problem when injecting source file mp3/m3u8 data to appsrc element.

[Sujith reddy]

Hi All,

Here i wanted to know how to inject a source mp3/m3u8 file to pipeline 
appsrc.


my code is working for raw data i.e wav(PCM) file as injecting source.  
for mp3/m3u8 it is giving noise .


Here is mycode can anyone help on this to play mp3/m3u8 by injecting data to
appsrc.

/*****************


gcc llll.c -o playback-tutorial-7 `pkg-config --cflags --libs gstreamer-1.0
gstreamer-audio-1.0 gstreamer-app-1.0`
*******************/

#include <gstreamer-1.0/gst/gst.h>
#include <gst/audio/audio.h>
#include <string.h>
#include <stdio.h>

#define CHUNK_SIZE 4096   /* Amount of bytes we are sending in each buffer
*/
#define SAMPLE_RATE 48000 /* 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;//*app_decode,*audio_decode;
  GstElement *app_queue, *audio_convert2,  *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 */
 FILE *fp,*fp1;
  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 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,r;
  GstMapInfo map;
  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 (CHUNK_SIZE,
GST_SECOND, SAMPLE_RATE);

  /* Generate some psychodelic waveforms */
  gst_buffer_map (buffer, &map, GST_MAP_WRITE);
 r=fread(map.data,2,CHUNK_SIZE/2,data->fp);	
  gst_buffer_unmap (buffer, &map);
  data->num_samples += num_samples;

while(r==NULL)
gst_app_src_end_of_stream (data->app_source);   



  /* Push the buffer into the appsrc */
 g_signal_emit_by_name (data->app_source, "push-buffer", buffer, &ret);
  // gst_app_src_end_of_stream (data->app_source);
 //gst_app_src_push_buffer (data->app_source, buffer);
  /* 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 void new_sample (GstElement *sink, CustomData *data) {
	
	//printf("sujith1111111");
	GstSample *sample;
	///////////////////////////////////////////////////////
	GstBuffer *buffer;
	GstMapInfo map;
	g_signal_emit_by_name (data ->app_sink, "pull-sample", &sample,NULL);
	if (sample)
	{
		buffer = gst_sample_get_buffer (sample);
	
		gst_buffer_map (buffer, &map, GST_MAP_READ);

		g_print("\n here size=%d\n",map.size);
		fwrite(map.data,1,map.size,data->fp1); ///data is written to a file
		gst_buffer_unmap (buffer,&map);
		gst_sample_unref(sample);

		/////////////////////////////////////////////////
	 }
}

/* 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_app_pad;
  GstPad *queue_audio_pad, *queue_app_pad;
  GstAudioInfo info;
  GstCaps *audio_caps;
  GstBus *bus;

  /* Initialize cumstom data structure */
  memset (&data, 0, sizeof (data));
  //data.fp=fopen("/songs/ChoosiChudangane.mp3","rb");
                 data.fp= fopen("./Deviceconnected.raw","rb");
 data.fp1 = fopen("1.raw","wb");
  /* 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.app_decode = gst_element_factory_make ("decodebin", "app_decode");
  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.app_queue = gst_element_factory_make ("queue", "app_queue");
  //data.audio_decode = gst_element_factory_make ("decodebin",
"audio_decode");
  data.audio_convert2 = gst_element_factory_make ("audioconvert",
"audio_convert2");
  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.audio_convert2 ||
      !data.app_queue || !data.app_sink ) //||!data.audio_decode||
!data.app_decode
  {
    g_printerr ("Not all elements could be created.\n");
    return -1;
  }

 
  /* 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);
 // g_free (audio_caps_text);

  /* 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.app_queue, data.audio_convert2,data.app_sink,
NULL);//,data.audio_decode,data.app_decode
  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.app_queue,
data.audio_convert2,data.app_sink, NULL) != TRUE )//,data.app_decode 
,data.audio_decode
     {
    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_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_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_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_app_pad);
  gst_object_unref (tee_audio_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;
}

Thanks
Sujith
  



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