if absent from comand line) [ --netsim-bandwidth (simulates a network download bandwidth limit) 于是按照第一种方式进行 参数添加 ./arm-linux-mediastream --local 8888 --remote 127.0.0.1:88 88 OK运行正常了 下面是运行信息。。。 ortp-message-audio_stream_process_rtcp: interarrival jitter=119 , lost packets percentage since last report=0.000000, round trip time=0.000000 seconds ortp-message-oRTP-stats: RTP stats : ortp-message- number of rtp packet sent=150 ortp-message- number of rtp bytes sent=25800 bytes ortp-message- number of rtp packet received=150 ortp-message- number of rtp bytes received=25800 bytes ortp-message- number of incoming rtp bytes successfully delivered to the application=25284 ortp-message- number of rtp packet lost=0 ortp-message- number of rtp packets received too late=0 ortp-message- number of bad formatted rtp packets=0 ortp-message- number of packet discarded because of queue overflow=0 ortp-message-Bandwidth usage: download=81.290281 kbits/sec, upload=81.288664 kbits/sec ortp-message-Receiving RTCP SR ortp-message-Receiving RTCP SDES ortp-message-Found CNAME=unknown@unknown ortp-message-Found TOOL=oRTP-0.18.0 ortp-message-Found NOTE=This is free sofware (LGPL) ! ortp-message-Quality indicator : 4.888437 运行正常了。。。。。。
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