static void
no_more_pads (GstElement * element, GstSplitMuxPartReader * reader)
{
GstClockTime duration = GST_CLOCK_TIME_NONE;
GList *cur;
/* Query the minimum duration of any pad in this piece and store it.
* FIXME: Only consider audio and video */
SPLITMUX_PART_LOCK (reader);
for (cur = g_list_first (reader->pads); cur != NULL; cur = g_list_next (cur)) {
GstPad *target = GST_PAD_CAST (cur->data);
if (target) {
gint64 cur_duration;
if (gst_pad_peer_query_duration (target, GST_FORMAT_TIME, &cur_duration)) {
GST_INFO_OBJECT (reader,
"file %s pad %" GST_PTR_FORMAT " duration %" GST_TIME_FORMAT,
reader->path, target, GST_TIME_ARGS (cur_duration));
if (cur_duration < duration)
duration = cur_duration;
}
}
}
GST_INFO_OBJECT (reader, "file %s duration %" GST_TIME_FORMAT,
reader->path, GST_TIME_ARGS (duration));
reader->duration = (GstClockTime) duration;
reader->no_more_pads = TRUE;
check_if_pads_collected (reader);
SPLITMUX_PART_UNLOCK (reader);
}
static void
gst_wavpack_dec_post_tags (GstWavpackDec * dec)
{
GstTagList *list;
GstFormat format_time = GST_FORMAT_TIME, format_bytes = GST_FORMAT_BYTES;
gint64 duration, size;
/* try to estimate the average bitrate */
if (gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
format_bytes, &size) &&
gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
format_time, &duration) && size > 0 && duration > 0) {
guint64 bitrate;
list = gst_tag_list_new_empty ();
bitrate = gst_util_uint64_scale (size, 8 * GST_SECOND, duration);
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE,
(guint) bitrate, NULL);
gst_audio_decoder_merge_tags (GST_AUDIO_DECODER (dec), list,
GST_TAG_MERGE_REPLACE);
}
}
/***********************************************************************************
* Query
***********************************************************************************/
static gboolean mpegts_demuxer_sink_query (GstPad *pad, GstObject *parent, GstQuery *query)
{
gboolean result = TRUE;
MpegTSDemuxer *demuxer = MPEGTS_DEMUXER(parent);
switch (GST_QUERY_TYPE(query))
{
case GST_QUERY_DURATION:
{
GstFormat format;
gst_query_parse_duration(query, &format, NULL);
if (format == GST_FORMAT_TIME)
result = gst_pad_peer_query(pad, query);
else if (format == GST_FORMAT_BYTES)
{
g_mutex_lock(&demuxer->lock);
int bit_rate = (demuxer->context != NULL) ? demuxer->context->bit_rate : 0;
g_mutex_unlock(&demuxer->lock);
if (bit_rate > 0)
{
gint64 duration = GST_CLOCK_TIME_NONE;
if (gst_pad_peer_query_duration(pad, GST_FORMAT_TIME, &duration))
{
// Approximate duration in bytes for a certain time duration and bit rate.
if (duration != GST_CLOCK_TIME_NONE)
duration = (double)(duration * bit_rate) / GST_SECOND / 8;
gst_query_set_duration(query, format, duration);
}
else
result = FALSE;
}
else
result = gst_pad_peer_query(pad, query);
}
break;
}
default:
result = gst_pad_peer_query(pad, query);
break;
}
return result;
}
static gboolean
gst_raw_parse_sink_activatemode (GstPad * sinkpad, GstObject * parent,
GstPadMode mode, gboolean active)
{
GstRawParse *rp = GST_RAW_PARSE (parent);
gboolean result;
switch (mode) {
case GST_PAD_MODE_PULL:
if (active) {
GstFormat format;
gint64 duration;
/* get the duration in bytes */
format = GST_FORMAT_BYTES;
result = gst_pad_peer_query_duration (sinkpad, format, &duration);
if (result) {
GST_DEBUG_OBJECT (rp, "got duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (duration));
rp->upstream_length = duration;
/* convert to time */
gst_raw_parse_convert (rp, format, duration, GST_FORMAT_TIME,
&duration);
} else {
rp->upstream_length = -1;
duration = -1;
}
rp->segment.duration = duration;
rp->push_stream_start = TRUE;
result = gst_raw_parse_handle_seek_pull (rp, NULL);
rp->mode = mode;
} else {
result = gst_pad_stop_task (sinkpad);
}
return result;
case GST_PAD_MODE_PUSH:
rp->mode = mode;
return TRUE;
default:
return FALSE;
}
}
static gboolean
gst_interleave_src_query_duration (GstInterleave * self, GstQuery * query)
{
gint64 max;
gboolean res;
GstFormat format;
GstIterator *it;
gboolean done;
/* parse format */
gst_query_parse_duration (query, &format, NULL);
max = -1;
res = TRUE;
done = FALSE;
/* Take maximum of all durations */
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (self));
while (!done) {
GstIteratorResult ires;
GValue item = { 0, };
ires = gst_iterator_next (it, &item);
switch (ires) {
case GST_ITERATOR_DONE:
done = TRUE;
break;
case GST_ITERATOR_OK:
{
GstPad *pad = GST_PAD_CAST (g_value_dup_object (&item));
gint64 duration;
/* ask sink peer for duration */
res &= gst_pad_peer_query_duration (pad, format, &duration);
/* take max from all valid return values */
if (res) {
/* valid unknown length, stop searching */
if (duration == -1) {
max = duration;
done = TRUE;
}
/* else see if bigger than current max */
else if (duration > max)
max = duration;
}
gst_object_unref (pad);
g_value_unset (&item);
break;
}
case GST_ITERATOR_RESYNC:
max = -1;
res = TRUE;
gst_iterator_resync (it);
break;
default:
res = FALSE;
done = TRUE;
break;
}
}
gst_iterator_free (it);
if (res) {
/* If in bytes format we have to multiply with the number of channels
* to get the correct results. All other formats should be fine */
if (format == GST_FORMAT_BYTES && max != -1)
max *= self->channels;
/* and store the max */
GST_DEBUG_OBJECT (self, "Total duration in format %s: %"
GST_TIME_FORMAT, gst_format_get_name (format), GST_TIME_ARGS (max));
gst_query_set_duration (query, format, max);
}
return res;
}
static gboolean
gst_frei0r_mixer_src_query_duration (GstFrei0rMixer * self, GstQuery * query)
{
gint64 min;
gboolean res;
GstFormat format;
GstIterator *it;
gboolean done;
/* parse format */
gst_query_parse_duration (query, &format, NULL);
min = -1;
res = TRUE;
done = FALSE;
/* Take minimum of all durations */
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (self));
while (!done) {
GstIteratorResult ires;
GValue item = { 0 };
ires = gst_iterator_next (it, &item);
switch (ires) {
case GST_ITERATOR_DONE:
done = TRUE;
break;
case GST_ITERATOR_OK:
{
GstPad *pad = g_value_get_object (&item);
gint64 duration;
/* ask sink peer for duration */
res &= gst_pad_peer_query_duration (pad, format, &duration);
/* take min from all valid return values */
if (res) {
/* valid unknown length, stop searching */
if (duration == -1) {
min = duration;
done = TRUE;
}
/* else see if smaller than current min */
else if (duration < min)
min = duration;
}
g_value_reset (&item);
break;
}
case GST_ITERATOR_RESYNC:
min = -1;
res = TRUE;
gst_iterator_resync (it);
break;
default:
res = FALSE;
done = TRUE;
break;
}
g_value_unset (&item);
}
gst_iterator_free (it);
if (res) {
/* and store the min */
GST_DEBUG_OBJECT (self, "Total duration in format %s: %"
GST_TIME_FORMAT, gst_format_get_name (format), GST_TIME_ARGS (min));
gst_query_set_duration (query, format, min);
}
return res;
}
static void
gst_raw_parse_loop (GstElement * element)
{
GstRawParse *rp = GST_RAW_PARSE (element);
GstRawParseClass *rp_class = GST_RAW_PARSE_GET_CLASS (rp);
GstFlowReturn ret;
GstBuffer *buffer;
gint size;
if (!gst_raw_parse_set_src_caps (rp))
goto no_caps;
if (rp->start_segment) {
GST_DEBUG_OBJECT (rp, "sending start segment");
gst_pad_push_event (rp->srcpad, rp->start_segment);
rp->start_segment = NULL;
}
if (rp_class->multiple_frames_per_buffer && rp->framesize < 4096)
size = 4096 - (4096 % rp->framesize);
else
size = rp->framesize;
if (rp->segment.rate >= 0) {
if (rp->offset + size > rp->upstream_length) {
GstFormat fmt = GST_FORMAT_BYTES;
if (!gst_pad_peer_query_duration (rp->sinkpad, fmt, &rp->upstream_length)) {
GST_WARNING_OBJECT (rp,
"Could not get upstream duration, trying to pull frame by frame");
size = rp->framesize;
} else if (rp->upstream_length < rp->offset + rp->framesize) {
ret = GST_FLOW_EOS;
goto pause;
} else if (rp->offset + size > rp->upstream_length) {
size = rp->upstream_length - rp->offset;
size -= size % rp->framesize;
}
}
} else {
if (rp->offset == 0) {
ret = GST_FLOW_EOS;
goto pause;
} else if (rp->offset < size) {
size -= rp->offset;
}
rp->offset -= size;
}
buffer = NULL;
ret = gst_pad_pull_range (rp->sinkpad, rp->offset, size, &buffer);
if (ret != GST_FLOW_OK) {
GST_DEBUG_OBJECT (rp, "pull_range (%" G_GINT64_FORMAT ", %u) "
"failed, flow: %s", rp->offset, size, gst_flow_get_name (ret));
buffer = NULL;
goto pause;
}
if (gst_buffer_get_size (buffer) < size) {
GST_DEBUG_OBJECT (rp, "Short read at offset %" G_GINT64_FORMAT
", got only %" G_GSIZE_FORMAT " of %u bytes", rp->offset,
gst_buffer_get_size (buffer), size);
if (size > rp->framesize) {
gst_buffer_set_size (buffer, gst_buffer_get_size (buffer) -
gst_buffer_get_size (buffer) % rp->framesize);
} else {
gst_buffer_unref (buffer);
buffer = NULL;
ret = GST_FLOW_EOS;
goto pause;
}
}
ret = gst_raw_parse_push_buffer (rp, buffer);
if (ret != GST_FLOW_OK)
goto pause;
return;
/* ERRORS */
no_caps:
{
GST_ERROR_OBJECT (rp, "could not negotiate caps");
ret = GST_FLOW_NOT_NEGOTIATED;
goto pause;
}
pause:
{
const gchar *reason = gst_flow_get_name (ret);
GST_LOG_OBJECT (rp, "pausing task, reason %s", reason);
gst_pad_pause_task (rp->sinkpad);
if (ret == GST_FLOW_EOS) {
if (rp->segment.flags & GST_SEEK_FLAG_SEGMENT) {
GstClockTime stop;
GST_LOG_OBJECT (rp, "Sending segment done");
if ((stop = rp->segment.stop) == -1)
stop = rp->segment.duration;
gst_element_post_message (GST_ELEMENT_CAST (rp),
gst_message_new_segment_done (GST_OBJECT_CAST (rp),
rp->segment.format, stop));
gst_pad_push_event (rp->srcpad,
gst_event_new_segment_done (rp->segment.format, stop));
} else {
GST_LOG_OBJECT (rp, "Sending EOS, at end of stream");
gst_pad_push_event (rp->srcpad, gst_event_new_eos ());
}
} else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
GST_ELEMENT_ERROR (rp, STREAM, FAILED,
("Internal data stream error."),
("stream stopped, reason %s", reason));
gst_pad_push_event (rp->srcpad, gst_event_new_eos ());
}
return;
}
}
static GstFlowReturn
mpegts_base_scan (MpegTSBase * base)
{
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *buf = NULL;
guint i;
gboolean done = FALSE;
MpegTSPacketizerPacketReturn pret;
gint64 tmpval;
gint64 upstream_size, seek_pos, reverse_limit;
GstFormat format;
guint initial_pcr_seen;
GST_DEBUG ("Scanning for initial sync point");
/* Find initial sync point and at least 5 PCR values */
for (i = 0; i < 20 && !done; i++) {
GST_DEBUG ("Grabbing %d => %d", i * 65536, (i + 1) * 65536);
ret = gst_pad_pull_range (base->sinkpad, i * 65536, 65536, &buf);
if (G_UNLIKELY (ret == GST_FLOW_EOS))
break;
if (G_UNLIKELY (ret != GST_FLOW_OK))
goto beach;
/* Push to packetizer */
mpegts_packetizer_push (base->packetizer, buf);
buf = NULL;
if (mpegts_packetizer_has_packets (base->packetizer)) {
if (base->seek_offset == -1) {
/* Mark the initial sync point and remember the packetsize */
base->seek_offset = base->packetizer->offset;
GST_DEBUG ("Sync point is now %" G_GUINT64_FORMAT, base->seek_offset);
base->packetsize = base->packetizer->packet_size;
}
while (1) {
/* Eat up all packets */
pret = mpegts_packetizer_process_next_packet (base->packetizer);
if (pret == PACKET_NEED_MORE)
break;
if (pret != PACKET_BAD && base->packetizer->nb_seen_offsets >= 5) {
GST_DEBUG ("Got enough initial PCR");
done = TRUE;
break;
}
}
}
}
initial_pcr_seen = base->packetizer->nb_seen_offsets;
if (G_UNLIKELY (initial_pcr_seen == 0))
goto no_initial_pcr;
GST_DEBUG ("Seen %d initial PCR", initial_pcr_seen);
/* Now send data from the end */
/* Get the size of upstream */
format = GST_FORMAT_BYTES;
if (!gst_pad_peer_query_duration (base->sinkpad, format, &tmpval))
goto beach;
upstream_size = tmpval;
/* The scanning takes place on the last 2048kB. Considering PCR should
* be present at least every 100ms, this should cope with streams
* up to 160Mbit/s */
reverse_limit = MAX (0, upstream_size - 2097152);
/* Find last PCR value, searching backwards by chunks of 300 MPEG-ts packets */
for (seek_pos = MAX (0, upstream_size - 56400);
seek_pos >= reverse_limit; seek_pos -= 56400) {
mpegts_packetizer_clear (base->packetizer);
GST_DEBUG ("Grabbing %" G_GUINT64_FORMAT " => %" G_GUINT64_FORMAT, seek_pos,
seek_pos + 56400);
ret = gst_pad_pull_range (base->sinkpad, seek_pos, 56400, &buf);
if (G_UNLIKELY (ret == GST_FLOW_EOS))
break;
if (G_UNLIKELY (ret != GST_FLOW_OK))
goto beach;
/* Push to packetizer */
mpegts_packetizer_push (base->packetizer, buf);
buf = NULL;
if (mpegts_packetizer_has_packets (base->packetizer)) {
pret = PACKET_OK;
/* Eat up all packets, really try to get last PCR(s) */
while (pret != PACKET_NEED_MORE)
pret = mpegts_packetizer_process_next_packet (base->packetizer);
if (base->packetizer->nb_seen_offsets > initial_pcr_seen) {
GST_DEBUG ("Got last PCR(s) (total seen:%d)",
base->packetizer->nb_seen_offsets);
break;
}
}
}
beach:
mpegts_packetizer_clear (base->packetizer);
return ret;
no_initial_pcr:
mpegts_packetizer_clear (base->packetizer);
GST_WARNING_OBJECT (base, "Couldn't find any PCR within the first %d bytes",
10 * 65536);
return GST_FLOW_ERROR;
}
static GstFlowReturn
gst_real_audio_demux_parse_header (GstRealAudioDemux * demux)
{
const guint8 *data;
gchar *codec_name = NULL;
GstCaps *caps = NULL;
GstEvent *event;
gchar *stream_id;
guint avail;
g_assert (demux->ra_version == 4 || demux->ra_version == 3);
avail = gst_adapter_available (demux->adapter);
if (avail < 16)
return GST_FLOW_OK;
if (!gst_real_audio_demux_get_data_offset_from_header (demux))
return GST_FLOW_ERROR; /* shouldn't happen */
GST_DEBUG_OBJECT (demux, "data_offset = %u", demux->data_offset);
if (avail + 6 < demux->data_offset) {
GST_DEBUG_OBJECT (demux, "Need %u bytes, but only %u available now",
demux->data_offset - 6, avail);
return GST_FLOW_OK;
}
data = gst_adapter_map (demux->adapter, demux->data_offset - 6);
g_assert (data);
switch (demux->ra_version) {
case 3:
demux->fourcc = GST_RM_AUD_14_4;
demux->packet_size = 20;
demux->sample_rate = 8000;
demux->channels = 1;
demux->sample_width = 16;
demux->flavour = 1;
demux->leaf_size = 0;
demux->height = 0;
break;
case 4:
demux->flavour = GST_READ_UINT16_BE (data + 16);
/* demux->frame_size = GST_READ_UINT32_BE (data + 36); */
demux->leaf_size = GST_READ_UINT16_BE (data + 38);
demux->height = GST_READ_UINT16_BE (data + 34);
demux->packet_size = GST_READ_UINT32_BE (data + 18);
demux->sample_rate = GST_READ_UINT16_BE (data + 42);
demux->sample_width = GST_READ_UINT16_BE (data + 46);
demux->channels = GST_READ_UINT16_BE (data + 48);
demux->fourcc = GST_READ_UINT32_LE (data + 56);
demux->pending_tags = gst_rm_utils_read_tags (data + 63,
demux->data_offset - 63, gst_rm_utils_read_string8);
if (demux->pending_tags)
gst_tag_list_set_scope (demux->pending_tags, GST_TAG_SCOPE_GLOBAL);
break;
default:
g_assert_not_reached ();
#if 0
case 5:
demux->flavour = GST_READ_UINT16_BE (data + 16);
/* demux->frame_size = GST_READ_UINT32_BE (data + 36); */
demux->leaf_size = GST_READ_UINT16_BE (data + 38);
demux->height = GST_READ_UINT16_BE (data + 34);
demux->sample_rate = GST_READ_UINT16_BE (data + 48);
demux->sample_width = GST_READ_UINT16_BE (data + 52);
demux->n_channels = GST_READ_UINT16_BE (data + 54);
demux->fourcc = RMDEMUX_FOURCC_GET (data + 60);
break;
#endif
}
GST_INFO_OBJECT (demux, "packet_size = %u", demux->packet_size);
GST_INFO_OBJECT (demux, "sample_rate = %u", demux->sample_rate);
GST_INFO_OBJECT (demux, "sample_width = %u", demux->sample_width);
GST_INFO_OBJECT (demux, "channels = %u", demux->channels);
GST_INFO_OBJECT (demux, "fourcc = '%" GST_FOURCC_FORMAT "' (%08X)",
GST_FOURCC_ARGS (demux->fourcc), demux->fourcc);
switch (demux->fourcc) {
case GST_RM_AUD_14_4:
caps = gst_caps_new_simple ("audio/x-pn-realaudio", "raversion",
G_TYPE_INT, 1, NULL);
demux->byterate_num = 1000;
demux->byterate_denom = 1;
break;
case GST_RM_AUD_28_8:
/* FIXME: needs descrambling */
caps = gst_caps_new_simple ("audio/x-pn-realaudio", "raversion",
G_TYPE_INT, 2, NULL);
break;
case GST_RM_AUD_DNET:
caps = gst_caps_new_simple ("audio/x-ac3", "rate", G_TYPE_INT,
demux->sample_rate, NULL);
if (demux->packet_size == 0 || demux->sample_rate == 0)
goto broken_file;
demux->byterate_num = demux->packet_size * demux->sample_rate;
demux->byterate_denom = 1536;
break;
/* Sipro/ACELP.NET Voice Codec (MIME unknown) */
case GST_RM_AUD_SIPR:
caps = gst_caps_new_empty_simple ("audio/x-sipro");
break;
default:
GST_WARNING_OBJECT (demux, "unknown fourcc %08X", demux->fourcc);
break;
}
if (caps == NULL)
goto unknown_fourcc;
gst_caps_set_simple (caps,
"flavor", G_TYPE_INT, demux->flavour,
"rate", G_TYPE_INT, demux->sample_rate,
"channels", G_TYPE_INT, demux->channels,
"width", G_TYPE_INT, demux->sample_width,
"leaf_size", G_TYPE_INT, demux->leaf_size,
"packet_size", G_TYPE_INT, demux->packet_size,
"height", G_TYPE_INT, demux->height, NULL);
GST_INFO_OBJECT (demux, "Adding source pad, caps %" GST_PTR_FORMAT, caps);
demux->srcpad = gst_pad_new_from_static_template (&src_template, "src");
gst_pad_set_event_function (demux->srcpad,
GST_DEBUG_FUNCPTR (gst_real_audio_demux_src_event));
gst_pad_set_query_function (demux->srcpad,
GST_DEBUG_FUNCPTR (gst_real_audio_demux_src_query));
gst_pad_set_active (demux->srcpad, TRUE);
gst_pad_use_fixed_caps (demux->srcpad);
stream_id =
gst_pad_create_stream_id (demux->srcpad, GST_ELEMENT_CAST (demux), NULL);
event = gst_pad_get_sticky_event (demux->sinkpad, GST_EVENT_STREAM_START, 0);
if (event) {
if (gst_event_parse_group_id (event, &demux->group_id))
demux->have_group_id = TRUE;
else
demux->have_group_id = FALSE;
gst_event_unref (event);
} else if (!demux->have_group_id) {
demux->have_group_id = TRUE;
demux->group_id = gst_util_group_id_next ();
}
event = gst_event_new_stream_start (stream_id);
if (demux->have_group_id)
gst_event_set_group_id (event, demux->group_id);
gst_pad_push_event (demux->srcpad, event);
g_free (stream_id);
gst_pad_set_caps (demux->srcpad, caps);
codec_name = gst_pb_utils_get_codec_description (caps);
gst_caps_unref (caps);
gst_element_add_pad (GST_ELEMENT (demux), demux->srcpad);
if (demux->byterate_num > 0 && demux->byterate_denom > 0) {
GstFormat bformat = GST_FORMAT_BYTES;
gint64 size_bytes = 0;
GST_INFO_OBJECT (demux, "byte rate = %u/%u = %u bytes/sec",
demux->byterate_num, demux->byterate_denom,
demux->byterate_num / demux->byterate_denom);
if (gst_pad_peer_query_duration (demux->sinkpad, bformat, &size_bytes)) {
demux->duration =
gst_real_demux_get_timestamp_from_offset (demux, size_bytes);
demux->upstream_size = size_bytes;
GST_INFO_OBJECT (demux, "upstream_size = %" G_GUINT64_FORMAT,
demux->upstream_size);
GST_INFO_OBJECT (demux, "duration = %" GST_TIME_FORMAT,
GST_TIME_ARGS (demux->duration));
}
}
demux->need_newsegment = TRUE;
if (codec_name) {
if (demux->pending_tags == NULL) {
demux->pending_tags = gst_tag_list_new_empty ();
gst_tag_list_set_scope (demux->pending_tags, GST_TAG_SCOPE_GLOBAL);
}
gst_tag_list_add (demux->pending_tags, GST_TAG_MERGE_REPLACE,
GST_TAG_AUDIO_CODEC, codec_name, NULL);
g_free (codec_name);
}
gst_adapter_unmap (demux->adapter);
gst_adapter_flush (demux->adapter, demux->data_offset - 6);
demux->state = REAL_AUDIO_DEMUX_STATE_DATA;
demux->need_newsegment = TRUE;
return GST_FLOW_OK;
/* ERRORS */
unknown_fourcc:
{
GST_ELEMENT_ERROR (GST_ELEMENT (demux), STREAM, DECODE, (NULL),
("Unknown fourcc '%" GST_FOURCC_FORMAT "'",
GST_FOURCC_ARGS (demux->fourcc)));
return GST_FLOW_ERROR;
}
broken_file:
{
GST_ELEMENT_ERROR (GST_ELEMENT (demux), STREAM, DECODE, (NULL),
("Broken file - invalid sample_rate or other header value"));
return GST_FLOW_ERROR;
}
}
static gboolean
gst_progress_report_do_query (GstProgressReport * filter, GstFormat format,
gint hh, gint mm, gint ss, GstBuffer * buf)
{
const gchar *format_name = NULL;
GstPad *sink_pad;
gint64 cur, total;
sink_pad = GST_BASE_TRANSFORM (filter)->sinkpad;
GST_LOG_OBJECT (filter, "querying using format %d (%s)", format,
gst_format_get_name (format));
if (filter->do_query || !buf) {
GST_LOG_OBJECT (filter, "using upstream query");
if (!gst_pad_peer_query_position (sink_pad, format, &cur) ||
!gst_pad_peer_query_duration (sink_pad, format, &total)) {
return FALSE;
}
} else {
GstBaseTransform *base = GST_BASE_TRANSFORM (filter);
GST_LOG_OBJECT (filter, "using buffer metadata");
if (format == GST_FORMAT_TIME && base->segment.format == GST_FORMAT_TIME) {
cur = gst_segment_to_stream_time (&base->segment, format,
GST_BUFFER_TIMESTAMP (buf));
total = base->segment.duration;
} else {
return FALSE;
}
}
switch (format) {
case GST_FORMAT_BYTES:
format_name = "bytes";
break;
case GST_FORMAT_BUFFERS:
format_name = "buffers";
break;
case GST_FORMAT_PERCENT:
format_name = "percent";
break;
case GST_FORMAT_TIME:
format_name = "seconds";
cur /= GST_SECOND;
total /= GST_SECOND;
break;
case GST_FORMAT_DEFAULT:{
GstCaps *caps;
format_name = "bogounits";
caps = gst_pad_get_current_caps (GST_BASE_TRANSFORM (filter)->sinkpad);
if (caps) {
if (gst_caps_is_fixed (caps) && !gst_caps_is_any (caps)) {
GstStructure *s = gst_caps_get_structure (caps, 0);
const gchar *mime_type = gst_structure_get_name (s);
if (g_str_has_prefix (mime_type, "video/") ||
g_str_has_prefix (mime_type, "image/")) {
format_name = "frames";
} else if (g_str_has_prefix (mime_type, "audio/")) {
format_name = "samples";
}
}
gst_caps_unref (caps);
}
break;
}
default:{
const GstFormatDefinition *details;
details = gst_format_get_details (format);
if (details) {
format_name = details->nick;
} else {
format_name = "unknown";
}
break;
}
}
if (!filter->silent) {
if (total > 0) {
g_print ("%s (%02d:%02d:%02d): %" G_GINT64_FORMAT " / %"
G_GINT64_FORMAT " %s (%4.1f %%)\n", GST_OBJECT_NAME (filter), hh,
mm, ss, cur, total, format_name, (gdouble) cur / total * 100.0);
} else {
g_print ("%s (%02d:%02d:%02d): %" G_GINT64_FORMAT " %s\n",
GST_OBJECT_NAME (filter), hh, mm, ss, cur, format_name);
}
}
gst_progress_report_post_progress (filter, format, cur, total);
return TRUE;
}
/* FIXME, the duration query should reflect how long you will produce
* data, that is the amount of stream time until you will emit EOS.
*
* For synchronized mixing this is always the max of all the durations
* of upstream since we emit EOS when all of them finished.
*
* We don't do synchronized mixing so this really depends on where the
* streams where punched in and what their relative offsets are against
* eachother which we can get from the first timestamps we see.
*
* When we add a new stream (or remove a stream) the duration might
* also become invalid again and we need to post a new DURATION
* message to notify this fact to the parent.
* For now we take the max of all the upstream elements so the simple
* cases work at least somewhat.
*/
static gboolean
gst_audio_aggregator_query_duration (GstAudioAggregator * aagg,
GstQuery * query)
{
gint64 max;
gboolean res;
GstFormat format;
GstIterator *it;
gboolean done;
GValue item = { 0, };
/* parse format */
gst_query_parse_duration (query, &format, NULL);
max = -1;
res = TRUE;
done = FALSE;
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (aagg));
while (!done) {
GstIteratorResult ires;
ires = gst_iterator_next (it, &item);
switch (ires) {
case GST_ITERATOR_DONE:
done = TRUE;
break;
case GST_ITERATOR_OK:
{
GstPad *pad = g_value_get_object (&item);
gint64 duration;
/* ask sink peer for duration */
res &= gst_pad_peer_query_duration (pad, format, &duration);
/* take max from all valid return values */
if (res) {
/* valid unknown length, stop searching */
if (duration == -1) {
max = duration;
done = TRUE;
}
/* else see if bigger than current max */
else if (duration > max)
max = duration;
}
g_value_reset (&item);
break;
}
case GST_ITERATOR_RESYNC:
max = -1;
res = TRUE;
gst_iterator_resync (it);
break;
default:
res = FALSE;
done = TRUE;
break;
}
}
g_value_unset (&item);
gst_iterator_free (it);
if (res) {
/* and store the max */
GST_DEBUG_OBJECT (aagg, "Total duration in format %s: %"
GST_TIME_FORMAT, gst_format_get_name (format), GST_TIME_ARGS (max));
gst_query_set_duration (query, format, max);
}
return res;
}
static GstFlowReturn
mpegts_base_scan (MpegTSBase * base)
{
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *buf = NULL;
guint i;
gboolean done = FALSE;
MpegTSPacketizerPacketReturn pret;
gint64 tmpval;
guint64 upstream_size, seek_pos;
GstFormat format;
guint initial_pcr_seen;
GST_DEBUG ("Scanning for initial sync point");
/* Find initial sync point and at least 5 PCR values */
for (i = 0; i < 10 && !done; i++) {
GST_DEBUG ("Grabbing %d => %d", i * 65536, 65536);
ret = gst_pad_pull_range (base->sinkpad, i * 65536, 65536, &buf);
if (G_UNLIKELY (ret != GST_FLOW_OK))
goto beach;
/* Push to packetizer */
mpegts_packetizer_push (base->packetizer, buf);
buf = NULL;
if (mpegts_packetizer_has_packets (base->packetizer)) {
if (base->seek_offset == -1) {
/* Mark the initial sync point and remember the packetsize */
base->seek_offset = base->packetizer->offset;
GST_DEBUG ("Sync point is now %" G_GUINT64_FORMAT, base->seek_offset);
base->packetsize = base->packetizer->packet_size;
}
while (1) {
/* Eat up all packets */
pret = mpegts_packetizer_process_next_packet (base->packetizer);
if (pret == PACKET_NEED_MORE)
break;
if (pret != PACKET_BAD &&
mpegts_packetizer_get_seen_pcr (base->packetizer) >= 5) {
GST_DEBUG ("Got enough initial PCR");
done = TRUE;
break;
}
}
}
}
initial_pcr_seen = mpegts_packetizer_get_seen_pcr (base->packetizer);
if (G_UNLIKELY (initial_pcr_seen == 0))
goto no_initial_pcr;
GST_DEBUG ("Seen %d initial PCR", initial_pcr_seen);
/* Now send data from the end */
mpegts_packetizer_clear (base->packetizer);
/* Get the size of upstream */
format = GST_FORMAT_BYTES;
if (!gst_pad_peer_query_duration (base->sinkpad, format, &tmpval))
goto beach;
upstream_size = tmpval;
done = FALSE;
/* Find last PCR value */
for (seek_pos = MAX (0, upstream_size - 655360);
seek_pos < upstream_size && !done; seek_pos += 65536) {
GST_DEBUG ("Grabbing %" G_GUINT64_FORMAT " => %d", seek_pos, 65536);
ret = gst_pad_pull_range (base->sinkpad, seek_pos, 65536, &buf);
if (G_UNLIKELY (ret != GST_FLOW_OK))
goto beach;
/* Push to packetizer */
mpegts_packetizer_push (base->packetizer, buf);
buf = NULL;
if (mpegts_packetizer_has_packets (base->packetizer)) {
while (1) {
/* Eat up all packets */
pret = mpegts_packetizer_process_next_packet (base->packetizer);
if (pret == PACKET_NEED_MORE)
break;
if (pret != PACKET_BAD &&
mpegts_packetizer_get_seen_pcr (base->packetizer) >
initial_pcr_seen) {
GST_DEBUG ("Got last PCR");
done = TRUE;
break;
}
}
}
}
beach:
mpegts_packetizer_clear (base->packetizer);
return ret;
no_initial_pcr:
mpegts_packetizer_clear (base->packetizer);
GST_WARNING_OBJECT (base, "Couldn't find any PCR within the first %d bytes",
10 * 65536);
return GST_FLOW_ERROR;
}
