<html> <head> <style></style></head> <body class='hmmessage'><div dir='ltr'>Hello I have use playbin2 with Gstreamer-0.10 for my app and it could read any kind of file. I want to port it to Gstreamer1.0. I have made some adjustments, as I could find from the doc. It can read only wave files now, but not .flac or .mp3 files! I have every possible plugins installed (I think). Could please help me with this? Thank you! Victor Here is the code for Gstreamer1.0 (when code for Gstreamer-0.10 is different, it is put after "//") : #include <stdlib.h> #include <stdio.h> #include <string.h> #include <gst/gst.h> #define AUDIOFREQ 44100 int main(int argc, char *argv[]){ gst_init (NULL, NULL); GstElement *src, *audioconvert, *audioconvert2, *audioconvert3, *spectrum, *flacenc, *sink, *equalizer, *equalizer2, *equalizer3, *pipeline, *BP_BRfilter, *playbin; GstBus *bus; GstCaps *caps; GstPad *audiopad; GMainLoop *loop; float limit = 10000; loop = g_main_loop_new(NULL, FALSE); pipeline = gst_bin_new ("pipeline"); g_assert (pipeline); playbin = gst_element_factory_make ("playbin", NULL); //GSTREAMER-0.10 : playbin = gst_element_factory_make ("playbin2", NULL); g_assert (playbin); g_object_set (G_OBJECT (playbin), "uri", "file:///home/victor/Music/main.wav", NULL); audioconvert = gst_element_factory_make ("audioconvert", NULL); g_assert (audioconvert); audioconvert2 = gst_element_factory_make ("audioconvert", NULL); g_assert (audioconvert2); audioconvert3 = gst_element_factory_make ("audioconvert", NULL); g_assert (audioconvert3); spectrum = gst_element_factory_make ("spectrum", "spectrum"); g_assert (spectrum); equalizer = gst_element_factory_make ("equalizer-nbands", "equalizer-nbands"); g_assert (equalizer); equalizer2 = gst_element_factory_make ("equalizer-nbands", "equalizer-nbands2"); g_assert (equalizer2); equalizer3 = gst_element_factory_make ("equalizer-nbands", "equalizer-nbands3"); g_assert (equalizer3); BP_BRfilter = gst_element_factory_make ("audiochebband", NULL); g_assert (BP_BRfilter); g_object_set (G_OBJECT (BP_BRfilter), "upper-frequency", limit, NULL); flacenc = gst_element_factory_make ("flacenc", NULL); g_assert (flacenc); sink = gst_element_factory_make("autoaudiosink", "sink"); g_assert (sink); gst_bin_add_many (GST_BIN (pipeline), audioconvert, equalizer, equalizer2, equalizer3, audioconvert2, BP_BRfilter, audioconvert3, spectrum, flacenc, sink, NULL); caps = gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, AUDIOFREQ, NULL); //GSTREAMER-0.10 : caps = gst_caps_new_simple ("audio/x-raw-int", "rate", G_TYPE_INT, AUDIOFREQ, NULL); g_assert(caps); if (!gst_element_link (audioconvert, equalizer)) //GSTREAMER-0.10 : if (!gst_element_link_filtered (audioconvert, equalizer, caps)) { fprintf (stderr, "can't link elements 2\n"); exit (1); } if (!gst_element_link_many (equalizer, equalizer2, equalizer3, audioconvert2, BP_BRfilter, audioconvert3, spectrum, sink, NULL)) { fprintf (stderr, "can't link elements\n"); exit (1); } audiopad = gst_element_get_static_pad (audioconvert, "sink"); gst_element_add_pad (pipeline, gst_ghost_pad_new (NULL, audiopad)); g_object_set(G_OBJECT(playbin), "audio-sink", pipeline, NULL); gst_object_unref (audiopad); gst_element_set_state (playbin, GST_STATE_PLAYING); g_main_loop_run (loop); gst_element_set_state (playbin, GST_STATE_NULL); } </div></body> </html>