Appsrc produces silent audio

dingoegret sephvelut at gmail.com
Sun Jun 25 22:28:13 UTC 2017


I have a simple application using appsrc. The produced file is blank. No
sound. How do timestamps work for buffers? If I set a buffer's pts to time
now() then wont that expire immediately after?



#include <gst/gst.h>
#include <string.h>
#include <ios>
#include <vector>
#include <fstream>
#include <gst/app/gstappsrc.h>
#include <iostream>

#define CHUNK_SIZE 10 * 1024   /* Amount of bytes we are sending in each
buffer */
#define SAMPLE_RATE 8000 /* Samples per second we are sending */

/* Structure to contain all our information, so we can pass it to callbacks
*/
typedef struct _CustomData {
    GstElement *pipeline;
    GstElement *app_source;
    GstElement *filesink;

    guint64 num_samples;   /* Number of samples generated so far (for
timestamp generation) */

    guint sourceid;        /* To control the GSource */

    GMainLoop *main_loop;  /* GLib's Main Loop */
} CustomData;

std::ifstream file_stream("/Users/seph/Desktop/intro.wav");
std::vector<char> chunk(CHUNK_SIZE);
unsigned long timestamp = 0;
/* 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).
 */
int c = 0;
static gboolean push_data(CustomData *data) {
    GstBuffer *buffer;
    GstFlowReturn ret;
    GstMapInfo map;
    unsigned long duration;

    /* Create a new empty buffer */
    buffer = gst_buffer_new_and_alloc (CHUNK_SIZE);

    GST_BUFFER_TIMESTAMP(buffer) = timestamp;
    duration = (unsigned long) (((double) CHUNK_SIZE / (SAMPLE_RATE * 1 *
(16 / 8))) * GST_SECOND);
    GST_BUFFER_DURATION (buffer) = duration;
    timestamp += duration;

    std::cout << timestamp << std::endl;

    gst_buffer_map(buffer, &map, GST_MAP_WRITE);
    file_stream.rdbuf()->sgetn(&chunk[0], chunk.size());
    map.data = (guint8 *) chunk.data();

    gst_buffer_unmap(buffer, &map);

    /* Push the buffer into the appsrc */
    g_signal_emit_by_name(data->app_source, "push-buffer", buffer, &ret);

    /* 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;
    }

    // ending prematurely. no reason really
    if(c == 4)
        gst_app_src_end_of_stream((GstAppSrc *) data->app_source);

    c++;
    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;
    }
}

/* 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;
    GstBus *bus;

    /* Initialize cumstom data structure */
    memset(&data, 0, sizeof(data));

    /* Initialize GStreamer */
    gst_init(&argc, &argv);

    /* setup pipeline */
    data.pipeline = gst_pipeline_new("pipeline");
    data.app_source = gst_element_factory_make("appsrc", NULL);
    data.filesink = gst_element_factory_make("filesink", NULL);

    g_assert(data.app_source);
    g_assert(data.filesink);
    g_assert(data.pipeline);

    gst_bin_add_many(GST_BIN(data.pipeline), data.app_source, data.filesink,
NULL);
    gst_element_link_many(data.app_source, data.filesink, NULL);

    g_object_set(G_OBJECT (data.filesink), "location",
"/Users/seph/Desktop/test.wav", NULL);
    //g_object_set (data.app_source, "caps", 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);

    /* 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);

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



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