gstreamer学习（3）——动态创建pipeline

最新推荐文章于 2024-01-13 15:41:59 发布

东楠爱学习

最新推荐文章于 2024-01-13 15:41:59 发布

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本文链接：https://blog.csdn.net/weixin_39592381/article/details/134786562

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实例

惯例，先上官网示例代码：

示例代码

#include <gst/gst.h>

/* Structure to contain all our information, so we can pass it to callbacks */
typedef struct _CustomData {
  GstElement *pipeline;
  GstElement *source;
  GstElement *convert;
  GstElement *resample;
  GstElement *sink;
} CustomData;

/* Handler for the pad-added signal */
static void pad_added_handler (GstElement *src, GstPad *pad, CustomData *data);

int main(int argc, char *argv[]) {
  CustomData data;
  GstBus *bus;
  GstMessage *msg;
  GstStateChangeReturn ret;
  gboolean terminate = FALSE;

  /* Initialize GStreamer */
  gst_init (&argc, &argv);

  /* Create the elements */
  data.source = gst_element_factory_make ("uridecodebin", "source");
  data.convert = gst_element_factory_make ("audioconvert", "convert");
  data.resample = gst_element_factory_make ("audioresample", "resample");
  data.sink = gst_element_factory_make ("autoaudiosink", "sink");

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

  if (!data.pipeline || !data.source || !data.convert || !data.resample || !data.sink) {
    g_printerr ("Not all elements could be created.\n");
    return -1;
  }

  /* Build the pipeline. Note that we are NOT linking the source at this
   * point. We will do it later. */
  gst_bin_add_many (GST_BIN (data.pipeline), data.source, data.convert, data.resample, data.sink, NULL);
  if (!gst_element_link_many (data.convert, data.resample, data.sink, NULL)) {
    g_printerr ("Elements could not be linked.\n");
    gst_object_unref (data.pipeline);
    return -1;
  }

  /* Set the URI to play */
  g_object_set (data.source, "uri", "https://gstreamer.freedesktop.org/data/media/sintel_trailer-480p.webm", NULL);

  /* Connect to the pad-added signal */
  g_signal_connect (data.source, "pad-added", G_CALLBACK (pad_added_handler), &data);

  /* Start playing */
  ret = gst_element_set_state (data.pipeline, GST_STATE_PLAYING);
  if (ret == GST_STATE_CHANGE_FAILURE) {
    g_printerr ("Unable to set the pipeline to the playing state.\n");
    gst_object_unref (data.pipeline);
    return -1;
  }

  /* Listen to the bus */
  bus = gst_element_get_bus (data.pipeline);
  do {
    msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
        GST_MESSAGE_STATE_CHANGED | GST_MESSAGE_ERROR | GST_MESSAGE_EOS);

    /* Parse message */
    if (msg != NULL) {
      GError *err;
      gchar *debug_info;

      switch (GST_MESSAGE_TYPE (msg)) {
        case GST_MESSAGE_ERROR:
          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);
          terminate = TRUE;
          break;
        case GST_MESSAGE_EOS:
          g_print ("End-Of-Stream reached.\n");
          terminate = TRUE;
          break;
        case GST_MESSAGE_STATE_CHANGED:
          /* We are only interested in state-changed messages from the pipeline */
          if (GST_MESSAGE_SRC (msg) == GST_OBJECT (data.pipeline)) {
            GstState old_state, new_state, pending_state;
            gst_message_parse_state_changed (msg, &old_state, &new_state, &pending_state);
            g_print ("Pipeline state changed from %s to %s:\n",
                gst_element_state_get_name (old_state), gst_element_state_get_name (new_state));
          }
          break;
        default:
          /* We should not reach here */
          g_printerr ("Unexpected message received.\n");
          break;
      }
      gst_message_unref (msg);
    }
  } while (!terminate);

  /* Free resources */
  gst_object_unref (bus);
  gst_element_set_state (data.pipeline, GST_STATE_NULL);
  gst_object_unref (data.pipeline);
  return 0;
}

/* This function will be called by the pad-added signal */
static void pad_added_handler (GstElement *src, GstPad *new_pad, CustomData *data) {
  GstPad *sink_pad = gst_element_get_static_pad (data->convert, "sink");
  GstPadLinkReturn ret;
  GstCaps *new_pad_caps = NULL;
  GstStructure *new_pad_struct = NULL;
  const gchar *new_pad_type = NULL;

  g_print ("Received new pad '%s' from '%s':\n", GST_PAD_NAME (new_pad), GST_ELEMENT_NAME (src));

  /* If our converter is already linked, we have nothing to do here */
  if (gst_pad_is_linked (sink_pad)) {
    g_print ("We are already linked. Ignoring.\n");
    goto exit;
  }

  /* Check the new pad's type */
  new_pad_caps = gst_pad_get_current_caps (new_pad);
  new_pad_struct = gst_caps_get_structure (new_pad_caps, 0);
  new_pad_type = gst_structure_get_name (new_pad_struct);
  if (!g_str_has_prefix (new_pad_type, "audio/x-raw")) {
    g_print ("It has type '%s' which is not raw audio. Ignoring.\n", new_pad_type);
    goto exit;
  }

  /* Attempt the link */
  ret = gst_pad_link (new_pad, sink_pad);
  if (GST_PAD_LINK_FAILED (ret)) {
    g_print ("Type is '%s' but link failed.\n", new_pad_type);
  } else {
    g_print ("Link succeeded (type '%s').\n", new_pad_type);
  }

exit:
  /* Unreference the new pad's caps, if we got them */
  if (new_pad_caps != NULL)
    gst_caps_unref (new_pad_caps);

  /* Unreference the sink pad */
  gst_object_unref (sink_pad);
}

运行结果

在这里插入图片描述
同时，耳机中会有音频播放（因为这个教程就是只把audio部分播放出来）。

概念

signals

信号用来对感兴趣的事件执行某些指定操作，它通过名字区分不同信号，并且每个GObject都有自己的信号。

GStreamer States

gstreamer有4种states，分别是：

state	描述
NULL	元素的初始状态
READY	元素已经准备好进入PAUSED状态了
PAUSED	元素被暂停，它已经准备好接收数据并处理它，sink元素只接收一个buffer并且阻塞住
PLAYING	元素进入播放状态，数据流此时已经运转起来了

需要注意的是，你不能从NULL状态直接流转到PLAYING状态，只能按照NULL --> READY --> PAUSED --> PLAYING状态时序进行流转，如果你将pipeline配置为PLAYING状态，gstreamer会为你进行中间状态的流转。
每个元素都会将自己当前的状态封装成一个消息放到pipeline的bus上（存疑，从代码中看是从bus中获取pipeline的状态，但是此处显然是说元素的状态会上报到bus，原文为

Every element puts messages on the bus regarding its current state, so we filter them out and only listen to messages coming from the pipeline.

代码解读

pipeline创建

/* Structure to contain all our information, so we can pass it to callbacks */
typedef struct _CustomData {
  GstElement *pipeline;
  GstElement *source;
  GstElement *convert;
  GstElement *resample;
  GstElement *sink;
} CustomData;

因为这个教程需要有一个回调函数，所以把所有数据放到同一个结构体中，方便传入。

/* Create the elements */
data.source = gst_element_factory_make ("uridecodebin", "source");
data.convert = gst_element_factory_make ("audioconvert", "convert");
data.resample = gst_element_factory_make ("audioresample", "resample");
data.sink = gst_element_factory_make ("autoaudiosink", "sink");

这里创建了4个元素，用来串pipeline。
uridecodebin这个元素，已经包含了必要的所有元素（sources, demuxers, decoders）用来将URL转换成audio/video数据流。

if (!gst_element_link_many (data.convert, data.resample, data.sink, NULL)) {
  g_printerr ("Elements could not be linked.\n");
  gst_object_unref (data.pipeline);
  return -1;
}

把除了source元素之外的三个元素link起来，source放在后面link，这也是dynamic pipeline的意义。

/* Set the URI to play */
g_object_set (data.source, "uri", "https://gstreamer.freedesktop.org/data/media/sintel_trailer-480p.webm", NULL);

配置source的URL。

/* Connect to the pad-added signal */
g_signal_connect (data.source, "pad-added", G_CALLBACK (pad_added_handler), &data);

为source元素的"pad-added"信号添加回调函数和对应的入参。

callback

uridecodebin模块在有足够的数据之后，它会创建source pad，这时候它会触发一个"pad-added"信号，并调用之前注册进去的回调函数。

GstPad *sink_pad = gst_element_get_static_pad (data->convert, "sink");

通过函数gst_element_get_static_pad()可以获取元素的对应pad。把这个sink pad和uridecodebin模块的source pad相连，就可以创建完整的pipeline。

/* If our converter is already linked, we have nothing to do here */
if (gst_pad_is_linked (sink_pad)) {
  g_print ("We are already linked. Ignoring.\n");
  goto exit;
}

当然，在连接sink pad前，先判断下这个pad是否已经link过了

/* Check the new pad's type */
new_pad_caps = gst_pad_get_current_caps (new_pad, NULL);
new_pad_struct = gst_caps_get_structure (new_pad_caps, 0);
new_pad_type = gst_structure_get_name (new_pad_struct);
if (!g_str_has_prefix (new_pad_type, "audio/x-raw")) {
  g_print ("It has type '%s' which is not raw audio. Ignoring.\n", new_pad_type);
  goto exit;
}

获取当前pad的属性，因为我们创建的pipeline是用来处理audio的，所以其他的pad都会默认忽略掉。
函数gst_pad_get_current_caps()用来获取当前pad的能力集。
通过函数gst_pad_query_caps()可以获取一个pad可以支持的所有能力。所以new_pad_caps可能包含多个能力，可以通过遍历的方式去遍历每个能力，直到它返回NULL。
因为我们知道uridecodebin的audio source pad只有一个能力，所以就直接获取它第一个能力即可。

/* Attempt the link */
ret = gst_pad_link (new_pad, sink_pad);
if (GST_PAD_LINK_FAILED (ret)) {
  g_print ("Type is '%s' but link failed.\n", new_pad_type);
} else {
  g_print ("Link succeeded (type '%s').\n", new_pad_type);
}

通过函数gst_pad_link()将两个pad连接到一起，跟函数gst_element_link()类似，link操作必须是一个source pad和一个sink pad相连，并且他们所属的元素同属于同一个bin。

case GST_MESSAGE_STATE_CHANGED:
  /* We are only interested in state-changed messages from the pipeline */
  if (GST_MESSAGE_SRC (msg) == GST_OBJECT (data.pipeline)) {
    GstState old_state, new_state, pending_state;
    gst_message_parse_state_changed (msg, &old_state, &new_state, &pending_state);
    g_print ("Pipeline state changed from %s to %s:\n",
        gst_element_state_get_name (old_state), gst_element_state_get_name (new_state));
  }
  break;

这些语句将当前pipeline的状态打印出来。在此处可以验证上面概念中的疑惑，是否总线上不止存在pipeline的状态，还存在各个元素的状态，通过添加以下代码验证：
在这里插入图片描述
运行结果为：

从上述打印可以看到，bus上的确同时存在pipeline和元素的状态，并且，当所有元素（或者最后一个元素）都处于某个状态时，pipeline状态才会对应的改变。因为pipeline也是一种特殊的元素，也就是说，只要是元素，都会将自身状态上报到bus。

课后练习

我们可以添加一个autovideosink模块，用来播放当前例程里的video，代码如下：

代码

#include <gst/gst.h>

#ifdef __APPLE__
#include <TargetConditionals.h>
#endif

/* Structure to contain all our information, so we can pass it to callbacks */
typedef struct _CustomData
{
  GstElement *pipeline;
  GstElement *source;
  GstElement *convert;
  GstElement *resample;
  GstElement *sink;
  GstElement *video_cvt;
  GstElement *video_sink;
} CustomData;

/* Handler for the pad-added signal */
static void pad_added_handler (GstElement * src, GstPad * pad,
    CustomData * data);

int
tutorial_main (int argc, char *argv[])
{
  CustomData data;
  GstBus *bus;
  GstMessage *msg;
  GstStateChangeReturn ret;
  gboolean terminate = FALSE;

  /* Initialize GStreamer */
  gst_init (&argc, &argv);

  /* Create the elements */
  data.source = gst_element_factory_make ("uridecodebin", "source");
  data.convert = gst_element_factory_make ("audioconvert", "convert");
  data.resample = gst_element_factory_make ("audioresample", "resample");
  data.sink = gst_element_factory_make ("autoaudiosink", "sink");
  data.video_cvt = gst_element_factory_make ("videoconvert", "video-cvt");
  data.video_sink = gst_element_factory_make ("autovideosink", "video-sink");

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

  if (!data.pipeline || !data.source || !data.convert || !data.resample
      || !data.sink || !data.video_cvt || !data.video_sink) {
    g_printerr ("Not all elements could be created.\n");
    return -1;
  }

  /* Build the pipeline. Note that we are NOT linking the source at this
   * point. We will do it later. */
  gst_bin_add_many (GST_BIN (data.pipeline), data.source, data.convert,
      data.resample, data.sink, data.video_cvt, data.video_sink, NULL);
  if (!gst_element_link_many (data.convert, data.resample, data.sink, NULL)) {
    g_printerr ("Audio elements could not be linked.\n");
    gst_object_unref (data.pipeline);
    return -1;
  }

  if (!gst_element_link_many (data.video_cvt, data.video_sink, NULL)) {
    g_printerr ("Video elements could not be linked.\n");
    gst_object_unref (data.pipeline);
    return -1;
  }

  /* Set the URI to play */
  g_object_set (data.source, "uri",
      "https://gstreamer.freedesktop.org/data/media/sintel_trailer-480p.webm",
      NULL);

  /* Connect to the pad-added signal */
  g_signal_connect (data.source, "pad-added", G_CALLBACK (pad_added_handler),
      &data);

  /* Start playing */
  ret = gst_element_set_state (data.pipeline, GST_STATE_PLAYING);
  if (ret == GST_STATE_CHANGE_FAILURE) {
    g_printerr ("Unable to set the pipeline to the playing state.\n");
    gst_object_unref (data.pipeline);
    return -1;
  }

  /* Listen to the bus */
  bus = gst_element_get_bus (data.pipeline);
  do {
    msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
        GST_MESSAGE_STATE_CHANGED | GST_MESSAGE_ERROR | GST_MESSAGE_EOS);

    /* Parse message */
    if (msg != NULL) {
      GError *err;
      gchar *debug_info;

      switch (GST_MESSAGE_TYPE (msg)) {
        case GST_MESSAGE_ERROR:
          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);
          terminate = TRUE;
          break;
        case GST_MESSAGE_EOS:
          g_print ("End-Of-Stream reached.\n");
          terminate = TRUE;
          break;
        case GST_MESSAGE_STATE_CHANGED:
          /* We are only interested in state-changed messages from the pipeline */
          if (GST_MESSAGE_SRC (msg) == GST_OBJECT (data.pipeline)) {
            GstState old_state, new_state, pending_state;
            gst_message_parse_state_changed (msg, &old_state, &new_state,
                &pending_state);
            g_print ("Pipeline state changed from %s to %s:\n",
                gst_element_state_get_name (old_state),
                gst_element_state_get_name (new_state));
          }
          if (GST_MESSAGE_SRC (msg) == GST_OBJECT (data.source)) {
            GstState old_state, new_state, pending_state;
            gst_message_parse_state_changed (msg, &old_state, &new_state,
                &pending_state);
            g_print ("Source state changed from %s to %s:\n",
                gst_element_state_get_name (old_state),
                gst_element_state_get_name (new_state));
          }
          if (GST_MESSAGE_SRC (msg) == GST_OBJECT (data.sink)) {
            GstState old_state, new_state, pending_state;
            gst_message_parse_state_changed (msg, &old_state, &new_state,
                &pending_state);
            g_print ("Sink state changed from %s to %s:\n",
                gst_element_state_get_name (old_state),
                gst_element_state_get_name (new_state));
          }
      
	  break;
        default:
          /* We should not reach here */
          g_printerr ("Unexpected message received.\n");
          break;
      }
      gst_message_unref (msg);
    }
  } while (!terminate);

  /* Free resources */
  gst_object_unref (bus);
  gst_element_set_state (data.pipeline, GST_STATE_NULL);
  gst_object_unref (data.pipeline);
  return 0;
}

/* This function will be called by the pad-added signal */
static void
pad_added_handler (GstElement * src, GstPad * new_pad, CustomData * data)
{
  GstPad *sink_pad;
  GstPadLinkReturn ret;
  GstCaps *new_pad_caps = NULL;
  GstStructure *new_pad_struct = NULL;
  const gchar *new_pad_type = NULL;

  g_print ("Received new pad '%s' from '%s':\n", GST_PAD_NAME (new_pad),
      GST_ELEMENT_NAME (src));

  /* Check the new pad's type */
  new_pad_caps = gst_pad_get_current_caps (new_pad);
  new_pad_struct = gst_caps_get_structure (new_pad_caps, 0);
  new_pad_type = gst_structure_get_name (new_pad_struct);

  /* if the audio elements */
  if (g_str_has_prefix (new_pad_type, "audio/x-raw")) {
    g_print ("It has type '%s' which is raw audio. Link it!\n",
        new_pad_type);
    /* Attempt the link */
    sink_pad = gst_element_get_static_pad (data->convert, "sink");
    /* If our converter is already linked, we have nothing to do here */
    if (gst_pad_is_linked (sink_pad)) {
      g_print ("We are already linked. Ignoring.\n");
      goto exit;
    }

    ret = gst_pad_link (new_pad, sink_pad);
    if (GST_PAD_LINK_FAILED (ret)) {
      g_print ("Type is '%s' but link failed.\n", new_pad_type);
    } else {
      g_print ("Link succeeded (type '%s').\n", new_pad_type);
    }
  }

  /* if the video elements */
  if (g_str_has_prefix (new_pad_type, "video/x-raw")) {
    g_print ("It has type '%s' which is raw video. Link it!\n",
        new_pad_type);
    /* Attempt the link */
    sink_pad = gst_element_get_static_pad (data->video_cvt, "sink");
    /* If our converter is already linked, we have nothing to do here */
    if (gst_pad_is_linked (sink_pad)) {
      g_print ("We are already linked. Ignoring.\n");
      goto exit;
    }

    ret = gst_pad_link (new_pad, sink_pad);
    if (GST_PAD_LINK_FAILED (ret)) {
      g_print ("Type is '%s' but link failed.\n", new_pad_type);
    } else {
      g_print ("Link succeeded (type '%s').\n", new_pad_type);
    }
  }

exit:
  /* Unreference the new pad's caps, if we got them */
  if (new_pad_caps != NULL)
    gst_caps_unref (new_pad_caps);

  /* Unreference the sink pad */
  gst_object_unref (sink_pad);
}

int
main (int argc, char *argv[])
{
#if defined(__APPLE__) && TARGET_OS_MAC && !TARGET_OS_IPHONE
  return gst_macos_main (tutorial_main, argc, argv, NULL);
#else
  return tutorial_main (argc, argv);
#endif
}

效果

在这里插入图片描述
这样，就完成了音频和视频通过两条流同时播放。

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gstreamer学习（3）——动态创建pipeline

此博客是在gstreamer官网学习并总结的学习概要，具体参考gstreamer官网教程：动态pipeline惯例，先上官网示例代码：运行结果同时，耳机中会有音频播放（因为这个教程就是只把audio部分播放出来）。信号用来对感兴趣的事件执行某些指定操作，它通过名字区分不同信号，并且每个GObject都有自己的信号。gstreamer有4种states，分别是：需要注意的是，你不能从NULL状态直接流转到PLAYING状态，只能按照NULL --> READY --> PAUSED --> PLAYIN
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