Encoding raw data from buffere and store encoded data in another buffer

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Encoding raw data from buffere and store encoded data in another buffer

devil coder
Hi guys,

I have very less experience in gstreamer api level code.
kindly help me or guide in right path.


I want to inject the buffer (two buffer's ) containing YUV data & PCM data
(read from .yuv & .pcm files) to gstreamer-1.0 pipe line.

I am getting error in linking the elements.
"Elements could not be linked"

#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <gst/app/gstappsrc.h>
#include <gst/base/gstpushsrc.h>
#include <gst/app/gstappsink.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((GstAppSrc *)(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 */

        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");
                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(map.data, 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 :


int main1(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("E:/pocVR/ConsoleApplication6/x64/Transformers1080p.pcm",
        if (data.fp == NULL)
                printf("\n not bale to open input file \n");
        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",
        data.audio_resample = gst_element_factory_make("audioresample",
        data.audio_sink = gst_element_factory_make("autoaudiosink",
        data.app_queue = gst_element_factory_make("queue", "app_queue");
        //data.audio_decode = gst_element_factory_make ("decodebin",
        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.audio_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,
        audio_caps = gst_audio_info_to_caps(&info);
        g_object_set(data.app_source, "caps", audio_caps, "format",GST_FORMAT_TIME,
        g_signal_connect(data.app_source, "need-data",
        g_signal_connect(data.app_source, "enough-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_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,
        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 ||
                        data.audio_convert2, data.app_sink, NULL) != TRUE)//,data.app_decode ,
                g_printerr("Elements could not be linked.\n");
        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");
                return -1;

        /* Instruct the bus to emit signals for each received message, and connect
        to the interesting signals */
        bus = gst_element_get_bus(data.pipeline);
        g_signal_connect(G_OBJECT(bus), "message::error", (GCallback)error_cb,

        /* 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);

        /* 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);

        /* Free resources */
        gst_element_set_state(data.pipeline, GST_STATE_NULL);
        return 0;

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