Internal data stream error when use streaming audio via Bluetooth

[ryan buf]

Hi,
I have a problem with streaming audio via bluetooth via avdtpsrc liblrary.
In the below, my code.

#include <gst/gst.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;
}


/* Main function for audio pipeline initialization and looping streaming
process */

gint main (gint argc, gchar **argv) {

GMainLoop *loop;

GstElement *pipeline;
GstElement *source;
GstElement *depay;
GstElement *parse;
GstElement *decoder;
GstElement *converter;
GstElement *queues;
GstElement *volume;
GstElement *sink;
GstElement *jitterbuffer;

GstBus *bus;
GstCaps *caps;
gboolean ret;

/* Initialisation */
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);

/* Create gstreamer elements:
*/

pipeline = gst_pipeline_new ("audio_stream");
gst_element_set_state (pipeline, GST_STATE_NULL);

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

source = gst_element_factory_make ("avdtpsrc", "audio_source");
g_return_val_if_fail (source, -1);
jitterbuffer = gst_element_factory_make("rtpjitterbuffer", "jitterbuffer");
depay = gst_element_factory_make("rtpsbcdepay", "depayloader");
parse = gst_element_factory_make("sbcparse", "parser");
decoder = gst_element_factory_make("sbcdec", "decoder");
volume = gst_element_factory_make("volume", "volume");
converter = gst_element_factory_make("audioconvert", "converter");
queues = gst_element_factory_make ("queue", "queues");
g_return_val_if_fail (queues, -1);
sink = gst_element_factory_make ("alsasink", "audio_sink");
g_return_val_if_fail (sink, -1);

if (!pipeline) {
g_printerr ("Audio: Pipeline couldn't be created\n");
return -1;
}
if (!source) {
    g_printerr ("Audio: alsasrc couldn't be created\n");
    return -1;
}
if (!sink) {
    g_printerr ("Audio: Output file couldn't be created\n");
    return -1;
}

// g_object_set (G_OBJECT (sink), "transport",
"/org/bluez/hci0/dev_2C_BA_BA_6D_27_FA", NULL);




/* Add elements to the bin before linking them. */
gst_bin_add (GST_BIN (pipeline), source);
gst_bin_add (GST_BIN (pipeline), jitterbuffer);
gst_bin_add (GST_BIN (pipeline), depay);
gst_bin_add (GST_BIN (pipeline), parse);
gst_bin_add (GST_BIN (pipeline), decoder);
gst_bin_add (GST_BIN (pipeline), converter);
gst_bin_add (GST_BIN (pipeline), queues);
gst_bin_add (GST_BIN (pipeline), sink);
g_object_set (G_OBJECT (jitterbuffer), "latency",0, NULL);
g_object_set (G_OBJECT (jitterbuffer), "drop-on-latency", TRUE, NULL);

g_object_set (G_OBJECT (source), "transport",
"/org/bluez/hci0/dev_98_10_E8_34_04_BE/fd0", NULL);
/* Link the pipeline */
caps = gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, "S16LE",
"layout", G_TYPE_STRING, "interleaved", "rate", G_TYPE_INT, (int)44100,
"channels", G_TYPE_INT, (int)2, NULL);
ret = gst_element_link_filtered (source, sink, caps);
    if (!ret) {
        g_print ("audio_source and sink couldn't be linked\n");
        gst_caps_unref (caps);
        return FALSE;
    }

gst_element_link_pads (queues, "src", sink, "sink");


/* Set the pipeline to "playing" state */
// g_print ("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;
}
When run it. It error
 "Running...
Error: Internal data stream error.
Returned, stopping playback
Deleting pipeline"

When I using cmd
gst-launch-1.0 avdtpsrc transport=/org/bluez/hci0/dev_98_10_E8_34_04_BE/fd0
! rtpjitterbuffer ! rtpsbcdepay ! sbcparse ! sbcdec ! volume  ! audioconvert
! alsasink

It working.

How to my code work?





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