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gstreamer/gst-libs/gst/audio/gstaudiocdsrc.c
Sebastian Dröge 8fe5dc53e0 audiocdsrc: Add the layout field to the caps
2012-01-05 10:34:24 +01:00

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/* GStreamer Audio CD Source Base Class
* Copyright (C) 2005 Tim-Philipp Müller <tim centricular net>
*
* 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
*
* <refsect2>
* <para>
* Provides a base class for CD digital audio (CDDA) sources, which handles
* things like seeking, querying, discid calculation, tags, and buffer
* timestamping.
* </para>
* <title>Using GstAudioCdSrc-based elements in applications</title>
* <para>
* GstAudioCdSrc registers two #GstFormat<!-- -->s 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").
* </para>
* <para>
* 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.
* </para>
* <para>
* 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.
* </para>
* <para>
* 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).
* </para>
* <title>Tags and meta-information</title>
* <para>
* 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.
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <stdlib.h> /* for strtol */
#include <gst/tag/tag.h>
#include <gst/audio/audio.h>
#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
};
struct _GstAudioCdSrcPrivate
{
GstAudioCdSrcMode mode;
gchar *device;
guint num_tracks;
guint num_all_tracks;
GstAudioCdSrcTrack *tracks;
gint cur_track; /* current track (starting from 0) */
gint prev_track; /* current track last time */
gint cur_sector; /* current sector */
gint seek_sector; /* -1 or sector to seek to */
gint uri_track;
gchar *uri;
guint32 discid; /* cddb disc id (for unit test) */
gchar mb_discid[32]; /* musicbrainz discid */
#if 0
GstIndex *index;
gint index_id;
#endif
gint toc_offset;
gboolean 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);
#if 0
static void gst_audio_cd_src_set_index (GstElement * src, GstIndex * index);
static GstIndex *gst_audio_cd_src_get_index (GstElement * src);
#endif
#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) ", " \
"layout = (string) interleaved, " \
"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");
g_type_class_add_private (klass, sizeof (GstAudioCdSrcPrivate));
/* 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 <n> 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));
#if 0
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);
#endif
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)
{
src->priv =
G_TYPE_INSTANCE_GET_PRIVATE (src, GST_TYPE_AUDIO_CD_SRC,
GstAudioCdSrcPrivate);
/* 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->priv->device = NULL;
src->priv->mode = GST_AUDIO_CD_SRC_MODE_NORMAL;
src->priv->uri_track = -1;
}
static void
gst_audio_cd_src_finalize (GObject * obj)
{
GstAudioCdSrc *cddasrc = GST_AUDIO_CD_SRC (obj);
g_free (cddasrc->priv->uri);
g_free (cddasrc->priv->device);
#if 0
if (cddasrc->priv->index)
gst_object_unref (cddasrc->priv->index);
#endif
G_OBJECT_CLASS (parent_class)->finalize (obj);
}
static void
gst_audio_cd_src_set_device (GstAudioCdSrc * src, const gchar * device)
{
if (src->priv->device)
g_free (src->priv->device);
src->priv->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->priv->device = g_strdup (rdsk_value);
g_free (rdsk_value);
} else {
#endif
src->priv->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->priv->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->priv->num_tracks > 0 && track > src->priv->num_tracks) {
g_warning ("Invalid track %u", track);
} else if (track > 0 && src->priv->tracks != NULL) {
src->priv->cur_sector = src->priv->tracks[track - 1].start;
src->priv->uri_track = track;
} else {
src->priv->uri_track = track; /* seek will be done in start() */
}
break;
}
case ARG_TOC_OFFSET:{
src->priv->toc_offset = g_value_get_int (value);
break;
}
case ARG_TOC_BIAS:{
src->priv->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)
{
#if 0
GstAudioCdSrcClass *klass = GST_AUDIO_CD_SRC_GET_CLASS (object);
#endif
GstAudioCdSrc *src = GST_AUDIO_CD_SRC (object);
GST_OBJECT_LOCK (src);
switch (prop_id) {
case ARG_MODE:
g_value_set_enum (value, src->priv->mode);
break;
case ARG_DEVICE:{
#if 0
if (src->priv->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;
}
}
#endif
if (src->priv->device == NULL)
g_value_set_string (value, DEFAULT_DEVICE);
else
g_value_set_string (value, src->priv->device);
break;
}
case ARG_TRACK:{
if (src->priv->num_tracks <= 0 && src->priv->uri_track > 0) {
g_value_set_uint (value, src->priv->uri_track);
} else {
g_value_set_uint (value, src->priv->cur_track + 1);
}
break;
}
case ARG_TOC_OFFSET:
g_value_set_int (value, src->priv->toc_offset);
break;
case ARG_TOC_BIAS:
g_value_set_boolean (value, src->priv->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->priv->num_tracks; ++i) {
if (sector >= src->priv->tracks[i].start
&& sector <= src->priv->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_FLAG_STARTED);
if (src_format == track_format) {
if (!started)
goto not_started;
if (src_val < 0 || src_val >= src->priv->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->priv->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_FLAG_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->priv->tracks != NULL);
if (dest_format == track_format) {
GST_LOG_OBJECT (src, "duration: %d tracks", src->priv->num_tracks);
gst_query_set_duration (query, track_format, src->priv->num_tracks);
return TRUE;
}
if (src->priv->cur_track < 0
|| src->priv->cur_track >= src->priv->num_tracks)
return FALSE;
if (src->priv->mode == GST_AUDIO_CD_SRC_MODE_NORMAL) {
sectors = src->priv->tracks[src->priv->cur_track].end -
src->priv->tracks[src->priv->cur_track].start + 1;
} else {
sectors = src->priv->tracks[src->priv->num_tracks - 1].end -
src->priv->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->priv->tracks != NULL);
if (dest_format == track_format) {
GST_LOG_OBJECT (src, "position: track %d", src->priv->cur_track);
gst_query_set_position (query, track_format, src->priv->cur_track);
return TRUE;
}
if (src->priv->cur_track < 0
|| src->priv->cur_track >= src->priv->num_tracks)
return FALSE;
if (src->priv->mode == GST_AUDIO_CD_SRC_MODE_NORMAL) {
pos_sector =
src->priv->cur_sector -
src->priv->tracks[src->priv->cur_track].start;
} else {
pos_sector = src->priv->cur_sector - src->priv->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->priv->cur_track >= 0
&& src->priv->cur_track < src->priv->num_tracks);
switch (src->priv->mode) {
case GST_AUDIO_CD_SRC_MODE_NORMAL:
seek_sector += src->priv->tracks[src->priv->cur_track].start;
break;
case GST_AUDIO_CD_SRC_MODE_CONTINUOUS:
seek_sector += src->priv->tracks[0].start;
break;
default:
g_return_val_if_reached (FALSE);
}
src->priv->cur_sector = (gint) seek_sector;
GST_DEBUG_OBJECT (src, "seek'd to sector %d", src->priv->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->priv->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->priv->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->priv->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->priv->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->priv->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->priv->cur_sector = src->priv->tracks[start].start;
GST_DEBUG_OBJECT (src, "starting at sector %d", src->priv->cur_sector);
if (src->priv->cur_track != start) {
src->priv->cur_track = (gint) start;
src->priv->uri_track = -1;
src->priv->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_FLAG_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->priv->uri);
if (GST_OBJECT_FLAG_IS_SET (GST_BASE_SRC (src), GST_BASE_SRC_FLAG_STARTED)) {
src->priv->uri =
g_strdup_printf ("cdda://%s#%d", src->priv->device,
(src->priv->uri_track > 0) ? src->priv->uri_track : 1);
} else {
src->priv->uri = g_strdup ("cdda://1");
}
GST_OBJECT_UNLOCK (src);
return g_strdup (src->priv->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->priv->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->priv->uri_track = strtol (track_number + 1, NULL, 10);
g_free (nuri);
} else {
if (*location == '\0')
src->priv->uri_track = 1;
else
src->priv->uri_track = strtol (location, NULL, 10);
}
if (src->priv->uri_track < 1)
goto failed;
if (src->priv->num_tracks > 0
&& src->priv->tracks != NULL
&& src->priv->uri_track > src->priv->num_tracks)
goto failed;
if (src->priv->uri_track > 0 && src->priv->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->priv->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->priv->num_tracks + 1, track->num, track->start,
track->end, (track->is_audio) ? "AUDIO" : "DATA ", track->tags);
if (src->priv->num_tracks > 0) {
guint end_of_previous_track =
src->priv->tracks[src->priv->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->priv->num_tracks;
src->priv->tracks =
g_renew (GstAudioCdSrcTrack, src->priv->tracks, src->priv->num_tracks);
src->priv->tracks[src->priv->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->priv->tracks[src->priv->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->priv->tracks[0].num);
g_string_append_printf (s, "%02X", src->priv->tracks[0].num);
g_checksum_update (sha, (guchar *) tmp, 2);
g_snprintf (tmp, sizeof (tmp), "%02X",
src->priv->tracks[src->priv->num_tracks - 1].num);
g_string_append_printf (s, " %02X",
src->priv->tracks[src->priv->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->priv->num_tracks) {
guint frame_offset = src->priv->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->priv->mb_discid) + 1);
memcpy (src->priv->mb_discid, ptr, i);
src->priv->mb_discid[i] = '\0';
free (ptr);
/* Replace '/', '+' and '=' by '_', '.' and '-' as specified on
* http://musicbrainz.org/doc/DiscIDCalculation
*/
for (ptr = src->priv->mb_discid; *ptr != '\0'; ptr++) {
if (*ptr == '/')
*ptr = '_';
else if (*ptr == '+')
*ptr = '.';
else if (*ptr == '=')
*ptr = '-';
}
GST_DEBUG_OBJECT (src, "musicbrainz-discid = %s", src->priv->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->priv->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->priv->num_tracks; ++i) {
if (1) { /* src->priv->tracks[i].is_audio) { */
if (num_audio_tracks == 0) {
first_sector = src->priv->tracks[i].start + CD_MSF_OFFSET;
}
last_sector = src->priv->tracks[i].end + CD_MSF_OFFSET + 1;
++num_audio_tracks;
lba_to_msf (src->priv->tracks[i].start + CD_MSF_OFFSET, NULL, NULL, NULL,
&secs);
len_secs =
(src->priv->tracks[i].end - src->priv->tracks[i].start + 1) / 75;
GST_DEBUG_OBJECT (src, "track %02u: lsn %6u (%02u:%02u), "
"length: %u seconds (%02u:%02u)",
num_audio_tracks, src->priv->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->priv->tracks[0].start + CD_MSF_OFFSET; */
lba_to_msf (first_sector, NULL, NULL, NULL, &start_secs);
/* last_sector = src->priv->tracks[src->priv->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->priv->discid = ((id % 0xff) << 24 | t << 8 | num_audio_tracks);
s = g_string_new (NULL);
g_string_append_printf (s, "%08x", src->priv->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->priv->num_tracks);
for (i = 0; i < src->priv->num_tracks; ++i) {
g_string_append_printf (s, " %u",
src->priv->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->priv->num_tracks; ++i) {
gint64 duration;
guint num_sectors;
if (src->priv->tracks[i].tags == NULL)
src->priv->tracks[i].tags = gst_tag_list_new_empty ();
num_sectors = src->priv->tracks[i].end - src->priv->tracks[i].start + 1;
gst_audio_cd_src_convert (src, sector_format, num_sectors,
GST_FORMAT_TIME, &duration);
gst_tag_list_add (src->priv->tracks[i].tags,
GST_TAG_MERGE_REPLACE,
GST_TAG_TRACK_NUMBER, i + 1,
GST_TAG_TRACK_COUNT, src->priv->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->priv->num_tracks, NULL);
#if 0
for (i = 0; i < src->priv->num_tracks; ++i) {
gst_tag_list_add (src->tags, GST_TAG_MERGE_APPEND,
GST_TAG_CDDA_TRACK_TAGS, src->priv->tracks[i].tags, NULL);
}
#endif
GST_DEBUG ("src->tags = %" GST_PTR_FORMAT, src->tags);
}
#if 0
static void
gst_audio_cd_src_add_index_associations (GstAudioCdSrc * src)
{
gint i;
for (i = 0; i < src->priv->num_tracks; i++) {
gint64 sector;
sector = src->priv->tracks[i].start;
gst_index_add_association (src->priv->index, src->priv->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->priv->index;
if (old == index) {
GST_OBJECT_UNLOCK (element);
return;
}
if (index)
gst_object_ref (index);
src->priv->index = index;
GST_OBJECT_UNLOCK (element);
if (old)
gst_object_unref (old);
if (index) {
gst_index_get_writer_id (index, GST_OBJECT (src), &src->priv->index_id);
gst_index_add_format (index, src->priv->index_id, track_format);
gst_index_add_format (index, src->priv->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->priv->index))
gst_object_ref (index);
GST_OBJECT_UNLOCK (element);
return index;
}
#endif
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->priv->discid = 0;
src->priv->mb_discid[0] = '\0';
g_assert (klass->open != NULL);
if (src->priv->device != NULL) {
device = g_strdup (src->priv->device);
}
#if 0
else if (klass->get_default_device != NULL) {
device = klass->get_default_device (src);
}
#endif
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)
goto open_failed;
if (src->priv->num_tracks == 0 || src->priv->tracks == NULL)
goto no_tracks;
/* 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->priv->toc_bias) {
src->priv->toc_offset -= src->priv->tracks[0].start;
}
for (i = 0; i < src->priv->num_tracks; ++i) {
src->priv->tracks[i].start += src->priv->toc_offset;
src->priv->tracks[i].end += src->priv->toc_offset;
}
#endif
/* now that we calculated the various disc IDs,
* sort the data tracks to end and ignore them */
src->priv->num_all_tracks = src->priv->num_tracks;
g_qsort_with_data (src->priv->tracks, src->priv->num_tracks,
sizeof (GstAudioCdSrcTrack), gst_audio_cd_src_track_sort_func, NULL);
while (src->priv->num_tracks > 0
&& !src->priv->tracks[src->priv->num_tracks - 1].is_audio)
--src->priv->num_tracks;
if (src->priv->num_tracks == 0)
goto no_tracks;
gst_audio_cd_src_add_tags (src);
#if 0
if (src->priv->index && GST_INDEX_IS_WRITABLE (src->priv->index))
gst_audio_cd_src_add_index_associations (src);
#endif
src->priv->cur_track = 0;
src->priv->prev_track = -1;
if (src->priv->uri_track > 0 && src->priv->uri_track <= src->priv->num_tracks) {
GST_LOG_OBJECT (src, "seek to track %d", src->priv->uri_track);
src->priv->cur_track = src->priv->uri_track - 1;
src->priv->uri_track = -1;
src->priv->mode = GST_AUDIO_CD_SRC_MODE_NORMAL;
}
src->priv->cur_sector = src->priv->tracks[src->priv->cur_track].start;
GST_LOG_OBJECT (src, "starting at sector %d", src->priv->cur_sector);
gst_audio_cd_src_update_duration (src);
return TRUE;
/* ERRORS */
open_failed:
{
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;
}
no_tracks:
{
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;
}
}
static void
gst_audio_cd_src_clear_tracks (GstAudioCdSrc * src)
{
if (src->priv->tracks != NULL) {
gint i;
for (i = 0; i < src->priv->num_all_tracks; ++i) {
if (src->priv->tracks[i].tags)
gst_tag_list_free (src->priv->tracks[i].tags);
}
g_free (src->priv->tracks);
src->priv->tracks = NULL;
}
src->priv->num_tracks = 0;
src->priv->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->priv->prev_track = -1;
src->priv->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->priv->mode) {
case GST_AUDIO_CD_SRC_MODE_NORMAL:
eos =
(src->priv->cur_sector > src->priv->tracks[src->priv->cur_track].end);
break;
case GST_AUDIO_CD_SRC_MODE_CONTINUOUS:
eos =
(src->priv->cur_sector >
src->priv->tracks[src->priv->num_tracks - 1].end);
src->priv->cur_track =
gst_audio_cd_src_get_track_from_sector (src, src->priv->cur_sector);
break;
default:
g_return_val_if_reached (GST_FLOW_ERROR);
}
if (eos) {
src->priv->prev_track = -1;
GST_DEBUG_OBJECT (src, "EOS at sector %d, cur_track=%d, mode=%d",
src->priv->cur_sector, src->priv->cur_track, src->priv->mode);
/* base class will send EOS for us */
return GST_FLOW_EOS;
}
if (src->priv->prev_track != src->priv->cur_track) {
GstTagList *tags;
tags =
gst_tag_list_merge (src->tags,
src->priv->tracks[src->priv->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->priv->prev_track = src->priv->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->priv->cur_sector);
buf = klass->read_sector (src, src->priv->cur_sector);
if (buf == NULL) {
GST_WARNING_OBJECT (src, "failed to read sector %u", src->priv->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->priv->cur_sector;
if (gst_pad_query_position (GST_BASE_SRC_PAD (src), GST_FORMAT_TIME,
&next_ts)) {
duration = (GstClockTime) (next_ts - qry_position);
}
--src->priv->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->priv->cur_sector, GST_TIME_ARGS (position));
++src->priv->cur_sector;
*buffer = buf;
return GST_FLOW_OK;
}
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