static GstFlowReturn
gst_png_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
{
GstPngParse *pngparse = GST_PNG_PARSE (parse);
if (!pngparse->sent_codec_tag) {
GstTagList *taglist;
GstCaps *caps;
taglist = gst_tag_list_new_empty ();
/* codec tag */
caps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (parse));
gst_pb_utils_add_codec_description_to_tag_list (taglist,
GST_TAG_VIDEO_CODEC, caps);
gst_caps_unref (caps);
gst_pad_push_event (GST_BASE_PARSE_SRC_PAD (pngparse),
gst_event_new_tag (taglist));
/* also signals the end of first-frame processing */
pngparse->sent_codec_tag = TRUE;
}
return GST_FLOW_OK;
}
static void
brasero_transcode_send_volume_event (BraseroTranscode *transcode)
{
BraseroTranscodePrivate *priv;
gdouble track_peak = 0.0;
gdouble track_gain = 0.0;
GstTagList *tag_list;
BraseroTrack *track;
GstEvent *event;
GValue *value;
priv = BRASERO_TRANSCODE_PRIVATE (transcode);
brasero_job_get_current_track (BRASERO_JOB (transcode), &track);
BRASERO_JOB_LOG (transcode, "Sending audio levels tags");
if (brasero_track_tag_lookup (track, BRASERO_TRACK_PEAK_VALUE, &value) == BRASERO_BURN_OK)
track_peak = g_value_get_double (value);
if (brasero_track_tag_lookup (track, BRASERO_TRACK_GAIN_VALUE, &value) == BRASERO_BURN_OK)
track_gain = g_value_get_double (value);
/* it's possible we fail */
tag_list = gst_tag_list_new (GST_TAG_TRACK_GAIN, track_gain,
GST_TAG_TRACK_PEAK, track_peak,
NULL);
/* NOTE: that event is goind downstream */
event = gst_event_new_tag (tag_list);
if (!gst_element_send_event (priv->convert, event))
BRASERO_JOB_LOG (transcode, "Couldn't send tags to rgvolume");
BRASERO_JOB_LOG (transcode, "Set volume level %lf %lf", track_gain, track_peak);
}
static GstFlowReturn
gst_mpegv_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
{
GstMpegvParse *mpvparse = GST_MPEGVIDEO_PARSE (parse);
GstTagList *taglist;
/* tag sending done late enough in hook to ensure pending events
* have already been sent */
if (G_UNLIKELY (mpvparse->send_codec_tag)) {
gchar *codec;
/* codec tag */
codec =
g_strdup_printf ("MPEG %d Video",
(mpvparse->config_flags & FLAG_MPEG2) ? 2 : 1);
taglist = gst_tag_list_new (GST_TAG_VIDEO_CODEC, codec, NULL);
g_free (codec);
gst_pad_push_event (GST_BASE_PARSE_SRC_PAD (mpvparse),
gst_event_new_tag (taglist));
mpvparse->send_codec_tag = FALSE;
}
/* usual clipping applies */
frame->flags |= GST_BASE_PARSE_FRAME_FLAG_CLIP;
return GST_FLOW_OK;
}
static gboolean
gst_mpeg_demux_parse_packhead (GstMPEGParse * mpeg_parse, GstBuffer * buffer)
{
GstMPEGDemux *demux = GST_MPEG_DEMUX (mpeg_parse);
guint8 *buf;
parent_class->parse_packhead (mpeg_parse, buffer);
buf = GST_BUFFER_DATA (buffer);
/* do something useful here */
if (demux->pending_tags) {
GstMPEGStream **streams;
guint i, num;
streams = demux->audio_stream;
num = GST_MPEG_DEMUX_NUM_AUDIO_STREAMS;
for (i = 0; i < num; ++i) {
if (streams[i] != NULL && streams[i]->tags != NULL)
gst_pad_push_event (streams[i]->pad,
gst_event_new_tag (gst_tag_list_copy (streams[i]->tags)));
}
demux->pending_tags = FALSE;
}
return TRUE;
}
GstEvent *
gst_kate_util_decoder_base_get_tag_event (GstKateDecoderBase * decoder)
{
if (!decoder->tags)
return NULL;
decoder->tags_changed = FALSE;
return gst_event_new_tag (gst_tag_list_ref (decoder->tags));
}
void notifyGstTagsOnPad(GstElement* element, GstPad* pad, GstTagList* tags)
{
#ifdef GST_API_VERSION_1
UNUSED_PARAM(element);
gst_pad_push_event(GST_PAD_CAST(pad), gst_event_new_tag(tags));
#else
gst_element_found_tags_for_pad(element, pad, tags);
#endif
}
/**
* gst_avi_subtitle_title_tag:
* @sub: subtitle element
* @title: the title of this subtitle stream
*
* Send an event to the srcpad of the @sub element with the title
* of the subtitle stream as a GST_TAG_TITLE
*/
static void
gst_avi_subtitle_title_tag (GstAviSubtitle * sub, gchar * title)
{
GstTagList *temp_list = gst_tag_list_new ();
gst_tag_list_add (temp_list, GST_TAG_MERGE_APPEND, GST_TAG_TITLE, title,
NULL);
gst_pad_push_event (sub->src, gst_event_new_tag (temp_list));
}
static VALUE
tag_initialize(VALUE self, VALUE taglist)
{
GstEvent *event;
event = gst_event_new_tag(RVAL2GST_STRUCT(taglist));
G_INITIALIZE(self, event);
return Qnil;
}
static GstFlowReturn
gst_tag_mux_render_start_tag (GstTagMux * mux)
{
GstTagMuxClass *klass;
GstBuffer *buffer;
GstTagList *taglist;
GstEvent *event;
GstFlowReturn ret;
GstSegment segment;
taglist = gst_tag_mux_get_tags (mux);
klass = GST_TAG_MUX_CLASS (G_OBJECT_GET_CLASS (mux));
if (klass->render_start_tag == NULL)
goto no_vfunc;
buffer = klass->render_start_tag (mux, taglist);
/* Null buffer is ok, just means we're not outputting anything */
if (buffer == NULL) {
GST_INFO_OBJECT (mux, "No start tag generated");
mux->priv->start_tag_size = 0;
return GST_FLOW_OK;
}
mux->priv->start_tag_size = gst_buffer_get_size (buffer);
GST_LOG_OBJECT (mux, "tag size = %" G_GSIZE_FORMAT " bytes",
mux->priv->start_tag_size);
/* Send newsegment event from byte position 0, so the tag really gets
* written to the start of the file, independent of the upstream segment */
gst_segment_init (&segment, GST_FORMAT_BYTES);
gst_pad_push_event (mux->priv->srcpad, gst_event_new_segment (&segment));
/* Send an event about the new tags to downstream elements */
/* gst_event_new_tag takes ownership of the list, so use a copy */
event = gst_event_new_tag (gst_tag_list_ref (taglist));
gst_pad_push_event (mux->priv->srcpad, event);
GST_BUFFER_OFFSET (buffer) = 0;
ret = gst_pad_push (mux->priv->srcpad, buffer);
mux->priv->current_offset = mux->priv->start_tag_size;
mux->priv->max_offset =
MAX (mux->priv->max_offset, mux->priv->current_offset);
return ret;
no_vfunc:
{
GST_ERROR_OBJECT (mux, "Subclass does not implement "
"render_start_tag vfunc!");
return GST_FLOW_ERROR;
}
}
static gboolean
gst_a2dp_sink_init_dynamic_elements (GstA2dpSink * self, GstCaps * caps)
{
GstStructure *structure;
GstEvent *event;
gboolean crc;
gchar *mode = NULL;
structure = gst_caps_get_structure (caps, 0);
/* first, we need to create our rtp payloader */
if (gst_structure_has_name (structure, "audio/x-sbc")) {
GST_LOG_OBJECT (self, "sbc media received");
if (!gst_a2dp_sink_init_rtp_sbc_element (self))
return FALSE;
} else if (gst_structure_has_name (structure, "audio/mpeg")) {
GST_LOG_OBJECT (self, "mp3 media received");
if (!gst_a2dp_sink_init_rtp_mpeg_element (self))
return FALSE;
} else {
GST_ERROR_OBJECT (self, "Unexpected media type");
return FALSE;
}
if (!gst_element_link (GST_ELEMENT (self->rtp), GST_ELEMENT (self->sink))) {
GST_ERROR_OBJECT (self, "couldn't link rtpsbcpay " "to avdtpsink");
return FALSE;
}
/* check if we should push the taglist FIXME should we push this?
* we can send the tags directly if needed */
if (self->taglist != NULL && gst_structure_has_name (structure, "audio/mpeg")) {
event = gst_event_new_tag (self->taglist);
/* send directly the crc */
if (gst_tag_list_get_boolean (self->taglist, "has-crc", &crc))
gst_avdtp_sink_set_crc (self->sink, crc);
if (gst_tag_list_get_string (self->taglist, "channel-mode", &mode))
gst_avdtp_sink_set_channel_mode (self->sink, mode);
gst_pad_send_event (self->ghostpad, event);
self->taglist = NULL;
g_free (mode);
}
g_object_set (self->rtp, "mtu",
gst_avdtp_sink_get_link_mtu (self->sink), NULL);
return TRUE;
}
static void
gst_vp9_dec_send_tags (GstVP9Dec * dec)
{
GstTagList *list;
list = gst_tag_list_new_empty ();
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_VIDEO_CODEC, "VP9 video", NULL);
gst_pad_push_event (GST_VIDEO_DECODER_SRC_PAD (dec),
gst_event_new_tag (list));
}
static inline void
_push_mandatory_events (GstAggregator * self)
{
GstAggregatorPrivate *priv = self->priv;
if (g_atomic_int_get (&self->priv->send_stream_start)) {
gchar s_id[32];
GST_INFO_OBJECT (self, "pushing stream start");
/* stream-start (FIXME: create id based on input ids) */
g_snprintf (s_id, sizeof (s_id), "agg-%08x", g_random_int ());
if (!gst_pad_push_event (self->srcpad, gst_event_new_stream_start (s_id))) {
GST_WARNING_OBJECT (self->srcpad, "Sending stream start event failed");
}
g_atomic_int_set (&self->priv->send_stream_start, FALSE);
}
if (self->priv->srccaps) {
GST_INFO_OBJECT (self, "pushing caps: %" GST_PTR_FORMAT,
self->priv->srccaps);
if (!gst_pad_push_event (self->srcpad,
gst_event_new_caps (self->priv->srccaps))) {
GST_WARNING_OBJECT (self->srcpad, "Sending caps event failed");
}
gst_caps_unref (self->priv->srccaps);
self->priv->srccaps = NULL;
}
if (g_atomic_int_get (&self->priv->send_segment)) {
if (!g_atomic_int_get (&self->priv->flush_seeking)) {
GstEvent *segev = gst_event_new_segment (&self->segment);
if (!self->priv->seqnum)
self->priv->seqnum = gst_event_get_seqnum (segev);
else
gst_event_set_seqnum (segev, self->priv->seqnum);
GST_DEBUG_OBJECT (self, "pushing segment %" GST_PTR_FORMAT, segev);
gst_pad_push_event (self->srcpad, segev);
g_atomic_int_set (&self->priv->send_segment, FALSE);
}
}
if (priv->tags && priv->tags_changed) {
gst_pad_push_event (self->srcpad,
gst_event_new_tag (gst_tag_list_ref (priv->tags)));
priv->tags_changed = FALSE;
}
}
static GstFlowReturn
vorbis_handle_type_packet (GstVorbisDec * vd)
{
GList *walk;
gint res;
g_assert (vd->initialized == FALSE);
#ifdef USE_TREMOLO
if (G_UNLIKELY ((res = vorbis_dsp_init (&vd->vd, &vd->vi))))
goto synthesis_init_error;
#else
if (G_UNLIKELY ((res = vorbis_synthesis_init (&vd->vd, &vd->vi))))
goto synthesis_init_error;
if (G_UNLIKELY ((res = vorbis_block_init (&vd->vd, &vd->vb))))
goto block_init_error;
#endif
vd->initialized = TRUE;
if (vd->pendingevents) {
for (walk = vd->pendingevents; walk; walk = g_list_next (walk))
gst_pad_push_event (vd->srcpad, GST_EVENT_CAST (walk->data));
g_list_free (vd->pendingevents);
vd->pendingevents = NULL;
}
if (vd->taglist) {
/* The tags have already been sent on the bus as messages. */
gst_pad_push_event (vd->srcpad, gst_event_new_tag (vd->taglist));
vd->taglist = NULL;
}
return GST_FLOW_OK;
/* ERRORS */
synthesis_init_error:
{
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("couldn't initialize synthesis (%d)", res));
return GST_FLOW_ERROR;
}
block_init_error:
{
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("couldn't initialize block (%d)", res));
return GST_FLOW_ERROR;
}
}
/* takes ownership of tag list */
static gboolean
gst_icydemux_send_tag_event (GstICYDemux * icydemux, GstTagList * tags)
{
GstEvent *event;
gst_element_post_message (GST_ELEMENT (icydemux),
gst_message_new_tag (GST_OBJECT (icydemux), gst_tag_list_copy (tags)));
event = gst_event_new_tag (tags);
GST_EVENT_TIMESTAMP (event) = 0;
GST_DEBUG_OBJECT (icydemux, "Sending tag event on src pad");
return gst_pad_push_event (icydemux->srcpad, event);
}
static GstFlowReturn
gst_real_audio_demux_parse_data (GstRealAudioDemux * demux)
{
GstFlowReturn ret = GST_FLOW_OK;
guint avail, unit_size;
avail = gst_adapter_available (demux->adapter);
if (demux->packet_size > 0)
unit_size = demux->packet_size;
else
unit_size = avail & 0xfffffff0; /* round down to next multiple of 16 */
GST_LOG_OBJECT (demux, "available = %u, unit_size = %u", avail, unit_size);
while (ret == GST_FLOW_OK && unit_size > 0 && avail >= unit_size) {
GstClockTime ts;
GstBuffer *buf;
buf = gst_adapter_take_buffer (demux->adapter, unit_size);
avail -= unit_size;
if (demux->need_newsegment) {
gst_pad_push_event (demux->srcpad,
gst_event_new_segment (&demux->segment));
demux->need_newsegment = FALSE;
}
if (demux->pending_tags) {
gst_pad_push_event (demux->srcpad,
gst_event_new_tag (demux->pending_tags));
demux->pending_tags = NULL;
}
if (demux->fourcc == GST_RM_AUD_DNET) {
buf = gst_rm_utils_descramble_dnet_buffer (buf);
}
ts = gst_real_demux_get_timestamp_from_offset (demux, demux->offset);
GST_BUFFER_TIMESTAMP (buf) = ts;
demux->segment.position = ts;
ret = gst_pad_push (demux->srcpad, buf);
}
return ret;
}
static void
avrcp_metadata_cb (GstAvrcpConnection * avrcp, GstTagList * taglist,
gpointer user_data)
{
GstAvdtpSrc *src = GST_AVDTP_SRC (user_data);
guint64 duration;
if (gst_tag_list_get_uint64 (taglist, GST_TAG_DURATION, &duration)) {
src->duration = duration;
gst_element_post_message (GST_ELEMENT (src),
gst_message_new_duration_changed (GST_OBJECT (src)));
}
gst_pad_push_event (GST_BASE_SRC_PAD (src),
gst_event_new_tag (gst_tag_list_copy (taglist)));
gst_element_post_message (GST_ELEMENT (src),
gst_message_new_tag (GST_OBJECT (src), taglist));
}
static void
gst_chromaprint_create_fingerprint (GstChromaprint * chromaprint)
{
GstTagList *tags;
if (chromaprint->duration <= 3)
return;
GST_DEBUG_OBJECT (chromaprint,
"Generating fingerprint based on %d seconds of audio",
chromaprint->duration);
chromaprint_finish (chromaprint->context);
chromaprint_get_fingerprint (chromaprint->context, &chromaprint->fingerprint);
chromaprint->record = FALSE;
tags = gst_tag_list_new (GST_TAG_CHROMAPRINT_FINGERPRINT,
chromaprint->fingerprint, NULL);
gst_pad_push_event (GST_BASE_TRANSFORM_SRC_PAD (chromaprint),
gst_event_new_tag (tags));
}
static gboolean on_activate(GstPad* pad, GstObject* parent) {
gst_pad_activate_mode(pad,GST_PAD_MODE_PUSH,TRUE);
GstTagList* list = gst_tag_list_new_empty();
GValue value;
memset(&value,0,sizeof(value));
g_value_init(&value,G_TYPE_DOUBLE);
fprintf(stderr,"bwub setting gain/peak %f %f %f\n",
g_activate_gain.peak, g_activate_gain.gain,
g_activate_gain.level);
g_value_set_double(&value,g_activate_gain.gain*2);
gst_tag_list_add_value(list, GST_TAG_MERGE_REPLACE, GST_TAG_TRACK_GAIN, &value);
gst_tag_list_add_value(list, GST_TAG_MERGE_REPLACE, GST_TAG_ALBUM_GAIN, &value);
g_value_set_double(&value,g_activate_gain.peak*2);
gst_tag_list_add_value(list, GST_TAG_MERGE_REPLACE, GST_TAG_TRACK_PEAK, &value);
gst_tag_list_add_value(list, GST_TAG_MERGE_REPLACE, GST_TAG_ALBUM_PEAK, &value);
g_value_set_double(&value,g_activate_gain.level);
gst_tag_list_add_value(list, GST_TAG_MERGE_REPLACE, GST_TAG_REFERENCE_LEVEL, &value);
assert(TRUE==gst_pad_send_event(pad,gst_event_new_tag(list)));
return TRUE;
}
// GStreamer event filter that removes any image metadata.
static gboolean remove_image(GstPad* pad,
GstObject* parent,
GstEvent* event) {
GstTagList* tags = NULL;
GstPad* sink = NULL;
gboolean ret = false;
switch (GST_EVENT_TYPE(event)) {
case GST_EVENT_TAG:
gst_event_parse_tag(event, &tags);
gst_tag_list_remove_tag(tags, "image");
event = gst_event_new_tag(tags);
break;
case GST_EVENT_CAPS:
sink = gst_element_get_static_pad(GST_ELEMENT(parent), "src");
ret = gst_pad_push_event(sink, event);
gst_object_unref(sink);
return ret;
default:
break;
}
return gst_pad_event_default(pad, parent, event);
}
static GstFlowReturn
gst_multipart_demux_chain (GstPad * pad, GstBuffer * buf)
{
GstMultipartDemux *multipart;
GstAdapter *adapter;
GstClockTime timestamp;
gint size = 1;
GstFlowReturn res;
multipart = GST_MULTIPART_DEMUX (gst_pad_get_parent (pad));
adapter = multipart->adapter;
res = GST_FLOW_OK;
timestamp = GST_BUFFER_TIMESTAMP (buf);
if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
gst_adapter_clear (adapter);
}
gst_adapter_push (adapter, buf);
while (gst_adapter_available (adapter) > 0) {
GstMultipartPad *srcpad;
GstBuffer *outbuf;
gboolean created;
gint datalen;
if (G_UNLIKELY (!multipart->header_completed)) {
if ((size = multipart_parse_header (multipart)) < 0) {
goto nodata;
} else {
gst_adapter_flush (adapter, size);
multipart->header_completed = TRUE;
}
}
if ((size = multipart_find_boundary (multipart, &datalen)) < 0) {
goto nodata;
}
/* Invalidate header info */
multipart->header_completed = FALSE;
multipart->content_length = -1;
if (G_UNLIKELY (datalen <= 0)) {
GST_DEBUG_OBJECT (multipart, "skipping empty content.");
gst_adapter_flush (adapter, size - datalen);
} else {
srcpad =
gst_multipart_find_pad_by_mime (multipart,
multipart->mime_type, &created);
outbuf = gst_adapter_take_buffer (adapter, datalen);
gst_adapter_flush (adapter, size - datalen);
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (srcpad->pad));
if (created) {
GstTagList *tags;
/* Push new segment, first buffer has 0 timestamp */
gst_pad_push_event (srcpad->pad,
gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0));
tags =
gst_tag_list_new_full (GST_TAG_CONTAINER_FORMAT, "Multipart", NULL);
gst_pad_push_event (srcpad->pad, gst_event_new_tag (tags));
GST_BUFFER_TIMESTAMP (outbuf) = 0;
} else {
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
}
GST_DEBUG_OBJECT (multipart,
"pushing buffer with timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
GST_DEBUG_OBJECT (multipart, "buffer has caps %" GST_PTR_FORMAT,
GST_BUFFER_CAPS (outbuf));
res = gst_pad_push (srcpad->pad, outbuf);
res = gst_multipart_combine_flows (multipart, srcpad, res);
if (res != GST_FLOW_OK)
break;
}
}
nodata:
gst_object_unref (multipart);
if (G_UNLIKELY (size == MULTIPART_DATA_ERROR))
return GST_FLOW_ERROR;
if (G_UNLIKELY (size == MULTIPART_DATA_EOS))
return GST_FLOW_UNEXPECTED;
return res;
}
static gboolean gst_a2dp_sink_init_dynamic_elements(GstA2dpSink *self,
GstCaps *caps)
{
GstStructure *structure;
GstEvent *event;
GstPad *capsfilterpad;
gboolean crc;
gchar *mode = NULL;
structure = gst_caps_get_structure(caps, 0);
/* before everything we need to remove fakesink */
gst_a2dp_sink_remove_fakesink(self);
/* first, we need to create our rtp payloader */
if (gst_structure_has_name(structure, "audio/x-sbc")) {
GST_LOG_OBJECT(self, "sbc media received");
if (!gst_a2dp_sink_init_rtp_sbc_element(self))
return FALSE;
} else if (gst_structure_has_name(structure, "audio/mpeg")) {
GST_LOG_OBJECT(self, "mp3 media received");
if (!gst_a2dp_sink_init_rtp_mpeg_element(self))
return FALSE;
} else {
GST_ERROR_OBJECT(self, "Unexpected media type");
return FALSE;
}
if (!gst_a2dp_sink_init_avdtp_sink(self))
return FALSE;
/* check if we should push the taglist FIXME should we push this?
* we can send the tags directly if needed */
if (self->taglist != NULL &&
gst_structure_has_name(structure, "audio/mpeg")) {
event = gst_event_new_tag(self->taglist);
/* send directly the crc */
if (gst_tag_list_get_boolean(self->taglist, "has-crc", &crc))
gst_avdtp_sink_set_crc(self->sink, crc);
if (gst_tag_list_get_string(self->taglist, "channel-mode",
&mode))
gst_avdtp_sink_set_channel_mode(self->sink, mode);
capsfilterpad = gst_ghost_pad_get_target(self->ghostpad);
gst_pad_send_event(capsfilterpad, event);
self->taglist = NULL;
g_free(mode);
}
if (!gst_avdtp_sink_set_device_caps(self->sink, caps))
return FALSE;
g_object_set(G_OBJECT(self->rtp), "mtu",
gst_avdtp_sink_get_link_mtu(self->sink), NULL);
/* we forward our new segment here if we have one */
if (self->newseg_event) {
gst_pad_send_event(GST_BASE_RTP_PAYLOAD_SINKPAD(self->rtp),
self->newseg_event);
self->newseg_event = NULL;
}
return TRUE;
}
static GstFlowReturn
gst_flac_tag_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstFlacTag *tag;
GstFlowReturn ret;
GstMapInfo map;
gsize size;
ret = GST_FLOW_OK;
tag = GST_FLAC_TAG (parent);
gst_adapter_push (tag->adapter, buffer);
/* Initial state, we don't even know if we are dealing with a flac file */
if (tag->state == GST_FLAC_TAG_STATE_INIT) {
GstBuffer *id_buffer;
if (gst_adapter_available (tag->adapter) < sizeof (FLAC_MAGIC))
goto cleanup;
id_buffer = gst_adapter_take_buffer (tag->adapter, FLAC_MAGIC_SIZE);
GST_DEBUG_OBJECT (tag, "looking for " FLAC_MAGIC " identifier");
if (gst_buffer_memcmp (id_buffer, 0, FLAC_MAGIC, FLAC_MAGIC_SIZE) == 0) {
GST_DEBUG_OBJECT (tag, "pushing " FLAC_MAGIC " identifier buffer");
ret = gst_pad_push (tag->srcpad, id_buffer);
if (ret != GST_FLOW_OK)
goto cleanup;
tag->state = GST_FLAC_TAG_STATE_METADATA_BLOCKS;
} else {
/* FIXME: does that work well with FLAC files containing ID3v2 tags ? */
gst_buffer_unref (id_buffer);
GST_ELEMENT_ERROR (tag, STREAM, WRONG_TYPE, (NULL), (NULL));
ret = GST_FLOW_ERROR;
}
}
/* The fLaC magic string has been skipped, try to detect the beginning
* of a metadata block
*/
if (tag->state == GST_FLAC_TAG_STATE_METADATA_BLOCKS) {
guint type;
gboolean is_last;
const guint8 *block_header;
g_assert (tag->metadata_block_size == 0);
g_assert (tag->metadata_last_block == FALSE);
/* The header of a flac metadata block is 4 bytes long:
* 1st bit: indicates whether this is the last metadata info block
* 7 next bits: 4 if vorbis comment block
* 24 next bits: size of the metadata to follow (big endian)
*/
if (gst_adapter_available (tag->adapter) < 4)
goto cleanup;
block_header = gst_adapter_map (tag->adapter, 4);
is_last = ((block_header[0] & 0x80) == 0x80);
type = block_header[0] & 0x7F;
size = (block_header[1] << 16)
| (block_header[2] << 8)
| block_header[3];
gst_adapter_unmap (tag->adapter);
/* The 4 bytes long header isn't included in the metadata size */
tag->metadata_block_size = size + 4;
tag->metadata_last_block = is_last;
GST_DEBUG_OBJECT (tag,
"got metadata block: %" G_GSIZE_FORMAT " bytes, type %d, "
"is vorbiscomment: %d, is last: %d",
size, type, (type == 0x04), is_last);
/* Metadata blocks of type 4 are vorbis comment blocks */
if (type == 0x04) {
tag->state = GST_FLAC_TAG_STATE_VC_METADATA_BLOCK;
} else {
tag->state = GST_FLAC_TAG_STATE_WRITING_METADATA_BLOCK;
}
}
/* Reads a metadata block */
if ((tag->state == GST_FLAC_TAG_STATE_WRITING_METADATA_BLOCK) ||
(tag->state == GST_FLAC_TAG_STATE_VC_METADATA_BLOCK)) {
GstBuffer *metadata_buffer;
if (gst_adapter_available (tag->adapter) < tag->metadata_block_size)
goto cleanup;
metadata_buffer = gst_adapter_take_buffer (tag->adapter,
tag->metadata_block_size);
/* clear the is-last flag, as the last metadata block will
* be the vorbis comment block which we will build ourselves.
*/
gst_buffer_map (metadata_buffer, &map, GST_MAP_READWRITE);
map.data[0] &= (~0x80);
gst_buffer_unmap (metadata_buffer, &map);
if (tag->state == GST_FLAC_TAG_STATE_WRITING_METADATA_BLOCK) {
GST_DEBUG_OBJECT (tag, "pushing metadata block buffer");
ret = gst_pad_push (tag->srcpad, metadata_buffer);
if (ret != GST_FLOW_OK)
goto cleanup;
} else {
tag->vorbiscomment = metadata_buffer;
}
tag->metadata_block_size = 0;
tag->state = GST_FLAC_TAG_STATE_METADATA_NEXT_BLOCK;
}
/* This state is mainly used to be able to stop as soon as we read
* a vorbiscomment block from the flac file if we are in an only output
* tags mode
*/
if (tag->state == GST_FLAC_TAG_STATE_METADATA_NEXT_BLOCK) {
/* Check if in the previous iteration we read a vorbis comment metadata
* block, and stop now if the user only wants to read tags
*/
if (tag->vorbiscomment != NULL) {
guint8 id_data[4];
/* We found some tags, try to parse them and notify the other elements
* that we encountered some tags
*/
GST_DEBUG_OBJECT (tag, "emitting vorbiscomment tags");
gst_buffer_extract (tag->vorbiscomment, 0, id_data, 4);
tag->tags = gst_tag_list_from_vorbiscomment_buffer (tag->vorbiscomment,
id_data, 4, NULL);
if (tag->tags != NULL) {
gst_pad_push_event (tag->srcpad,
gst_event_new_tag (gst_tag_list_copy (tag->tags)));
}
gst_buffer_unref (tag->vorbiscomment);
tag->vorbiscomment = NULL;
}
/* Skip to next state */
if (tag->metadata_last_block == FALSE) {
tag->state = GST_FLAC_TAG_STATE_METADATA_BLOCKS;
} else {
tag->state = GST_FLAC_TAG_STATE_ADD_VORBIS_COMMENT;
}
}
/* Creates a vorbis comment block from the metadata which was set
* on the gstreamer element, and add it to the flac stream
*/
if (tag->state == GST_FLAC_TAG_STATE_ADD_VORBIS_COMMENT) {
GstBuffer *buffer;
const GstTagList *user_tags;
GstTagList *merged_tags;
/* merge the tag lists */
user_tags = gst_tag_setter_get_tag_list (GST_TAG_SETTER (tag));
if (user_tags != NULL) {
merged_tags = gst_tag_list_merge (user_tags, tag->tags,
gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (tag)));
} else {
merged_tags = gst_tag_list_copy (tag->tags);
}
if (merged_tags == NULL) {
/* If we get a NULL list of tags, we must generate a padding block
* which is marked as the last metadata block, otherwise we'll
* end up with a corrupted flac file.
*/
GST_WARNING_OBJECT (tag, "No tags found");
buffer = gst_buffer_new_and_alloc (12);
if (buffer == NULL)
goto no_buffer;
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
memset (map.data, 0, map.size);
map.data[0] = 0x81; /* 0x80 = Last metadata block,
* 0x01 = padding block */
gst_buffer_unmap (buffer, &map);
} else {
guchar header[4];
guint8 fbit[1];
memset (header, 0, sizeof (header));
header[0] = 0x84; /* 0x80 = Last metadata block,
* 0x04 = vorbiscomment block */
buffer = gst_tag_list_to_vorbiscomment_buffer (merged_tags, header,
sizeof (header), NULL);
GST_DEBUG_OBJECT (tag, "Writing tags %" GST_PTR_FORMAT, merged_tags);
gst_tag_list_free (merged_tags);
if (buffer == NULL)
goto no_comment;
size = gst_buffer_get_size (buffer);
if ((size < 4) || ((size - 4) > 0xFFFFFF))
goto comment_too_long;
fbit[0] = 1;
/* Get rid of the framing bit at the end of the vorbiscomment buffer
* if it exists since libFLAC seems to lose sync because of this
* bit in gstflacdec
*/
if (gst_buffer_memcmp (buffer, size - 1, fbit, 1) == 0) {
buffer = gst_buffer_make_writable (buffer);
gst_buffer_resize (buffer, 0, size - 1);
}
}
/* The 4 byte metadata block header isn't accounted for in the total
* size of the metadata block
*/
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
map.data[1] = (((map.size - 4) & 0xFF0000) >> 16);
map.data[2] = (((map.size - 4) & 0x00FF00) >> 8);
map.data[3] = ((map.size - 4) & 0x0000FF);
gst_buffer_unmap (buffer, &map);
GST_DEBUG_OBJECT (tag, "pushing %" G_GSIZE_FORMAT " byte vorbiscomment "
"buffer", map.size);
ret = gst_pad_push (tag->srcpad, buffer);
if (ret != GST_FLOW_OK) {
goto cleanup;
}
tag->state = GST_FLAC_TAG_STATE_AUDIO_DATA;
}
static GstBuffer *
gst_tag_lib_mux_render_tag (GstTagLibMux * mux)
{
GstTagLibMuxClass *klass;
GstTagMergeMode merge_mode;
GstTagSetter *tagsetter;
GstBuffer *buffer;
const GstTagList *tagsetter_tags;
GstTagList *taglist;
GstEvent *event;
tagsetter = GST_TAG_SETTER (mux);
tagsetter_tags = gst_tag_setter_get_tag_list (tagsetter);
merge_mode = gst_tag_setter_get_tag_merge_mode (tagsetter);
GST_LOG_OBJECT (mux, "merging tags, merge mode = %d", merge_mode);
GST_LOG_OBJECT (mux, "event tags: %" GST_PTR_FORMAT, mux->event_tags);
GST_LOG_OBJECT (mux, "set tags: %" GST_PTR_FORMAT, tagsetter_tags);
taglist = gst_tag_list_merge (tagsetter_tags, mux->event_tags, merge_mode);
GST_LOG_OBJECT (mux, "final tags: %" GST_PTR_FORMAT, taglist);
klass = GST_TAG_LIB_MUX_CLASS (G_OBJECT_GET_CLASS (mux));
if (klass->render_tag == NULL)
goto no_vfunc;
buffer = klass->render_tag (mux, taglist);
if (buffer == NULL)
goto render_error;
mux->tag_size = GST_BUFFER_SIZE (buffer);
GST_LOG_OBJECT (mux, "tag size = %" G_GSIZE_FORMAT " bytes", mux->tag_size);
/* Send newsegment event from byte position 0, so the tag really gets
* written to the start of the file, independent of the upstream segment */
gst_pad_push_event (mux->srcpad,
gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_BYTES, 0, -1, 0));
/* Send an event about the new tags to downstream elements */
/* gst_event_new_tag takes ownership of the list, so no need to unref it */
event = gst_event_new_tag (taglist);
gst_pad_push_event (mux->srcpad, event);
GST_BUFFER_OFFSET (buffer) = 0;
return buffer;
no_vfunc:
{
GST_ERROR_OBJECT (mux, "Subclass does not implement render_tag vfunc!");
gst_tag_list_free (taglist);
return NULL;
}
render_error:
{
GST_ERROR_OBJECT (mux, "Failed to render tag");
gst_tag_list_free (taglist);
return NULL;
}
}
static gboolean
gst_musepack_stream_init (GstMusepackDec * musepackdec)
{
mpc_streaminfo i;
GstTagList *tags;
GstCaps *caps;
gchar *stream_id;
/* set up reading */
gst_musepack_init_reader (musepackdec->r, musepackdec);
musepackdec->d = mpc_demux_init (musepackdec->r);
if (!musepackdec->d) {
GST_ELEMENT_ERROR (musepackdec, STREAM, WRONG_TYPE, (NULL), (NULL));
return FALSE;
}
mpc_demux_get_info (musepackdec->d, &i);
stream_id = gst_pad_create_stream_id (musepackdec->srcpad,
GST_ELEMENT_CAST (musepackdec), NULL);
gst_pad_push_event (musepackdec->srcpad,
gst_event_new_stream_start (stream_id));
g_free (stream_id);
/* capsnego */
caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, GST_MPC_FORMAT,
"layout", G_TYPE_STRING, "interleaved",
"channels", G_TYPE_INT, i.channels,
"rate", G_TYPE_INT, i.sample_freq, NULL);
gst_pad_use_fixed_caps (musepackdec->srcpad);
if (!gst_pad_set_caps (musepackdec->srcpad, caps)) {
GST_ELEMENT_ERROR (musepackdec, CORE, NEGOTIATION, (NULL), (NULL));
return FALSE;
}
g_atomic_int_set (&musepackdec->bps, 4 * i.channels);
g_atomic_int_set (&musepackdec->rate, i.sample_freq);
musepackdec->segment.position = 0;
musepackdec->segment.duration = mpc_streaminfo_get_length_samples (&i);
/* send basic tags */
tags = gst_tag_list_new_empty ();
gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE,
GST_TAG_AUDIO_CODEC, "Musepack", NULL);
if (i.encoder[0] != '\0' && i.encoder_version > 0) {
gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE,
GST_TAG_ENCODER, i.encoder,
GST_TAG_ENCODER_VERSION, i.encoder_version, NULL);
}
if (i.bitrate > 0) {
gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE,
GST_TAG_BITRATE, i.bitrate, NULL);
} else if (i.average_bitrate > 0.0) {
gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE,
GST_TAG_BITRATE, (guint) i.average_bitrate, NULL);
}
if (i.gain_title != 0 || i.gain_album != 0) {
gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE,
GST_TAG_TRACK_GAIN, (gdouble) i.gain_title / 100.0,
GST_TAG_ALBUM_GAIN, (gdouble) i.gain_album / 100.0, NULL);
}
if (i.peak_title != 0 && i.peak_title != 32767 &&
i.peak_album != 0 && i.peak_album != 32767) {
gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE,
GST_TAG_TRACK_PEAK, (gdouble) i.peak_title / 32767.0,
GST_TAG_ALBUM_PEAK, (gdouble) i.peak_album / 32767.0, NULL);
}
GST_LOG_OBJECT (musepackdec, "Posting tags: %" GST_PTR_FORMAT, tags);
gst_pad_push_event (musepackdec->srcpad, gst_event_new_tag (tags));
return TRUE;
}
static GstFlowReturn
gst_lv2_source_fill (GstBaseSrc * base, guint64 offset,
guint length, GstBuffer * buffer)
{
GstLV2Source *lv2 = (GstLV2Source *) base;
GstLV2SourceClass *klass = (GstLV2SourceClass *) GST_BASE_SRC_GET_CLASS (lv2);
GstLV2Class *lv2_class = &klass->lv2;
GstLV2Group *lv2_group;
GstLV2Port *lv2_port;
GstClockTime next_time;
gint64 next_sample, next_byte;
guint bytes, samples;
GstElementClass *eclass;
GstMapInfo map;
gint samplerate, bpf;
guint j, k, l;
gfloat *out = NULL, *cv = NULL, *mem;
gfloat val;
/* example for tagging generated data */
if (!lv2->tags_pushed) {
GstTagList *taglist;
taglist = gst_tag_list_new (GST_TAG_DESCRIPTION, "lv2 wave", NULL);
eclass = GST_ELEMENT_CLASS (parent_class);
if (eclass->send_event)
eclass->send_event (GST_ELEMENT (base), gst_event_new_tag (taglist));
else
gst_tag_list_unref (taglist);
lv2->tags_pushed = TRUE;
}
if (lv2->eos_reached) {
GST_INFO_OBJECT (lv2, "eos");
return GST_FLOW_EOS;
}
samplerate = GST_AUDIO_INFO_RATE (&lv2->info);
bpf = GST_AUDIO_INFO_BPF (&lv2->info);
/* if no length was given, use our default length in samples otherwise convert
* the length in bytes to samples. */
if (length == -1)
samples = lv2->samples_per_buffer;
else
samples = length / bpf;
/* if no offset was given, use our next logical byte */
if (offset == -1)
offset = lv2->next_byte;
/* now see if we are at the byteoffset we think we are */
if (offset != lv2->next_byte) {
GST_DEBUG_OBJECT (lv2, "seek to new offset %" G_GUINT64_FORMAT, offset);
/* we have a discont in the expected sample offset, do a 'seek' */
lv2->next_sample = offset / bpf;
lv2->next_time =
gst_util_uint64_scale_int (lv2->next_sample, GST_SECOND, samplerate);
lv2->next_byte = offset;
}
/* check for eos */
if (lv2->check_seek_stop &&
(lv2->sample_stop > lv2->next_sample) &&
(lv2->sample_stop < lv2->next_sample + samples)
) {
/* calculate only partial buffer */
lv2->generate_samples_per_buffer = lv2->sample_stop - lv2->next_sample;
next_sample = lv2->sample_stop;
lv2->eos_reached = TRUE;
GST_INFO_OBJECT (lv2, "eos reached");
} else {
/* calculate full buffer */
lv2->generate_samples_per_buffer = samples;
next_sample = lv2->next_sample + (lv2->reverse ? (-samples) : samples);
}
bytes = lv2->generate_samples_per_buffer * bpf;
next_byte = lv2->next_byte + (lv2->reverse ? (-bytes) : bytes);
next_time = gst_util_uint64_scale_int (next_sample, GST_SECOND, samplerate);
GST_LOG_OBJECT (lv2, "samplerate %d", samplerate);
GST_LOG_OBJECT (lv2,
"next_sample %" G_GINT64_FORMAT ", ts %" GST_TIME_FORMAT, next_sample,
GST_TIME_ARGS (next_time));
gst_buffer_set_size (buffer, bytes);
GST_BUFFER_OFFSET (buffer) = lv2->next_sample;
GST_BUFFER_OFFSET_END (buffer) = next_sample;
if (!lv2->reverse) {
GST_BUFFER_TIMESTAMP (buffer) = lv2->timestamp_offset + lv2->next_time;
GST_BUFFER_DURATION (buffer) = next_time - lv2->next_time;
} else {
GST_BUFFER_TIMESTAMP (buffer) = lv2->timestamp_offset + next_time;
GST_BUFFER_DURATION (buffer) = lv2->next_time - next_time;
}
gst_object_sync_values (GST_OBJECT (lv2), GST_BUFFER_TIMESTAMP (buffer));
lv2->next_time = next_time;
lv2->next_sample = next_sample;
lv2->next_byte = next_byte;
GST_LOG_OBJECT (lv2, "generating %u samples at ts %" GST_TIME_FORMAT,
samples, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
/* multi channel outputs */
lv2_group = &lv2_class->out_group;
if (lv2_group->ports->len > 1) {
out = g_new0 (gfloat, samples * lv2_group->ports->len);
for (j = 0; j < lv2_group->ports->len; ++j) {
lv2_port = &g_array_index (lv2_group->ports, GstLV2Port, j);
lilv_instance_connect_port (lv2->lv2.instance, lv2_port->index,
out + (j * samples));
GST_LOG_OBJECT (lv2, "connected port %d/%d", j, lv2_group->ports->len);
}
} else {
lv2_port = &g_array_index (lv2_group->ports, GstLV2Port, 0);
lilv_instance_connect_port (lv2->lv2.instance, lv2_port->index,
(gfloat *) map.data);
GST_LOG_OBJECT (lv2, "connected port 0");
}
/* cv ports */
cv = g_new (gfloat, samples * lv2_class->num_cv_in);
for (j = k = 0; j < lv2_class->control_in_ports->len; j++) {
lv2_port = &g_array_index (lv2_class->control_in_ports, GstLV2Port, j);
if (lv2_port->type != GST_LV2_PORT_CV)
continue;
mem = cv + (k * samples);
val = lv2->lv2.ports.control.in[j];
/* FIXME: use gst_control_binding_get_value_array */
for (l = 0; l < samples; l++)
mem[l] = val;
lilv_instance_connect_port (lv2->lv2.instance, lv2_port->index, mem);
k++;
}
lilv_instance_run (lv2->lv2.instance, samples);
if (lv2_group->ports->len > 1) {
gst_lv2_source_interleave_data (lv2_group->ports->len,
(gfloat *) map.data, samples, out);
g_free (out);
}
g_free (cv);
gst_buffer_unmap (buffer, &map);
return GST_FLOW_OK;
}
static gboolean
gme_setup (GstGmeDec * gme)
{
gme_info_t *info;
gme_err_t gme_err = NULL;
GstTagList *taglist;
guint64 total_duration;
guint64 fade_time;
GstBuffer *buffer;
GstSegment seg;
GstMapInfo map;
if (!gst_adapter_available (gme->adapter) || !gme_negotiate (gme)) {
return FALSE;
}
buffer =
gst_adapter_take_buffer (gme->adapter,
gst_adapter_available (gme->adapter));
gst_buffer_map (buffer, &map, GST_MAP_READ);
gme_err = gme_open_data (map.data, map.size, &gme->player, 32000);
gst_buffer_unmap (buffer, &map);
gst_buffer_unref (buffer);
if (gme_err || !gme->player) {
if (gme->player) {
gme_delete (gme->player);
gme->player = NULL;
}
GST_ELEMENT_ERROR (gme, STREAM, DEMUX, (NULL), ("%s", gme_err));
return FALSE;
}
gme_err = gme_track_info (gme->player, &info, 0);
taglist = gst_tag_list_new_empty ();
if (info->song && *info->song)
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_TITLE,
info->song, NULL);
if (info->author && *info->author)
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_ARTIST,
info->author, NULL);
/* Prefer the name of the official soundtrack over the name of the game (since this is
* how track numbers are derived)
*/
if (info->game && *info->game)
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_ALBUM, info->game,
NULL);
if (info->comment && *info->comment)
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_COMMENT,
info->comment, NULL);
if (info->dumper && *info->dumper)
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_CONTACT,
info->dumper, NULL);
if (info->copyright && *info->copyright)
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_COPYRIGHT,
info->copyright, NULL);
if (info->system && *info->system)
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_ENCODER,
info->system, NULL);
gme->total_duration = total_duration =
gst_util_uint64_scale_int (info->play_length + (info->loop_length >
0 ? 8000 : 0), GST_MSECOND, 1);
fade_time = info->loop_length > 0 ? info->play_length : 0;
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
GST_TAG_DURATION, total_duration, NULL);
gst_pad_push_event (gme->srcpad, gst_event_new_tag (taglist));
g_free (info);
#ifdef HAVE_LIBGME_ACCURACY
/* TODO: Is it worth it to make this optional? */
gme_enable_accuracy (gme->player, 1);
#endif
gme_start_track (gme->player, 0);
if (fade_time)
gme_set_fade (gme->player, fade_time);
gst_segment_init (&seg, GST_FORMAT_TIME);
gst_pad_push_event (gme->srcpad, gst_event_new_segment (&seg));
gst_pad_start_task (gme->srcpad, (GstTaskFunction) gst_gme_play, gme->srcpad,
NULL);
gme->initialized = TRUE;
gme->seeking = FALSE;
gme->seekpoint = 0;
return gme->initialized;
}
static void
gst_rg_analysis_handle_eos (GstRgAnalysis * filter)
{
gboolean album_processing = (filter->num_tracks > 0);
gboolean album_finished = (filter->num_tracks == 1);
gboolean album_skipping = album_processing && filter->skip;
filter->has_track_gain = FALSE;
filter->has_track_peak = FALSE;
if (album_finished) {
filter->ignore_tags = FALSE;
filter->skip = FALSE;
filter->has_album_gain = FALSE;
filter->has_album_peak = FALSE;
} else if (!album_skipping) {
filter->skip = FALSE;
}
/* We might have just fully processed a track because it has
* incomplete tags. If we do album processing and allow skipping
* (not forced), prevent switching to skipping if a later track with
* full tags comes along: */
if (!filter->forced && album_processing && !album_finished)
filter->ignore_tags = TRUE;
if (!filter->skip) {
GstTagList *tag_list = NULL;
gboolean track_success;
gboolean album_success = FALSE;
track_success = gst_rg_analysis_track_result (filter, &tag_list);
if (album_finished)
album_success = gst_rg_analysis_album_result (filter, &tag_list);
else if (!album_processing)
rg_analysis_reset_album (filter->ctx);
if (track_success || album_success) {
GST_LOG_OBJECT (filter, "posting tag list with results");
gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND,
GST_TAG_REFERENCE_LEVEL, filter->reference_level, NULL);
/* This steals our reference to the list: */
gst_pad_push_event (GST_BASE_TRANSFORM_SRC_PAD (GST_BASE_TRANSFORM
(filter)), gst_event_new_tag (gst_tag_list_ref (tag_list)));
}
}
if (album_processing) {
filter->num_tracks--;
if (!album_finished) {
GST_DEBUG_OBJECT (filter, "album not finished yet (num-tracks is now %u)",
filter->num_tracks);
} else {
GST_DEBUG_OBJECT (filter, "album finished (num-tracks is now 0)");
}
}
if (album_processing)
g_object_notify (G_OBJECT (filter), "num-tracks");
}
static GstFlowReturn
gst_ladspa_source_type_fill (GstBaseSrc * base, guint64 offset,
guint length, GstBuffer * buffer)
{
GstLADSPASource *ladspa = GST_LADSPA_SOURCE (base);
GstClockTime next_time;
gint64 next_sample, next_byte;
gint bytes, samples;
GstElementClass *eclass;
GstMapInfo map;
gint samplerate, bpf;
/* example for tagging generated data */
if (!ladspa->tags_pushed) {
GstTagList *taglist;
taglist = gst_tag_list_new (GST_TAG_DESCRIPTION, "ladspa wave", NULL);
eclass = GST_ELEMENT_CLASS (gst_ladspa_source_type_parent_class);
if (eclass->send_event)
eclass->send_event (GST_ELEMENT (base), gst_event_new_tag (taglist));
else
gst_tag_list_unref (taglist);
ladspa->tags_pushed = TRUE;
}
if (ladspa->eos_reached) {
GST_INFO_OBJECT (ladspa, "eos");
return GST_FLOW_EOS;
}
samplerate = GST_AUDIO_INFO_RATE (&ladspa->info);
bpf = GST_AUDIO_INFO_BPF (&ladspa->info);
/* if no length was given, use our default length in samples otherwise convert
* the length in bytes to samples. */
if (length == -1)
samples = ladspa->samples_per_buffer;
else
samples = length / bpf;
/* if no offset was given, use our next logical byte */
if (offset == -1)
offset = ladspa->next_byte;
/* now see if we are at the byteoffset we think we are */
if (offset != ladspa->next_byte) {
GST_DEBUG_OBJECT (ladspa, "seek to new offset %" G_GUINT64_FORMAT, offset);
/* we have a discont in the expected sample offset, do a 'seek' */
ladspa->next_sample = offset / bpf;
ladspa->next_time =
gst_util_uint64_scale_int (ladspa->next_sample, GST_SECOND, samplerate);
ladspa->next_byte = offset;
}
/* check for eos */
if (ladspa->check_seek_stop &&
(ladspa->sample_stop > ladspa->next_sample) &&
(ladspa->sample_stop < ladspa->next_sample + samples)
) {
/* calculate only partial buffer */
ladspa->generate_samples_per_buffer =
ladspa->sample_stop - ladspa->next_sample;
next_sample = ladspa->sample_stop;
ladspa->eos_reached = TRUE;
} else {
/* calculate full buffer */
ladspa->generate_samples_per_buffer = samples;
next_sample =
ladspa->next_sample + (ladspa->reverse ? (-samples) : samples);
}
bytes = ladspa->generate_samples_per_buffer * bpf;
next_byte = ladspa->next_byte + (ladspa->reverse ? (-bytes) : bytes);
next_time = gst_util_uint64_scale_int (next_sample, GST_SECOND, samplerate);
GST_LOG_OBJECT (ladspa, "samplerate %d", samplerate);
GST_LOG_OBJECT (ladspa,
"next_sample %" G_GINT64_FORMAT ", ts %" GST_TIME_FORMAT, next_sample,
GST_TIME_ARGS (next_time));
gst_buffer_set_size (buffer, bytes);
GST_BUFFER_OFFSET (buffer) = ladspa->next_sample;
GST_BUFFER_OFFSET_END (buffer) = next_sample;
if (!ladspa->reverse) {
GST_BUFFER_TIMESTAMP (buffer) =
ladspa->timestamp_offset + ladspa->next_time;
GST_BUFFER_DURATION (buffer) = next_time - ladspa->next_time;
} else {
GST_BUFFER_TIMESTAMP (buffer) = ladspa->timestamp_offset + next_time;
GST_BUFFER_DURATION (buffer) = ladspa->next_time - next_time;
}
gst_object_sync_values (GST_OBJECT (ladspa), GST_BUFFER_TIMESTAMP (buffer));
ladspa->next_time = next_time;
ladspa->next_sample = next_sample;
ladspa->next_byte = next_byte;
GST_LOG_OBJECT (ladspa, "generating %u samples at ts %" GST_TIME_FORMAT,
ladspa->generate_samples_per_buffer,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
gst_ladspa_transform (&ladspa->ladspa, map.data,
ladspa->generate_samples_per_buffer, NULL);
gst_buffer_unmap (buffer, &map);
return GST_FLOW_OK;
}
static GstFlowReturn
gst_multipart_demux_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstMultipartDemux *multipart;
GstAdapter *adapter;
gint size = 1;
GstFlowReturn res;
multipart = GST_MULTIPART_DEMUX (parent);
adapter = multipart->adapter;
res = GST_FLOW_OK;
if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
GSList *l;
for (l = multipart->srcpads; l != NULL; l = l->next) {
GstMultipartPad *srcpad = l->data;
srcpad->discont = TRUE;
}
gst_adapter_clear (adapter);
}
gst_adapter_push (adapter, buf);
while (gst_adapter_available (adapter) > 0) {
GstMultipartPad *srcpad;
GstBuffer *outbuf;
gboolean created;
gint datalen;
if (G_UNLIKELY (!multipart->header_completed)) {
if ((size = multipart_parse_header (multipart)) < 0) {
goto nodata;
} else {
gst_adapter_flush (adapter, size);
multipart->header_completed = TRUE;
}
}
if ((size = multipart_find_boundary (multipart, &datalen)) < 0) {
goto nodata;
}
/* Invalidate header info */
multipart->header_completed = FALSE;
multipart->content_length = -1;
if (G_UNLIKELY (datalen <= 0)) {
GST_DEBUG_OBJECT (multipart, "skipping empty content.");
gst_adapter_flush (adapter, size - datalen);
} else {
GstClockTime ts;
srcpad =
gst_multipart_find_pad_by_mime (multipart,
multipart->mime_type, &created);
ts = gst_adapter_prev_pts (adapter, NULL);
outbuf = gst_adapter_take_buffer (adapter, datalen);
gst_adapter_flush (adapter, size - datalen);
if (created) {
GstTagList *tags;
GstSegment segment;
gst_segment_init (&segment, GST_FORMAT_TIME);
/* Push new segment, first buffer has 0 timestamp */
gst_pad_push_event (srcpad->pad, gst_event_new_segment (&segment));
tags = gst_tag_list_new (GST_TAG_CONTAINER_FORMAT, "Multipart", NULL);
gst_tag_list_set_scope (tags, GST_TAG_SCOPE_GLOBAL);
gst_pad_push_event (srcpad->pad, gst_event_new_tag (tags));
}
outbuf = gst_buffer_make_writable (outbuf);
if (srcpad->last_ts == GST_CLOCK_TIME_NONE || srcpad->last_ts != ts) {
GST_BUFFER_TIMESTAMP (outbuf) = ts;
srcpad->last_ts = ts;
} else {
GST_BUFFER_TIMESTAMP (outbuf) = GST_CLOCK_TIME_NONE;
}
if (srcpad->discont) {
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
srcpad->discont = FALSE;
} else {
GST_BUFFER_FLAG_UNSET (outbuf, GST_BUFFER_FLAG_DISCONT);
}
GST_DEBUG_OBJECT (multipart,
"pushing buffer with timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
res = gst_pad_push (srcpad->pad, outbuf);
res = gst_multipart_combine_flows (multipart, srcpad, res);
if (res != GST_FLOW_OK)
break;
}
}
nodata:
if (G_UNLIKELY (size == MULTIPART_DATA_ERROR))
return GST_FLOW_ERROR;
if (G_UNLIKELY (size == MULTIPART_DATA_EOS))
return GST_FLOW_EOS;
return res;
}
static gboolean gst_uade_raw_dec_load_from_custom(GstNonstreamAudioDecoder *dec, guint initial_subsong, G_GNUC_UNUSED GstNonstreamAudioSubsongMode initial_subsong_mode, GstClockTime *initial_position, GstNonstreamAudioOutputMode *initial_output_mode, gint *initial_num_loops)
{
GstUadeRawDec *uade_raw_dec;
int ret;
GstTagList *tags;
struct uade_config *config;
gchar *tmpstr;
uade_raw_dec = GST_UADE_RAW_DEC(dec);
g_assert(uade_raw_dec->state == NULL);
if (uade_raw_dec->location == NULL)
{
GST_ERROR_OBJECT(uade_raw_dec, "no location set -> nothing to play");
return FALSE;
}
GST_TRACE_OBJECT(uade_raw_dec, "attempting to load music file \"%s\"", uade_raw_dec->location);
config = uade_new_config();
uade_config_set_option(config, UC_ONE_SUBSONG, NULL);
uade_config_set_option(config, UC_NO_EP_END, NULL);
uade_config_set_option(config, UC_UADECORE_FILE, uade_raw_dec->uadecore_file);
uade_config_set_option(config, UC_BASE_DIR, uade_raw_dec->base_directory);
switch (uade_raw_dec->filter_type)
{
case GST_UADE_FILTER_TYPE_A500: uade_config_set_option(config, UC_FILTER_TYPE, "a500"); break;
case GST_UADE_FILTER_TYPE_A1200: uade_config_set_option(config, UC_FILTER_TYPE, "a1200"); break;
default: break;
}
switch (uade_raw_dec->headphone_mode)
{
case GST_UADE_HEADPHONE_MODE_NONE: uade_config_set_option(config, UC_NO_HEADPHONES, NULL); break;
case GST_UADE_HEADPHONE_MODE_1: uade_config_set_option(config, UC_HEADPHONES, NULL); break;
case GST_UADE_HEADPHONE_MODE_2: uade_config_set_option(config, UC_HEADPHONES2, NULL); break;
default: break;
}
if (!(uade_raw_dec->use_filter))
uade_config_set_option(config, UC_NO_FILTER, NULL); /* this must be called AFTER the filter type is set */
tmpstr = g_strdup_printf("%f", uade_raw_dec->gain);
uade_config_set_option(config, UC_GAIN, tmpstr);
g_free(tmpstr);
if (!(uade_raw_dec->use_postprocessing))
uade_config_set_option(config, UC_NO_POSTPROCESSING, NULL);
tmpstr = g_strdup_printf("%f", uade_raw_dec->panning);
uade_config_set_option(config, UC_PANNING_VALUE, tmpstr);
g_free(tmpstr);
uade_raw_dec->state = uade_new_state(config);
free(config);
/* Set output format */
gst_nonstream_audio_decoder_set_output_format_simple(
GST_NONSTREAM_AUDIO_DECODER(uade_raw_dec),
uade_get_sampling_rate(uade_raw_dec->state),
GST_AUDIO_FORMAT_S16,
2
);
ret = uade_play(uade_raw_dec->location, -1, uade_raw_dec->state);
if (!ret)
{
GST_ERROR_OBJECT(uade_raw_dec, "uade_play failed");
return FALSE;
}
GST_TRACE_OBJECT(uade_raw_dec, "loading successful, retrieving song information");
uade_raw_dec->playback_started = TRUE;
uade_raw_dec->info = uade_get_song_info(uade_raw_dec->state);
if (uade_raw_dec->info == NULL)
{
GST_ERROR_OBJECT(uade_raw_dec, "uade_get_song_info failed");
return FALSE;
}
GST_INFO_OBJECT(uade_raw_dec, "min subsong: %d max subsong: %d", uade_raw_dec->info->subsongs.min, uade_raw_dec->info->subsongs.max);
uade_raw_dec->current_subsong = CLAMP(((int)initial_subsong) + uade_raw_dec->info->subsongs.min, uade_raw_dec->info->subsongs.min, uade_raw_dec->info->subsongs.max);
if (uade_seek(UADE_SEEK_SUBSONG_RELATIVE, 0, uade_raw_dec->current_subsong, uade_raw_dec->state) != 0)
{
GST_ERROR_OBJECT(uade_raw_dec, "seeking to initial subsong failed");
return FALSE;
}
*initial_position = 0;
*initial_output_mode = GST_NONSTREM_AUDIO_OUTPUT_MODE_STEADY;
tags = gst_tag_list_new_empty();
if (uade_raw_dec->info->modulename[0] != 0)
gst_tag_list_add(tags, GST_TAG_MERGE_APPEND, GST_TAG_TITLE, uade_raw_dec->info->modulename, NULL);
if (uade_raw_dec->info->formatname[0] != 0)
gst_tag_list_add(tags, GST_TAG_MERGE_APPEND, GST_TAG_CONTAINER_FORMAT, uade_raw_dec->info->formatname, NULL);
if (uade_raw_dec->info->playername[0] != 0)
gst_tag_list_add(tags, GST_TAG_MERGE_APPEND, GST_TAG_APPLICATION_NAME, uade_raw_dec->info->playername, NULL);
gst_pad_push_event(GST_NONSTREAM_AUDIO_DECODER_SRC_PAD(uade_raw_dec), gst_event_new_tag(tags));
return TRUE;
}
