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`
/* 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。
咱們在本文中瞭解了: