GStreamer基礎教程09 - Appsrc及Appsink

時間  2019-11-20
標籤 gstreamer 基礎 教程 appsrc appsink 简体版
原文   原文鏈接

摘要

在咱們前面的文章中，咱們的Pipline都是使用GStreamer自帶的插件去產生/消費數據。在實際的狀況中，咱們的數據源可能沒有相應的gstreamer插件，但咱們又須要將數據發送到GStreamer Pipeline中。GStreamer爲咱們提供了Appsrc以及Appsink插件，用於處理這種狀況，本文將介紹如何使用這些插件來實現數據與應用程序的交互。html

 

Appsrc與Appsink

GStreamer提供了多種方法使得應用程序與GStreamer Pipeline之間能夠進行數據交互，咱們這裏介紹的是最簡單的一種方式：appsrc與appsink。ios

  • appsrc：

用於將應用程序的數據發送到Pipeline中。應用程序負責數據的生成，並將其做爲GstBuffer傳輸到Pipeline中。
appsrc有2中模式，拉模式和推模式。在拉模式下，appsrc會在須要數據時，經過指定接口從應用程序中獲取相應數據。在推模式下，則須要由應用程序主動將數據推送到Pipeline中，應用程序能夠指定在Pipeline的數據隊列滿時是否阻塞相應調用，或經過監聽enough-data和need-data信號來控制數據的發送。多線程

  • appsink：

用於從Pipeline中提取數據，併發送到應用程序中。併發


　　appsrc和appsink須要經過特殊的API才能與Pipeline進行數據交互，相應的接口能夠查看官方文檔，在編譯的時候還需鏈接gstreamer-app庫。app

 

GstBuffer

　　在GStreamer Pipeline中的plugin間傳輸的數據塊被稱爲buffer，在GStreamer內部對應於GstBuffer。Buffer由Source Pad產生，並由Sink Pad消耗。一個Buffer只表示一塊數據，不一樣的buffer可能包含不一樣大小，不一樣時間長度的數據。同時，某些Element中可能對Buffer進行拆分或合併，因此GstBuffer中可能包含不止一個內存數據，實際的內存數據在GStreamer系統中經過GstMemory對象進行描述，所以，GstBuffer能夠包含多個GstMemory對象。
　　每一個GstBuffer都有相應的時間戳以及時間長度，用於描述這個buffer的解碼時間以及顯示時間。ide

 

示例代碼

本例在GStreamer基礎教程08 - 多線程示例上進行擴展，首先使用appsrc替代audiotestsrc用於產生audio數據，另外增長一個新的分支，將tee產生的數據發送到應用程序，由應用程序決定如何處理收到的數據。Pipeline的示意圖以下：函數

#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_get_request_pad (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_get_request_pad (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_get_request_pad (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;
}

保存以上代碼，執行下列編譯命令便可獲得可執行程序：oop

gcc basic-tutorial-9.c -o basic-tutorial-9 `pkg-config --cflags --libs gstreamer-1.0 gstreamer-audio-1.0 `

Note：本例在編譯時沒有鏈接gstreamer-app-1.0的庫是由於咱們使用的是經過信號的方式，由appsrc自動處理buffer，因此無需在編譯時鏈接相應庫。在源碼分析部分會詳述。源碼分析

 

源碼分析

　　與上一示例相同，首先對所需Element進行實例化，同時將Element的Always Pad鏈接起來，並與tee的Request Pad相連。此外咱們還對appsrc及appsink進行了相應的配置：post

/* 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);

　　首先須要對appsrc的caps進行設定，指定咱們會產生何種類型的數據，這樣GStreamer會在鏈接階段檢查後續的Element是否支持此數據類型。這裏的 caps必須爲GstCaps對象，咱們能夠經過gst_caps_from_string()或gst_audio_info_to_caps ()獲得相應的實例。
　　咱們同時監聽了「need-data」與「enough-data」事件，這2個事件由appsrc在須要數據和緩衝區滿時觸發，使用這2個事件能夠方便的控制什麼時候產生數據與中止數據。

 

/* 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);

　　對於appsink，咱們監聽「new-sample」事件，用於appsink在收到數據時的處理。同時咱們須要顯式的使能「new-sample」事件，由於這個事件默認是處於關閉狀態。

　　Pipeline的播放，中止及消息處理與其餘示例相同，再也不復述。咱們接下來將查看咱們監聽事件的回調函數。

 

/* 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);
  }
}

　　appsrc會在其內部的數據隊列即將缺少數據時調用此回調函數，這裏咱們經過註冊一個GLib的idle函數來向appsrc填充數據，GLib的主循環在「idle」狀態時會循環調用 push_data，用於向appsrc填充數據。這只是一種向appsrc填充數據的方式，咱們能夠在任意線程中想appsrc填充數據。
　　咱們保存了g_idle_add()的返回值，以便後續用於中止數據寫入。

/* 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;
  }
}

　　stop_feed函數會在appsrc內部數據隊列滿時被調用。這裏咱們僅僅經過g_source_remove() 將先前註冊的idle處理函數從GLib的主循環中移除（idle處理函數是被實現爲一個GSource）。

 

/* This method is called by the idle GSource in the mainloop, to feed CHUNK_SIZE bytes into appsrc.
 * The ide 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 (num_samples, GST_SECOND, SAMPLE_RATE);

  /* Generate some psychodelic waveforms */
  raw = (gint16 *)GST_BUFFER_DATA (buffer);

　　此函數會將真實的數據填充到appsrc的數據隊列中，首先經過gst_buffer_new_and_alloc()分配一個GstBuffer對象，而後經過產生的採樣數量計算這塊buffre所對應的時間戳及事件長度。
　　gst_util_uint64_scale(val, num, denom)函數用於計算 val * num / denom，此函數內部會對數據範圍進行檢測，避免溢出的問題。
　　GstBuffer的數據指針能夠經過GST_BUFFER_DATA 宏獲取，在寫數據時須要避免超出內存分配大小。本文將跳過audio波形生成的函數，其內容不是本文介紹的重點。

 

/* 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);

　　在咱們準備好數據後，咱們這裏經過「push-buffer」事件通知appsrc數據就緒，並釋放咱們申請的buffer。 另一種方式爲經過調用gst_app_src_push_buffer() 向appsrc填充數據，這種方式就須要在編譯時連接gstreamer-app-1.0庫，同時gst_app_src_push_buffer() 會接管GstBuffer的全部權，調用者無需釋放buffer。在全部數據都發送完成後，咱們能夠調用gst_app_src_end_of_stream()向Pipeline寫入EOS事件。

/* 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;
}

　　當appsink獲得數據時會調用new_sample函數，咱們使用「pull-sample」信號提取sample，這裏僅輸出一個」*「代表此函數被調用。除此以外，咱們一樣可使用gst_app_sink_pull_sample ()獲取Sample。獲得GstSample以後，咱們能夠經過gst_sample_get_buffer()獲得Sample中所包含的GstBuffer，再使用GST_BUFFER_DATA， GST_BUFFER_SIZE 等接口訪問其中的數據。使用完後，獲得的GstSample一樣須要經過gst_sample_unref()進行釋放。
　　須要注意的是，在某些Pipeline裏獲得的GstBuffer可能會和source中填充的GstBuffer有所差別，由於Pipeline中的Element可能對Buffer進行各類處理（此例中不存在此種狀況，由於在appsrc與appsink之間只存在一個tee）。

 

總結

在本文中，咱們介紹了:

  • 如何經過appsrc向Pipeline中寫入數據
  • 如何經過appsink取得Pipeline中的數據
  • 如何獲取/填充GstBuffer中對應的數據

後續咱們將繼續學習有關GStreamer的其餘知識。

 

引用

https://gstreamer.freedesktop.org/documentation/tutorials/basic/short-cutting-the-pipeline.html?gi-language=c

  

做者： John.Leng
本文版權歸做者全部，歡迎轉載。商業轉載請聯繫做者得到受權，非商業轉載請在文章頁面明顯位置給出原文鏈接.
相關文章
相關標籤/搜索
每日一句
    每一个你不满意的现在，都有一个你没有努力的曾经。
本站公眾號
   歡迎關注本站公眾號,獲取更多信息
聯繫我們 最近搜索 最新文章 沪ICP备13005482号-10 MyBatis教程 SQL 教程 MySQL教程 Java 教程 Thymeleaf 教程 Hibernate教程 Spring教程 Redis教程