gstreamer flacenc How to add duration information to flac files? - gstreamer

Question

I'm using gstreamer(gst-launch-1.0 version 1.8.3) to record flac files. the command line looks like this:
gst-launch-1.0 -v alsasrc ! flacenc ! filesink location="output.flac"
mediainfo output.flac
mediainfo result
The above picture shows the results of using the mediainfo.
This file can play at media player, but it does not support navigation and play time.
I think that there is no duration information.
player screen
How to add duration imformation to flac files?

Answer 1

I think .flac is a very basic stream format. It does not support random access or carry a duration. You can't know the exact duration unless you parse the complete file. Some players may make a "best effort" approach here and seek to roughly the file position to where you out the slider to but its nothing the format itself provides.
I think for seeking you are supposed to put .flac into a container like .ogg. This is actually very similar to .aac files which should be put into .mp4.
So try gst-launch-1.0 -e -v alsasrc ! flacenc ! oggmux ! filesink location="output.ogg".
$ mediainfo output.ogg
General
Complete name : output.ogg
Format : Ogg
Format/Info : Free Lossless Audio Codec
File size : 598 KiB
Duration : 7 s 941 ms
Overall bit rate mode : Variable
Overall bit rate : 617 kb/s
Audio
ID : 256729656 (0xF4D6238)
Format : FLAC
Format/Info : Free Lossless Audio Codec
Duration : 7 s 941 ms
Bit rate mode : Variable
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 44.1 kHz
Bit depth : 16 bits
Writing library : libFLAC 1.3.2 (UTC 2017-01-01)

Answer 2

I made some modifications to user199309's answer
https://stackoverflow.com/a/47569428/5564626
compile: g++ -o test test.c $(pkg-config --cflags --libs gstreamer-1.0)
#include <stdio.h>
#include <gst/gst.h>
#define GLIB_DISABLE_DEPRECATION_WARNINGS
static GstElement *pipeline;
static GstPad *queue_src_pad;
static GstElement *bins[2];
static GstPad *bin_pads[2];
static GstElement *filesink[2];
static GMainLoop *loop;
static GstElement *flacenc[2];
static size_t current_bin = 0;
static int current_file = 0;
static GstPadProbeReturn
pad_probe_cb(GstPad * pad, GstPadProbeInfo * info, gpointer user_data) {
gst_pad_remove_probe(pad, GST_PAD_PROBE_INFO_ID (info));
gst_pad_unlink(queue_src_pad, bin_pads[current_bin]);
gst_pad_send_event(bin_pads[current_bin], gst_event_new_eos());
gst_element_set_state(bins[current_bin], GST_STATE_NULL);
gst_object_ref(bins[current_bin]);
gst_bin_remove(GST_BIN(pipeline), bins[current_bin]);
current_file++;
current_bin = (current_file % 2);
{
char file_location[32];
sprintf(file_location, "recording_%ld.flac", current_file);
g_object_set(G_OBJECT(
filesink[current_bin]), "location", file_location, NULL);
printf("now writing to %s\n", file_location);
}
gst_bin_add(GST_BIN(pipeline), bins[current_bin]);
gst_pad_link(queue_src_pad, bin_pads[current_bin]);
gst_element_set_state(bins[current_bin], GST_STATE_PLAYING);
gst_element_sync_state_with_parent(bins[current_bin]);
return GST_PAD_PROBE_OK;
}
static gboolean timeout_cb(gpointer user_data) {
gst_pad_add_probe(queue_src_pad, GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM,
pad_probe_cb, NULL, NULL);
return TRUE;
}
static gboolean
bus_cb (GstBus *bus,
GstMessage *msg,
gpointer data)
{
GMainLoop *loop = (GMainLoop *)data;
g_print("Got %s message\n", GST_MESSAGE_TYPE_NAME(msg));
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_EOS:
g_print ("End of stream\n");
//g_main_loop_quit (loop);
break;
case GST_MESSAGE_ERROR: {
gchar *debug;
GError *error;
gst_message_parse_error(msg, &error, &debug);
g_free(debug);
g_printerr("Error: %s\n", error->message);
g_error_free(error);
g_main_loop_quit(loop);
break;
}
default:
break;
}
return TRUE;
}
gint main(gint argc, gchar *argv[])
{
GstElement *audiosrc;
GstElement *queue;
GstBus *bus;
GMainLoop *loop;
pthread_t libusb_tid;
guint bus_watch_id;
gst_init (&argc, &argv);
audiosrc = gst_element_factory_make("alsasrc", "audiosrc");
queue = gst_element_factory_make("queue", "queue");
bins[0] = gst_bin_new("bin0");
bins[1] = gst_bin_new("bin1");
flacenc[0] = gst_element_factory_make("flacenc", "flacenc0");
flacenc[1] = gst_element_factory_make("flacenc", "flacenc1");
filesink[0] = gst_element_factory_make("filesink", "filesink0");
filesink[1] = gst_element_factory_make("filesink", "filesink1");
pipeline = gst_pipeline_new("test-pipeline");
if (!pipeline || !audiosrc || !queue
|| !flacenc[0] || !filesink[0]
|| !flacenc[1] || !filesink[1]
) {
g_printerr ("not all elements could be created\n");
//return -1;
}
gst_bin_add_many(GST_BIN(bins[0]), flacenc[0], filesink[0], NULL);
gst_bin_add_many(GST_BIN(bins[1]), flacenc[1], filesink[1], NULL);
gst_bin_add_many(GST_BIN(pipeline), audiosrc, queue, bins[0], NULL);
g_assert(gst_element_link(audiosrc, queue));
g_assert(gst_element_link_many(flacenc[0], filesink[0], NULL));
g_assert(gst_element_link_many(flacenc[1], filesink[1], NULL));
GstPad* pad = gst_element_get_static_pad(flacenc[0], "sink");
gst_element_add_pad(bins[0], gst_ghost_pad_new("sink", pad));
gst_object_unref(pad);
GstPad* pad2 = gst_element_get_static_pad(flacenc[1], "sink");
gst_element_add_pad(bins[1], gst_ghost_pad_new("sink", pad2));
gst_object_unref(pad2);
bin_pads[0] = gst_element_get_static_pad(bins[0], "sink");
bin_pads[1] = gst_element_get_static_pad(bins[1], "sink");
current_bin = 0;
gst_element_link(queue, bins[current_bin]);
g_object_set(filesink[current_bin], "location", "recording_0.flac", NULL);
queue_src_pad = gst_element_get_static_pad(queue, "src");
bus = gst_element_get_bus(pipeline);
bus_watch_id = gst_bus_add_watch(bus, bus_cb, loop);
gst_element_set_state(pipeline, GST_STATE_PLAYING);
loop = g_main_loop_new(NULL, FALSE);
g_timeout_add_seconds(10, timeout_cb, NULL);
g_main_loop_run (loop);
gst_object_unref(bus);
gst_element_set_state(pipeline, GST_STATE_NULL);
gst_object_unref(pipeline);
return 0;
}