gstreamer-基础教程8-appsrc和appsink的应用

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);

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)

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 <gst/gst.h>
#include <gst/audio/audio.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 */

/* 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;
}

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`

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

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

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