Debugging information: gstbasesrc.c(2948): gst_base_src_loop (): /GstPipeline:test-pipeline/GstAppSrc:audio_source: streaming task paused, reason not-negotiated (-4)

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Debugging information: gstbasesrc.c(2948): gst_base_src_loop (): /GstPipeline:test-pipeline/GstAppSrc:audio_source: streaming task paused, reason not-negotiated (-4)

Sujith reddy
Hi All,

I am getting a below error when i am compiling the code. can anyone help on
this error.Here i am attaching the code.

****************error snippet start***************

*Received new pad 'src_0' from 'app_decode':
  Link succeeded (type 'audio/x-raw').
Error received from element audio_source: Internal data flow error.
Debugging information: gstbasesrc.c(2948): gst_base_src_loop ():
streaming task paused, reason not-negotiated (-4)*
*****************error snippet end***************

*****************code start***************
for compiling use below command.........

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;
        GstElement *app_queue, *audio_convert2,  *app_sink;

        guint64 num_samples;   /* Number of samples generated so far (for timestamp
generation) */
        guint sourceid;        /* To control the GSource */
        FILE *fp,*fp1;
        GMainLoop *main_loop;  /* GLib's Main Loop */
} CustomData;
static void pad_added_handler (GstElement *src, GstPad *new_pad, CustomData
*data) {
        GstPad *sink_pad = gst_element_get_static_pad (data->tee, "sink");
        GstPadLinkReturn ret;
        GstCaps *new_pad_caps = NULL;
        GstStructure *new_pad_struct = NULL;
        const gchar *new_pad_type = NULL;

        g_print ("Received new pad '%s' from '%s':\n", GST_PAD_NAME (new_pad),

        /* If our converter is already linked, we have nothing to do here */
        if (gst_pad_is_linked (sink_pad)) {
                g_print ("  We are already linked. Ignoring.\n");
                goto exit;

        /* Check the new pad's type */
        new_pad_caps = gst_pad_query_caps (new_pad, NULL);
        new_pad_struct = gst_caps_get_structure (new_pad_caps, 0);
        new_pad_type = gst_structure_get_name (new_pad_struct);
        if (!g_str_has_prefix (new_pad_type, "audio/x-raw")) {
                g_print ("  It has type '%s' which is not raw audio. Ignoring.\n",
                goto exit;

/* Attempt the link */
        ret = gst_pad_link (new_pad, sink_pad);
        if (GST_PAD_LINK_FAILED (ret)) {
        g_print ("  Type is '%s' but link failed.\n", new_pad_type);
        } else {
        g_print ("  Link succeeded (type '%s').\n", new_pad_type);

        /* Unreference the new pad's caps, if we got them */
        if (new_pad_caps != NULL)
        gst_caps_unref (new_pad_caps);

        /* Unreference the sink pad */
        gst_object_unref (sink_pad);

/* 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; /* 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_BUFFER_DURATION (buffer) = gst_util_uint64_scale (CHUNK_SIZE,

        /* Generate some psychodelic waveforms */
        gst_buffer_map (buffer, &map, GST_MAP_WRITE);
        gst_buffer_unmap (buffer, &map);
        data->num_samples += num_samples;
        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
* 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) {

        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(,1,map.size,data->fp1); ///data is written to a file
                gst_buffer_unmap (buffer,&map);


/* 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 :
        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.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.app_decode = gst_element_factory_make ("decodebin", "app_decode");
        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",
        data.audio_resample = gst_element_factory_make ("audioresample",
        data.audio_sink = gst_element_factory_make ("autoaudiosink", "audio_sink");
        data.app_queue = gst_element_factory_make ("queue", "app_queue");
        data.audio_convert2 = gst_element_factory_make ("audioconvert",
        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.app_decode ) //
        g_printerr ("Not all elements could be created.\n");
        return -1;

        /* 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.app_decode,
data.tee, data.audio_queue, data.audio_convert1, data.audio_resample,
        data.audio_sink, data.app_queue, data.audio_convert2,data.app_sink,
        if (gst_element_link_many (data.app_source, data.app_decode, 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
        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
        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,
        gst_object_unref (bus);
        /* Configure appsrc */
        gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, SAMPLE_RATE, 1,
        audio_caps = gst_audio_info_to_caps (&info);
        //g_object_set (data.app_source, "caps", audio_caps, "format",
        g_object_set (data.app_source,  "format", GST_FORMAT_TIME, NULL);

        g_signal_connect (data.app_source, "need-data", G_CALLBACK (start_feed),
        g_signal_connect (data.app_source, "enough-data", G_CALLBACK (stop_feed),
      /*configure decodebin*/
    g_signal_connect (data.app_decode, "pad-added", G_CALLBACK
(pad_added_handler), &data);

        /* Configure appsink */
        g_object_set (data.app_sink, "emit-signals", TRUE, "caps", audio_caps,
        g_signal_connect (data.app_sink, "new-sample", G_CALLBACK (new_sample),
        gst_caps_unref (audio_caps);
        // g_free (audio_caps_text);

        /* Start playing the pipeline */
        gst_element_set_state (data.pipeline, GST_STATE_PLAYING);
        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;

*******************code end **********************


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