GStreamer appsrc 到文件示例

Question

I'd like to write audio buffers to file.

The problem is that the file is always empty or the applicaition stops immediately with Internal data flow error .

Examples are:

https://gist.github.com/nzjrs/725122/16ceee88aafae389bab207818e0661328921e1ab

and

http://gstreamer-devel.966125.n4.nabble.com/appsrc-to-filesink-td970865.html

Composed this from them:

#include <string.h>

#include <gst/gst.h>

#include <gst/app/gstappsrc.h>

/*
 * an example application of using appsrc in push mode to create a video file.
 * from buffers we push into the pipeline.
 */

/* Video resolution: 80 x 60 x 4 = 80x60 pixels, 32 bpp (4 bytes per pixel) = 19200 bytes */
#define BUFFER_SIZE  19200

/* 300 frames = 10 seconds of video, since we are going to save at 30fps (se the video_caps next) */
#define TOTAL_FRAMES 30000

#define QUEUED_FRAMES 30

/* these are the caps we are going to pass through the appsrc */
const gchar *video_caps =
    "video/x-raw,format=RGB,width=80,height=60,bpp=32,endianness=4321,depth=24,red_mask=65280,green_mask=16711680,blue_mask=-16777216,framerate=30/1";

typedef struct
{
  GMainLoop *loop;
  GstElement *pipeline;
  GstElement *source;
  guint source_id;
  guint num_frame;
} AppData;

/* This method is called by the idle GSource in the mainloop. We feed 1 buffer
 * of BUFFER_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_buffer (AppData * app)
{
  gpointer raw_buffer;
  GstBuffer *app_buffer;
  GstMemory *mem;
  GstFlowReturn ret;

  app->num_frame++;

  if (app->num_frame >= TOTAL_FRAMES) {
    /* we are EOS, send end-of-stream and remove the source */
    g_signal_emit_by_name (app->source, "end-of-stream", &ret);
    return FALSE;
  }

  app_buffer = gst_buffer_new();

  mem = gst_allocator_alloc (NULL, BUFFER_SIZE, NULL);

  gst_buffer_append_memory(app_buffer, mem);

  gst_buffer_set_size(app_buffer, BUFFER_SIZE);

  /* Setting the correct timestamp for the buffer is very important, otherwise the
   * resulting video file won't be created correctly */
  GST_BUFFER_TIMESTAMP(app_buffer) = (GstClockTime)((app->num_frame / 30.0) * 1e9);

  /* push new buffer */
  g_signal_emit_by_name (app->source, "push-buffer", app_buffer, &ret);

  gst_buffer_unref(app_buffer);

  if (ret != GST_FLOW_OK) {
    /* some error, stop sending data */
    return FALSE;
  }

  return TRUE;
}

/* This signal callback is called when appsrc needs data, we add an idle handler
 * to the mainloop to start pushing data into the appsrc */
static void
start_feed (GstElement * pipeline, guint size, AppData * app)
{
  if (app->source_id == 0) {
        g_print ("start feeding at frame %i\n", app->num_frame);
    app->source_id = g_idle_add ((GSourceFunc) push_buffer, app);
  }
}

/* This callback is called when appsrc has enough data and we can stop sending.
 * We remove the idle handler from the mainloop */
static void
stop_feed (GstElement * pipeline, AppData * app)
{
  if (app->source_id != 0) {
        g_print ("stop feeding at frame %i\n", app->num_frame);
    g_source_remove (app->source_id);
    app->source_id = 0;
  }
}

/* called when we get a GstMessage from the pipeline when we get EOS, we
 * exit the mainloop and this testapp. */
static gboolean
on_pipeline_message (GstBus * bus, GstMessage * message, AppData * app)
{
  GstState state, pending;

  switch (GST_MESSAGE_TYPE (message)) {
    case GST_MESSAGE_EOS:
      g_print ("Received End of Stream message\n");
      g_main_loop_quit (app->loop);
      break;
    case GST_MESSAGE_ERROR:
      {
        g_print ("Received error\n");

        GError *err = NULL;
        gchar *dbg_info = NULL;

        gst_message_parse_error (message, &err, &dbg_info);
        g_printerr ("ERROR from element %s: %s\n",
            GST_OBJECT_NAME (message->src), err->message);
        g_printerr ("Debugging info: %s\n", (dbg_info) ? dbg_info : "none");
        g_error_free (err);
        g_free (dbg_info);
      }

      g_main_loop_quit (app->loop);
      break;
    case GST_MESSAGE_STATE_CHANGED:
          gst_element_get_state(app->source, &state, &pending, GST_CLOCK_TIME_NONE);
      /* g_print ("State changed from %i to %i\n", state, pending); */
      break;
        default:
      break;
  }
  return TRUE;
}

int
main (int argc, char *argv[])
{
  AppData *app = NULL;
  gchar *string = NULL;
  GstBus *bus = NULL;
  GstElement *appsrc = NULL;

  gst_init (&argc, &argv);

  app = g_new0 (AppData, 1);

  app->loop = g_main_loop_new (NULL, FALSE);

  /* setting up pipeline, we push video data into this pipeline that will
   * then be recorded to an avi file, encoded with the h.264 codec*/
  string =
      g_strdup_printf ("appsrc is-live=true name=source caps=\"%s\" ! ffmpegcolorspace ! video/x-raw,format=(fourcc)I420,width=80,height=60 ! queue ! videorate ! video/x-raw,format=YUV,framerate=30/1 ! h264enc ! queue ! avimux ! queue ! filesink location=test.avi",
      video_caps);
  app->pipeline = gst_parse_launch (string, NULL);
  g_free (string);

  if (app->pipeline == NULL) {
    g_print ("Bad pipeline\n");
    return -1;
  }

  appsrc = gst_bin_get_by_name (GST_BIN (app->pipeline), "source");
  /* configure for time-based format */
  g_object_set (appsrc, "format", GST_FORMAT_TIME, NULL);
  /* setting maximum of bytes queued. default is 200000 */
  gst_app_src_set_max_bytes((GstAppSrc *)appsrc, QUEUED_FRAMES * BUFFER_SIZE);
  /* uncomment the next line to block when appsrc has buffered enough */
  /* g_object_set (appsrc, "block", TRUE, NULL); */
  app->source = appsrc;

  /* add watch for messages */
  bus = gst_element_get_bus (app->pipeline);
  gst_bus_add_watch (bus, (GstBusFunc) on_pipeline_message, app);
  gst_object_unref (bus);

  /* configure the appsrc, we will push data into the appsrc from the
   * mainloop */
  g_signal_connect (app->source, "need-data", G_CALLBACK (start_feed), app);
  g_signal_connect (app->source, "enough-data", G_CALLBACK (stop_feed), app);

  /* go to playing and wait in a mainloop */
  gst_element_set_state (app->pipeline, GST_STATE_PLAYING);

  /* this mainloop is stopped when we receive an error or EOS */
  g_print ("Creating movie...\n");
  g_main_loop_run (app->loop);
  g_print ("Done.\n");

  gst_app_src_end_of_stream (GST_APP_SRC (app->source));

  gst_element_set_state (app->pipeline, GST_STATE_NULL);

  /* Cleaning up */
  gst_object_unref (app->source);
  gst_object_unref (app->pipeline);
  g_main_loop_unref (app->loop);
  g_free (app);

  return 0;
}

CMakeLists.txt (put FindGStreamer.cmake into modules/ directory of the project):

cmake_minimum_required(VERSION 2.8.11)

project(myproject)

set(CMAKE_MODULE_PATH ${CMAKE_MODULE_PATH} "${CMAKE_SOURCE_DIR}/modules/")

find_package(GStreamer 1.0 REQUIRED COMPONENTS
    app
    audio
    video
)

add_executable(${PROJECT_NAME}
    main.c
)

target_include_directories(${PROJECT_NAME} PRIVATE ${GSTREAMER_INCLUDE_DIRS})
target_include_directories(${PROJECT_NAME} PRIVATE ${GSTREAMER_APP_INCLUDE_DIRS})

target_link_libraries(${PROJECT_NAME}
    ${GSTREAMER_LIBRARIES}
    ${GSTREAMER_APP_LIBRARIES}
    ${GSTREAMER_AUDIO_LIBRARIES}
    ${GSTREAMER_VIDEO_LIBRARIES}
    glib-2.0
    gobject-2.0
)

Actually, pipeline for my program is:

appsrc is-live=true name=mysource ! flacenc ! oggmux ! filesink location=flac-audio.ogg

but I'm trying to make the examples work as they are

Answer 1

Launching with --gst-debug-level=2 was useful. Was complaining about the caps string.

#include <string.h>

#include <gst/gst.h>

#include <gst/app/gstappsrc.h>

/*
 * an example application of using appsrc in push mode to create a file.
 * from buffers we push into the pipeline.
 */

/* S16LE 10ms frame audio */
#define BUFFER_SIZE 160

/* 300 frames = 5 minutes */
#define TOTAL_FRAMES 30000

#define QUEUED_FRAMES 30

typedef struct
{
  GMainLoop *loop;
  GstElement *pipeline;
  GstElement *source;
  guint source_id;
  guint num_frame;
} AppData;

/* This method is called by the idle GSource in the mainloop. We feed 1 buffer
 * of BUFFER_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_buffer (AppData * app)
{
  gpointer raw_buffer;
  GstBuffer *app_buffer;
  GstMemory *mem;
  GstFlowReturn ret;

  app->num_frame++;

  if (app->num_frame >= TOTAL_FRAMES) {
    /* we are EOS, send end-of-stream and remove the source */
    g_signal_emit_by_name (app->source, "end-of-stream", &ret);
    return FALSE;
  }

  app_buffer = gst_buffer_new();

  mem = gst_allocator_alloc (NULL, BUFFER_SIZE, NULL);

  gst_buffer_append_memory(app_buffer, mem);

  gst_buffer_set_size(app_buffer, BUFFER_SIZE);

  /* Setting the correct timestamp for the buffer is very important, otherwise the
   * resulting file won't be created correctly */
  GST_BUFFER_TIMESTAMP(app_buffer) = (GstClockTime)((app->num_frame / 100.0) * 1e9);

  /* push new buffer */
  g_signal_emit_by_name (app->source, "push-buffer", app_buffer, &ret);

  gst_buffer_unref(app_buffer);

  if (ret != GST_FLOW_OK) {
    /* some error, stop sending data */
    return FALSE;
  }

  return TRUE;
}

/* This signal callback is called when appsrc needs data, we add an idle handler
 * to the mainloop to start pushing data into the appsrc */
static void
start_feed (GstElement * pipeline, guint size, AppData * app)
{
  if (app->source_id == 0) {
        g_print ("start feeding at frame %i\n", app->num_frame);
    app->source_id = g_idle_add ((GSourceFunc) push_buffer, app);
  }
}

/* This callback is called when appsrc has enough data and we can stop sending.
 * We remove the idle handler from the mainloop */
static void
stop_feed (GstElement * pipeline, AppData * app)
{
  if (app->source_id != 0) {
        g_print ("stop feeding at frame %i\n", app->num_frame);
    g_source_remove (app->source_id);
    app->source_id = 0;
  }
}

/* called when we get a GstMessage from the pipeline when we get EOS, we
 * exit the mainloop and this testapp. */
static gboolean
on_pipeline_message (GstBus * bus, GstMessage * message, AppData * app)
{
  GstState state, pending;

  switch (GST_MESSAGE_TYPE (message)) {
    case GST_MESSAGE_EOS:
      g_print ("Received End of Stream message\n");
      g_main_loop_quit (app->loop);
      break;
    case GST_MESSAGE_ERROR:
      {
        g_print ("Received error\n");

        GError *err = NULL;
        gchar *dbg_info = NULL;

        gst_message_parse_error (message, &err, &dbg_info);
        g_printerr ("ERROR from element %s: %s\n",
            GST_OBJECT_NAME (message->src), err->message);
        g_printerr ("Debugging info: %s\n", (dbg_info) ? dbg_info : "none");
        g_error_free (err);
        g_free (dbg_info);
      }

      g_main_loop_quit (app->loop);
      break;
    case GST_MESSAGE_STATE_CHANGED:
          gst_element_get_state(app->source, &state, &pending, GST_CLOCK_TIME_NONE);
      /* g_print ("State changed from %i to %i\n", state, pending); */
      break;
        default:
      break;
  }
  return TRUE;
}

int
main (int argc, char *argv[])
{
  AppData *app = NULL;
  GstBus *bus = NULL;
  GstElement *appsrc = NULL;

  gst_init (&argc, &argv);

  app = g_new0 (AppData, 1);

  app->loop = g_main_loop_new (NULL, FALSE);

  /* setting up pipeline, we push media data into this pipeline that will
   * then be recorded to a file, encoded with a codec*/
  app->pipeline = gst_parse_launch ("appsrc is-live=true name=source caps=audio/x-raw,format=S16LE,rate=8000,channels=1,layout=interleaved ! flacenc ! oggmux ! filesink location=flac-audio.ogg", NULL);

  if (app->pipeline == NULL) {
    g_print ("Bad pipeline\n");
    return -1;
  }

  appsrc = gst_bin_get_by_name (GST_BIN (app->pipeline), "source");

  /* setting maximum of bytes queued */
  gst_app_src_set_max_bytes((GstAppSrc *)appsrc, QUEUED_FRAMES * BUFFER_SIZE);

  /* uncomment the next line to block when appsrc has buffered enough */
  /* g_object_set (appsrc, "block", TRUE, NULL); */
  app->source = appsrc;

  /* add watch for messages */
  bus = gst_element_get_bus (app->pipeline);
  gst_bus_add_watch (bus, (GstBusFunc) on_pipeline_message, app);
  gst_object_unref (bus);

  /* configure the appsrc, we will push data into the appsrc from the
   * mainloop */
  g_signal_connect (app->source, "need-data", G_CALLBACK (start_feed), app);
  g_signal_connect (app->source, "enough-data", G_CALLBACK (stop_feed), app);

  /* go to playing and wait in a mainloop */
  gst_element_set_state (app->pipeline, GST_STATE_PLAYING);

  /* this mainloop is stopped when we receive an error or EOS */
  g_print ("Creating movie...\n");
  g_main_loop_run (app->loop);
  g_print ("Done.\n");

  gst_app_src_end_of_stream (GST_APP_SRC (app->source));

  gst_element_set_state (app->pipeline, GST_STATE_NULL);

  /* Cleaning up */
  gst_object_unref (app->source);
  gst_object_unref (app->pipeline);
  g_main_loop_unref (app->loop);
  g_free (app);

  return 0;
}