/* GStreamer Audio CD Source Base Class * Copyright (C) 2005 Tim-Philipp Müller * * 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. */ /* TODO: * * - in ::start(), we want to post a tags message with an array or a list * of tagslists of all tracks, so that applications know at least the * number of tracks and all track durations immediately without having * to do any querying. We have to decide what type and name to use for * this array of track taglists. * * - FIX cddb discid calculation algorithm for mixed mode CDs - do we use * offsets and duration of ALL tracks (data + audio) for the CDDB ID * calculation, or only audio tracks? * * - Do we really need properties for the TOC bias/offset stuff? Wouldn't * environment variables make much more sense? Do we need this at all * (does it only affect ancient hardware?) */ /** * SECTION:gstaudiocdsrc * @short_description: Base class for Audio CD sources * * * * Provides a base class for CD digital audio (CDDA) sources, which handles * things like seeking, querying, discid calculation, tags, and buffer * timestamping. * * Using GstAudioCdSrc-based elements in applications * * GstAudioCdSrc registers two #GstFormats of its own, namely * the "track" format and the "sector" format. Applications will usually * only find the "track" format interesting. You can retrieve that #GstFormat * for use in seek events or queries with gst_format_get_by_nick("track"). * * * In order to query the number of tracks, for example, an application would * set the CDDA source element to READY or PAUSED state and then query the * the number of tracks via gst_element_query_duration() using the track * format acquired above. Applications can query the currently playing track * in the same way. * * * Alternatively, applications may retrieve the currently playing track and * the total number of tracks from the taglist that will posted on the bus * whenever the CD is opened or the currently playing track changes. The * taglist will contain GST_TAG_TRACK_NUMBER and GST_TAG_TRACK_COUNT tags. * * * Applications playing back CD audio using playbin and cdda://n URIs should * issue a seek command in track format to change between tracks, rather than * setting a new cdda://n+1 URI on playbin (as setting a new URI on playbin * involves closing and re-opening the CD device, which is much much slower). * * Tags and meta-information * * CDDA sources will automatically emit a number of tags, details about which * can be found in the libgsttag documentation. Those tags are: * #GST_TAG_CDDA_CDDB_DISCID, #GST_TAG_CDDA_CDDB_DISCID_FULL, * #GST_TAG_CDDA_MUSICBRAINZ_DISCID, #GST_TAG_CDDA_MUSICBRAINZ_DISCID_FULL, * among others. * * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include /* for strtol */ #include #include #include "gstaudiocdsrc.h" #include "gst/gst-i18n-plugin.h" GST_DEBUG_CATEGORY_STATIC (gst_audio_cd_src_debug); #define GST_CAT_DEFAULT gst_audio_cd_src_debug #define DEFAULT_DEVICE "/dev/cdrom" #define CD_FRAMESIZE_RAW (2352) #define SECTORS_PER_SECOND (75) #define SECTORS_PER_MINUTE (75*60) #define SAMPLES_PER_SECTOR (CD_FRAMESIZE_RAW >> 2) #define TIME_INTERVAL_FROM_SECTORS(sectors) ((SAMPLES_PER_SECTOR * sectors * GST_SECOND) / 44100) #define SECTORS_FROM_TIME_INTERVAL(dtime) (dtime * 44100 / (SAMPLES_PER_SECTOR * GST_SECOND)) enum { ARG_0, ARG_MODE, ARG_DEVICE, ARG_TRACK, ARG_TOC_OFFSET, ARG_TOC_BIAS }; static void gst_audio_cd_src_uri_handler_init (gpointer g_iface, gpointer iface_data); static void gst_audio_cd_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static void gst_audio_cd_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_audio_cd_src_finalize (GObject * obj); static gboolean gst_audio_cd_src_query (GstBaseSrc * src, GstQuery * query); static gboolean gst_audio_cd_src_handle_event (GstBaseSrc * basesrc, GstEvent * event); static gboolean gst_audio_cd_src_do_seek (GstBaseSrc * basesrc, GstSegment * segment); static gboolean gst_audio_cd_src_start (GstBaseSrc * basesrc); static gboolean gst_audio_cd_src_stop (GstBaseSrc * basesrc); static GstFlowReturn gst_audio_cd_src_create (GstPushSrc * pushsrc, GstBuffer ** buf); static gboolean gst_audio_cd_src_is_seekable (GstBaseSrc * basesrc); static void gst_audio_cd_src_update_duration (GstAudioCdSrc * src); static void gst_audio_cd_src_set_index (GstElement * src, GstIndex * index); static GstIndex *gst_audio_cd_src_get_index (GstElement * src); #define gst_audio_cd_src_parent_class parent_class G_DEFINE_TYPE_WITH_CODE (GstAudioCdSrc, gst_audio_cd_src, GST_TYPE_PUSH_SRC, G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_audio_cd_src_uri_handler_init)); #define SRC_CAPS \ "audio/x-raw, " \ "format = (string) " GST_AUDIO_NE(S16) ", " \ "rate = (int) 44100, " \ "channels = (int) 2" \ static GstStaticPadTemplate gst_audio_cd_src_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (SRC_CAPS) ); /* our two formats */ static GstFormat track_format; static GstFormat sector_format; GType gst_audio_cd_src_mode_get_type (void) { static GType mode_type; /* 0 */ static const GEnumValue modes[] = { {GST_AUDIO_CD_SRC_MODE_NORMAL, "Stream consists of a single track", "normal"}, {GST_AUDIO_CD_SRC_MODE_CONTINUOUS, "Stream consists of the whole disc", "continuous"}, {0, NULL, NULL} }; if (mode_type == 0) mode_type = g_enum_register_static ("GstAudioCdSrcMode", modes); return mode_type; } static void gst_audio_cd_src_class_init (GstAudioCdSrcClass * klass) { GstElementClass *element_class; GstPushSrcClass *pushsrc_class; GstBaseSrcClass *basesrc_class; GObjectClass *gobject_class; gobject_class = (GObjectClass *) klass; element_class = (GstElementClass *) klass; basesrc_class = (GstBaseSrcClass *) klass; pushsrc_class = (GstPushSrcClass *) klass; GST_DEBUG_CATEGORY_INIT (gst_audio_cd_src_debug, "audiocdsrc", 0, "Audio CD source base class"); /* our very own formats */ track_format = gst_format_register ("track", "CD track"); sector_format = gst_format_register ("sector", "CD sector"); /* register CDDA tags */ gst_tag_register_musicbrainz_tags (); #if 0 ///// FIXME: what type to use here? /////// gst_tag_register (GST_TAG_CDDA_TRACK_TAGS, GST_TAG_FLAG_META, GST_TYPE_TAG_LIST, "track-tags", "CDDA taglist for one track", gst_tag_merge_use_first); ///////////// FIXME: right function??? /////// #endif gobject_class->set_property = gst_audio_cd_src_set_property; gobject_class->get_property = gst_audio_cd_src_get_property; gobject_class->finalize = gst_audio_cd_src_finalize; g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_DEVICE, g_param_spec_string ("device", "Device", "CD device location", NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MODE, g_param_spec_enum ("mode", "Mode", "Mode", GST_TYPE_AUDIO_CD_SRC_MODE, GST_AUDIO_CD_SRC_MODE_NORMAL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_TRACK, g_param_spec_uint ("track", "Track", "Track", 1, 99, 1, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); #if 0 /* Do we really need this toc adjustment stuff as properties? does the user * have a chance to set it in practice, e.g. when using sound-juicer, rb, * totem, whatever? Shouldn't we rather use environment variables * for this? (tpm) */ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_TOC_OFFSET, g_param_spec_int ("toc-offset", "Table of contents offset", "Add sectors to the values reported", G_MININT, G_MAXINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_TOC_BIAS, g_param_spec_boolean ("toc-bias", "Table of contents bias", "Assume that the beginning offset of track 1 as reported in the TOC " "will be addressed as LBA 0. Necessary for some Toshiba drives to " "get track boundaries", FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); #endif gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_audio_cd_src_src_template)); element_class->set_index = GST_DEBUG_FUNCPTR (gst_audio_cd_src_set_index); element_class->get_index = GST_DEBUG_FUNCPTR (gst_audio_cd_src_get_index); basesrc_class->start = GST_DEBUG_FUNCPTR (gst_audio_cd_src_start); basesrc_class->stop = GST_DEBUG_FUNCPTR (gst_audio_cd_src_stop); basesrc_class->query = GST_DEBUG_FUNCPTR (gst_audio_cd_src_query); basesrc_class->event = GST_DEBUG_FUNCPTR (gst_audio_cd_src_handle_event); basesrc_class->do_seek = GST_DEBUG_FUNCPTR (gst_audio_cd_src_do_seek); basesrc_class->is_seekable = GST_DEBUG_FUNCPTR (gst_audio_cd_src_is_seekable); pushsrc_class->create = GST_DEBUG_FUNCPTR (gst_audio_cd_src_create); } static void gst_audio_cd_src_init (GstAudioCdSrc * src) { /* we're not live and we operate in time */ gst_base_src_set_format (GST_BASE_SRC (src), GST_FORMAT_TIME); gst_base_src_set_live (GST_BASE_SRC (src), FALSE); GST_OBJECT_FLAG_SET (src, GST_ELEMENT_FLAG_INDEXABLE); src->device = NULL; src->mode = GST_AUDIO_CD_SRC_MODE_NORMAL; src->uri_track = -1; } static void gst_audio_cd_src_finalize (GObject * obj) { GstAudioCdSrc *cddasrc = GST_AUDIO_CD_SRC (obj); g_free (cddasrc->uri); g_free (cddasrc->device); if (cddasrc->index) gst_object_unref (cddasrc->index); G_OBJECT_CLASS (parent_class)->finalize (obj); } static void gst_audio_cd_src_set_device (GstAudioCdSrc * src, const gchar * device) { if (src->device) g_free (src->device); src->device = NULL; if (!device) return; /* skip multiple slashes */ while (*device == '/' && *(device + 1) == '/') device++; #ifdef __sun /* * On Solaris, /dev/rdsk is used for accessing the CD device, but some * applications pass in /dev/dsk, so correct. */ if (strncmp (device, "/dev/dsk", 8) == 0) { gchar *rdsk_value; rdsk_value = g_strdup_printf ("/dev/rdsk%s", device + 8); src->device = g_strdup (rdsk_value); g_free (rdsk_value); } else { #endif src->device = g_strdup (device); #ifdef __sun } #endif } static void gst_audio_cd_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudioCdSrc *src = GST_AUDIO_CD_SRC (object); GST_OBJECT_LOCK (src); switch (prop_id) { case ARG_MODE:{ src->mode = g_value_get_enum (value); break; } case ARG_DEVICE:{ const gchar *dev = g_value_get_string (value); gst_audio_cd_src_set_device (src, dev); break; } case ARG_TRACK:{ guint track = g_value_get_uint (value); if (src->num_tracks > 0 && track > src->num_tracks) { g_warning ("Invalid track %u", track); } else if (track > 0 && src->tracks != NULL) { src->cur_sector = src->tracks[track - 1].start; src->uri_track = track; } else { src->uri_track = track; /* seek will be done in start() */ } break; } case ARG_TOC_OFFSET:{ src->toc_offset = g_value_get_int (value); break; } case ARG_TOC_BIAS:{ src->toc_bias = g_value_get_boolean (value); break; } default:{ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } GST_OBJECT_UNLOCK (src); } static void gst_audio_cd_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAudioCdSrcClass *klass = GST_AUDIO_CD_SRC_GET_CLASS (object); GstAudioCdSrc *src = GST_AUDIO_CD_SRC (object); GST_OBJECT_LOCK (src); switch (prop_id) { case ARG_MODE: g_value_set_enum (value, src->mode); break; case ARG_DEVICE:{ if (src->device == NULL && klass->get_default_device != NULL) { gchar *d = klass->get_default_device (src); if (d != NULL) { g_value_set_string (value, DEFAULT_DEVICE); g_free (d); break; } } if (src->device == NULL) g_value_set_string (value, DEFAULT_DEVICE); else g_value_set_string (value, src->device); break; } case ARG_TRACK:{ if (src->num_tracks <= 0 && src->uri_track > 0) { g_value_set_uint (value, src->uri_track); } else { g_value_set_uint (value, src->cur_track + 1); } break; } case ARG_TOC_OFFSET: g_value_set_int (value, src->toc_offset); break; case ARG_TOC_BIAS: g_value_set_boolean (value, src->toc_bias); break; default:{ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } GST_OBJECT_UNLOCK (src); } static gint gst_audio_cd_src_get_track_from_sector (GstAudioCdSrc * src, gint sector) { gint i; for (i = 0; i < src->num_tracks; ++i) { if (sector >= src->tracks[i].start && sector <= src->tracks[i].end) return i; } return -1; } static gboolean gst_audio_cd_src_convert (GstAudioCdSrc * src, GstFormat src_format, gint64 src_val, GstFormat dest_format, gint64 * dest_val) { gboolean started; GST_LOG_OBJECT (src, "converting value %" G_GINT64_FORMAT " from %s into %s", src_val, gst_format_get_name (src_format), gst_format_get_name (dest_format)); if (src_format == dest_format) { *dest_val = src_val; return TRUE; } started = GST_OBJECT_FLAG_IS_SET (GST_BASE_SRC (src), GST_BASE_SRC_STARTED); if (src_format == track_format) { if (!started) goto not_started; if (src_val < 0 || src_val >= src->num_tracks) { GST_DEBUG_OBJECT (src, "track number %d out of bounds", (gint) src_val); goto wrong_value; } src_format = GST_FORMAT_DEFAULT; src_val = src->tracks[src_val].start * SAMPLES_PER_SECTOR; } else if (src_format == sector_format) { src_format = GST_FORMAT_DEFAULT; src_val = src_val * SAMPLES_PER_SECTOR; } if (src_format == dest_format) { *dest_val = src_val; goto done; } switch (src_format) { case GST_FORMAT_BYTES: /* convert to samples (4 bytes per sample) */ src_val = src_val >> 2; /* fallthrough */ case GST_FORMAT_DEFAULT:{ switch (dest_format) { case GST_FORMAT_BYTES:{ if (src_val < 0) { GST_DEBUG_OBJECT (src, "sample source value negative"); goto wrong_value; } *dest_val = src_val << 2; /* 4 bytes per sample */ break; } case GST_FORMAT_TIME:{ *dest_val = gst_util_uint64_scale_int (src_val, GST_SECOND, 44100); break; } default:{ gint64 sector = src_val / SAMPLES_PER_SECTOR; if (dest_format == sector_format) { *dest_val = sector; } else if (dest_format == track_format) { if (!started) goto not_started; *dest_val = gst_audio_cd_src_get_track_from_sector (src, sector); } else { goto unknown_format; } break; } } break; } case GST_FORMAT_TIME:{ gint64 sample_offset; if (src_val == GST_CLOCK_TIME_NONE) { GST_DEBUG_OBJECT (src, "source time value invalid"); goto wrong_value; } sample_offset = gst_util_uint64_scale_int (src_val, 44100, GST_SECOND); switch (dest_format) { case GST_FORMAT_BYTES:{ *dest_val = sample_offset << 2; /* 4 bytes per sample */ break; } case GST_FORMAT_DEFAULT:{ *dest_val = sample_offset; break; } default:{ gint64 sector = sample_offset / SAMPLES_PER_SECTOR; if (dest_format == sector_format) { *dest_val = sector; } else if (dest_format == track_format) { if (!started) goto not_started; *dest_val = gst_audio_cd_src_get_track_from_sector (src, sector); } else { goto unknown_format; } break; } } break; } default:{ goto unknown_format; } } done: { GST_LOG_OBJECT (src, "returning %" G_GINT64_FORMAT, *dest_val); return TRUE; } unknown_format: { GST_DEBUG_OBJECT (src, "conversion failed: %s", "unsupported format"); return FALSE; } wrong_value: { GST_DEBUG_OBJECT (src, "conversion failed: %s", "source value not within allowed range"); return FALSE; } not_started: { GST_DEBUG_OBJECT (src, "conversion failed: %s", "cannot do this conversion, device not open"); return FALSE; } } static gboolean gst_audio_cd_src_query (GstBaseSrc * basesrc, GstQuery * query) { GstAudioCdSrc *src = GST_AUDIO_CD_SRC (basesrc); gboolean started; started = GST_OBJECT_FLAG_IS_SET (basesrc, GST_BASE_SRC_STARTED); GST_LOG_OBJECT (src, "handling %s query", gst_query_type_get_name (GST_QUERY_TYPE (query))); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_DURATION:{ GstFormat dest_format; gint64 dest_val; guint sectors; gst_query_parse_duration (query, &dest_format, NULL); if (!started) return FALSE; g_assert (src->tracks != NULL); if (dest_format == track_format) { GST_LOG_OBJECT (src, "duration: %d tracks", src->num_tracks); gst_query_set_duration (query, track_format, src->num_tracks); return TRUE; } if (src->cur_track < 0 || src->cur_track >= src->num_tracks) return FALSE; if (src->mode == GST_AUDIO_CD_SRC_MODE_NORMAL) { sectors = src->tracks[src->cur_track].end - src->tracks[src->cur_track].start + 1; } else { sectors = src->tracks[src->num_tracks - 1].end - src->tracks[0].start + 1; } /* ... and convert into final format */ if (!gst_audio_cd_src_convert (src, sector_format, sectors, dest_format, &dest_val)) { return FALSE; } gst_query_set_duration (query, dest_format, dest_val); GST_LOG ("duration: %u sectors, %" G_GINT64_FORMAT " in format %s", sectors, dest_val, gst_format_get_name (dest_format)); break; } case GST_QUERY_POSITION:{ GstFormat dest_format; gint64 pos_sector; gint64 dest_val; gst_query_parse_position (query, &dest_format, NULL); if (!started) return FALSE; g_assert (src->tracks != NULL); if (dest_format == track_format) { GST_LOG_OBJECT (src, "position: track %d", src->cur_track); gst_query_set_position (query, track_format, src->cur_track); return TRUE; } if (src->cur_track < 0 || src->cur_track >= src->num_tracks) return FALSE; if (src->mode == GST_AUDIO_CD_SRC_MODE_NORMAL) { pos_sector = src->cur_sector - src->tracks[src->cur_track].start; } else { pos_sector = src->cur_sector - src->tracks[0].start; } if (!gst_audio_cd_src_convert (src, sector_format, pos_sector, dest_format, &dest_val)) { return FALSE; } gst_query_set_position (query, dest_format, dest_val); GST_LOG ("position: sector %u, %" G_GINT64_FORMAT " in format %s", (guint) pos_sector, dest_val, gst_format_get_name (dest_format)); break; } case GST_QUERY_CONVERT:{ GstFormat src_format, dest_format; gint64 src_val, dest_val; gst_query_parse_convert (query, &src_format, &src_val, &dest_format, NULL); if (!gst_audio_cd_src_convert (src, src_format, src_val, dest_format, &dest_val)) { return FALSE; } gst_query_set_convert (query, src_format, src_val, dest_format, dest_val); break; } default:{ GST_DEBUG_OBJECT (src, "unhandled query, chaining up to parent class"); return GST_BASE_SRC_CLASS (parent_class)->query (basesrc, query); } } return TRUE; } static gboolean gst_audio_cd_src_is_seekable (GstBaseSrc * basesrc) { return TRUE; } static gboolean gst_audio_cd_src_do_seek (GstBaseSrc * basesrc, GstSegment * segment) { GstAudioCdSrc *src = GST_AUDIO_CD_SRC (basesrc); gint64 seek_sector; GST_DEBUG_OBJECT (src, "segment %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT, GST_TIME_ARGS (segment->start), GST_TIME_ARGS (segment->stop)); if (!gst_audio_cd_src_convert (src, GST_FORMAT_TIME, segment->start, sector_format, &seek_sector)) { GST_WARNING_OBJECT (src, "conversion failed"); return FALSE; } /* we should only really be called when open */ g_assert (src->cur_track >= 0 && src->cur_track < src->num_tracks); switch (src->mode) { case GST_AUDIO_CD_SRC_MODE_NORMAL: seek_sector += src->tracks[src->cur_track].start; break; case GST_AUDIO_CD_SRC_MODE_CONTINUOUS: seek_sector += src->tracks[0].start; break; default: g_return_val_if_reached (FALSE); } src->cur_sector = (gint) seek_sector; GST_DEBUG_OBJECT (src, "seek'd to sector %d", src->cur_sector); return TRUE; } static gboolean gst_audio_cd_src_handle_track_seek (GstAudioCdSrc * src, gdouble rate, GstSeekFlags flags, GstSeekType start_type, gint64 start, GstSeekType stop_type, gint64 stop) { GstBaseSrc *basesrc = GST_BASE_SRC (src); GstEvent *event; if ((flags & GST_SEEK_FLAG_SEGMENT) == GST_SEEK_FLAG_SEGMENT) { gint64 start_time = -1; gint64 stop_time = -1; if (src->mode != GST_AUDIO_CD_SRC_MODE_CONTINUOUS) { GST_DEBUG_OBJECT (src, "segment seek in track format is only " "supported in CONTINUOUS mode, not in mode %d", src->mode); return FALSE; } switch (start_type) { case GST_SEEK_TYPE_SET: if (!gst_audio_cd_src_convert (src, track_format, start, GST_FORMAT_TIME, &start_time)) { GST_DEBUG_OBJECT (src, "cannot convert track %d to time", (gint) start); return FALSE; } break; case GST_SEEK_TYPE_END: if (!gst_audio_cd_src_convert (src, track_format, src->num_tracks - start - 1, GST_FORMAT_TIME, &start_time)) { GST_DEBUG_OBJECT (src, "cannot convert track %d to time", (gint) start); return FALSE; } start_type = GST_SEEK_TYPE_SET; break; case GST_SEEK_TYPE_NONE: start_time = -1; break; default: g_return_val_if_reached (FALSE); } switch (stop_type) { case GST_SEEK_TYPE_SET: if (!gst_audio_cd_src_convert (src, track_format, stop, GST_FORMAT_TIME, &stop_time)) { GST_DEBUG_OBJECT (src, "cannot convert track %d to time", (gint) stop); return FALSE; } break; case GST_SEEK_TYPE_END: if (!gst_audio_cd_src_convert (src, track_format, src->num_tracks - stop - 1, GST_FORMAT_TIME, &stop_time)) { GST_DEBUG_OBJECT (src, "cannot convert track %d to time", (gint) stop); return FALSE; } stop_type = GST_SEEK_TYPE_SET; break; case GST_SEEK_TYPE_NONE: stop_time = -1; break; default: g_return_val_if_reached (FALSE); } GST_LOG_OBJECT (src, "seek segment %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT, GST_TIME_ARGS (start_time), GST_TIME_ARGS (stop_time)); /* send fake segment seek event in TIME format to * base class, which will hopefully handle the rest */ event = gst_event_new_seek (rate, GST_FORMAT_TIME, flags, start_type, start_time, stop_type, stop_time); return GST_BASE_SRC_CLASS (parent_class)->event (basesrc, event); } /* not a segment seek */ if (start_type == GST_SEEK_TYPE_NONE) { GST_LOG_OBJECT (src, "start seek type is NONE, nothing to do"); return TRUE; } if (stop_type != GST_SEEK_TYPE_NONE) { GST_WARNING_OBJECT (src, "ignoring stop seek type (expected NONE)"); } if (start < 0 || start >= src->num_tracks) { GST_DEBUG_OBJECT (src, "invalid track %" G_GINT64_FORMAT, start); return FALSE; } GST_DEBUG_OBJECT (src, "seeking to track %" G_GINT64_FORMAT, start + 1); src->cur_sector = src->tracks[start].start; GST_DEBUG_OBJECT (src, "starting at sector %d", src->cur_sector); if (src->cur_track != start) { src->cur_track = (gint) start; src->uri_track = -1; src->prev_track = -1; gst_audio_cd_src_update_duration (src); } else { GST_DEBUG_OBJECT (src, "is current track, just seeking back to start"); } /* send fake segment seek event in TIME format to * base class (so we get a newsegment etc.) */ event = gst_event_new_seek (rate, GST_FORMAT_TIME, flags, GST_SEEK_TYPE_SET, 0, GST_SEEK_TYPE_NONE, -1); return GST_BASE_SRC_CLASS (parent_class)->event (basesrc, event); } static gboolean gst_audio_cd_src_handle_event (GstBaseSrc * basesrc, GstEvent * event) { GstAudioCdSrc *src = GST_AUDIO_CD_SRC (basesrc); gboolean ret = FALSE; GST_LOG_OBJECT (src, "handling %s event", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_SEEK:{ GstSeekType start_type, stop_type; GstSeekFlags flags; GstFormat format; gdouble rate; gint64 start, stop; if (!GST_OBJECT_FLAG_IS_SET (basesrc, GST_BASE_SRC_STARTED)) { GST_DEBUG_OBJECT (src, "seek failed: device not open"); break; } gst_event_parse_seek (event, &rate, &format, &flags, &start_type, &start, &stop_type, &stop); if (format == sector_format) { GST_DEBUG_OBJECT (src, "seek in sector format not supported"); break; } if (format == track_format) { ret = gst_audio_cd_src_handle_track_seek (src, rate, flags, start_type, start, stop_type, stop); } else { GST_LOG_OBJECT (src, "let base class handle seek in %s format", gst_format_get_name (format)); event = gst_event_ref (event); ret = GST_BASE_SRC_CLASS (parent_class)->event (basesrc, event); } break; } default:{ GST_LOG_OBJECT (src, "let base class handle event"); ret = GST_BASE_SRC_CLASS (parent_class)->event (basesrc, event); break; } } return ret; } static GstURIType gst_audio_cd_src_uri_get_type (GType type) { return GST_URI_SRC; } static const gchar *const * gst_audio_cd_src_uri_get_protocols (GType type) { static const gchar *protocols[] = { "cdda", NULL }; return protocols; } static gchar * gst_audio_cd_src_uri_get_uri (GstURIHandler * handler) { GstAudioCdSrc *src = GST_AUDIO_CD_SRC (handler); GST_OBJECT_LOCK (src); /* FIXME: can we get rid of all that here and just return a copy of the * existing URI perhaps? */ g_free (src->uri); if (GST_OBJECT_FLAG_IS_SET (GST_BASE_SRC (src), GST_BASE_SRC_STARTED)) { src->uri = g_strdup_printf ("cdda://%s#%d", src->device, (src->uri_track > 0) ? src->uri_track : 1); } else { src->uri = g_strdup ("cdda://1"); } GST_OBJECT_UNLOCK (src); return g_strdup (src->uri); } /* Note: gst_element_make_from_uri() might call us with just 'cdda://' as * URI and expects us to return TRUE then (and this might be in any state) */ /* We accept URIs of the format cdda://(device#track)|(track) */ static gboolean gst_audio_cd_src_uri_set_uri (GstURIHandler * handler, const gchar * uri, GError ** error) { GstAudioCdSrc *src = GST_AUDIO_CD_SRC (handler); const gchar *location; gchar *track_number; GST_OBJECT_LOCK (src); location = uri + 7; track_number = g_strrstr (location, "#"); src->uri_track = 0; /* FIXME 0.11: ignore URI fragments that look like device paths for * the benefit of rhythmbox and possibly other applications. */ if (track_number && track_number[1] != '/') { gchar *device, *nuri = g_strdup (uri); track_number = nuri + (track_number - uri); *track_number = '\0'; device = gst_uri_get_location (nuri); gst_audio_cd_src_set_device (src, device); g_free (device); src->uri_track = strtol (track_number + 1, NULL, 10); g_free (nuri); } else { if (*location == '\0') src->uri_track = 1; else src->uri_track = strtol (location, NULL, 10); } if (src->uri_track < 1) goto failed; if (src->num_tracks > 0 && src->tracks != NULL && src->uri_track > src->num_tracks) goto failed; if (src->uri_track > 0 && src->tracks != NULL) { GST_OBJECT_UNLOCK (src); gst_pad_send_event (GST_BASE_SRC_PAD (src), gst_event_new_seek (1.0, track_format, GST_SEEK_FLAG_FLUSH, GST_SEEK_TYPE_SET, src->uri_track - 1, GST_SEEK_TYPE_NONE, -1)); } else { /* seek will be done in start() */ GST_OBJECT_UNLOCK (src); } GST_LOG_OBJECT (handler, "successfully handled uri '%s'", uri); return TRUE; failed: { GST_OBJECT_UNLOCK (src); GST_DEBUG_OBJECT (src, "cannot handle URI '%s'", uri); g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI, "Could not handle CDDA URI"); return FALSE; } } static void gst_audio_cd_src_uri_handler_init (gpointer g_iface, gpointer iface_data) { GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface; iface->get_type = gst_audio_cd_src_uri_get_type; iface->get_uri = gst_audio_cd_src_uri_get_uri; iface->set_uri = gst_audio_cd_src_uri_set_uri; iface->get_protocols = gst_audio_cd_src_uri_get_protocols; } /** * gst_audio_cd_src_add_track: * @src: a #GstAudioCdSrc * @track: address of #GstAudioCdSrcTrack to add * * CDDA sources use this function from their start vfunc to announce the * available data and audio tracks to the base source class. The caller * should allocate @track on the stack, the base source will do a shallow * copy of the structure (and take ownership of the taglist if there is one). * * Returns: FALSE on error, otherwise TRUE. */ gboolean gst_audio_cd_src_add_track (GstAudioCdSrc * src, GstAudioCdSrcTrack * track) { g_return_val_if_fail (GST_IS_AUDIO_CD_SRC (src), FALSE); g_return_val_if_fail (track != NULL, FALSE); g_return_val_if_fail (track->num > 0, FALSE); GST_DEBUG_OBJECT (src, "adding track %2u (%2u) [%6u-%6u] [%5s], tags: %" GST_PTR_FORMAT, src->num_tracks + 1, track->num, track->start, track->end, (track->is_audio) ? "AUDIO" : "DATA ", track->tags); if (src->num_tracks > 0) { guint end_of_previous_track = src->tracks[src->num_tracks - 1].end; if (track->start <= end_of_previous_track) { GST_WARNING ("track %2u overlaps with previous tracks", track->num); return FALSE; } } GST_OBJECT_LOCK (src); ++src->num_tracks; src->tracks = g_renew (GstAudioCdSrcTrack, src->tracks, src->num_tracks); src->tracks[src->num_tracks - 1] = *track; GST_OBJECT_UNLOCK (src); return TRUE; } static void gst_audio_cd_src_update_duration (GstAudioCdSrc * src) { GstBaseSrc *basesrc; gint64 dur; basesrc = GST_BASE_SRC (src); if (!gst_pad_query_duration (GST_BASE_SRC_PAD (src), GST_FORMAT_TIME, &dur)) { dur = GST_CLOCK_TIME_NONE; } basesrc->segment.duration = dur; gst_element_post_message (GST_ELEMENT (src), gst_message_new_duration (GST_OBJECT (src), GST_FORMAT_TIME, -1)); GST_LOG_OBJECT (src, "duration updated to %" GST_TIME_FORMAT, GST_TIME_ARGS (dur)); } #define CD_MSF_OFFSET 150 /* the cddb hash function */ static guint cddb_sum (gint n) { guint ret; ret = 0; while (n > 0) { ret += (n % 10); n /= 10; } return ret; } static void gst_audio_cd_src_calculate_musicbrainz_discid (GstAudioCdSrc * src) { GString *s; GChecksum *sha; guchar digest[20]; gchar *ptr; gchar tmp[9]; gulong i; guint leadout_sector; gsize digest_len; s = g_string_new (NULL); leadout_sector = src->tracks[src->num_tracks - 1].end + 1 + CD_MSF_OFFSET; /* generate SHA digest */ sha = g_checksum_new (G_CHECKSUM_SHA1); g_snprintf (tmp, sizeof (tmp), "%02X", src->tracks[0].num); g_string_append_printf (s, "%02X", src->tracks[0].num); g_checksum_update (sha, (guchar *) tmp, 2); g_snprintf (tmp, sizeof (tmp), "%02X", src->tracks[src->num_tracks - 1].num); g_string_append_printf (s, " %02X", src->tracks[src->num_tracks - 1].num); g_checksum_update (sha, (guchar *) tmp, 2); g_snprintf (tmp, sizeof (tmp), "%08X", leadout_sector); g_string_append_printf (s, " %08X", leadout_sector); g_checksum_update (sha, (guchar *) tmp, 8); for (i = 0; i < 99; i++) { if (i < src->num_tracks) { guint frame_offset = src->tracks[i].start + CD_MSF_OFFSET; g_snprintf (tmp, sizeof (tmp), "%08X", frame_offset); g_string_append_printf (s, " %08X", frame_offset); g_checksum_update (sha, (guchar *) tmp, 8); } else { g_checksum_update (sha, (guchar *) "00000000", 8); } } digest_len = 20; g_checksum_get_digest (sha, (guint8 *) & digest, &digest_len); /* re-encode to base64 */ ptr = g_base64_encode (digest, digest_len); g_checksum_free (sha); i = strlen (ptr); g_assert (i < sizeof (src->mb_discid) + 1); memcpy (src->mb_discid, ptr, i); src->mb_discid[i] = '\0'; free (ptr); /* Replace '/', '+' and '=' by '_', '.' and '-' as specified on * http://musicbrainz.org/doc/DiscIDCalculation */ for (ptr = src->mb_discid; *ptr != '\0'; ptr++) { if (*ptr == '/') *ptr = '_'; else if (*ptr == '+') *ptr = '.'; else if (*ptr == '=') *ptr = '-'; } GST_DEBUG_OBJECT (src, "musicbrainz-discid = %s", src->mb_discid); GST_DEBUG_OBJECT (src, "musicbrainz-discid-full = %s", s->str); gst_tag_list_add (src->tags, GST_TAG_MERGE_REPLACE, GST_TAG_CDDA_MUSICBRAINZ_DISCID, src->mb_discid, GST_TAG_CDDA_MUSICBRAINZ_DISCID_FULL, s->str, NULL); g_string_free (s, TRUE); } static void lba_to_msf (guint sector, guint * p_m, guint * p_s, guint * p_f, guint * p_secs) { guint m, s, f; m = sector / SECTORS_PER_MINUTE; sector = sector % SECTORS_PER_MINUTE; s = sector / SECTORS_PER_SECOND; f = sector % SECTORS_PER_SECOND; if (p_m) *p_m = m; if (p_s) *p_s = s; if (p_f) *p_f = f; if (p_secs) *p_secs = s + (m * 60); } static void gst_audio_cd_src_calculate_cddb_id (GstAudioCdSrc * src) { GString *s; guint first_sector = 0, last_sector = 0; guint start_secs, end_secs, secs, len_secs; guint total_secs, num_audio_tracks; guint id, t, i; id = 0; total_secs = 0; num_audio_tracks = 0; /* FIXME: do we use offsets and duration of ALL tracks (data + audio) * for the CDDB ID calculation, or only audio tracks? */ for (i = 0; i < src->num_tracks; ++i) { if (1) { /* src->tracks[i].is_audio) { */ if (num_audio_tracks == 0) { first_sector = src->tracks[i].start + CD_MSF_OFFSET; } last_sector = src->tracks[i].end + CD_MSF_OFFSET + 1; ++num_audio_tracks; lba_to_msf (src->tracks[i].start + CD_MSF_OFFSET, NULL, NULL, NULL, &secs); len_secs = (src->tracks[i].end - src->tracks[i].start + 1) / 75; GST_DEBUG_OBJECT (src, "track %02u: lsn %6u (%02u:%02u), " "length: %u seconds (%02u:%02u)", num_audio_tracks, src->tracks[i].start + CD_MSF_OFFSET, secs / 60, secs % 60, len_secs, len_secs / 60, len_secs % 60); id += cddb_sum (secs); total_secs += len_secs; } } /* first_sector = src->tracks[0].start + CD_MSF_OFFSET; */ lba_to_msf (first_sector, NULL, NULL, NULL, &start_secs); /* last_sector = src->tracks[src->num_tracks-1].end + CD_MSF_OFFSET; */ lba_to_msf (last_sector, NULL, NULL, NULL, &end_secs); GST_DEBUG_OBJECT (src, "first_sector = %u = %u secs (%02u:%02u)", first_sector, start_secs, start_secs / 60, start_secs % 60); GST_DEBUG_OBJECT (src, "last_sector = %u = %u secs (%02u:%02u)", last_sector, end_secs, end_secs / 60, end_secs % 60); t = end_secs - start_secs; GST_DEBUG_OBJECT (src, "total length = %u secs (%02u:%02u), added title " "lengths = %u seconds (%02u:%02u)", t, t / 60, t % 60, total_secs, total_secs / 60, total_secs % 60); src->discid = ((id % 0xff) << 24 | t << 8 | num_audio_tracks); s = g_string_new (NULL); g_string_append_printf (s, "%08x", src->discid); gst_tag_list_add (src->tags, GST_TAG_MERGE_REPLACE, GST_TAG_CDDA_CDDB_DISCID, s->str, NULL); g_string_append_printf (s, " %u", src->num_tracks); for (i = 0; i < src->num_tracks; ++i) { g_string_append_printf (s, " %u", src->tracks[i].start + CD_MSF_OFFSET); } g_string_append_printf (s, " %u", t); gst_tag_list_add (src->tags, GST_TAG_MERGE_REPLACE, GST_TAG_CDDA_CDDB_DISCID_FULL, s->str, NULL); GST_DEBUG_OBJECT (src, "cddb discid = %s", s->str); g_string_free (s, TRUE); } static void gst_audio_cd_src_add_tags (GstAudioCdSrc * src) { gint i; /* fill in details for each track */ for (i = 0; i < src->num_tracks; ++i) { gint64 duration; guint num_sectors; if (src->tracks[i].tags == NULL) src->tracks[i].tags = gst_tag_list_new_empty (); num_sectors = src->tracks[i].end - src->tracks[i].start + 1; gst_audio_cd_src_convert (src, sector_format, num_sectors, GST_FORMAT_TIME, &duration); gst_tag_list_add (src->tracks[i].tags, GST_TAG_MERGE_REPLACE, GST_TAG_TRACK_NUMBER, i + 1, GST_TAG_TRACK_COUNT, src->num_tracks, GST_TAG_DURATION, duration, NULL); } /* now fill in per-album tags and include each track's tags * in the album tags, so that interested parties can retrieve * the relevant details for each track in one go */ /* /////////////////////////////// FIXME should we rather insert num_tracks * tags by the name of 'track-tags' and have the caller use * gst_tag_list_get_value_index() rather than use tag names incl. * the track number ?? *//////////////////////////////////////// gst_tag_list_add (src->tags, GST_TAG_MERGE_REPLACE, GST_TAG_TRACK_COUNT, src->num_tracks, NULL); #if 0 for (i = 0; i < src->num_tracks; ++i) { gst_tag_list_add (src->tags, GST_TAG_MERGE_APPEND, GST_TAG_CDDA_TRACK_TAGS, src->tracks[i].tags, NULL); } #endif GST_DEBUG ("src->tags = %" GST_PTR_FORMAT, src->tags); } static void gst_audio_cd_src_add_index_associations (GstAudioCdSrc * src) { gint i; for (i = 0; i < src->num_tracks; i++) { gint64 sector; sector = src->tracks[i].start; gst_index_add_association (src->index, src->index_id, GST_ASSOCIATION_FLAG_KEY_UNIT, track_format, i, /* here we count from 0 */ sector_format, sector, GST_FORMAT_TIME, (gint64) (((CD_FRAMESIZE_RAW >> 2) * sector * GST_SECOND) / 44100), GST_FORMAT_BYTES, (gint64) (sector << 2), GST_FORMAT_DEFAULT, (gint64) ((CD_FRAMESIZE_RAW >> 2) * sector), NULL); } } static void gst_audio_cd_src_set_index (GstElement * element, GstIndex * index) { GstAudioCdSrc *src = GST_AUDIO_CD_SRC (element); GstIndex *old; GST_OBJECT_LOCK (element); old = src->index; if (old == index) { GST_OBJECT_UNLOCK (element); return; } if (index) gst_object_ref (index); src->index = index; GST_OBJECT_UNLOCK (element); if (old) gst_object_unref (old); if (index) { gst_index_get_writer_id (index, GST_OBJECT (src), &src->index_id); gst_index_add_format (index, src->index_id, track_format); gst_index_add_format (index, src->index_id, sector_format); } } static GstIndex * gst_audio_cd_src_get_index (GstElement * element) { GstAudioCdSrc *src = GST_AUDIO_CD_SRC (element); GstIndex *index; GST_OBJECT_LOCK (element); if ((index = src->index)) gst_object_ref (index); GST_OBJECT_UNLOCK (element); return index; } static gint gst_audio_cd_src_track_sort_func (gconstpointer a, gconstpointer b, gpointer foo) { GstAudioCdSrcTrack *track_a = ((GstAudioCdSrcTrack *) a); GstAudioCdSrcTrack *track_b = ((GstAudioCdSrcTrack *) b); /* sort data tracks to the end, and audio tracks by track number */ if (track_a->is_audio == track_b->is_audio) return (gint) track_a->num - (gint) track_b->num; if (track_a->is_audio) { return -1; } else { return 1; } } static gboolean gst_audio_cd_src_start (GstBaseSrc * basesrc) { GstAudioCdSrcClass *klass = GST_AUDIO_CD_SRC_GET_CLASS (basesrc); GstAudioCdSrc *src = GST_AUDIO_CD_SRC (basesrc); gboolean ret; gchar *device = NULL; src->discid = 0; src->mb_discid[0] = '\0'; g_assert (klass->open != NULL); if (src->device != NULL) { device = g_strdup (src->device); } else if (klass->get_default_device != NULL) { device = klass->get_default_device (src); } if (device == NULL) device = g_strdup (DEFAULT_DEVICE); GST_LOG_OBJECT (basesrc, "opening device %s", device); src->tags = gst_tag_list_new_empty (); ret = klass->open (src, device); g_free (device); device = NULL; if (!ret) { GST_DEBUG_OBJECT (basesrc, "failed to open device"); /* subclass (should have) posted an error message with the details */ gst_audio_cd_src_stop (basesrc); return FALSE; } if (src->num_tracks == 0 || src->tracks == NULL) { GST_DEBUG_OBJECT (src, "no tracks"); GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (_("This CD has no audio tracks")), (NULL)); gst_audio_cd_src_stop (basesrc); return FALSE; } /* need to calculate disc IDs before we ditch the data tracks */ gst_audio_cd_src_calculate_cddb_id (src); gst_audio_cd_src_calculate_musicbrainz_discid (src); #if 0 /* adjust sector offsets if necessary */ if (src->toc_bias) { src->toc_offset -= src->tracks[0].start; } for (i = 0; i < src->num_tracks; ++i) { src->tracks[i].start += src->toc_offset; src->tracks[i].end += src->toc_offset; } #endif /* now that we calculated the various disc IDs, * sort the data tracks to end and ignore them */ src->num_all_tracks = src->num_tracks; g_qsort_with_data (src->tracks, src->num_tracks, sizeof (GstAudioCdSrcTrack), gst_audio_cd_src_track_sort_func, NULL); while (src->num_tracks > 0 && !src->tracks[src->num_tracks - 1].is_audio) --src->num_tracks; if (src->num_tracks == 0) { GST_DEBUG_OBJECT (src, "no audio tracks"); GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (_("This CD has no audio tracks")), (NULL)); gst_audio_cd_src_stop (basesrc); return FALSE; } gst_audio_cd_src_add_tags (src); if (src->index && GST_INDEX_IS_WRITABLE (src->index)) gst_audio_cd_src_add_index_associations (src); src->cur_track = 0; src->prev_track = -1; if (src->uri_track > 0 && src->uri_track <= src->num_tracks) { GST_LOG_OBJECT (src, "seek to track %d", src->uri_track); src->cur_track = src->uri_track - 1; src->uri_track = -1; src->mode = GST_AUDIO_CD_SRC_MODE_NORMAL; } src->cur_sector = src->tracks[src->cur_track].start; GST_LOG_OBJECT (src, "starting at sector %d", src->cur_sector); gst_audio_cd_src_update_duration (src); return TRUE; } static void gst_audio_cd_src_clear_tracks (GstAudioCdSrc * src) { if (src->tracks != NULL) { gint i; for (i = 0; i < src->num_all_tracks; ++i) { if (src->tracks[i].tags) gst_tag_list_free (src->tracks[i].tags); } g_free (src->tracks); src->tracks = NULL; } src->num_tracks = 0; src->num_all_tracks = 0; } static gboolean gst_audio_cd_src_stop (GstBaseSrc * basesrc) { GstAudioCdSrcClass *klass = GST_AUDIO_CD_SRC_GET_CLASS (basesrc); GstAudioCdSrc *src = GST_AUDIO_CD_SRC (basesrc); g_assert (klass->close != NULL); klass->close (src); gst_audio_cd_src_clear_tracks (src); if (src->tags) { gst_tag_list_free (src->tags); src->tags = NULL; } src->prev_track = -1; src->cur_track = -1; return TRUE; } static GstFlowReturn gst_audio_cd_src_create (GstPushSrc * pushsrc, GstBuffer ** buffer) { GstAudioCdSrcClass *klass = GST_AUDIO_CD_SRC_GET_CLASS (pushsrc); GstAudioCdSrc *src = GST_AUDIO_CD_SRC (pushsrc); GstBuffer *buf; gboolean eos; GstClockTime position = GST_CLOCK_TIME_NONE; GstClockTime duration = GST_CLOCK_TIME_NONE; gint64 qry_position; g_assert (klass->read_sector != NULL); switch (src->mode) { case GST_AUDIO_CD_SRC_MODE_NORMAL: eos = (src->cur_sector > src->tracks[src->cur_track].end); break; case GST_AUDIO_CD_SRC_MODE_CONTINUOUS: eos = (src->cur_sector > src->tracks[src->num_tracks - 1].end); src->cur_track = gst_audio_cd_src_get_track_from_sector (src, src->cur_sector); break; default: g_return_val_if_reached (GST_FLOW_ERROR); } if (eos) { src->prev_track = -1; GST_DEBUG_OBJECT (src, "EOS at sector %d, cur_track=%d, mode=%d", src->cur_sector, src->cur_track, src->mode); /* base class will send EOS for us */ return GST_FLOW_EOS; } if (src->prev_track != src->cur_track) { GstTagList *tags; tags = gst_tag_list_merge (src->tags, src->tracks[src->cur_track].tags, GST_TAG_MERGE_REPLACE); GST_LOG_OBJECT (src, "announcing tags: %" GST_PTR_FORMAT, tags); gst_pad_push_event (GST_BASE_SRC_PAD (src), gst_event_new_tag (tags)); src->prev_track = src->cur_track; gst_audio_cd_src_update_duration (src); g_object_notify (G_OBJECT (src), "track"); } GST_LOG_OBJECT (src, "asking for sector %u", src->cur_sector); buf = klass->read_sector (src, src->cur_sector); if (buf == NULL) { GST_WARNING_OBJECT (src, "failed to read sector %u", src->cur_sector); return GST_FLOW_ERROR; } if (gst_pad_query_position (GST_BASE_SRC_PAD (src), GST_FORMAT_TIME, &qry_position)) { gint64 next_ts = 0; position = (GstClockTime) qry_position; ++src->cur_sector; if (gst_pad_query_position (GST_BASE_SRC_PAD (src), GST_FORMAT_TIME, &next_ts)) { duration = (GstClockTime) (next_ts - qry_position); } --src->cur_sector; } /* fallback duration: 4 bytes per sample, 44100 samples per second */ if (duration == GST_CLOCK_TIME_NONE) { duration = gst_util_uint64_scale_int (gst_buffer_get_size (buf) >> 2, GST_SECOND, 44100); } GST_BUFFER_TIMESTAMP (buf) = position; GST_BUFFER_DURATION (buf) = duration; GST_LOG_OBJECT (src, "pushing sector %d with timestamp %" GST_TIME_FORMAT, src->cur_sector, GST_TIME_ARGS (position)); ++src->cur_sector; *buffer = buf; return GST_FLOW_OK; }