如何修复Gstreamer来捕获麦克风音频和缓冲区或转储作为原始文件，当我说它没有保存任何东西?

I am trying to capture the microphone audio and save it as a file. But its not working, i can only play the file while assign. How can i enable the microphone and buffer it or save or dump as raw .odd/vorbis ?

我试图捕获麦克风音频并将其保存为文件。但是它不工作，我只能在分配时播放文件。如何启用麦克风和缓冲，或保存或转储为raw .odd/vorbis ?

#include <gst/gst.h>
#include <glib.h>


static gboolean
bus_call (GstBus     *bus,
          GstMessage *msg,
          gpointer    data)
{
  GMainLoop *loop = (GMainLoop *) data;

  switch (GST_MESSAGE_TYPE (msg)) 
  {

    case GST_MESSAGE_EOS:
      g_print ("End of stream\n");
      g_main_loop_quit (loop);
      break;

    case GST_MESSAGE_ERROR: {
      gchar  *debug;
      GError *error;

      gst_message_parse_error (msg, &error, &debug);
      g_free (debug);

      g_printerr ("Error: %s\n", error->message);
      g_error_free (error);

      g_main_loop_quit (loop);
      break;
    }
    default:
      break;
  }

  return TRUE;
}


static void
on_pad_added (GstElement *element,
              GstPad     *pad,
              gpointer    data)
{
  GstPad *sinkpad;
  GstElement *decoder = (GstElement *) data;

  /* We can now link this pad with the vorbis-decoder sink pad */
  g_print ("Dynamic pad created, linking demuxer/decoder\n");

  sinkpad = gst_element_get_static_pad (decoder, "sink");

  gst_pad_link (pad, sinkpad);

  gst_object_unref (sinkpad);
}



int
main (int   argc,
      char *argv[])
{
  GMainLoop *loop;

  GstElement *pipeline, *source, *demuxer, *decoder, *conv, *sink;
  GstBus *bus;

  /* Initialisation */
  gst_init (&argc, &argv);

  loop = g_main_loop_new (NULL, FALSE);


  /* Check input arguments */
  if (argc != 2) {
    g_printerr ("Usage: %s <Ogg/Vorbis filename>\n", argv[0]);
    return -1;
  }


  /* Create gstreamer elements */
  pipeline = gst_pipeline_new ("audio-player");
  source   = gst_element_factory_make ("filesrc",       "file-source");
  demuxer  = gst_element_factory_make ("oggdemux",      "ogg-demuxer");
  decoder  = gst_element_factory_make ("vorbisdec",     "vorbis-decoder");
  conv     = gst_element_factory_make ("audioconvert",  "converter");
  sink     = gst_element_factory_make ("autoaudiosink", "audio-output");

  if (!pipeline || !source || !demuxer || !decoder || !conv || !sink) {
    g_printerr ("One element could not be created. Exiting.\n");
    return -1;
  }

  /* Set up the pipeline */

  /* we set the input filename to the source element */
  g_object_set (G_OBJECT (source), "location", argv[1], NULL);

  /* we add a message handler */
  bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
  gst_bus_add_watch (bus, bus_call, loop);
  gst_object_unref (bus);

  /* we add all elements into the pipeline */
  /* file-source | ogg-demuxer | vorbis-decoder | converter | alsa-output */
  gst_bin_add_many (GST_BIN (pipeline),
                    source, demuxer, decoder, conv, sink, NULL);

  /* we link the elements together */
  /* file-source -> ogg-demuxer ~> vorbis-decoder -> converter -> alsa-output */
  gst_element_link (source, demuxer);
  gst_element_link_many (decoder, conv, sink, NULL);
  g_signal_connect (demuxer, "pad-added", G_CALLBACK (on_pad_added), decoder);

  /* note that the demuxer will be linked to the decoder dynamically.
     The reason is that Ogg may contain various streams (for example
     audio and video). The source pad(s) will be created at run time,
     by the demuxer when it detects the amount and nature of streams.
     Therefore we connect a callback function which will be executed
     when the "pad-added" is emitted.*/


  /* Set the pipeline to "playing" state*/
  g_print ("Now playing: %s\n", argv[1]);
  gst_element_set_state (pipeline, GST_STATE_PLAYING);


  /* Iterate */
  g_print ("Running...\n");
  g_main_loop_run (loop);


  /* Out of the main loop, clean up nicely */
  g_print ("Returned, stopping playback\n");
  gst_element_set_state (pipeline, GST_STATE_NULL);

  g_print ("Deleting pipeline\n");
  gst_object_unref (GST_OBJECT (pipeline));

  return 0;
}

2 个解决方案

#1

what's the question actually?

这个问题实际上是什么?

on linux with pulseaudio it's as simple as

在使用pulseaudio的linux上，它很简单。

$ gst-launch pulsesrc ! filesink location=dump.raw
$ gst-launch pulsesrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=dump.ogg

#2

you can also use the following pipeline :

您还可以使用以下管道:

gst-launch osssrc device=<mic i/p dev> ! audioconvert ! vorbisenc ! oggmux ! filesink location=dump.ogg

#1