gstreamer-基础教程8

基础教程 8：Short-cutting the pipeline goal

这个教程通过appsrc元素，将应用程序数据注入 GStreamer 管道，并且使用appsink元素将 GStreamer 数据提取回应用程序。通过本教程，可以学习到：

• 如何将外部数据注入通用 GStreamer 管道。

• 如何从通用 GStreamer 管道中提取数据。

• 如何访问和操作这些数据。

说明

• appsrc将生成音频数据
• tee分成三路：
  • audio sink一路播放音频
  • video sink一路显示波形图像
  • app sink一路通知用户已收到数据
    

关于appsrc和appsink元素的配置：

/* Configure appsrc */
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, SAMPLE_RATE, 1, NULL);
audio_caps = gst_audio_info_to_caps (&info);
g_object_set (data.app_source, "caps", audio_caps, NULL);
g_signal_connect (data.app_source, "need-data", G_CALLBACK (start_feed), &data);
g_signal_connect (data.app_source, "enough-data", G_CALLBACK (stop_feed), &data);
1
2
3
4
5
6

在appsrc上，需要设置的第一个属性appsrc是caps，它指定元素将要生成的数据类型，因此 GStreamer 可以检查是否可以与下游元素链接（也就是说，下游元素是否会理解这种数据）。该属性必须是一个GstCaps对象，它可以很容易地从gst_caps_from_string()获得.

然后我们连接到need-data和enough-data信号，它们分别在appsrc内部数据队列运行不足或者满时触发。

/* Configure appsink */
g_object_set (data.app_sink, "emit-signals", TRUE, "caps", audio_caps, NULL);
g_signal_connect (data.app_sink, "new-sample", G_CALLBACK (new_sample), &data);
gst_caps_unref (audio_caps)
1
2
3
4

在appsink上，我们连接new-sample信号到new_sample处理函数，appsink接收到新的buffer时会触发这个signal。此外，需要通过该emit-signals属性启用信号发射，因为默认情况下它是禁用的。

全部代码路径:
gst-docs/examples/tutorials

#include 
#include 
#include 

#define CHUNK_SIZE 1024         /* Amount of bytes we are sending in each buffer */
#define SAMPLE_RATE 44100       /* Samples per second we are sending */

/* Structure to contain all our information, so we can pass it to callbacks */
typedef struct _CustomData
{
  GstElement *pipeline, *app_source, *tee, *audio_queue, *audio_convert1,
      *audio_resample, *audio_sink;
  GstElement *video_queue, *audio_convert2, *visual, *video_convert,
      *video_sink;
  GstElement *app_queue, *app_sink;

  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 idle 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;
  GstMapInfo map;
  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 (num_samples, GST_SECOND, SAMPLE_RATE);

  /* Generate some psychodelic waveforms */
  gst_buffer_map (buffer, &map, GST_MAP_WRITE);
  raw = (gint16 *) map.data;
  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);
  }
  gst_buffer_unmap (buffer, &map);
  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;
  }
}

/* The appsink has received a buffer */
static GstFlowReturn
new_sample (GstElement * sink, CustomData * data)
{
  GstSample *sample;

  /* Retrieve the buffer */
  g_signal_emit_by_name (sink, "pull-sample", &sample);
  if (sample) {
    /* The only thing we do in this example is print a * to indicate a received buffer */
    g_print ("*");
    gst_sample_unref (sample);
    return GST_FLOW_OK;
  }

  return GST_FLOW_ERROR;
}

/* 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);
}

int
main (int argc, char *argv[])
{
  CustomData data;
  GstPad *tee_audio_pad, *tee_video_pad, *tee_app_pad;
  GstPad *queue_audio_pad, *queue_video_pad, *queue_app_pad;
  GstAudioInfo info;
  GstCaps *audio_caps;
  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 elements */
  data.app_source = gst_element_factory_make ("appsrc", "audio_source");
  data.tee = gst_element_factory_make ("tee", "tee");
  data.audio_queue = gst_element_factory_make ("queue", "audio_queue");
  data.audio_convert1 =
      gst_element_factory_make ("audioconvert", "audio_convert1");
  data.audio_resample =
      gst_element_factory_make ("audioresample", "audio_resample");
  data.audio_sink = gst_element_factory_make ("autoaudiosink", "audio_sink");
  data.video_queue = gst_element_factory_make ("queue", "video_queue");
  data.audio_convert2 =
      gst_element_factory_make ("audioconvert", "audio_convert2");
  data.visual = gst_element_factory_make ("wavescope", "visual");
  data.video_convert =
      gst_element_factory_make ("videoconvert", "video_convert");
  data.video_sink = gst_element_factory_make ("autovideosink", "video_sink");
  data.app_queue = gst_element_factory_make ("queue", "app_queue");
  data.app_sink = gst_element_factory_make ("appsink", "app_sink");

  /* Create the empty pipeline */
  data.pipeline = gst_pipeline_new ("test-pipeline");

  if (!data.pipeline || !data.app_source || !data.tee || !data.audio_queue
      || !data.audio_convert1 || !data.audio_resample || !data.audio_sink
      || !data.video_queue || !data.audio_convert2 || !data.visual
      || !data.video_convert || !data.video_sink || !data.app_queue
      || !data.app_sink) {
    g_printerr ("Not all elements could be created.\n");
    return -1;
  }

  /* Configure wavescope */
  g_object_set (data.visual, "shader", 0, "style", 0, NULL);

  /* Configure appsrc */
  gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, SAMPLE_RATE, 1, NULL);
  audio_caps = gst_audio_info_to_caps (&info);
  g_object_set (data.app_source, "caps", audio_caps, "format", GST_FORMAT_TIME,
      NULL);
  g_signal_connect (data.app_source, "need-data", G_CALLBACK (start_feed),
      &data);
  g_signal_connect (data.app_source, "enough-data", G_CALLBACK (stop_feed),
      &data);

  /* Configure appsink */
  g_object_set (data.app_sink, "emit-signals", TRUE, "caps", audio_caps, NULL);
  g_signal_connect (data.app_sink, "new-sample", G_CALLBACK (new_sample),
      &data);
  gst_caps_unref (audio_caps);

  /* Link all elements that can be automatically linked because they have "Always" pads */
  gst_bin_add_many (GST_BIN (data.pipeline), data.app_source, data.tee,
      data.audio_queue, data.audio_convert1, data.audio_resample,
      data.audio_sink, data.video_queue, data.audio_convert2, data.visual,
      data.video_convert, data.video_sink, data.app_queue, data.app_sink, NULL);
  if (gst_element_link_many (data.app_source, data.tee, NULL) != TRUE
      || gst_element_link_many (data.audio_queue, data.audio_convert1,
          data.audio_resample, data.audio_sink, NULL) != TRUE
      || gst_element_link_many (data.video_queue, data.audio_convert2,
          data.visual, data.video_convert, data.video_sink, NULL) != TRUE
      || gst_element_link_many (data.app_queue, data.app_sink, NULL) != TRUE) {
    g_printerr ("Elements could not be linked.\n");
    gst_object_unref (data.pipeline);
    return -1;
  }

  /* Manually link the Tee, which has "Request" pads */
  tee_audio_pad = gst_element_request_pad_simple (data.tee, "src_%u");
  g_print ("Obtained request pad %s for audio branch.\n",
      gst_pad_get_name (tee_audio_pad));
  queue_audio_pad = gst_element_get_static_pad (data.audio_queue, "sink");
  tee_video_pad = gst_element_request_pad_simple (data.tee, "src_%u");
  g_print ("Obtained request pad %s for video branch.\n",
      gst_pad_get_name (tee_video_pad));
  queue_video_pad = gst_element_get_static_pad (data.video_queue, "sink");
  tee_app_pad = gst_element_request_pad_simple (data.tee, "src_%u");
  g_print ("Obtained request pad %s for app branch.\n",
      gst_pad_get_name (tee_app_pad));
  queue_app_pad = gst_element_get_static_pad (data.app_queue, "sink");
  if (gst_pad_link (tee_audio_pad, queue_audio_pad) != GST_PAD_LINK_OK ||
      gst_pad_link (tee_video_pad, queue_video_pad) != GST_PAD_LINK_OK ||
      gst_pad_link (tee_app_pad, queue_app_pad) != GST_PAD_LINK_OK) {
    g_printerr ("Tee could not be linked\n");
    gst_object_unref (data.pipeline);
    return -1;
  }
  gst_object_unref (queue_audio_pad);
  gst_object_unref (queue_video_pad);
  gst_object_unref (queue_app_pad);

  /* 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);

  /* Release the request pads from the Tee, and unref them */
  gst_element_release_request_pad (data.tee, tee_audio_pad);
  gst_element_release_request_pad (data.tee, tee_video_pad);
  gst_element_release_request_pad (data.tee, tee_app_pad);
  gst_object_unref (tee_audio_pad);
  gst_object_unref (tee_video_pad);
  gst_object_unref (tee_app_pad);

  /* Free resources */
  gst_element_set_state (data.pipeline, GST_STATE_NULL);
  gst_object_unref (data.pipeline);
  return 0;
}
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270

编译

这里除了用到gstreamer-1.0的基础库，还需要指定audio的库，所以pkg-config后面要跟gstreamer-audio-1.0参数：

gcc basic-tutorial-8.c -o tutorial-8 `pkg-config --cflags --libs gstreamer-audio-1.0`
1

编译后，运行tutorial-8，就可以看到下图的效果：

pkg-config的用法

$ pkg-config --cflags --libs gstreamer-audio-1.0

-pthread -I/usr/local/include/gstreamer-1.0 -I/usr/local/include/orc-0.4 -I/usr/local/include/gstreamer-1.0 -I/usr/include/glib-2.0 -I/usr/lib/x86_64-linux-gnu/glib-2.0/include -L/usr/local/lib/x86_64-linux-gnu -lgstaudio-1.0 -lgstbase-1.0 -lgstreamer-1.0 -lgobject-2.0 -lglib-2.0
1
2
3

pkg-config后面的参数可以跟那些，完全取决于安装或者编译生成的.pc文件有哪些，比如用gstreamer最近的版本编译的pc文件有下面这么多，那么都可以通过pkg-config命令查到。

gst-editing-services-1.0.pc  gstreamer-base-1.0.pc          gstreamer-gl-egl-1.0.pc         gstreamer-pbutils-1.0.pc       gstreamer-riff-1.0.pc         gstreamer-tag-1.0.pc         orc-0.4.pc
gstreamer-1.0.pc             gstreamer-check-1.0.pc         gstreamer-gl-prototypes-1.0.pc  gstreamer-photography-1.0.pc   gstreamer-rtp-1.0.pc          gstreamer-transcoder-1.0.pc  orc-test-0.4.pc
gstreamer-allocators-1.0.pc  gstreamer-codecparsers-1.0.pc  gstreamer-gl-x11-1.0.pc         gstreamer-play-1.0.pc          gstreamer-rtsp-1.0.pc         gstreamer-video-1.0.pc
gstreamer-app-1.0.pc         gstreamer-controller-1.0.pc    gstreamer-insertbin-1.0.pc      gstreamer-player-1.0.pc        gstreamer-rtsp-server-1.0.pc  gstreamer-webrtc-1.0.pc
gstreamer-audio-1.0.pc       gstreamer-fft-1.0.pc           gstreamer-mpegts-1.0.pc         gstreamer-plugins-bad-1.0.pc   gstreamer-sctp-1.0.pc         gst-validate-1.0.pc
gstreamer-bad-audio-1.0.pc   gstreamer-gl-1.0.pc            gstreamer-net-1.0.pc            gstreamer-plugins-base-1.0.pc  gstreamer-sdp-1.0.pc          nice.pc
1
2
3
4
5
6

比如:

pkg-config --cflags --libs gstreamer-play-1.0
pkg-config --cflags --libs gstreamer-rtsp-server-1.0
pkg-config --cflags --libs pkg-config --cflags --libs gstreamer-webrtc-1.0
1
2
3