GStreamer基礎教程08 - 多線程

摘要

  GStreamer框架會自動處理多線程的邏輯,但在某些狀況下,咱們仍然須要根據實際的狀況本身將部分Pipeline在單獨的線程中執行,本文將介紹如何處理這種狀況。html

GStreamer多線程

  GStreamer框架是一個支持多線程的框架,線程會根據Pipeline的須要自動建立和銷燬,例如,將媒體流與應用線程解耦,應用線程不會被GStreamer的處理阻塞。並且,GStreamer的插件還能夠建立本身所需的線程用於媒體的處理,例如:在一個4核的CPU上,視頻解碼插件能夠建立4個線程來最大化利用CPU資源。
  此外,在建立Pipeline時,咱們還能夠指定某個Pipeline的分支在不一樣的線程中執行(例如,使audio、video同時在不一樣的線程中進行解碼)。這是經過queue Element來實現的,queue的sink pad僅僅將數據放入隊列,另一個線程從隊列中取出數據,並傳遞到下一個Element。queue一般也被用於做爲數據緩衝,緩衝區大小能夠經過queue的屬性進行配置。ios

  在上面的示例Pipeline中,souce是audiotestsrc,會產生一個相應的audio信號,而後使用tee Element將數據分爲兩路,一路被用於播放,經過聲卡輸出,另外一路被用於轉換爲視頻波形,用於輸出到屏幕。
示例圖中的紅色陰影部分表示位於同一個線程中,queue會建立單獨的線程,因此上面的Pipeline使用了3個線程完成相應的功能。擁有多個sink的Pipeline一般須要多個線程,由於在多個sync間進行同步的時候,sink會阻塞當前所在線程直到所等待的事件發生。多線程

示例代碼

示例代碼將建立上圖所示的Pipeline。框架

#include <gst/gst.h>

int main(int argc, char *argv[]) {
  GstElement *pipeline, *audio_source, *tee, *audio_queue, *audio_convert, *audio_resample, *audio_sink;
  GstElement *video_queue, *visual, *video_convert, *video_sink;
  GstBus *bus;
  GstMessage *msg;
  GstPad *tee_audio_pad, *tee_video_pad;
  GstPad *queue_audio_pad, *queue_video_pad;

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

  /* Create the elements */
  audio_source = gst_element_factory_make ("audiotestsrc", "audio_source");
  tee = gst_element_factory_make ("tee", "tee");
  audio_queue = gst_element_factory_make ("queue", "audio_queue");
  audio_convert = gst_element_factory_make ("audioconvert", "audio_convert");
  audio_resample = gst_element_factory_make ("audioresample", "audio_resample");
  audio_sink = gst_element_factory_make ("autoaudiosink", "audio_sink");
  video_queue = gst_element_factory_make ("queue", "video_queue");
  visual = gst_element_factory_make ("wavescope", "visual");
  video_convert = gst_element_factory_make ("videoconvert", "csp");
  video_sink = gst_element_factory_make ("autovideosink", "video_sink");

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

  if (!pipeline || !audio_source || !tee || !audio_queue || !audio_convert || !audio_resample || !audio_sink ||
      !video_queue || !visual || !video_convert || !video_sink) {
    g_printerr ("Not all elements could be created.\n");
    return -1;
  }

  /* Configure elements */
  g_object_set (audio_source, "freq", 215.0f, NULL);
  g_object_set (visual, "shader", 0, "style", 1, NULL);

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

  /* Manually link the Tee, which has "Request" pads */
  tee_audio_pad = gst_element_get_request_pad (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 (audio_queue, "sink");
  tee_video_pad = gst_element_get_request_pad (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 (video_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) {
    g_printerr ("Tee could not be linked.\n");
    gst_object_unref (pipeline);
    return -1;
  }
  gst_object_unref (queue_audio_pad);
  gst_object_unref (queue_video_pad);

  /* Start playing the pipeline */
  gst_element_set_state (pipeline, GST_STATE_PLAYING);

  /* Wait until error or EOS */
  bus = gst_element_get_bus (pipeline);
  msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE, GST_MESSAGE_ERROR | GST_MESSAGE_EOS);

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

  /* Free resources */
  if (msg != NULL)
    gst_message_unref (msg);
  gst_object_unref (bus);
  gst_element_set_state (pipeline, GST_STATE_NULL);

  gst_object_unref (pipeline);
  return 0;
}

保存以上代碼,執行下列編譯命令便可獲得可執行程序:ide

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

源碼分析

/* Create the elements */
audio_source = gst_element_factory_make ("audiotestsrc", "audio_source");
tee = gst_element_factory_make ("tee", "tee");
audio_queue = gst_element_factory_make ("queue", "audio_queue");
audio_convert = gst_element_factory_make ("audioconvert", "audio_convert");
audio_resample = gst_element_factory_make ("audioresample", "audio_resample");
audio_sink = gst_element_factory_make ("autoaudiosink", "audio_sink");
video_queue = gst_element_factory_make ("queue", "video_queue");
visual = gst_element_factory_make ("wavescope", "visual");
video_convert = gst_element_factory_make ("videoconvert", "video_convert");
video_sink = gst_element_factory_make ("autovideosink", "video_sink");

  首先建立所需的Element:audiotestsrc會產生測試的音頻波形數據。wavescope 會將輸入的音頻數據轉換爲波形圖像。audioconvert,audioresample,videoconvert保證了Pipeline中各個Element之間的數據能夠互相兼容,使得Pipeline可以被正確的link起來,若是不須要對數據進行轉換,這些Element會直接將數據發送到下一個Element,這種狀況下的性能影響能夠忽略不計。函數

/* Configure elements */
g_object_set (audio_source, "freq", 215.0f, NULL);
g_object_set (visual, "shader", 0, "style", 1, NULL);

  這裏修改相應Element的參數,使得輸出結果更直觀。「freq」會設置audiotestsrc輸出波形的頻率爲215Hz,設置「shader」和「style」使得波形更加連續。其餘的參數能夠經過gst-inspect查看。源碼分析

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

  這裏咱們使用gst_element_link_many 將多個Element鏈接起來,須要注意的是,這裏咱們只鏈接了擁有Always Pad的Eelement。雖然gst_element_link_many() 可以在內部處理Request Pad的狀況,但咱們仍然須要單獨釋放Request Pad,若是直接使用此函數鏈接全部的Element,這樣容易忘記釋放Request Pad。所以咱們使用下面的代碼單獨處理Request Pad。性能

/* Manually link the Tee, which has "Request" pads */
tee_audio_pad = gst_element_get_request_pad (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 (audio_queue, "sink");
tee_video_pad = gst_element_get_request_pad (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 (video_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) {
  g_printerr ("Tee could not be linked.\n");
  gst_object_unref (pipeline);
  return -1;
}
gst_object_unref (queue_audio_pad);
gst_object_unref (queue_video_pad);

  爲了可以鏈接到Request Pad,咱們須要主動的向Element取得相應的Pad。因爲一個Element能夠提供不一樣的Request Pad,因此咱們須要指定所需的「Pad Template」,Element提供的Pad Template能夠經過gst-inspect查看。從下面的結果能夠發現,tee提供了2種類型的模板, 」sink「 和「src_%u"。測試

$ gst-inspect-1.0  tee
...
Pad Templates:
  SRC template: 'src_%u'
    Availability: On request
      Has request_new_pad() function: gst_tee_request_new_pad
    Capabilities:
      ANY

  SINK template: 'sink'
    Availability: Always
    Capabilities:
      ANY
...

  因爲咱們這裏須要的是2個Source Pad,因此咱們經過gst_element_get_request_pad (tee, "src_%u")獲取兩個Request Pad分別用於audio和video。queue的Sink Pad是Alwasy Pad,因此咱們直接使用gst_element_get_static_pad 獲取其Sink Pad。最後再經過gst_pad_link()將其鏈接起來,在gst_element_link()和gst_element_link_many()內部也是使用此函數鏈接兩個Element的Pad。spa

須要注意的是,咱們經過Element獲取到的Pad的引用計數會自動增長,所以咱們須要調用gst_object_unref()釋放相關的引用,對於Request Pad,咱們須要在Pipeline執行完成後進行釋放。

 

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

除了播放完成後正常的資源釋放外,咱們還要對Request進行釋放,須要首先調用gst_element_release_request_pad(),最後再釋放相應的對象。

總結

咱們在本文中瞭解了:

  • 如何經過queue讓Pipeline運行在多個線程上。
  • 如何經過gst_element_get_request_pad(), gst_pad_link() gst_element_release_request_pad() 對Request Pad進行操做。
  • 如何使用tee將一路媒體數據分爲多路。

 

引用

https://gstreamer.freedesktop.org/documentation/tutorials/basic/multithreading-and-pad-availability.html?gi-language=c

做者: John.Leng
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