~audio-recorder/audio-recorder/trunk

/*
 * Copyright (c) Linux community.
 *
 * This library is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Library General Public
 * License as published by the Free Software Foundation; either
 * version 2 of the License, or (at your option) any later version.
 *
 * This library is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.	 See the GNU
 * Library General Public License for more details.
 *
 * You should have received a copy of the GNU Library General Public
 * License along with this library; if not, write to the
 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
 * Boston, MA 02111-1307, USA.
*/
#include <math.h>
#include "gst-recorder.h"
//#include "gst-recorder-ext.h"
#include "rec-manager.h"
#include "media-profiles.h"
#include "log.h"
#include "support.h"
#include "dconf.h"
#include "utility.h"
#include "pulseaudio.h"

#include "timer.h"

#include "gst-pipeline.h"

#define AUDIO_RECORDER 101
#define AUDIO_RECORDER_ERR1 -1

// GStreamer pipeline
static GstElement *g_pipeline;

// Flag to test if EOS (end of stream) message was seen
static gboolean g_got_EOS_message = FALSE;

static gboolean rec_level_message_cb(GstBus *bus, GstMessage *message, void *user_data);

GstElement *rec_create_pipeline(PipelineParms *parms, GError **error);

void rec_set_state_to_null();

static gchar *rec_create_filename(gchar *track, gchar *artist, gchar *album);
static gchar *rec_generate_unique_filename();
static gchar *check_audio_folder(gchar *audio_folder);
static gchar *rec_get_media_profile_id();

void rec_module_init() {
    LOG_DEBUG("Init gst-recorder.c.\n");

    g_pipeline = NULL;
}

void rec_module_exit() {
    LOG_DEBUG("Clean up gst-recorder.c.\n");

    // Stop evt. recording
    rec_stop_recording(FALSE);
}

void rec_set_state_to_null() {
    // Copyright notice:
    // I have copied this function from gnome-media-2.30.0/grecord.

    LOG_DEBUG("\n--------- rec_set_state_to_null() ----------\n");

    // Set state of pipeline to GST_STATE_NULL.
    GstMessage *msg;
    GstState cur_state, pending;
    GstBus *bus;

    if (!GST_IS_PIPELINE(g_pipeline)) return;

    gst_element_get_state(g_pipeline, &cur_state, &pending, 0);

    if (cur_state == GST_STATE_NULL && pending == GST_STATE_VOID_PENDING)
        return;

    if (cur_state == GST_STATE_NULL && pending != GST_STATE_VOID_PENDING) {
        gst_element_set_state (g_pipeline, GST_STATE_NULL);
        return;
    }

    gst_element_set_state(g_pipeline, GST_STATE_READY);
    gst_element_get_state(g_pipeline, NULL, NULL, -1);

    bus = gst_element_get_bus(g_pipeline);
    if (GST_IS_BUS(bus)) {
        while ((msg = gst_bus_pop(bus))) {
            gst_bus_async_signal_func(bus, msg, NULL);

            if (msg) {
                gst_message_unref(msg);
            }
        }
        gst_object_unref(bus);
    }

    gst_element_set_state(g_pipeline, GST_STATE_NULL);
}

void rec_pause_recording()  {
    // Valid pipeline?
    if (!GST_IS_PIPELINE(g_pipeline)) return;

    // Get recording state
    gint state = -1;
    gint pending = -1;
    rec_get_state(&state, &pending);

    // Already paused?
    if (state == GST_STATE_PAUSED) return;

    LOG_DEBUG("\n--------- rec_pause_recording() ----------\n");

    // Reset timer
    timer_module_reset(GST_STATE_PAUSED);

    // Pause the stream
    gst_element_set_state(g_pipeline, GST_STATE_PAUSED);
}

void rec_continue_recording()  {
    // Valid pipeline?
    if (!GST_IS_PIPELINE(g_pipeline)) return;

    // Get recording state
    gint state = -1;
    gint pending = -1;
    rec_get_state(&state, &pending);

    // Already playing?
    if (state == GST_STATE_PLAYING) return;

    LOG_DEBUG("\n--------- rec_continue_recording() ----------\n");

    // Reset timer
    timer_module_reset(GST_STATE_PLAYING);

    // Continue recording
    gst_element_set_state(g_pipeline, GST_STATE_PLAYING);
}

void rec_update_gui() {
    // Recording state has changed (PLAYING, PAUSED, STOPPED/NULL).

    // Set widgets and GUI.
    rec_manager_update_gui();
}

gboolean rec_start_recording() {
    LOG_DEBUG("\n--------- rec_start_recording() ----------\n");

    // Get recording state
    gint state = -1;
    gint pending = -1;
    rec_get_state(&state, &pending);

    // Already recording?
    if (state == GST_STATE_PLAYING) {
        return TRUE;

        // Is it paused?
    } else if (state == GST_STATE_PAUSED) {
        // Continue recording
        rec_continue_recording();
        return TRUE;
    }

    // Call stop
    rec_stop_recording(FALSE);

    // Reset timer (prepare for GST_STATE_PLAYING)
    timer_module_reset(GST_STATE_PLAYING);

    // Create a new GStreamer pipeline and start recording.

    // Clear static variables (call with NULLs)
    rec_level_message_cb(NULL, NULL, NULL);

    // Variables
    gboolean ret = FALSE;

    // Parms to construct a pipeline
    PipelineParms *parms = g_malloc0(sizeof(PipelineParms));

    gchar *track_name = NULL;
    gchar *artist_name = NULL;
    gchar *album_name = NULL;
    gint track_pos = 0;
    gint track_len = 0;

    // Get data from GConf registry.
    // Track name
    conf_get_string_value("track/track-name", &track_name);
    str_trim(track_name);
    purify_filename(track_name, TRUE/*purify_all*/);

    // Artist
    conf_get_string_value("track/artist-name", &artist_name);
    str_trim(artist_name);
    purify_filename(artist_name, TRUE/*purify_all*/);

    // Album
    conf_get_string_value("track/album-name", &album_name);
    str_trim(album_name);
    purify_filename(album_name, TRUE/*purify_all*/);

    conf_get_int_value("track/track-pos", &track_pos);
    conf_get_int_value("track/track-len", &track_len);

    LOG_DEBUG("Start recording. track=%s artist=%s album=%s track pos/len=%d/%d.\n", track_name, artist_name, album_name, track_pos, track_len);

    // Get last (saved) track name
    gchar *last_track_name = NULL;
    conf_get_string_value("track/last-track-name", &last_track_name);
    str_trim(last_track_name);

    // Append to file?
    conf_get_boolean_value("append-to-file", &parms->append);

    if (parms->append && g_file_test(last_track_name, G_FILE_TEST_IS_REGULAR)) {
        // Record to an existing filename
        parms->filename = g_strdup(last_track_name);
    }
    // Did we get a track name (output filename) from a Media Player, Skype?
    else if (str_length(track_name, 2048) > 0) {
        // We have track, album and artist names. Create filename from these.
        parms->filename = rec_create_filename(track_name, artist_name, album_name);

    } else  {
        // Simply generate a new, unique filename from date+time pattern
        parms->filename = rec_generate_unique_filename();
    }

    // Purify the final filename, remove illegal characters. Edit in place.
    purify_filename(parms->filename, FALSE);

    // Can we write to path/filename?
    if (!is_file_writable(parms->filename)) {
        gchar *msg = g_strdup_printf(_("Cannot write to file \"%s\".\n"), parms->filename);

        // Show error text
        LOG_DEBUG("Cannot write to file \"%s\".\n", parms->filename);

        rec_manager_set_error_text(msg);
        g_free(msg);

        goto LBL_1;
    }

    // Save the last file name
    conf_save_string_value("track/last-track-name", parms->filename);

    // Get the saved media profile id (aac, mp3, cdlossless, cdlossy, etc).
    gchar *media_profile_id = rec_get_media_profile_id();

    // Get partial pipline from GConf for this media_profile_id
    parms->profile_str = media_profiles_get_pipeline(media_profile_id);
    parms->file_ext = media_profiles_get_extension(media_profile_id);
    g_free(media_profile_id);



#if 0
    // Debugging:
    gchar *device_id = NULL;
    conf_get_string_value("audio-device-id", &device_id);

    gchar *device_name = NULL;
    conf_get_string_value("audio-device-name", &device_name);

    gint device_type = -1;
    conf_get_int_value("audio-device-type", &device_type);

    LOG_DEBUG("audio-device-id=%s\n", audio_device_id);
    LOG_DEBUG("audio-device-name=%s\n", audio_device_name);
    LOG_DEBUG("audio-device-type=%d\n", audio_device_type);

    g_free(device_name);
    g_free(device_id);
#endif

    // Set/show current filename
    rec_manager_set_filename_label(parms->filename);

    // Get audio source and device list
    gchar *audio_source = NULL;
    parms->dev_list = audio_sources_get_device_NEW(&audio_source);
    parms->source = audio_source; 

    // Now build a GStreamer pipeline with these values
    GError *error = NULL;
    g_pipeline = rec_create_pipeline(parms, &error);

    ret = TRUE;

    // Ok?
    if (error) {
        // Got errors.

        // Display error message in the GUI (set red label)
        rec_manager_set_error_text(error->message);

        // Stop and reset everything
        rec_stop_and_reset();

        g_error_free(error);

        ret = FALSE;
    } else {
        // Alles Ok. Clear error messages.
        rec_manager_set_error_text(NULL);
    }

    LOG_DEBUG("------------------------\n");

LBL_1:

    g_free(track_name);
    g_free(artist_name);
    g_free(album_name);

    pipeline_free_parms(parms);
    parms = NULL;

    return ret;
}

gint64 rec_get_stream_time() {
    // Return current recording time in seconds

    GstFormat format = GST_FORMAT_TIME;
    gint64 val = -1;
    gint secs = 0L;

    // Valid pipeline?
    if (!GST_IS_ELEMENT(g_pipeline)) return 0L;

    // Stream/recording time
    if (gst_element_query_position(g_pipeline, &format, &val) && val != -1) {
        secs = val / GST_SECOND;
    }
    return secs;
}

void rec_stop_recording(gboolean delete_file) {
    // Stop recording, terminate pipeline

    if (!GST_IS_PIPELINE(g_pipeline)) return;

    // Get recording state
    gint state = -1;
    gint pending = -1;
    rec_get_state(&state, &pending);

    // Already stopped?
    if (state != GST_STATE_NULL) {
        // Reset timer
        timer_module_reset(GST_STATE_NULL);
    }

    LOG_DEBUG("rec_stop_recording(%s)\n", (delete_file ? "delete_file=TRUE" : "delete_file=FALSE"));

    // Send EOS message. This will terminate the stream/file properly. This is very important for ACC (.m4a) files.
    gst_element_send_event(g_pipeline, gst_event_new_eos());
    gst_element_send_event(g_pipeline, gst_event_new_eos());

    g_usleep(GST_USECOND * 5);

    // Set pipeline state to NULL
    rec_set_state_to_null();

    // Shutdown the entire pipeline
    gst_element_set_state(g_pipeline, GST_STATE_NULL);

    // Destroy it
    if (GST_IS_OBJECT(g_pipeline))
        gst_object_unref(GST_OBJECT(g_pipeline));

    g_pipeline = NULL;

    LOG_DEBUG("--------- Pipeline closed and destroyed ----------\n\n");

    // Delete the recorded file?
    if (delete_file) {
        // Get last_track_name
        gchar *filename = NULL;
        conf_get_string_value("track/last-track-name", &filename);

        LOG_DEBUG("Deleted file:\"%s\"\n", filename);

        // Remove it
        g_remove(filename);
        g_free(filename);

        // Erase last_track_name
        conf_save_string_value("track/last-track-name", "");

        // Update filename in the GUI
        rec_manager_set_filename_label("");
    }

    // Notice:
    // The rec_state_changed_cb() function will reset the GUI by calling rec_manager_update_gui();
}

void rec_stop_and_reset() {
    // Stop recording
    rec_stop_recording(FALSE);

    // Reset window and its widgets
    rec_manager_update_gui();
}

void rec_get_state(gint *state, gint *pending) {
    /*
    state:
    GST_STATE_VOID_PENDING
    GST_STATE_NULL
    GST_STATE_READY
    GST_STATE_PAUSED
    GST_STATE_PLAYING
    */
    *state = GST_STATE_NULL;
    *pending = GST_STATE_NULL;

    // The pipeline has been created?
    if (!GST_IS_ELEMENT(g_pipeline)) return;

    // The state is?
    // Ref: http://www.gstreamer.net/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-get-state
    gst_element_get_state(g_pipeline, (GstState*)state, (GstState*)pending, 0);
}

gboolean rec_is_recording() {
    // Get recording state
    gint state = -1;
    gint pending = -1;
    rec_get_state(&state, &pending);

    gboolean ret = FALSE;

    // Is playing?
    switch (state) {
    case GST_STATE_PLAYING:
        ret = TRUE;
        break;
    }
    return ret;
}

static gboolean rec_level_message_cb(GstBus *bus, GstMessage *message, void *user_data) {
    if (!GST_IS_MESSAGE(message)) return TRUE;

    static guint64 last_stream_time_t = 0L;
    static guint64 last_stream_time_fz = 0L;

    // Calling with NULL arguments?
    // This will reset the static variables, then exit.
    if (!message) {
        last_stream_time_t = 0L;
        last_stream_time_fz = 0L;
        return TRUE;
    }

    guint64 stream_time = 0L;

    if (message->type == GST_MESSAGE_ELEMENT) {
        const GstStructure *s = gst_message_get_structure (message);
        const gchar *name = gst_structure_get_name(s);

        if (g_str_equal(name, "level")) {
            // Ref: http://www.gstreamer.net/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-level.html

            GstClockTime endtime;
            if (!gst_structure_get_clock_time(s, "endtime", &endtime)) {
                LOG_ERROR("Pipeline error: Cannot read endtime.\n");
            } else {
                stream_time = GST_TIME_AS_SECONDS(endtime);
            }

            const GValue *list;
            list = gst_structure_get_value(s, "rms");

            gint channels = gst_value_list_get_size(list);
            if (channels < 1) goto LBL_1;

            const GValue *value;
            list = gst_structure_get_value (s, "peak");
            value = gst_value_list_get_value (list, 0);
            gdouble peak_dB = g_value_get_double(value);

            gdouble norm_peak = exp(peak_dB / 20);
            if (norm_peak > 1.0) norm_peak = 1.0;

            // Update level (progress) bar
            rec_manager_update_level_bar(norm_peak * 0.95, peak_dB);  // (0.95, just cheating a bit, space for value-text)

            // Update the time label (GUI).
            if (stream_time - last_stream_time_t >= 1/*seconds*/) {
                guint hours = (guint)(stream_time / 3600);
                stream_time = stream_time - (hours*3600);

                // Count only to 23 hours
                // if (hours > 99) hours = 23;

                guint minutes = (guint)(stream_time / 60);
                guint seconds = stream_time - (minutes*60);

                // Show stream time
                gchar *time_txt = g_strdup_printf("%02d:%02d:%02d", hours, minutes, seconds);
                rec_manager_set_time_label(time_txt);
                g_free(time_txt);

                // Save last stream_time
                last_stream_time_t = stream_time;
            }


            // Update the time label (GUI).
            if (stream_time < 10 || (stream_time - last_stream_time_fz > 3/*seconds*/)) {
                // Update more frequently if stream_time < 10 seconds, after that, update each 3.rd second.

                gchar *filename = rec_manager_get_output_filename();
                gchar *size_txt = NULL;

                // Get file size
                struct stat fstat;
                if (stat(filename, &fstat))
                    LOG_ERROR("Cannot get file information of %s.\n", filename);
                else
                    size_txt = format_file_size(fstat.st_size);

                // Show file size
                rec_manager_set_size_label(size_txt);

                g_free(filename);
                g_free(size_txt);

                // Save last stream_time
                last_stream_time_fz = stream_time;
            }
        }
    }

LBL_1:
    // TRUE: Continue calling this function
    return TRUE;
}

static void rec_state_changed_cb(GstBus *bus, GstMessage *msg, void *userdata) {
    if (!GST_IS_MESSAGE(msg)) return;

    // We are only interested in the top-level pipeline.
    if (GST_MESSAGE_SRC(msg) != GST_OBJECT(g_pipeline)) return;

    GstState old_state, new_state;
    gst_message_parse_state_changed(msg, &old_state, &new_state, NULL);

    GstState state, pending;
    gst_element_get_state(g_pipeline, &state, &pending, 0);
    LOG_DEBUG("Pipeline state changed from %s to: %s  (stat=%s pending=%s).\n", 
              gst_element_state_get_name(old_state), gst_element_state_get_name(new_state), 
              gst_element_state_get_name(state), gst_element_state_get_name(pending));

    // Pipeline state changed from PAUSED to: READY  (stat=READY pending=VOID_PENDING).

    switch (new_state) {

    case GST_STATE_PLAYING:
        // We are playing. Inform the GUI.
        rec_manager_update_gui();
        break;

    case GST_STATE_READY:
        // Recording may have stopped. Inform the GUI.
        rec_manager_update_gui();
        break;

    case GST_STATE_PAUSED:
        // Goes from GST_STATE_PLAYING to GST_STATE_PAUSED state?
        if (old_state == GST_STATE_PLAYING) {

            // Recording has paused. Inform the GUI.
            rec_manager_update_gui();

        }
        break;

    case GST_STATE_NULL:
        // Recording has stopped. Inform the GUI.
        rec_manager_update_gui();
        break;

    default:
        break;
    }
}

static void rec_pipeline_error_cb(GstBus *bus, GstMessage *msg, void *userdata) {
    if (!GST_IS_MESSAGE(msg)) return;

    GError *error = NULL;
    gchar *dbg = NULL;

    gst_message_parse_error(msg, &error, &dbg);
    g_return_if_fail(error != NULL);

    LOG_DEBUG("\nGot pipeline error: %s.\n", error->message);

    if (dbg) {
        g_free(dbg);
    }

    if (error->code == GST_RESOURCE_ERROR_BUSY) {
        g_error_free(error);
        return;
    }

    if (error) {
        g_error_free(error);
    }
}

static void rec_eos_msg_cb(GstBus *bus, GstMessage *msg, void *userdata) {
    LOG_DEBUG("Got EOS message. Finishing recording.\n");

    // We've seen EOS. Set g_got_EOS_message to TRUE.
    g_got_EOS_message = TRUE;
}

GstElement *rec_create_pipeline(PipelineParms *parms, GError **error) {
    // Create a GStreamer pipeline for audio recording.
    gchar *err_msg = NULL;

    // Print debug info
    LOG_DEBUG("----------------------------\n");
    LOG_DEBUG("rec_create_pipeline, The parameters are:\n");
    LOG_DEBUG("audio source=%s\n", parms->source);
    LOG_DEBUG("device list is:\n");

    GList *n = g_list_first(parms->dev_list);
    while (n) {
        gchar *device = (gchar*)n->data;
        (void)device; // Avoid unused var message

        LOG_DEBUG("\t%s\n", device);
        n = g_list_next(n);
    }

    LOG_DEBUG("profile from GConf=%s\n",parms->profile_str);
    LOG_DEBUG("filename=%s\n", parms->filename);
    LOG_DEBUG("append to file=%s\n", (parms->append ? "TRUE" : "FALSE"));

    // Create pipeline from the parms
    GstElement *pipeline = pipeline_create(parms, &err_msg);

    // Errors?
    if (!GST_IS_PIPELINE(pipeline) || err_msg) {
        goto LBL_1;
    }

    // Get "filesink" and set location (file name) and append mode
    GstElement *filesink = gst_bin_get_by_name(GST_BIN(pipeline), "filesink");

    if (!GST_IS_ELEMENT(filesink)) {
        err_msg = g_strdup_printf(_("Cannot find audio element %s.\n"), "filesink");
        goto LBL_1;
    }

    g_object_set(G_OBJECT(filesink), "location", parms->filename, NULL);
    g_object_set(G_OBJECT(filesink), "append", parms->append, NULL);
    g_object_unref(filesink);

    // Add a message handler
    GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));

    // gst_bus_add_watch (bus, bus_info_cb, NULL);
    gst_bus_add_signal_watch(bus);

    // Detect state changes
    g_signal_connect(bus, "message::state-changed", G_CALLBACK(rec_state_changed_cb), NULL);

    // Monitor sound level/amplitude
    g_signal_connect(bus, "message::element", G_CALLBACK(rec_level_message_cb), NULL);

    // Catch error messages
    g_signal_connect(bus, "message::error", G_CALLBACK(rec_pipeline_error_cb), NULL);

    // EOS
    g_signal_connect(bus, "message::eos", G_CALLBACK(rec_eos_msg_cb), NULL);

    gst_object_unref(bus);

    // Assume all is OK
    gboolean ret = TRUE;

    // Set the pipeline to "playing" state
    if (gst_element_set_state(pipeline, GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
        err_msg = g_strdup(_("Cannot start reading from the stream/pipeline.\n"));
        ret = FALSE;
    } else {
        LOG_DEBUG("Pipeline is OK. Starting recording to %s.\n", parms->filename);
    }

    // gst_element_get_state(pipeline, NULL, NULL, GST_CLOCK_TIME_NONE);


    // Are we finally ok?
    if (ret) {
        return pipeline;
    }

    // Error occured
LBL_1:
    // Set the error object
    if (!err_msg)
        err_msg = g_strdup_printf(_("Cannot create audio pipeline. %s.\n"), "");

    LOG_ERROR(err_msg);

    // Destroy pipeline
    if (G_IS_OBJECT(pipeline)) {
        gst_element_set_state(pipeline, GST_STATE_NULL);
        gst_object_unref(GST_OBJECT(pipeline));
    }

    // Set error
    g_set_error(error,
                AUDIO_RECORDER,      /* Error domain */
                AUDIO_RECORDER_ERR1, /* Error code */
                err_msg, "");

    g_free(err_msg);

    // Return NULL
    return NULL;
}

gchar *rec_get_output_filename() {
    // Return the current output filename

    // Valid pipeline?
    if (!GST_IS_ELEMENT(g_pipeline)) return NULL;

    // Get the "filesink" element (by name)
    GstElement *filesink = gst_bin_get_by_name(GST_BIN(g_pipeline), "filesink");

    // Got filesink?
    if (!GST_IS_ELEMENT(filesink)) return NULL;

    gchar *filename = NULL;
    g_object_get(G_OBJECT(filesink), "location", &filename, NULL);
    g_object_unref(filesink);

    // The caller should g_free() the value
    return filename;
}

// ------------------------------------------------------------
// Support functions
// ------------------------------------------------------------

static gchar *rec_get_media_profile_id() {
    // Return the saved media profile id. Typical values are: "aac", "mp3", "cdlossless" and "cdlossy".
    // See gconf-editor, key: system -> gstreamer -> 0.10 -> audio -> profiles.
    gchar *id = NULL;
    conf_get_string_value("media-format", &id);

    // Is this id real?
    GMAudioProfile *media_profile = media_profiles_get_profile(id);

    if (!media_profile) {
        g_free(id);
        id = g_strdup("cdlossy"); // = .ogg audio format
    }

    // The caller should g_free() this value
    return id;
}

static gchar *check_audio_folder(gchar *audio_folder) {
    // Create the folder, this may fail.
    g_mkdir(audio_folder, 0755);

    // Directory exists?
    if (!g_file_test(audio_folder, G_FILE_TEST_IS_DIR)) {
        // Replace the old audio_folder
        g_free(audio_folder);

        // Set it to $HOME
        audio_folder = get_home_dir();
    }

    return audio_folder;
}

static gchar *rec_generate_unique_filename() {
    // Generate a new unique filename.

    // filename = file_prefix + pattern + "." + extension.

    // Get the saved media profile id (aac, mp3, cdlossless, cdlossy, etc).
    gchar *media_profile_id = rec_get_media_profile_id();

    // Take the file extension from media_profile_id; .ogg, .mp3, etc.
    gchar *file_ext = media_profiles_get_extension(media_profile_id);

    // Last used filename (if we need to add_to_file)
    gchar *last_track_name = NULL;
    conf_get_string_value("track/last-track-name", &last_track_name);
    str_trim(last_track_name);

    // Audio folder
    gchar *audio_folder = get_audio_folder();

    // Check if audio_folder is writable
    audio_folder = check_audio_folder(audio_folder);

    // Pattern to generate the filename (includes base + date + time pattern)
    gchar *filename_pattern = get_filename_pattern(FALSE);

    gboolean append_to_file = FALSE;
    conf_get_boolean_value("append-to-file", &append_to_file);

    LOG_DEBUG("last_track_name=%s  pattern=%s  append_to_file=%d\n", last_track_name, filename_pattern, append_to_file);

    // Auto generate filename?
    if ((!append_to_file) || str_length(last_track_name, 4000) < 2) {
        // Generate a filename from the pattern
        g_free(last_track_name);
        last_track_name = substitute_time_and_date_pattern(filename_pattern);

        // Invalid pattern?
        if (str_length(last_track_name, 4000) < 1) {
            // Filename not given, set it to "Some filename"
            last_track_name= g_strdup(_("Some filename"));
        }
    }

    // Split the last filename
    gchar *path=NULL, *basename=NULL, *ext=NULL;
    split_filename3(last_track_name, &path, &basename, &ext);

    // Remove old and set real (.ogg, .mp3, etc.) file extension
    gchar *s = g_strdup_printf("%s.%s", basename, file_ext);
    g_free(path);
    g_free(basename);
    g_free(ext);

    // Build the entire filename; path + file.ext
    gchar *fname = g_build_filename(audio_folder, s, NULL);

    // Free the values
    g_free(s);
    g_free(media_profile_id);
    g_free(file_ext);
    g_free(last_track_name);
    g_free(audio_folder);
    g_free(filename_pattern);

    LOG_DEBUG("Auto generated filename is:%s.\n", fname);

    // Return filename. The caller should g_free() this value.
    return fname;
}

static gchar *rec_create_filename(gchar *track, gchar *artist, gchar *album) {
    // Note: we are not using album name here. Maybe in the next version.

    gchar *filename = NULL;

    // Do we have a track name?
    if (str_length(track, 2048) < 2) {
        // No.
        // Auto-generate a new filename from the date+time pattern
        return rec_generate_unique_filename();
    }

    // Get Audio/ folder
    gchar *audio_folder = get_audio_folder();
    audio_folder = check_audio_folder(audio_folder);

    // Get the saved media profile id (aac, mp3, cdlossless, cdlossy, etc).
    gchar *media_profile_id = rec_get_media_profile_id();

    // Take the file extension from media_profile_id; .ogg, .mp3, etc.
    gchar *file_ext = media_profiles_get_extension(media_profile_id);

    // Make track + file extension
    gchar *fname = g_strdup_printf("%s.%s", track, file_ext);
    gchar *path = NULL;

    // Do we have artist name?
    if (str_length(artist, 1024) > 0) {
        // Build path: audio_folder / artist
        // Eg. "/home/moma/Audio/Salomon Burke/" or from Skype "/home/moma/Audio/Skype recordings/"
        path = g_build_filename(audio_folder, artist, NULL);

        // Create directory
        if (g_mkdir_with_parents(path, 0755) == -1) {
            // Ref: http://library.gnome.org/devel/glib/unstable/glib-File-Utilities.html#g-mkdir-with-parents
            LOG_ERROR("Cannot create path \"%s\"\n", path);
            g_free(path);
            path = NULL;
        }
    }

    if (path) {
        // audio_folder / artist + track.ext
        filename = g_build_filename(path, fname, NULL);
    } else {
        // audio_folder + track.ext
        filename = g_build_filename(audio_folder, fname, NULL);
    }

    // TODO: Should we delete existing audio files?
    // Delete existing file
    g_remove(filename);

    // Filename was write-protected?
    if (g_file_test(filename, G_FILE_TEST_IS_REGULAR)) {
        // Yes.
        // Generate new, unique name by using the pattern

        g_free(filename);

        // Pattern to generate a unique filename (date + time pattern)
        gchar *filename_pattern = get_filename_pattern(TRUE/*remove "rec-" from the pattern*/);

        // Re-build the filename
        gchar *t = g_strdup_printf("%s/%s-%s.%s", (path ? path : audio_folder), track, filename_pattern, file_ext);

        // Substitute date+time values
        filename = substitute_time_and_date_pattern(t);

        g_free(t);
        g_free(filename_pattern);
    }

    g_free(fname);
    g_free(path);

    g_free(media_profile_id);
    g_free(file_ext);
    g_free(audio_folder);

    // The caller should g_free() this value
    return filename;
}

/*
void rec_treshold_message(GstClockTime timestamp, gboolean above, gdouble threshold) {
    // Listener module calls this function to notify the recorder about level change.
    // The signal level is below/above the threshold.
 
    g_print("Timestamp=%" GST_TIME_FORMAT "  above=%s  threshold=%3.3f\n", 
                GST_TIME_ARGS(timestamp), (above ? "TRUE" : "FALSE") , threshold);

}
*/