[Gstreamer/Qt] Gstreamer(7) - buffer, GLib

Application와 gstreamer의 pipeline 간에 데이터를 주고받는 내용에 대해 살펴봅니다. 이 예제와 basic tutorial 7이 큰 차이가 없어 이 포스팅에서 한번에 다루겠습니다.

Basic tutorial 8: Short-cutting the pipeline

Application에서 pipeline으로 데이터를 넣을 수 있는 elements를 appsrc, 그 반대를 appsink라고 합니다. 데이터를 넣는 appsrc는 pull mode / push mode가 있는데 pull의 경우 주기적으로 데이터를 넣어주고 push의 경우 원할 때 넣는 방식으로 진행됩니다. 데이터는 Gstbuffer를 통해 파이프라인을 통과합니다.

QT -= gui

CONFIG += c++17 console
CONFIG -= app_bundle

# You can make your code fail to compile if it uses deprecated APIs.
# In order to do so, uncomment the following line.
#DEFINES += QT_DISABLE_DEPRECATED_BEFORE=0x060000    # disables all the APIs deprecated before Qt 6.0.0

SOURCES += \
        main.cpp

DEPENDPATH += \
    $$PWD/include

INCLUDEPATH += \
    $$PWD/include \
    $$PWD/include/gstreamer-1.0 \
    $$PWD/include/glib-2.0/ \
    $$PWD/include/glib-2.0/include \
    $$PWD/include/orc-0.4 \
    $$PWD/lib/glib-2.0/include \

win32: LIBS += -L$$PWD/lib/ -lgstreamer-1.0 -lgobject-2.0 -lglib-2.0 -lintl -lgstaudio-1.0 -lgstbase-1.0

DESTDIR += \
    $$PWD/bin

# Default rules for deployment.
qnx: target.path = /tmp/$${TARGET}/bin
else: unix:!android: target.path = /opt/$${TARGET}/bin
!isEmpty(target.path): INSTALLS += target

gstaudio-1.0 dll 사용이 필요해 pro 파일도 일부 수정하였습니다.

#include <QCoreApplication>
#include <QDebug>
#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[])
{
    QCoreApplication a(argc, 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 custom 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_request_pad_simple (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_request_pad_simple (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_request_pad_simple (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 a.exec();
}

 

예제의 pipeline은 아래와 같습니다. Tee라는 elements를 통해 하나의 appsrc에서 오는 데이터를 3 갈래로 나눠줍니다.

/* 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");

queue를 사용하면 multi thread 환경처럼 사용할 수 있습니다.

appsrc elements에 audio caps를 설정하고 set해줍니다.

/* 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);

여기서 need-data, enough-data signal이 나오는데 source가 data가 필요할때 need-data signal을 이제 충분히 데이터를 받았을 때 enough-data signal을 발생합니다. 이 기준은 내부 데이터 queue에 있는 데이터 양으로 판단합니다.

appsrc 내부 queue의 데이터가 필요할 때 g_idle_add를 통해 등록한 GLib 함수를 호출하게 됩니다. 이 함수는 mainloop가 idle 상태 즉 우선순위가 급한 일이 없을 때 호출됩니다. 

/* 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);
    }
}

push_data 함수에서는 buffer를 생성하여 wave 데이터를 생성하고 push-buffer signal을 발생시켜 buffer를 app_src elements의 source pad로 전달합니다. 이 source pad의 buffer는 결국 Tee->App_queue를 통해 app_sink로 전달될 것입니다.

/* 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;
}

app_sink에 new-sample signal을 등록해서 sample이 들어올 때 new_sample 함수가 호출됩니다. 

/* 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);

/* 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;
}

내부 queue가 어느정도 찼다고 판단되면 stop_feed 함수가 호출되고 등록한 함수를 제거합니다.

/* 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;
    }
}

위와 같이 GLib를 사용하기 위해서는 main loop를 run상태로 설정해줘야 합니다.

/* 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);