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
@H_502_25@#include <gst/gst.h> #include <string.h> #define CHUNK_SIZE 1024 /* Amount of bytes we are sending in each buffer */ #define SAMPLE_RATE 44100 /* Samples per second we are sending */ #define AUdio_CAPS "audio/x-raw-int,channels=1,rate=%d,signed=(boolean)true,width=16,depth=16,endianness=BYTE_ORDER" /* Structure to contain all our information,so we can pass it to callbacks */ typedef struct _CustomData { GstElement *pipeline; GstElement *app_source; guint64 num_samples; /* Number of samples generated so far (for timestamp generation) */ gfloat a,b,c,d; /* For waveform generation */ guint sourceid; /* To control the GSource */ GMainLoop *main_loop; /* GLib's Main Loop */ } CustomData; /* This method is called by the idle GSource in the mainloop,to Feed CHUNK_SIZE bytes into appsrc. * The ide handler is added to the mainloop when appsrc requests us to start sending data (need-data signal) * and is removed when appsrc has enough data (enough-data signal). */ static gboolean push_data (CustomData *data) { GstBuffer *buffer; GstFlowReturn ret; int i; gint16 *raw; gint num_samples = CHUNK_SIZE / 2; /* Because each sample is 16 bits */ gfloat freq; /* Create a new empty buffer */ buffer = gst_buffer_new_and_alloc (CHUNK_SIZE); /* Set its timestamp and duration */ 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); /* Generate some psychodelic waveforms */ raw = (gint16 *)GST_BUFFER_DATA (buffer); data->c += data->d; data->d -= data->c / 1000; freq = 1100 + 1000 * data->d; for (i = 0; i < num_samples; i++) { data->a += data->b; data->b -= data->a / freq; raw[i] = (gint16)(500 * data->a); } data->num_samples += num_samples; /* Push the buffer into the appsrc */ g_signal_emit_by_name (data->app_source,"push-buffer",buffer,&ret); /* Free the buffer now that we are done with it */ gst_buffer_unref (buffer); if (ret != GST_FLOW_OK) { /* We got some error,stop sending data */ return FALSE; } return TRUE; } /* This signal callback triggers when appsrc needs data. Here,we add an idle handler * to the mainloop to start pushing data into the appsrc */ static void start_Feed (GstElement *source,guint size,CustomData *data) { if (data->sourceid == 0) { g_print ("Start Feeding\n"); data->sourceid = g_idle_add ((GSourceFunc) push_data,data); } } /* This callback triggers when appsrc has enough data and we can stop sending. * We remove the idle handler from the mainloop */ static void stop_Feed (GstElement *source,CustomData *data) { if (data->sourceid != 0) { g_print ("Stop Feeding\n"); g_source_remove (data->sourceid); data->sourceid = 0; } } /* This function is called when an error message is posted on the bus */ static void error_cb (GstBus *bus,GstMessage *msg,CustomData *data) { GError *err; gchar *debug_info; /* Print error details on the screen */ gst_message_parse_error (msg,&err,&debug_info); g_printerr ("Error received from element %s: %s\n",GST_OBJECT_NAME (msg->src),err->message); g_printerr ("Debugging information: %s\n",debug_info ? debug_info : "none"); g_clear_error (&err); g_free (debug_info); g_main_loop_quit (data->main_loop); } /* This function is called when playbin2 has created the appsrc element,so we have * a chance to configure it. */ static void source_setup (GstElement *pipeline,GstElement *source,CustomData *data) { gchar *audio_caps_text; GstCaps *audio_caps; g_print ("Source has been created. Configuring.\n"); data->app_source = source; /* Configure appsrc */ 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); g_signal_connect (source,"enough-data",G_CALLBACK (stop_Feed),data); gst_caps_unref (audio_caps); g_free (audio_caps_text); } int main(int argc,char *argv[]) { CustomData data; GstBus *bus; /* Initialize cumstom data structure */ memset (&data,sizeof (data)); data.b = 1; /* For waveform generation */ data.d = 1; /* Initialize GStreamer */ gst_init (&argc,&argv); /* Create the playbin2 element */ data.pipeline = gst_parse_launch ("playbin2 uri=appsrc://",NULL); g_signal_connect (data.pipeline,"source-setup",G_CALLBACK (source_setup),&data); /* Instruct the bus to emit signals for each received message,and connect to the interesting signals */ 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); /* Start playing the pipeline */ gst_element_set_state (data.pipeline,GST_STATE_PLAYING); /* Create a GLib Main Loop and set it to run */ data.main_loop = g_main_loop_new (NULL,FALSE); g_main_loop_run (data.main_loop); /* Free resources */ gst_element_set_state (data.pipeline,GST_STATE_NULL); gst_object_unref (data.pipeline); return 0; }把appsrc用作pipeline的source，仅仅把playbin2的UIR设置成appsrc://即可。
/* Create the playbin2 element */
data.pipeline = gst_parse_launch ("playbin2 uri=appsrc://",NULL);
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。
g_signal_connect (data.pipeline,&data);
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。
/* This function is called when playbin2 has created the appsrc element,data);
gst_caps_unref (audio_caps);
g_free (audio_caps_text);
}
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。
原文链接：https://www.f2er.com/javaschema/285705.html