GStreamer播放教程03——pipeline的快捷访问

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

#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, GST_SECOND, 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, 0, 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, "source-setup", G_CALLBACK (source_setup), &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, 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);}

      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。