【GStreamer开发】GStreamer播放教程03——pipeline的快捷访问

发布时间：2020-12-14 01:51:14 所属栏目：百科来源：网络整理

导读：目的《GStreamer08——pipeline的快捷访问》展示了一个应用如何用appsrc和appsink这两个特殊的element在pipeline中手动输入/提取数据。playbin2也允许使用这两个element，但连接它们的方法有所不同。连接appsink到playbin2的方法在后面还会提到。这里我们主

目的

《GStreamer08——pipeline的快捷访问》展示了一个应用如何用appsrc和appsink这两个特殊的element在pipeline中手动输入/提取数据。playbin2也允许使用这两个element，但连接它们的方法有所不同。连接appsink到playbin2的方法在后面还会提到。这里我们主要讲述：

如何把appsrc连接到playbin2

如何配置appsrc

一个playbin2波形发生器

[objc]view plaincopy 
   
 <spanstyle="font-size:14px;">#include<gst/gst.h>
#include<string.h>
#defineCHUNK_SIZE1024/*Amountofbyteswearesendingineachbuffer*/
#defineSAMPLE_RATE44100/*Samplespersecondwearesending*/
#defineAUDIO_CAPS"audio/x-raw-int,channels=1,rate=%d,signed=(boolean)true,width=16,depth=16,endianness=BYTE_ORDER"
/*Structuretocontainallourinformation,sowecanpassittocallbacks*/
typedefstruct_CustomData{
GstElement*pipeline;
GstElement*app_source;
guint64num_samples;/*Numberofsamplesgeneratedsofar(fortimestampgeneration)*/
gfloata,b,c,d;/*Forwaveformgeneration*/
guintsourceid;/*TocontroltheGSource*/
GMainLoop*main_loop;/*GLib'sMainLoop*/
}CustomData;
/*ThismethodiscalledbytheidleGSourceinthemainloop,tofeedCHUNK_SIZEbytesintoappsrc.
*Theidehandlerisaddedtothemainloopwhenappsrcrequestsustostartsendingdata(need-datasignal)
*andisremovedwhenappsrchasenoughdata(enough-datasignal).
*/
staticgbooleanpush_data(CustomData*data){
GstBuffer*buffer;
GstFlowReturnret;
inti;
gint16*raw;
gintnum_samples=CHUNK_SIZE/2;/*Becauseeachsampleis16bits*/
gfloatfreq;
/*Createanewemptybuffer*/
buffer=gst_buffer_new_and_alloc(CHUNK_SIZE);
/*Setitstimestampandduration*/
GST_BUFFER_TIMESTAMP(buffer)=gst_util_uint64_scale(data->num_samples,GST_SECOND,SAMPLE_RATE);
GST_BUFFER_DURATION(buffer)=gst_util_uint64_scale(CHUNK_SIZE,SAMPLE_RATE);
/*Generatesomepsychodelicwaveforms*/
raw=(gint16*)GST_BUFFER_DATA(buffer);
data->c+=data->d;
data->d-=data->c/1000;
freq=1100+11000*data->d;
for(i=0;i<num_samples;i++){
data->a+=data->b;
data->b-=data->a/freq;
raw[i]=(gint16)(5500*data->a);
}
data->num_samples+=num_samples;
/*Pushthebufferintotheappsrc*/
g_signal_emit_by_name(data->app_source,"push-buffer",buffer,&ret);
/*Freethebuffernowthatwearedonewithit*/
gst_buffer_unref(buffer);
if(ret!=GST_FLOW_OK){
/*Wegotsomeerror,stopsendingdata*/
returnFALSE;
returnTRUE;
}
/*Thissignalcallbacktriggerswhenappsrcneedsdata.Here,weaddanidlehandler
*tothemainlooptostartpushingdataintotheappsrc*/
staticvoidstart_feed(GstElement*source,guintsize,CustomData*data){
if(data->sourceid==0){
g_print("Startfeedingn");
data->sourceid=g_idle_add((GSourceFunc)push_data,data);
/*Thiscallbacktriggerswhenappsrchasenoughdataandwecanstopsending.
*Weremovetheidlehandlerfromthemainloop*/
staticvoidstop_feed(GstElement*source,CustomData*data){
if(data->sourceid!=0){
g_print("Stopfeedingn");
g_source_remove(data->sourceid);
data->sourceid=0;
/*Thisfunctioniscalledwhenanerrormessageispostedonthebus*/
staticvoiderror_cb(GstBus*bus,GstMessage*msg,0)">GError*err;
gchar*debug_info;
/*Printerrordetailsonthescreen*/
gst_message_parse_error(msg,&err,&debug_info);
g_printerr("Errorreceivedfromelement%s:%sn",GST_OBJECT_NAME(msg->src),err->message);
g_printerr("Debugginginformation:%sn",debug_info?debug_info:"none");
g_clear_error(&err);
g_free(debug_info);
g_main_loop_quit(data->main_loop);
/*Thisfunctioniscalledwhenplaybin2hascreatedtheappsrcelement,sowehave
*achancetoconfigureit.*/
staticvoidsource_setup(GstElement*pipeline,GstElement*source,0)">gchar*audio_caps_text;
GstCaps*audio_caps;
g_print("Sourcehasbeencreated.Configuring.n");
data->app_source=source;
/*Configureappsrc*/
audio_caps_text=g_strdup_printf(AUDIO_CAPS,SAMPLE_RATE);
audio_caps=gst_caps_from_string(audio_caps_text);
g_object_set(source,"caps",audio_caps,NULL);
g_signal_connect(source,"need-data",G_CALLBACK(start_feed),data);
"enough-data",G_CALLBACK(stop_feed),data);
gst_caps_unref(audio_caps);
g_free(audio_caps_text);
intmain(intargc,charchar*argv[]){
CustomDatadata;
GstBus*bus;
/*Initializecumstomdatastructure*/
memset(&data,0,sizeof(data));
data.b=1;/*Forwaveformgeneration*/
data.d=1;
/*InitializeGStreamer*/
gst_init(&argc,&argv);
/*Createtheplaybin2element*/
data.pipeline=gst_parse_launch("playbin2uri=appsrc://",0)">g_signal_connect(data.pipeline,"source-setup",G_CALLBACK(source_setup),&data);
/*Instructthebustoemitsignalsforeachreceivedmessage,andconnecttotheinterestingsignals*/
bus=gst_element_get_bus(data.pipeline);
gst_bus_add_signal_watch(bus);
g_signal_connect(G_OBJECT(bus),"message::error",(GCallback)error_cb,&data);
gst_object_unref(bus);
/*Startplayingthepipeline*/
gst_element_set_state(data.pipeline,GST_STATE_PLAYING);
/*CreateaGLibMainLoopandsetittorun*/
data.main_loop=g_main_loop_new(NULL,FALSE);
g_main_loop_run(data.main_loop);
/*Freeresources*/
gst_object_unref(data.pipeline);
return0;
</span>

把appsrc用作pipeline的source，仅仅把playbin2的UIR设置成appsrc://即可。

[objc]view plaincopy 
   
<spanstyle="font-size:14px;">/*Createtheplaybin2element*/NULL);</span>

playbin2创建一个内部的appsrc element并且发送source-setup信号来通知应用进行设置。

[objc]view plaincopy 
   
<spanstyle="font-size:14px;">g_signal_connect(data.pipeline,&data);</span>

特别地，设置appsrc的caps属性是很重要的，因为一旦这个信号的处理返回，playbin2就会根据返回值来初始化下一个element。

[objc]view plaincopy 
   
<spanstyle="font-size:14px;">/*Thisfunctioniscalledwhenplaybin2hascreatedtheappsrcelement,sowehave*achancetoconfigureit.*/
gchar*audio_caps_text;
GstCaps*audio_caps;
g_print("Sourcehasbeencreated.Configuring.n");
data->app_source=source;
/*Configureappsrc*/
audio_caps=gst_caps_from_string(audio_caps_text);
NULL);
gst_caps_unref(audio_caps);
g_free(audio_caps_text);
}</span>

appsrc的配置和《GStreamer08——pipeline的快捷访问》里面一样：caps设置成audio/x-raw-int，注册两个回调，这样element可以在需要/停止给它推送数据时通知应用。具体细节请参考《GStreamer08——pipeline的快捷访问》。

在这个点之后，playbin2接管处理了剩下的pipeline，应用仅仅需要生成数据即可。

至于使用appsink来从从playbin2里面提取数据，在后面的教程里面再讲述。

（编辑：李大同）

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