gstreamer应用笔记

gstreamer官网

https://gstreamer.freedesktop.org/

应用手册

https://gstreamer.freedesktop.org/documentation/index.html

一、getreamer安装(ubuntu)

gstreamer0.10和gstreamer1.0两个版本容易混淆

sudo add-apt-repository ppa:mc3man/trusty-media

sudo apt-get update

sudo apt-get install build-essential dpkg-dev flex bison autotools-dev automake liborc-dev autopoint libtool gtk-doc-tools libgstreamer1.0-dev

sudo apt-get install libgstreamer0.10-0 libgstreamer0.10-dev gstreamer0.10-tools gstreamer0.10-plugins-base libgstreamer-plugins-base0.10-dev gstreamer0.10-plugins-good gstreamer0.10-plugins-ugly gstreamer0.10-plugins-bad gstreamer0.10-ffmpeg

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

若有需要还可以再安装如下gst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

安装音频库

sudo apt-get install gstreamer1.0-alsa

安装ffmpeg多媒体库

gst-ffmpeg
--enable-liba52       enable GPLed liba52 support [default=no]
--enable-liba52bin    open liba52.so.0 at runtime [default=no]
--enable-libamr-nb    enable libamr-nb floating point audio codec
--enable-libamr-wb    enable libamr-wb floating point audio codec
--enable-libfaac      enable FAAC support via libfaac [default=no]
--enable-libfaad      enable FAAD support via libfaad [default=no]
--enable-libfaadbin   open libfaad.so.0 at runtime [default=no]
--enable-libgsm       enable GSM support via libgsm [default=no]
--enable-libmp3lame   enable MP3 encoding via libmp3lame
--enable-libvorbis    enable Vorbis encoding via libvorbis,
                     native implementation exists [default=no]
--enable-libx264      enable H.264 encoding via x264 [default=no]
--enable-libxvid      enable Xvid encoding via xvidcore,
                      native MPEG-4/Xvid encoder exists [default=no]

插件太多了，几百上千个

List of Elements and Plugins

https://gstreamer.freedesktop.org/documentation/plugins.html

一劳永逸

sudo apt-get install gstreamer-plugins-*

sudo apt-get install gstreamer-*

二、 千里之行，始于"hello,world"

Tutorials

Basic tutorials

Basic tutorial 1: Hello world!

Basic tutorial 2: GStreamer concepts

Basic tutorial 3: Dynamic pipelines

Basic tutorial 4: Time management

Basic tutorial 5: GUI toolkit integration

Basic tutorial 6: Media formats and Pad Capabilities

Basic tutorial 7: Multithreading and Pad Availability

Basic tutorial 8: Short-cutting the pipeline

Basic tutorial 9: Media information gathering

Basic tutorial 10: GStreamer tools

Basic tutorial 11: Debugging tools

Basic tutorial 12: Streaming

Basic tutorial 13: Playback speed

Basic tutorial 14: Handy elements

Basic tutorial 16: Platform-specific elements

除了这16个入门samples，后面的pipelines Command line tools 和 Plugin 插件开发也是多媒体类应用极好的教材

官网才是最应该多关注的地方

https://gstreamer.freedesktop.org/documentation/tutorials/basic/hello-world.html

wiki手册也非常全面，几乎所有应用方向都说明

http://wiki.oz9aec.net/index.php?title=Gstreamer_cheat_sheet

IBM社区gstreamer教程

https://www.ibm.com/developerworks/cn/linux/l-gstreamer/

三、 gstreamer 进阶...

1、播放视频文件

   以MP4格式为例，其它格式可以 通过gst-inspect-1.0 |  grep  查找对应的demux，decode，sink等插件，当然也可以使用auto开头的插件，或者playbin会自动选择播放，只是没有自己指定那么灵活，方便调试和验证一些功能。

    1）硬解(vaapi)播放MP4文件：

     gst-launch-1.0 filesrc location=FilePath/test.mp4 ! qtdemux ! vaapidecode ! vaapisink

    2） 软解，只要将解码器vaapidecode换成avdec_h264，播放器vaapisink换成 ximagesink即可

2、播放RTSP视频流

    1） 硬解。

     gst-launch-1.0 rtspsrc location=rtsp://username:passwd@ipaddr:port  latency=0 ! rtph264depay  !  capsfilter caps="video/x-h264"  ! h264parse  ! vaapidecode  !  vaapipostproc  width=800 height=600  !  vaapisink sync=false

    2）软解。

   gst-launch-1.0 rtspsrc location=rtsp://username:passwd@ipaddr:port  latency=0 ! rtph264depay ! capsfilter caps="video/x-h264" ! h264parse ! avdec_h264 ! videoconvert ! videoscale ! video/x-raw,width=800,height=600 ! ximagesink

3、 播放Udp视频流

    Udp播放需要根据发送端数据源封装格式来决定采用哪些Gstreamer插件，如果进行了RTP封装，则需要先用rtph264depay进行解包，如果包含自定义帧头的情况，应该编程对帧头进行处理，不然会显示异常，比如部分花屏现象，以下是对裸流进行播放。    

1）硬解

     gst-launch-1.0 udpsrc port=2101 ! h264parse ! vaapidecode ! vaapisink

2）软解

     gst-launch-1.0  udpsrc port=2101 ! h264parse ! avdec_h264 !  autovideosink

 

参考https://blog.csdn.net/manjiao4651538/article/details/80227966

4、gstreamer rtsp推流/拉流

    1）gstreamer rtsp拉流播放

    https://blog.csdn.net/yang_quan_yang/article/details/78846134

    2）gstereamer rtsp推流

    https://blog.csdn.net/zhuwei622/article/details/80348916

    3) On the Raspberry:

 $ gst-launch-1.0 rtspsrc location=rtsp://192.168.2.112:8080/stream.sdp ! rtph264depay ! h264parse ! omxh264dec ! autovideosink

5、rtpbin Network/RTP

send

Encode and payload H263 video captured from a v4l2src. Encode and payload AMR audio generated from audiotestsrc. The video is sent to session 0 in rtpbin and the audio is sent to session 1. Video packets are sent on UDP port 5000 and audio packets on port 5002. The video RTCP packets for session 0 are sent on port 5001 and the audio RTCP packets for session 0 are sent on port 5003. RTCP packets for session 0 are received on port 5005 and RTCP for session 1 is received on port 5007. Since RTCP packets from the sender should be sent as soon as possible and do not participate in preroll, sync=false and async=false is configured on udpsink

gst-launch-1.0 rtpbin name=rtpbin 
        v4l2src ! videoconvert ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 
                  rtpbin.send_rtp_src_0 ! udpsink port=5000                            
                  rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false    
                  udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0                           
        audiotestsrc ! amrnbenc ! rtpamrpay ! rtpbin.send_rtp_sink_1                   
                  rtpbin.send_rtp_src_1 ! udpsink port=5002                            
                  rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false    
                  udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1

recv

Receive H263 on port 5000, send it through rtpbin in session 0, depayload, decode and display the video. Receive AMR on port 5002, send it through rtpbin in session 1, depayload, decode and play the audio. Receive server RTCP packets for session 0 on port 5001 and RTCP packets for session 1 on port 5003. These packets will be used for session management and synchronisation. Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1 on port 5007.

gst-launch-1.0 -v rtpbin name=rtpbin                                          
    udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" 
            port=5000 ! rtpbin.recv_rtp_sink_0                                
        rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink                    
     udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0                               
     rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false        
    udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,octet-align=(string)1" 
            port=5002 ! rtpbin.recv_rtp_sink_1                                
        rtpbin. ! rtpamrdepay ! amrnbdec ! alsasink                           
     udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1                               
     rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false

参考：

https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-rtpbin.html

GStreamer RTP Streaming

https://community.nxp.com/docs/DOC-94646 

6、录音

录音:

gst-launch -e pulsesrc ! audioconvert ! lamemp3enc target=1 bitrate=64 cbr=true ! filesink location=audio.mp3

gst-launch -e pulsesrc device="alsa_input.pci-0000_02_02.0.analog-stereo" ! audioconvert ! 
   lamemp3enc target=1 bitrate=64 cbr=true ! filesink location=audio.mp3

 播放录音：

gst-launch-1.0 filesrc location=audio.mp3 ! decodebin ! audioconvert ! audioresample ! autoaudiosink

还是上面的wiki

http://wiki.oz9aec.net/index.php?title=Gstreamer_cheat_sheet

7、录视频

命令：gst-launch-1.0 -e rtspsrc location=rtsp://admin:admin@192.168.1.2 ! rtph264depay ! "video/x-h264, stream-format=byte-stream" ! filesink location=test.264

说明：主要是用gst-lanuch工具连接相关插件将rtsp video stream 保存为.264文件，然后可以利用相关播放器（如：kmpplayer）进行播放，亦可以供live555MediaServer生成rtsp stream；（"video/x-h264, stream-format=byte-stream"这个caps一定要连接才行）
原文：https://blog.csdn.net/u010005508/article/details/52710302

8、视频收发(监控，预览)

send:

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=127.0.0.1 port=5000 sync=false

recv:
gst-launch  udpsrc  port=5000 ! application/x-rtp, clock-rate=90000,payload=96 ! rtph263pdepay queue-delay=0 ! ffdec_h263 ! xvimagesink

以Freescale平台为例，实时码流收发命令行如下：

Server侧（发送方）：

gst-launch -v videotestsrc ! video/x-raw-yuv,width=640,height=480 ! vpuenc codec=avc ! rtph264pay pt=96 ! udpsink host=127.0.0.1 port=1234

Client侧（接收方）：

gst-launch -vvv udpsrc port=1234 caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264" ! rtph264depay ! vpudec ! mfw_isink

9、音频收发(语音对讲)

模拟声音数据

1）send.sh

gst-launch-1.0 rtpbin name=rtpbin 
        audiotestsrc ! amrnbenc ! rtpamrpay ! rtpbin.send_rtp_sink_1                   
                  rtpbin.send_rtp_src_1 ! udpsink port=5002                            
                  rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false    
                  udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1

2)recv.sh

gst-launch-1.0 -v rtpbin name=rtpbin                                          
    udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,octet-align=(string)1" 
            port=5002 ! rtpbin.recv_rtp_sink_1                                
        rtpbin. ! rtpamrdepay ! amrnbdec ! alsasink                           
     udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1                               
     rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false

真实声卡

1) send.sh

gst-launch-1.0 rtpbin name=rtpbin 
        pulsesrc ! amrnbenc ! rtpamrpay ! rtpbin.send_rtp_sink_1                   
                  rtpbin.send_rtp_src_1 ! udpsink port=5002                            
                  rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false    
                  udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1

2) recv.sh

gst-launch-1.0 -v rtpbin name=rtpbin                                          
    udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,octet-align=(string)1" 
            port=5002 ! rtpbin.recv_rtp_sink_1                                
        rtpbin. ! rtpamrdepay ! amrnbdec ! alsasink                           
     udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1                               
     rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false

模拟声音和实际声卡只有发送端采集程序不同，模拟采集是audiotestsrc，实际声卡采集是pulsesrc

中国移动和对讲amr实时语音解码播放

gst-launch-1.0 udpsrc port=6000 caps="application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)AMR, payload=(int)106" ! rtpamrdepay ! decodebin name=decoder ! queue ! audioconvert ! autoaudiosink

 tcpdump -i eth0 -w dump.pcap

gstreamer中通过UDP(RTP)远程播放MP3

send.sh

gst-launch-1.0 -v filesrc location = Hopy_Always.mp3 ! decodebin ! audioconvert ! rtpL16pay ! udpsink host=127.0.0.1 port=6000

recv.sh

gst-launch-1.0 udpsrc port=6000 caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, channels=(int)2' ! rtpjitterbuffer latency=400 ! rtpL16depay ! pulsesink

Gstreamer 测试udpsink udpsrc播放mp3文件

https://blog.csdn.net/zhujinghao_09/article/details/8513962

10、gstreamer 播放mp3源码(播放器) ，入门开发极好的samples

https://blog.csdn.net/fireroll/article/details/49126827

https://www.cnblogs.com/274914765qq/p/5090299.html

11、Gstreamer的音视频同步

https://blog.csdn.net/maeom/article/details/7729840

12、播放音频

gst-launch-1.0 playbin uri=file:///home/dong/Hopy_Always.mp3
gst-launch-1.0 filesrc location=Hopy_Always.mp3 ! decodebin ! audioconvert ! audioresample ! autoaudiosink

13、Gstreamer视频传输测试gst-launch

https://blog.csdn.net/meng_tianshi/article/details/80142005

14、How to listen to the pulseaudio RTP Stream and play

https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Network/RTP/

15、Gstreamer cheat sheet —— Picture in Picture / Video Wall / Text Overlay / Time Overlay ... ... ..

http://wiki.oz9aec.net/index.php?title=Gstreamer_cheat_sheet

16、用树莓派做 RTMP 流直播服务器，可推送至斗鱼直播

http://shumeipai.nxez.com/2017/11/01/build-rtmp-stream-live-server-with-raspberry-pi.html

17、gstreamer学习笔记：通过gst-launch工具抓取播放的音频数据并通过upd传输

gst-launch数据转换(pcm,aac,ts), rtp收发

https://blog.csdn.net/u010312436/article/details/53335579

18、gstreamer实现摄像头的远程采集，udp传输，本地显示和保存为AVI文件 发送端

send

https://blog.csdn.net/zhujinghao_09/article/details/8528802

recv

https://blog.csdn.net/zhujinghao_09/article/details/8528879

19、QtGStreamer dvr

https://blog.csdn.net/lg1259156776/article/details/53413877

https://blog.csdn.net/xueyeguiren8/article/details/54581536

20、基于Gstreamer的实时视频流的分发

https://blog.csdn.net/sdjhs/article/details/51444934

21、gstreamer学习笔记：将音视频合成MPEG2-TS流并打包通过rtp传输

https://blog.csdn.net/u010312436/article/details/53668083

22、gstreamer之RTSP Server一个进程提供多路不同视频

https://blog.csdn.net/quantum7/article/details/82999132

23、GStreamer资料整理(包括摄像头采集,视频保存,远程监控,流媒体RTP传输)

https://blog.csdn.net/wzwxiaozheng/article/details/6099397

24、使用GStreamer作v4l2摄像头采集和输出到YUV文件及屏幕的相关测试

https://blog.csdn.net/shallon_luo/article/details/5400708

25、Gstreamer中添加x265编解码器

https://blog.csdn.net/songwater/article/details/34855883

 

26、Gstreamer One Liners

https://metalab.at/wiki/Gstreamer_One_Liners

 

ARM平台基于嵌入式Linux Gstreamer 使用

https://www.eefocus.com/toradex/blog/16-05/379143_e4fcb.html

 

常见gstreamer pipeline 命令—— TI 3730 dvsdk

https://blog.csdn.net/songwater/article/details/34800017

 

gstreamer中的好东西，appsink和appsrc

https://blog.csdn.net/jack0106/article/details/5909935

 

基于DM3730平台的gstreamer音视频传输调试

https://blog.csdn.net/goalietech/article/details/24887955

 

gstreamer appsrc appsink应用

gstreamer向appsrc发送帧画面的代码

https://blog.csdn.net/quantum7/article/details/82226608

gstreamer向appsrc发送编码数据的代码

https://blog.csdn.net/quantum7/article/details/82250524

gstreamer学习笔记：分享几个appsink和appsrc的example

https://blog.csdn.net/u010312436/article/details/53610599

 

Here are two basic send/receive  h264 video stream pipelines:

gst-launch-0.10 v4l2src ! ffmpegcolorspace ! videoscale ! video/x-raw-yuv,width=640,height=480 ! vpuenc ! h264parse ! rtph264pay ! udpsink host=localhost port=5555

gst-launch-0.10 udpsrc port=5555 ! application/x-rtp,encoding-name=H264,payload=96 ! rtph264depay ! h264parse ! ffdec_h264 ! videoconvert ! ximagesink

gstreamer使用进阶

https://blog.csdn.net/jack0106/article/details/5592557

 

# 整理了这么多，梳理一下指令，组织一下模块代码，应付常规的多媒体应用绰绰有余了！