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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("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", "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; } |
Hi,
On Mon, Jun 26, 2017 at 7:28 AM, dingoegret <[hidden email]> wrote: > > 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? Do you mean that "/Users/seph/Desktop/test.wav" is size 0? filesink seems to write to the file in my env. what do you see if you set GST_DEBUG=filesink:6 ? regards, -- yashi _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
It writes to test.wav - places bytes in there - but there is no actual audio data. Just a blank file filled with bytes
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On Tue, Jun 27, 2017 at 8:35 PM, dingoegret <[hidden email]> wrote:
> > It writes to test.wav - places bytes in there - but there is no actual audio > data. Just a blank file filled with bytes ah, 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); This does not work. memcpy() it, or use gst_buffer_fill(). -- yashi _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Sorry it took me so long to reply. Thank you for your help : )
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