GStreamer基础教程08——pipeline的快捷访问

目标

      GStreamer建立的pipeline不需要完全关闭。有多种方法可以让数据在任何时候送到pipeline中或者从pipeline中取出。本教程会展示：

      如何把外部数据送到pipeline中

      如何把数据从pipeline中取出

      如何操作这些数据

介绍

      有几种方法可以让应用通过pipeline和数据流交互。本教程讲述了最简单的一种，因为使用了专门为这个而创建的element。

      专门让应用可以往pipeline里面传入数据的element时appsrc，而appsink就正好相反，让应用可以从pipeline中获得数据。为了避免混淆，我们可以这么来理解，appsrc是一个普通的source element，不过它的数据都是来自外太空，而appsink是一个普通的sink element，数据从这里出去的就消失不见了。

      appsrc和appsink用得非常多，所以他们都自己提供API，你只要连接了gstreamer-app库，那么就可以访问到。在本教程里，我们会使用一种简单地方法通过信号来实现。

      appsrc可以有不同的工作模式：在pull模式，在需要时向应用请求数据；在push模式，应用根据自己的节奏把数据推送过来。而且，在push模式，如果已经有了足够的数据，应用可以在push时被阻塞，或者可以经由enough-data和need-data信号来控制。本教程中得例子就采用了这种信号控制的方式，其他没有提及的方法可以在appsrc的文档中查阅。

Buffers

      通过pipeline传递的大块数据被称为buffers。因为本例子会制造数据同时也消耗数据，所以我们需要了解GstBuffer。

      Source Pads负责制造buffer，这些buffer被sink pad消耗掉。GStreamer在一个个element之间传递这些buffer。

      一个buffer只能简单地描述一小片数据，不要认为我们所有的buffer都是一样大小的。而且，buffer有一个时间戳和有效期，这个就描述了什么时候buffer里的数据需要渲染出来。时间戳是个非常复杂和精深的话题，但目前这个简单地解释也足够了。

      作为一个例子，一个filesrc会提供“ANY”属性的buffers并且没有时间戳信息。在demux（《GStreamer基础教程03——动态pipeline》）之后，buffers会有一些特定的cap了，比如"video/x-h264"，在解码后，每一个buffer都会包含一帧有原始caps的视频帧（比如：video/x-raw-yuv），并且有非常明确地时间戳用来指示这一帧在什么时候显示。

教程

      本教程是上一篇教程（《GStreamer基础教程07——多线程和Pad的有效性》）在两个方面的扩展：第一是用appsrc来取代audiotestsrc来生成音频数据；第二是在tee里新加了一个分支，这样流入audio sink和波形显示的数据同样复制了一份传给appsink。这个appsink就把信息回传给应用，应用就可以通知用户收到了数据或者做其他更复杂的工作。

一个粗糙的波形发生器

#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, *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;
  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;
  }
}

/* The appsink has received a buffer */
static void new_buffer (GstElement *sink, CustomData *data) {
  GstBuffer *buffer;

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

/* 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;
  GstPadTemplate *tee_src_pad_template;
  GstPad *tee_audio_pad, *tee_video_pad, *tee_app_pad;
  GstPad *queue_audio_pad, *queue_video_pad, *queue_app_pad;
  gchar *audio_caps_text;
  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 ("ffmpegcolorspace", "csp");
  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 */
  audio_caps_text = g_strdup_printf (AUDIO_CAPS, SAMPLE_RATE);
  audio_caps = gst_caps_from_string (audio_caps_text);
  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);

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

  /* 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_src_pad_template = gst_element_class_get_pad_template (GST_ELEMENT_GET_CLASS (data.tee), "src%d");
  tee_audio_pad = gst_element_request_pad (data.tee, tee_src_pad_template, NULL, NULL);
  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 (data.tee, tee_src_pad_template, NULL, NULL);
  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 (data.tee, tee_src_pad_template, NULL, NULL);
  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;
}

工作流程

      创建pipeline段的代码就是上一篇的教程中得例子的扩大版。包括初始或所有的element，连接有Always Pad的element然后手动连接tee element的Request Pad。

      下面我们关注一下appsrc和appsink这两个element的配置：

  /* Configure appsrc */
  audio_caps_text = g_strdup_printf (AUDIO_CAPS, SAMPLE_RATE);
  audio_caps = gst_caps_from_string (audio_caps_text);
  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);

      appsrc里面第一个需要关注的属性是caps。它说明了element准备生成的数据的类型，这样GStreamer就可以检查下游的element看看是否支持。这个属性必须是一个GstCaps对象，这个对象可以很方便的由gst_caps_from_string()来生成。

      然后我们把need-data和enough-data信号和回调连接起来，这样在appsrc内部的队列里面数据不足或将要满地时候会发送信号，我们就用这些信号来启动/停止我们的信号发生过程。

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

      关于appsink的配置，我们连接了new-buffer的信号，这个信号在每次收到buffer的时候发出。当然，这个信号的发出需要emit-signals这个信号属性被开启（默认是关闭的）。

      启动pipeline，等到消息和最后的清理资源都和以前的没什么区别。让我们关注我们刚刚注册的回调吧。

/* 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);
  }
}

      这个函数在appsrc内部队列将要空的时候调用，在这里我们做的事情仅仅是用g_idle_add()方法注册一个GLib的idle函数，这个函数会给appsrc输入数据知道内部队列满为止。一个GLib的idle函数是一个GLib在主循环在“idle”时会调用的方法，也就是说，当时没有更高优先级的任务运行。

      这只是appsrc多种发出数据方法中的一个。特别需要指出的是，buffer不是必须要在主线程中用GLib方法来传递给appsrc的，你也不是一定要用need-data和enough-data信号来同步appsrc的（据说这样最方便）。

      我们记录下g_idle_add()的返回的sourceid，这样后面可以关掉它。

/* 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;
  }
}

      这个函数当appsrc内部的队列满的时候调用，所以我们需要停止发送数据。这里我们简单地用g_source_remove()来把idle函数移走。

/* 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;
  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);

      这个函数给appsrc发送数据。它被GLib调用的次数和频率我们不加以控制，但我们会在它任务完成时关闭它（appsrc内部队列满）。

      这里第一步是用gst_buffer_new_and_alloc()方法和给定的大小创建一个新buffer（例子中是1024字节）。

      我们计算我们生成的采样数据的数据量，把数据存在CustomData.num_samples里面，这样我们可以用GstBuffer提供的GST_BUFFER_TIMESTAMP宏来生成buffer的时间戳。

      gst_util_uint64_scale是一个工具函数，用来缩放数据，确保不会溢出。

      这些给buffer的数据可以用GstBuffer提供的GST_BUFFER_DATA宏来访问。

      我们会跳过波形的生成部分，因为这不是本教程要讲述的内容。

  /* 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);

      一旦我们的buffer已经准备好，我们把带着这个buffer的push-buffer信号传给appsrc，然后就调用gst_buffer_unref()方法，因为我们不会再用到它了。

/* The appsink has received a buffer */
static void new_buffer (GstElement *sink, CustomData *data) {
  GstBuffer *buffer;

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

      最后，这个函数在appsink收到buffer时被调用。我们使用了pull-buffer的信号来重新获得buffer，因为是例子，所以仅仅在屏幕上打印一些内容。我们可以用GstBuffer的GST_BUFFER_DATA宏来获得数据指针和用GST_BUFFER_SIZE宏来获得数据大小。请记住，这里的buffer不是一定要和我们在push_data函数里面创建的buffer完全一致的，在传输路径上得任何一个element都可能对buffer进行一些改变。（这个例子中仅仅是在appsrc和appsink中间通过一个tee element，所以buffer没有变化）。

    请不要忘记调用gst_buffer_unref()来释放buffer，就讲这么多吧。