ubuntu下 gstreamer 的配置及播放音视频例子（已经验证）

官方网址： http://gstreamer.freedesktop.org

Gstreamer安装:

使用sudo apt-get install 安装

sudo apt-get install libgstreamer0.10-dev gstreamer-tools gstreamer0.10-tools gstreamer0.10-doc
sudo apt-get install gstreamer0.10-plugins-base gstreamer0.10-plugins-good gstreamer0.10-plugins-ugly gstreamer0.10-plugins-bad gstreamer0.10-plugins-bad-multiverse

安装了如下的gst的插件：

gstreamer0.10-tools

gstreamer0.10-x

gstreamer0.10-plugins-base

gstreamer0.10-plugins-good

gstreamer0.10-plugins-ugly

gstreamer0.10-plugins-bad

gstreamer0.10-ffmpeg

gstreamer0.10-alsa

gstreamer0.10-schroedinger

gstreamer0.10-pulseaudio

有可能需要安装的软件：

sudo apt-get install bison

sudo apt-get install flex

sudo apt-get install zlib1g

mad解码插件

apt-get install libmad0-dev

apt-get install gstreamer0.10-plugins-ugly

Initializing GStreamer

[cpp]  view plain  copy

 

1. #include <stdio.h>  
2. #include <gst/gst.h>  
3.   
4. int main (int   argc,   char *argv[])  
5. {  
6.   const gchar *nano_str;  
7.   guint major, minor, micro, nano;  
8.   
9.   gst_init (&argc, &argv);  
10.   
11.   gst_version (&major, &minor, µ, &nano);  
12.   
13.   if (nano == 1)  
14.     nano_str = "(CVS)";  
15.   else if (nano == 2)  
16.     nano_str = "(Prerelease)";  
17.   else  
18.     nano_str = "";  
19.   
20.   printf ("This program is linked against GStreamer %d.%d.%d %s\n",  
21.           major, minor, micro, nano_str);  
22.   
23.   return 0;  
24. }  

[cpp]  view plain  copy

 

1. #include <gst/gst.h>  
2. #include <glib.h>  
3. //定义消息处理函数,  
4. static gboolean bus_call(GstBus *bus,GstMessage *msg,gpointer data)  
5. {  
6.     GMainLoop *loop = (GMainLoop *) data;//这个是主循环的指针，在接受EOS消息时退出循环  
7.     switch (GST_MESSAGE_TYPE(msg))  
8.     {  
9.         case GST_MESSAGE_EOS:  
10.             g_print("End of stream\n");  
11.             g_main_loop_quit(loop);  
12.             break;  
13.         case GST_MESSAGE_ERROR:  
14.         {  
15.                gchar *debug;  
16.                GError *error;  
17.                gst_message_parse_error(msg,&error,&debug);  
18.                g_free(debug);  
19.                g_printerr("ERROR:%s\n",error->message);  
20.                g_error_free(error);  
21.                g_main_loop_quit(loop);  
22.   
23.                 break;  
24.         }  
25.         default:  
26.              break;  
27.     }  
28.     return TRUE;  
29. }  
30.   
31. int main(int argc,char *argv[])  
32. {  
33.     GMainLoop *loop;  
34.     GstElement *pipeline,*source,*decoder,*sink;//定义组件  
35.     GstBus *bus;  
36.   
37.     gst_init(&argc,&argv);  
38.     loop = g_main_loop_new(NULL,FALSE);//创建主循环，在执行 g_main_loop_run后正式开始循环  
39.   
40.     if(argc != 2)  
41.     {  
42.         g_printerr("Usage:%s <mp3 filename>\n",argv[0]);  
43.         return -1;  
44.     }  
45.     //创建管道和组件  
46.     pipeline = gst_pipeline_new("audio-player");  
47.     source = gst_element_factory_make("filesrc","file-source");  
48.     decoder = gst_element_factory_make("mad","mad-decoder");  
49.     sink = gst_element_factory_make("autoaudiosink","audio-output");  
50.   
51.     if(!pipeline||!source||!decoder||!sink){  
52.         g_printerr("One element could not be created.Exiting.\n");  
53.         return -1;  
54.     }  
55.     //设置 source的location 参数。即 文件地址.  
56.     g_object_set(G_OBJECT(source),"location",argv[1],NULL);  
57.     //得到 管道的消息总线  
58.     bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));  
59.    //添加消息监视器  
60.     gst_bus_add_watch(bus,bus_call,loop);  
61.     gst_object_unref(bus);  
62.     //把组件添加到管道中.管道是一个特殊的组件，可以更好的让数据流动  
63.     gst_bin_add_many(GST_BIN(pipeline),source,decoder,sink,NULL);  
64.    //依次连接组件  
65.    gst_element_link_many(source,decoder,sink,NULL);  
66.    //开始播放  
67.     gst_element_set_state(pipeline,GST_STATE_PLAYING);  
68.     g_print("Running\n");  
69.     //开始循环  
70.     g_main_loop_run(loop);  
71.     g_print("Returned,stopping playback\n");  
72.     gst_element_set_state(pipeline,GST_STATE_NULL);  
73.     gst_object_unref(GST_OBJECT(pipeline));  
74.     return 0;  
75. }  

编译运行

gcc -Wall test2.c -o test2  $(pkg-config --cflags --libs gstreamer-0.10)
（注意！！！很多地方写的这条命令 ：gcc -Wall $(pkg-config --cflags --libs gstreamer-0.10) -g test2.c -o test2 ，根本编译不过！）

编译时借助了 pkg-config 
pkg-config --cflags --libs gstreamer-0.10 会把gstreamer-0.10编译所依赖的库的路径 头文件的路径全部帮你找出来，不用你再依次写出。
(上面这条pkg-config --cflags --libs gstreamer-0.10 就相当于：
-pthread -I/usr/include/glib-2.0 -I/usr/lib/i386-Linux-gnu/glib-2.0/include -I/usr/include/gstreamer-0.10 -I/usr/include/libxml2  -pthread -L/usr/lib/i386-linux-gnu -lgstreamer-0.10 -lgobject-2.0 -lgmodule-2.0 -lxml2 -lgthread-2.0 -lrt -lglib-2.0)

./test2  ./abc.mp3

使用命令的方式，一样的效果

#gst-launch-0.10 filesrc location=/aaron/gstreamer/test/abc.mp3 ! mad ! autoaudiosink

播放视频：

Ubuntu 12.04

gst-launch-0.10 filesrc location=/tmp/video_stream_pipe ! decodebin ! ffmpegcolorspace ! videoscale ! ximagesink sync=false

Ubuntu 13.10

gst-launch-1.0 filesrc location=/tmp/video_stream_pipe ! decodebin ! videoconvert ! xvimagesink sync=false
gst-launch-1.0 filesrc location=/tmp/video_stream_pipe ! decodebin ! videoconvert ! ximagesink sync=false

以下摘至：http://linux.die.net/man/1/gst-launch-0.10

gst-launch-0.10(1) - Linux man page

Name

gst-launch - build and run a GStreamer pipeline

Synopsis

gst-launch [OPTION...] PIPELINE-DESCRIPTION

Description

gst-launch is a tool that builds and runs basicGStreamer pipelines.

In simple form, a PIPELINE-DESCRIPTION is a list of elements separated by exclamation marks (!). Properties may be appended to elements, in the formproperty=value.

For a complete description of possible PIPELINE-DESCRIPTIONS see the section pipeline descriptionbelow or consult the GStreamer documentation.

Please note that gst-launch is primarily a debugging tool for developers and users. You should not build applications on top of it. For applications, use the gst_parse_launch() function of the GStreamer API as an easy way to construct pipelines from pipeline descriptions.

Options

gst-launch accepts the following options:

--help

Print help synopsis and available FLAGS

-v, --verbose

Output status information and property notifications

-q, --quiet

Do not print any progress information

-m, --messages

Output messages posted on the pipeline's bus

-t, --tags

Output tags (also known as metadata)

-o FILE, --output=FILE

Save XML representation of pipeline to FILE and exit

-f, --no_fault

Do not install a fault handler

-T, --trace

Print memory allocation traces. The feature must be enabled at compile time to work.

Gstreamer Options

gst-launch also accepts the following options that are common to all GStreamer applications:

--gst-version

Prints the version string of the  GStreamer core library.

--gst-fatal-warnings

Causes  GStreamer to abort if a warning message occurs. This is equivalent to setting the environment variable G_DEBUG to 'fatal_warnings' (see the section  environment variables below for further information).

--gst-debug=STRING

A comma separated list of category_name:level pairs to specify debugging levels for each category. Level is in the range 0-5 where 0 will show no messages, and 5 will show all messages. The wildcard * can be used to match category names.

Use --gst-debug-help to show category names

Example: GST_CAT:5,GST_ELEMENT_*:3,oggdemux:5

--gst-debug-level=LEVEL

Sets the threshold for printing debugging messages. A higher level will print more messages. The useful range is 0-5, with the default being 0.

--gst-debug-no-color

GStreamer normally prints debugging messages so that the messages are color-coded when printed to a terminal that handles ANSI escape sequences. Using this option causes  GStreamer to print messages without color. Setting the GST_DEBUG_NO_COLOR environment variable will achieve the same thing.

--gst-debug-disable

Disables debugging.

--gst-debug-help

Prints a list of available debug categories and their default debugging level.

--gst-plugin-spew

GStreamer info flags to set Enable printout of errors while loading  GStreamer plugins

--gst-plugin-path=PATH

Add directories separated with ':' to the plugin search path

--gst-plugin-load=PLUGINS

Preload plugins specified in a comma-separated list. Another way to specify plugins to preload is to use the environment variable GST_PLUGIN_PATH

Pipeline Description

A pipeline consists elements and links. Elements can be put into bins of different sorts. Elements, links and bins can be specified in a pipeline description in any order.

Elements

ELEMENTTYPE [PROPERTY1 ...]

Creates an element of type ELEMENTTYPE and sets the PROPERTIES.

Properties

PROPERTY=VALUE ...

Sets the property to the specified value. You can use gst-inspect(1) to find out about properties and allowed values of different elements.
Enumeration properties can be set by name, nick or value.

Bins

[BINTYPE.] ( [PROPERTY1 ...] PIPELINE-DESCRIPTION )

Specifies that a bin of type BINTYPE is created and the given properties are set. Every element between the braces is put into the bin. Please note the dot that has to be used after the BINTYPE. You will almost never need this functionality, it is only really useful for applications using the gst_launch_parse() API with 'bin' as bintype. That way it is possible to build partial pipelines instead of a full-fledged top-level pipeline.

Links

[[SRCELEMENT].[PAD1,...]] ! [[SINKELEMENT].[PAD1,...]] [[SRCELEMENT].[PAD1,...]] ! CAPS ! [[SINKELEMENT].[PAD1,...]]

Links the element with name SRCELEMENT to the element with name SINKELEMENT, using the caps specified in CAPS as a filter. Names can be set on elements with the name property. If the name is omitted, the element that was specified directly in front of or after the link is used. This works across bins. If a padname is given, the link is done with these pads. If no pad names are given all possibilities are tried and a matching pad is used. If multiple padnames are given, both sides must have the same number of pads specified and multiple links are done in the given order.
So the simplest link is a simple exclamation mark, that links the element to the left of it to the element right of it.

Caps

MIMETYPE [, PROPERTY[, PROPERTY ...]]] [; CAPS[; CAPS ...]]

Creates a capability with the given mimetype and optionally with given properties. The mimetype can be escaped using " or '. If you want to chain caps, you can add more caps in the same format afterwards.

Properties

NAME=[(TYPE)]VALUE
in lists and ranges: [(TYPE)]VALUE

Sets the requested property in capabilities. The name is an alphanumeric value and the type can have the following case-insensitive values:
- i or int for integer values or ranges
- f or float for float values or ranges
- 4 or fourcc for FOURCC values
- b, bool or boolean for boolean values
- s, str or string for strings
- fraction for fractions (framerate, pixel-aspect-ratio)
- l or list for lists
If no type was given, the following order is tried: integer, float, boolean, string.
Integer values must be parsable by strtol(), floats by strtod(). FOURCC values may either be integers or strings. Boolean values are (case insensitive) yes, no, true or false and may like strings be escaped with " or '.
Ranges are in this format: [ VALUE, VALUE ]
Lists use this format: ( VALUE [, VALUE ...] )

Pipeline Control

A pipeline can be controlled by signals. SIGUSR2 will stop the pipeline (GST_STATE_NULL); SIGUSR1 will put it back to play (GST_STATE_PLAYING). By default, the pipeline will start in the playing state.
There are currently no signals defined to Go into the ready or pause (GST_STATE_READY and GST_STATE_PAUSED) state explicitely.

Pipeline Examples

The examples below assume that you have the correct plug-ins available. In general, "osssink" can be substituted with another audio output plug-in such as "esdsink", "alsasink", "osxaudiosink", or "artsdsink". Likewise, "xvimagesink" can be substituted with "ximagesink", "sdlvideosink", "osxvideosink", or "aasink". Keep in mind though that different sinks might accept different formats and even the same sink might accept different formats on different machines, so you might need to add converter elements like audioconvert and audioresample (for audio) or ffmpegcolorspace (for video) in front of the sink to make things work.

Audio playback

gst-launch filesrc location=music.mp3 ! mad ! audioconvert ! audioresample ! osssink
Play the mp3 music file "music.mp3" using a libmad-based plug-in and output to an OSS device

gst-launch filesrc location=music.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! osssink
Play an Ogg Vorbis format file

gst-launch gnomevfssrc location=music.mp3 ! mad ! osssink
gst-launch gnomevfssrc location=http://domain.com/music.mp3 ! mad ! audioconvert ! audioresample ! osssink
Play an mp3 file or an http stream using GNOME-VFS

gst-launch gnomevfssrc location=smb://computer/music.mp3 ! mad ! audioconvert ! audioresample ! osssink
Use GNOME-VFS to play an mp3 file located on an SMB server

Format conversion

gst-launch filesrc location=music.mp3 ! mad ! audioconvert ! vorbisenc ! oggmux ! filesink location=music.ogg
Convert an mp3 music file to an Ogg Vorbis file

gst-launch filesrc location=music.mp3 ! mad ! audioconvert ! flacenc ! filesink location=test.flac
Convert to the FLAC format

Other

gst-launch filesrc location=music.wav ! wavparse ! audioconvert ! audioresample ! osssink
Plays a .WAV file that contains raw audio data (PCM).

gst-launch filesrc location=music.wav ! wavparse ! audioconvert ! vorbisenc ! oggmux ! filesink location=music.ogg
gst-launch filesrc location=music.wav ! wavparse ! audioconvert ! lame ! filesink location=music.mp3
Convert a .WAV file containing raw audio data into an Ogg Vorbis or mp3 file

gst-launch cdparanoiasrc mode=continuous ! audioconvert ! lame ! id3v2mux ! filesink location=cd.mp3
rips all tracks from compact disc and convert them into a single mp3 file

gst-launch cdparanoiasrc track=5 ! audioconvert ! lame ! id3v2mux ! filesink location=track5.mp3
rips track 5 from the CD and converts it into a single mp3 file

Using gst-inspect(1), it is possible to discover settings like the above for cdparanoiasrc that will tell it to rip the entire cd or only tracks of it. Alternatively, you can use an URI and gst-launch-0.10 will find an element (such as cdparanoia) that supports that protocol for you, e.g.: gst-launch cdda://5 ! lame vbr=new vbr-quality=6 ! filesink location=track5.mp3

gst-launch osssrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=input.ogg
records sound from your audio input and encodes it into an ogg file

Video

gst-launch filesrc location=JB_FF9_TheGravityOfLove.mpg ! dvddemux ! mpeg2dec ! xvimagesink
Display only the video portion of an MPEG-1 video file, outputting to an X display window

gst-launch filesrc location=/flflfj.vob ! dvddemux ! mpeg2dec ! sdlvideosink
Display the video portion of a .vob file (used on DVDs), outputting to an SDL window

gst-launch filesrc location=movie.mpg ! dvddemux name=demuxer demuxer. ! queue ! mpeg2dec ! sdlvideosink demuxer. ! queue ! mad ! audioconvert ! audioresample ! osssink
Play both video and audio portions of an MPEG movie

gst-launch filesrc location=movie.mpg ! mpegdemux name=demuxer demuxer. ! queue ! mpeg2dec ! ffmpegcolorspace ! sdlvideosink demuxer. ! queue ! mad ! audioconvert ! audioresample ! osssink
Play an AVI movie with an external text subtitle stream

This example also shows how to refer to specific pads by name if an element (here: textoverlay) has multiple sink or source pads.

gst-launch textoverlay name=overlay ! ffmpegcolorspace ! videoscale ! autovideosink filesrc location=movie.avi ! decodebin2 ! ffmpegcolorspace ! overlay.video_sink filesrc location=movie.srt ! subparse ! overlay.text_sink

Play an AVI movie with an external text subtitle stream using playbin2

gst-launch playbin2 uri=file:///path/to/movie.avi suburi=file:///path/to/movie.srt

Network streaming

Stream video using RTP and network elements.

gst-launch v4l2src ! video/x-raw-yuv,width=128,height=96,format='(fourcc)'UYVY ! ffmpegcolorspace ! ffenc_h263 ! video/x-h263 ! rtph263ppay pt=96 ! udpsink host=192.168.1.1 port=5000 sync=false
Use this command on the receiver

gst-launch udpsrc port=5000 ! application/x-rtp, clock-rate=90000,payload=96 ! rtph263pdepay queue-delay=0 ! ffdec_h263 ! xvimagesink
This command would be run on the transmitter

Diagnostic

gst-launch -v fakesrc num-buffers=16 ! fakesink
Generate a null stream and ignore it (and print out details).

gst-launch audiotestsrc ! audioconvert ! audioresample ! osssink
Generate a pure sine tone to test the audio output

gst-launch videotestsrc ! xvimagesink
gst-launch videotestsrc ! ximagesink
Generate a familiar test pattern to test the video output

Automatic linking

You can use the decodebin element to automatically select the right elements to get a working pipeline.

gst-launch filesrc location=musicfile ! decodebin ! audioconvert ! audioresample ! osssink
Play any supported audio format

gst-launch filesrc location=videofile ! decodebin name=decoder decoder. ! queue ! audioconvert ! audioresample ! osssink decoder. ! ffmpegcolorspace ! xvimagesink
Play any supported video format with video and audio output. Threads are used automatically. To make this even easier, you can use the playbin element:

gst-launch playbin uri=file:///home/joe/foo.avi

Filtered connections

These examples show you how to use filtered caps.

gst-launch videotestsrc ! 'video/x-raw-yuv,format=(fourcc)YUY2;video/x-raw-yuv,format=(fourcc)YV12' ! xvimagesink
Show a test image and use the YUY2 or YV12 video format for this.

gst-launch osssrc ! 'audio/x-raw-int,rate=[32000,64000],width=[16,32],depth={16,24,32},signed=(boolean)true' ! wavenc ! filesink location=recording.wav
record audio and write it to a .wav file. Force usage of signed 16 to 32 bit samples and a sample rate between 32kHz and 64KHz.

Environment Variables

GST_DEBUG

Comma-separated list of debug categories and levels, e.g. GST_DEBUG=totem:4,typefind:5

GST_DEBUG_NO_COLOR

When this environment variable is set, coloured debug output is disabled.

GST_DEBUG_DUMP_DOT_DIR

When set to a filesystem path, store dot files of pipeline graphs there.

GST_REGISTRY

Path of the plugin registry file. Default is ~/.gstreamer-0.10/registry-CPU.xml where CPU is the machine/cpu type GStreamer was compiled for, e.g. 'i486', 'i686', 'x86-64', 'ppc', etc. (check the output of "uname -i" and "uname -m" for details).

GST_REGISTRY_UPDATE

Set to "no" to force GStreamer to assume that no plugins have changed, been added or been removed. This will make GStreamer skip the initial check whether a rebuild of the registry cache is required or not. This may be useful in embedded environments where the installed plugins never change. Do not use this option in any other setup.

GST_PLUGIN_PATH

Specifies a list of directories to scan for additional plugins. These take precedence over the system plugins.

GST_PLUGIN_SYSTEM_PATH

Specifies a list of plugins that are always loaded by default. If not set, this defaults to the system-installed path, and the plugins installed in the user's home directory

OIL_CPU_FLAGS

Useful liboil environment variable. Set OIL_CPU_FLAGS=0 when valgrind or other debugging tools trip over liboil's CPU detection (quite a few important GStreamer plugins like videotestsrc, audioconvert or audioresample use liboil).

G_DEBUG

Useful GLib environment variable. Set G_DEBUG=fatal_warnings to make GStreamer programs abort when a critical warning such as an assertion failure occurs. This is useful if you want to find out which part of the code caused that warning to be triggered and under what circumstances. Simply set G_DEBUG as mentioned above and run the program in gdb (or let it core dump). Then get a stack trace in the usual way.

Files

~/.gstreamer-0.10/registry-*.xml

The xml plugin database; can be deleted at any time, will be re-created automatically when it does not exist yet or plugins change.