static GstStateChangeReturn
gst_musepackdec_change_state (GstElement * element, GstStateChange transition)
{
GstMusepackDec *musepackdec = GST_MUSEPACK_DEC (element);
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_segment_init (&musepackdec->segment, GST_FORMAT_DEFAULT);
gst_segment_set_last_stop (&musepackdec->segment, GST_FORMAT_DEFAULT, 0);
break;
default:
break;
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_segment_init (&musepackdec->segment, GST_FORMAT_UNDEFINED);
musepackdec->offset = 0;
musepackdec->rate = 0;
musepackdec->bps = 0;
break;
default:
break;
}
return ret;
}
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;
const guint8 *data;
GstBuffer *buf = NULL;
buf = gst_buffer_new_and_alloc (unit_size);
gst_buffer_set_caps (buf, GST_PAD_CAPS (demux->srcpad));
data = gst_adapter_peek (demux->adapter, unit_size);
memcpy (GST_BUFFER_DATA (buf), data, unit_size);
gst_adapter_flush (demux->adapter, unit_size);
avail -= unit_size;
if (demux->need_newsegment) {
gst_pad_push_event (demux->srcpad,
gst_event_new_new_segment_full (FALSE, demux->segment.rate,
demux->segment.applied_rate, GST_FORMAT_TIME,
demux->segment.start, demux->segment.stop, demux->segment.time));
demux->need_newsegment = FALSE;
}
if (demux->pending_tags) {
gst_element_found_tags_for_pad (GST_ELEMENT (demux), demux->srcpad,
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;
gst_segment_set_last_stop (&demux->segment, GST_FORMAT_TIME, ts);
ret = gst_pad_push (demux->srcpad, buf);
}
return ret;
}
void QGstreamerPlayerSession::processNewBuffer(GstBuffer *buf)
{
GstClockTime last_stop, duration;
last_stop = GST_BUFFER_TIMESTAMP (buf);
if (GST_CLOCK_TIME_IS_VALID (last_stop)) {
duration = GST_BUFFER_DURATION (buf);
if (GST_CLOCK_TIME_IS_VALID (duration)) {
last_stop += duration;
}
gst_segment_set_last_stop(&m_segment, m_segment.format, last_stop);
}
}
static GstFlowReturn
gst_selector_pad_chain (GstPad * pad, GstBuffer * buf)
{
RsnStreamSelector *sel;
GstFlowReturn res;
GstPad *active_sinkpad;
RsnSelectorPad *selpad;
GstClockTime timestamp;
GstSegment *seg;
sel = RSN_STREAM_SELECTOR (gst_pad_get_parent (pad));
selpad = GST_SELECTOR_PAD_CAST (pad);
seg = &selpad->segment;
active_sinkpad = rsn_stream_selector_get_active (sel, pad);
timestamp = GST_BUFFER_TIMESTAMP (buf);
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
GST_DEBUG_OBJECT (sel, "received timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
gst_segment_set_last_stop (seg, seg->format, timestamp);
}
/* Ignore buffers from pads except the selected one */
if (pad != active_sinkpad)
goto ignore;
/* if we have a pending segment, push it out now */
if (selpad->segment_pending) {
gst_pad_push_event (sel->srcpad, gst_event_new_new_segment_full (FALSE,
seg->rate, seg->applied_rate, seg->format, seg->start, seg->stop,
seg->time));
selpad->segment_pending = FALSE;
}
/* forward */
GST_DEBUG_OBJECT (sel, "Forwarding buffer %p from pad %s:%s", buf,
GST_DEBUG_PAD_NAME (pad));
res = gst_pad_push (sel->srcpad, buf);
done:
gst_object_unref (sel);
return res;
/* dropped buffers */
ignore:
{
GST_DEBUG_OBJECT (sel, "Ignoring buffer %p from pad %s:%s",
buf, GST_DEBUG_PAD_NAME (pad));
gst_buffer_unref (buf);
res = GST_FLOW_NOT_LINKED;
goto done;
}
}
GstFlowReturn
gst_vdp_mpeg_dec_push_video_buffer (GstVdpMpegDec * mpeg_dec,
GstVdpVideoBuffer * buffer)
{
gint64 byterate;
if (GST_BUFFER_TIMESTAMP (buffer) == GST_CLOCK_TIME_NONE
&& GST_CLOCK_TIME_IS_VALID (mpeg_dec->next_timestamp)) {
GST_BUFFER_TIMESTAMP (buffer) = mpeg_dec->next_timestamp;
} else if (GST_BUFFER_TIMESTAMP (buffer) == GST_CLOCK_TIME_NONE) {
GST_BUFFER_TIMESTAMP (buffer) = gst_util_uint64_scale (mpeg_dec->frame_nr,
GST_SECOND * mpeg_dec->fps_d, mpeg_dec->fps_n);
}
if (mpeg_dec->seeking) {
GstEvent *event;
event = gst_event_new_new_segment (FALSE,
mpeg_dec->segment.rate, GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP (buffer),
mpeg_dec->segment.stop, GST_BUFFER_TIMESTAMP (buffer));
gst_pad_push_event (mpeg_dec->src, event);
mpeg_dec->seeking = FALSE;
}
mpeg_dec->next_timestamp = GST_BUFFER_TIMESTAMP (buffer) +
GST_BUFFER_DURATION (buffer);
gst_segment_set_last_stop (&mpeg_dec->segment, GST_FORMAT_TIME,
GST_BUFFER_TIMESTAMP (buffer));
mpeg_dec->accumulated_duration += GST_BUFFER_DURATION (buffer);
mpeg_dec->accumulated_size += GST_BUFFER_SIZE (buffer);
byterate = gst_util_uint64_scale (mpeg_dec->accumulated_size, GST_SECOND,
mpeg_dec->accumulated_duration);
GST_DEBUG ("byterate: %" G_GINT64_FORMAT, mpeg_dec->byterate);
mpeg_dec->byterate = (mpeg_dec->byterate + byterate) / 2;
gst_buffer_set_caps (GST_BUFFER (buffer), GST_PAD_CAPS (mpeg_dec->src));
GST_DEBUG_OBJECT (mpeg_dec,
"Pushing buffer with timestamp: %" GST_TIME_FORMAT
" frame_nr: %" G_GINT64_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_BUFFER_OFFSET (buffer));
return gst_pad_push (mpeg_dec->src, GST_BUFFER (buffer));
}
gboolean
gst_kate_util_decoder_base_update_segment (GstKateDecoderBase * decoder,
GstElement * element, GstBuffer * buf)
{
gint64 clip_start = 0, clip_stop = 0;
gboolean in_seg;
if (decoder->kate_flushing) {
GST_LOG_OBJECT (element, "Kate pad flushing, buffer ignored");
return FALSE;
}
if (G_LIKELY (GST_BUFFER_TIMESTAMP_IS_VALID (buf))) {
GstClockTime stop;
if (G_LIKELY (GST_BUFFER_DURATION_IS_VALID (buf)))
stop = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
else
stop = GST_CLOCK_TIME_NONE;
in_seg = gst_segment_clip (&decoder->kate_segment, GST_FORMAT_TIME,
GST_BUFFER_TIMESTAMP (buf), stop, &clip_start, &clip_stop);
} else {
in_seg = TRUE;
}
if (in_seg) {
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
gst_segment_set_last_stop (&decoder->kate_segment, GST_FORMAT_TIME,
clip_start);
}
} else {
GST_INFO_OBJECT (element, "Kate buffer not in segment, ignored");
}
return in_seg;
}
static GstFlowReturn
gst_aiff_parse_stream_data (GstAiffParse * aiff)
{
GstBuffer *buf = NULL;
GstFlowReturn res = GST_FLOW_OK;
guint64 desired, obtained;
GstClockTime timestamp, next_timestamp, duration;
guint64 pos, nextpos;
iterate_adapter:
GST_LOG_OBJECT (aiff,
"offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
G_GINT64_FORMAT, aiff->offset, aiff->end_offset, aiff->dataleft);
/* Get the next n bytes and output them */
if (aiff->dataleft == 0 || aiff->dataleft < aiff->bytes_per_sample)
goto found_eos;
/* scale the amount of data by the segment rate so we get equal
* amounts of data regardless of the playback rate */
desired =
MIN (gst_guint64_to_gdouble (aiff->dataleft),
MAX_BUFFER_SIZE * aiff->segment.abs_rate);
if (desired >= aiff->bytes_per_sample && aiff->bytes_per_sample > 0)
desired -= (desired % aiff->bytes_per_sample);
GST_LOG_OBJECT (aiff, "Fetching %" G_GINT64_FORMAT " bytes of data "
"from the sinkpad", desired);
if (aiff->streaming) {
guint avail = gst_adapter_available (aiff->adapter);
if (avail < desired) {
GST_LOG_OBJECT (aiff, "Got only %d bytes of data from the sinkpad",
avail);
return GST_FLOW_OK;
}
buf = gst_adapter_take_buffer (aiff->adapter, desired);
} else {
if ((res = gst_pad_pull_range (aiff->sinkpad, aiff->offset,
desired, &buf)) != GST_FLOW_OK)
goto pull_error;
}
/* If we have a pending close/start segment, send it now. */
if (G_UNLIKELY (aiff->close_segment != NULL)) {
gst_pad_push_event (aiff->srcpad, aiff->close_segment);
aiff->close_segment = NULL;
}
if (G_UNLIKELY (aiff->start_segment != NULL)) {
gst_pad_push_event (aiff->srcpad, aiff->start_segment);
aiff->start_segment = NULL;
}
obtained = GST_BUFFER_SIZE (buf);
/* our positions in bytes */
pos = aiff->offset - aiff->datastart;
nextpos = pos + obtained;
/* update offsets, does not overflow. */
GST_BUFFER_OFFSET (buf) = pos / aiff->bytes_per_sample;
GST_BUFFER_OFFSET_END (buf) = nextpos / aiff->bytes_per_sample;
if (aiff->bps > 0) {
/* and timestamps if we have a bitrate, be careful for overflows */
timestamp = uint64_ceiling_scale (pos, GST_SECOND, (guint64) aiff->bps);
next_timestamp =
uint64_ceiling_scale (nextpos, GST_SECOND, (guint64) aiff->bps);
duration = next_timestamp - timestamp;
/* update current running segment position */
gst_segment_set_last_stop (&aiff->segment, GST_FORMAT_TIME, next_timestamp);
} else {
/* no bitrate, all we know is that the first sample has timestamp 0, all
* other positions and durations have unknown timestamp. */
if (pos == 0)
timestamp = 0;
else
timestamp = GST_CLOCK_TIME_NONE;
duration = GST_CLOCK_TIME_NONE;
/* update current running segment position with byte offset */
gst_segment_set_last_stop (&aiff->segment, GST_FORMAT_BYTES, nextpos);
}
if (aiff->discont) {
GST_DEBUG_OBJECT (aiff, "marking DISCONT");
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
aiff->discont = FALSE;
}
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = duration;
gst_buffer_set_caps (buf, aiff->caps);
GST_LOG_OBJECT (aiff,
"Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
", size:%u", GST_TIME_ARGS (timestamp), GST_TIME_ARGS (duration),
GST_BUFFER_SIZE (buf));
if ((res = gst_pad_push (aiff->srcpad, buf)) != GST_FLOW_OK)
goto push_error;
if (obtained < aiff->dataleft) {
aiff->offset += obtained;
aiff->dataleft -= obtained;
} else {
aiff->offset += aiff->dataleft;
aiff->dataleft = 0;
}
/* Iterate until need more data, so adapter size won't grow */
if (aiff->streaming) {
GST_LOG_OBJECT (aiff,
"offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, aiff->offset,
aiff->end_offset);
goto iterate_adapter;
}
return res;
/* ERROR */
found_eos:
{
GST_DEBUG_OBJECT (aiff, "found EOS");
return GST_FLOW_UNEXPECTED;
}
pull_error:
{
/* check if we got EOS */
if (res == GST_FLOW_UNEXPECTED)
goto found_eos;
GST_WARNING_OBJECT (aiff,
"Error getting %" G_GINT64_FORMAT " bytes from the "
"sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, aiff->dataleft);
return res;
}
push_error:
{
GST_INFO_OBJECT (aiff,
"Error pushing on srcpad %s:%s, reason %s, is linked? = %d",
GST_DEBUG_PAD_NAME (aiff->srcpad), gst_flow_get_name (res),
gst_pad_is_linked (aiff->srcpad));
return res;
}
}
static GstFlowReturn
gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
{
GstAudioRate *audiorate;
GstClockTime in_time, in_duration, in_stop, run_time;
guint64 in_offset, in_offset_end, in_samples;
guint in_size;
GstFlowReturn ret = GST_FLOW_OK;
audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));
/* need to be negotiated now */
if (audiorate->bytes_per_sample == 0)
goto not_negotiated;
/* we have a new pending segment */
if (audiorate->next_offset == -1) {
gint64 pos;
/* update the TIME segment */
gst_audio_rate_convert_segments (audiorate);
/* first buffer, we are negotiated and we have a segment, calculate the
* current expected offsets based on the segment.start, which is the first
* media time of the segment and should match the media time of the first
* buffer in that segment, which is the offset expressed in DEFAULT units.
*/
/* convert first timestamp of segment to sample position */
pos = gst_util_uint64_scale_int (audiorate->src_segment.start,
audiorate->rate, GST_SECOND);
GST_DEBUG_OBJECT (audiorate, "resync to offset %" G_GINT64_FORMAT, pos);
audiorate->next_offset = pos;
audiorate->next_ts = gst_util_uint64_scale_int (audiorate->next_offset,
GST_SECOND, audiorate->rate);
}
audiorate->in++;
in_time = GST_BUFFER_TIMESTAMP (buf);
if (in_time == GST_CLOCK_TIME_NONE) {
GST_DEBUG_OBJECT (audiorate, "no timestamp, using expected next time");
in_time = audiorate->next_ts;
}
in_size = GST_BUFFER_SIZE (buf);
in_samples = in_size / audiorate->bytes_per_sample;
/* get duration from the size because we can and it's more accurate */
in_duration =
gst_util_uint64_scale_int (in_samples, GST_SECOND, audiorate->rate);
in_stop = in_time + in_duration;
/* Figure out the total accumulated segment time. */
run_time = in_time + audiorate->src_segment.accum;
/* calculate the buffer offset */
in_offset = gst_util_uint64_scale_int (run_time, audiorate->rate, GST_SECOND);
in_offset_end = in_offset + in_samples;
GST_LOG_OBJECT (audiorate,
"in_time:%" GST_TIME_FORMAT ", run_time:%" GST_TIME_FORMAT
", in_duration:%" GST_TIME_FORMAT
", in_size:%u, in_offset:%lld, in_offset_end:%lld" ", ->next_offset:%lld",
GST_TIME_ARGS (in_time), GST_TIME_ARGS (run_time),
GST_TIME_ARGS (in_duration), in_size, in_offset, in_offset_end,
audiorate->next_offset);
/* do we need to insert samples */
if (in_offset > audiorate->next_offset) {
GstBuffer *fill;
gint fillsize;
guint64 fillsamples;
/* We don't want to allocate a single unreasonably huge buffer - it might
be hundreds of megabytes. So, limit each output buffer to one second of
audio */
fillsamples = in_offset - audiorate->next_offset;
while (fillsamples > 0) {
guint64 cursamples = MIN (fillsamples, audiorate->rate);
fillsamples -= cursamples;
fillsize = cursamples * audiorate->bytes_per_sample;
fill = gst_buffer_new_and_alloc (fillsize);
/* FIXME, 0 might not be the silence byte for the negotiated format. */
memset (GST_BUFFER_DATA (fill), 0, fillsize);
GST_DEBUG_OBJECT (audiorate, "inserting %lld samples", cursamples);
GST_BUFFER_OFFSET (fill) = audiorate->next_offset;
audiorate->next_offset += cursamples;
GST_BUFFER_OFFSET_END (fill) = audiorate->next_offset;
/* Use next timestamp, then calculate following timestamp based on
* offset to get duration. Neccesary complexity to get 'perfect'
* streams */
GST_BUFFER_TIMESTAMP (fill) = audiorate->next_ts;
audiorate->next_ts = gst_util_uint64_scale_int (audiorate->next_offset,
GST_SECOND, audiorate->rate);
GST_BUFFER_DURATION (fill) = audiorate->next_ts -
GST_BUFFER_TIMESTAMP (fill);
/* we created this buffer to fill a gap */
GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_GAP);
/* set discont if it's pending, this is mostly done for the first buffer
* and after a flushing seek */
if (audiorate->discont) {
GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_DISCONT);
audiorate->discont = FALSE;
}
gst_buffer_set_caps (fill, GST_PAD_CAPS (audiorate->srcpad));
ret = gst_pad_push (audiorate->srcpad, fill);
if (ret != GST_FLOW_OK)
goto beach;
audiorate->out++;
audiorate->add += cursamples;
if (!audiorate->silent)
g_object_notify (G_OBJECT (audiorate), "add");
}
} else if (in_offset < audiorate->next_offset) {
/* need to remove samples */
if (in_offset_end <= audiorate->next_offset) {
guint64 drop = in_size / audiorate->bytes_per_sample;
audiorate->drop += drop;
GST_DEBUG_OBJECT (audiorate, "dropping %lld samples", drop);
/* we can drop the buffer completely */
gst_buffer_unref (buf);
if (!audiorate->silent)
g_object_notify (G_OBJECT (audiorate), "drop");
goto beach;
} else {
guint64 truncsamples;
guint truncsize, leftsize;
GstBuffer *trunc;
/* truncate buffer */
truncsamples = audiorate->next_offset - in_offset;
truncsize = truncsamples * audiorate->bytes_per_sample;
leftsize = in_size - truncsize;
trunc = gst_buffer_create_sub (buf, truncsize, leftsize);
gst_buffer_unref (buf);
buf = trunc;
gst_buffer_set_caps (buf, GST_PAD_CAPS (audiorate->srcpad));
audiorate->drop += truncsamples;
}
}
/* Now calculate parameters for whichever buffer (either the original
* or truncated one) we're pushing. */
GST_BUFFER_OFFSET (buf) = audiorate->next_offset;
GST_BUFFER_OFFSET_END (buf) = in_offset_end;
GST_BUFFER_TIMESTAMP (buf) = audiorate->next_ts;
audiorate->next_ts = gst_util_uint64_scale_int (in_offset_end,
GST_SECOND, audiorate->rate);
GST_BUFFER_DURATION (buf) = audiorate->next_ts - GST_BUFFER_TIMESTAMP (buf);
if (audiorate->discont) {
/* we need to output a discont buffer, do so now */
GST_DEBUG_OBJECT (audiorate, "marking DISCONT on output buffer");
buf = gst_buffer_make_metadata_writable (buf);
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
audiorate->discont = FALSE;
} else if (GST_BUFFER_IS_DISCONT (buf)) {
/* else we make everything continuous so we can safely remove the DISCONT
* flag from the buffer if there was one */
GST_DEBUG_OBJECT (audiorate, "removing DISCONT from buffer");
buf = gst_buffer_make_metadata_writable (buf);
GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);
}
/* set last_stop on segment */
gst_segment_set_last_stop (&audiorate->src_segment, GST_FORMAT_TIME,
GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf));
ret = gst_pad_push (audiorate->srcpad, buf);
audiorate->out++;
audiorate->next_offset = in_offset_end;
beach:
gst_object_unref (audiorate);
return ret;
/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (audiorate, STREAM, FORMAT,
(NULL), ("pipeline error, format was not negotiated"));
return GST_FLOW_NOT_NEGOTIATED;
}
}
static gboolean
gst_musepack_stream_init (GstMusepackDec * musepackdec)
{
mpc_streaminfo i;
GstTagList *tags;
GstCaps *caps;
/* set up reading */
gst_musepack_init_reader (musepackdec->r, musepackdec);
#ifdef MPC_IS_OLD_API
/* streaminfo */
mpc_streaminfo_init (&i);
if (mpc_streaminfo_read (&i, musepackdec->r) < 0) {
GST_ELEMENT_ERROR (musepackdec, STREAM, WRONG_TYPE, (NULL), (NULL));
return FALSE;
}
/* decoding */
mpc_decoder_setup (musepackdec->d, musepackdec->r);
mpc_decoder_scale_output (musepackdec->d, 1.0);
if (!mpc_decoder_initialize (musepackdec->d, &i)) {
GST_ELEMENT_ERROR (musepackdec, STREAM, WRONG_TYPE, (NULL), (NULL));
return FALSE;
}
#else
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);
#endif
/* capsnego */
caps = gst_caps_from_string (BASE_CAPS);
gst_caps_set_simple (caps,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"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);
gst_segment_set_last_stop (&musepackdec->segment, GST_FORMAT_DEFAULT, 0);
gst_segment_set_duration (&musepackdec->segment, GST_FORMAT_DEFAULT,
mpc_streaminfo_get_length_samples (&i));
/* send basic tags */
tags = gst_tag_list_new ();
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_element_found_tags (GST_ELEMENT (musepackdec), tags);
return TRUE;
}
static gboolean
gst_musepackdec_handle_seek_event (GstMusepackDec * dec, GstEvent * event)
{
GstSeekType start_type, stop_type;
GstSeekFlags flags;
GstSegment segment;
GstFormat format;
gboolean flush;
gdouble rate;
gint64 start, stop;
gint samplerate;
gst_event_parse_seek (event, &rate, &format, &flags, &start_type, &start,
&stop_type, &stop);
if (format != GST_FORMAT_TIME && format != GST_FORMAT_DEFAULT) {
GST_DEBUG_OBJECT (dec, "seek failed: only TIME or DEFAULT format allowed");
return FALSE;
}
samplerate = g_atomic_int_get (&dec->rate);
if (format == GST_FORMAT_TIME) {
if (start_type != GST_SEEK_TYPE_NONE)
start = gst_util_uint64_scale_int (start, samplerate, GST_SECOND);
if (stop_type != GST_SEEK_TYPE_NONE)
stop = gst_util_uint64_scale_int (stop, samplerate, GST_SECOND);
}
flush = ((flags & GST_SEEK_FLAG_FLUSH) == GST_SEEK_FLAG_FLUSH);
if (flush)
gst_pad_push_event (dec->srcpad, gst_event_new_flush_start ());
else
gst_pad_pause_task (dec->sinkpad); /* not _stop_task()? */
GST_PAD_STREAM_LOCK (dec->sinkpad);
/* operate on segment copy until we know the seek worked */
segment = dec->segment;
gst_segment_set_seek (&segment, rate, GST_FORMAT_DEFAULT,
flags, start_type, start, stop_type, stop, NULL);
gst_pad_push_event (dec->sinkpad, gst_event_new_flush_stop ());
GST_DEBUG_OBJECT (dec, "segment: [%" G_GINT64_FORMAT "-%" G_GINT64_FORMAT
"] = [%" GST_TIME_FORMAT "-%" GST_TIME_FORMAT "]",
segment.start, segment.stop,
GST_TIME_ARGS (segment.start * GST_SECOND / dec->rate),
GST_TIME_ARGS (segment.stop * GST_SECOND / dec->rate));
GST_DEBUG_OBJECT (dec, "performing seek to sample %" G_GINT64_FORMAT,
segment.start);
if (segment.start < 0 || segment.start >= segment.duration) {
GST_WARNING_OBJECT (dec, "seek out of bounds");
goto failed;
}
#ifdef MPC_IS_OLD_API
if (!mpc_decoder_seek_sample (dec->d, segment.start))
goto failed;
#else
if (mpc_demux_seek_sample (dec->d, segment.start) != MPC_STATUS_OK)
goto failed;
#endif
if ((flags & GST_SEEK_FLAG_SEGMENT) == GST_SEEK_FLAG_SEGMENT) {
GST_DEBUG_OBJECT (dec, "posting SEGMENT_START message");
gst_element_post_message (GST_ELEMENT (dec),
gst_message_new_segment_start (GST_OBJECT (dec), GST_FORMAT_TIME,
gst_util_uint64_scale_int (segment.start, GST_SECOND, dec->rate)));
}
if (flush) {
gst_pad_push_event (dec->srcpad, gst_event_new_flush_stop ());
}
gst_segment_set_last_stop (&segment, GST_FORMAT_DEFAULT, segment.start);
dec->segment = segment;
gst_musepackdec_send_newsegment (dec);
GST_DEBUG_OBJECT (dec, "seek successful");
gst_pad_start_task (dec->sinkpad,
(GstTaskFunction) gst_musepackdec_loop, dec->sinkpad);
GST_PAD_STREAM_UNLOCK (dec->sinkpad);
return TRUE;
failed:
{
GST_WARNING_OBJECT (dec, "seek failed");
GST_PAD_STREAM_UNLOCK (dec->sinkpad);
return FALSE;
}
}
static GstFlowReturn
gst_selector_pad_chain (GstPad * pad, GstBuffer * buf)
{
GstInputSelector *sel;
GstFlowReturn res;
GstPad *active_sinkpad;
GstPad *prev_active_sinkpad;
GstSelectorPad *selpad;
GstClockTime start_time;
GstSegment *seg;
GstEvent *close_event = NULL, *start_event = NULL;
GstCaps *caps;
sel = GST_INPUT_SELECTOR (gst_pad_get_parent (pad));
selpad = GST_SELECTOR_PAD_CAST (pad);
seg = &selpad->segment;
GST_INPUT_SELECTOR_LOCK (sel);
/* wait or check for flushing */
if (gst_input_selector_wait (sel, pad))
goto flushing;
GST_LOG_OBJECT (pad, "getting active pad");
prev_active_sinkpad = sel->active_sinkpad;
active_sinkpad = gst_input_selector_activate_sinkpad (sel, pad);
/* update the segment on the srcpad */
start_time = GST_BUFFER_TIMESTAMP (buf);
if (GST_CLOCK_TIME_IS_VALID (start_time)) {
GST_LOG_OBJECT (pad, "received start time %" GST_TIME_FORMAT,
GST_TIME_ARGS (start_time));
if (GST_BUFFER_DURATION_IS_VALID (buf))
GST_LOG_OBJECT (pad, "received end time %" GST_TIME_FORMAT,
GST_TIME_ARGS (start_time + GST_BUFFER_DURATION (buf)));
GST_OBJECT_LOCK (pad);
gst_segment_set_last_stop (seg, seg->format, start_time);
GST_OBJECT_UNLOCK (pad);
}
/* Ignore buffers from pads except the selected one */
if (pad != active_sinkpad)
goto ignore;
if (G_UNLIKELY (sel->pending_close)) {
GstSegment *cseg = &sel->segment;
GST_DEBUG_OBJECT (sel,
"pushing close NEWSEGMENT update %d, rate %lf, applied rate %lf, "
"format %d, "
"%" G_GINT64_FORMAT " -- %" G_GINT64_FORMAT ", time %"
G_GINT64_FORMAT, TRUE, cseg->rate, cseg->applied_rate, cseg->format,
cseg->start, cseg->stop, cseg->time);
/* create update segment */
close_event = gst_event_new_new_segment_full (TRUE, cseg->rate,
cseg->applied_rate, cseg->format, cseg->start, cseg->stop, cseg->time);
sel->pending_close = FALSE;
}
/* if we have a pending segment, push it out now */
if (G_UNLIKELY (selpad->segment_pending)) {
GST_DEBUG_OBJECT (pad,
"pushing pending NEWSEGMENT update %d, rate %lf, applied rate %lf, "
"format %d, "
"%" G_GINT64_FORMAT " -- %" G_GINT64_FORMAT ", time %"
G_GINT64_FORMAT, FALSE, seg->rate, seg->applied_rate, seg->format,
seg->start, seg->stop, seg->time);
start_event = gst_event_new_new_segment_full (FALSE, seg->rate,
seg->applied_rate, seg->format, seg->start, seg->stop, seg->time);
selpad->segment_pending = FALSE;
}
GST_INPUT_SELECTOR_UNLOCK (sel);
if (prev_active_sinkpad != active_sinkpad && pad == active_sinkpad)
g_object_notify (G_OBJECT (sel), "active-pad");
if (close_event)
gst_pad_push_event (sel->srcpad, close_event);
if (start_event)
gst_pad_push_event (sel->srcpad, start_event);
if (selpad->discont) {
buf = gst_buffer_make_metadata_writable (buf);
GST_DEBUG_OBJECT (pad, "Marking discont buffer %p", buf);
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
selpad->discont = FALSE;
}
/* forward */
GST_LOG_OBJECT (pad, "Forwarding buffer %p", buf);
if ((caps = GST_BUFFER_CAPS (buf))) {
if (GST_PAD_CAPS (sel->srcpad) != caps)
gst_pad_set_caps (sel->srcpad, caps);
}
res = gst_pad_push (sel->srcpad, buf);
done:
gst_object_unref (sel);
return res;
/* dropped buffers */
ignore:
{
GST_DEBUG_OBJECT (pad, "Pad not active, discard buffer %p", buf);
/* when we drop a buffer, we're creating a discont on this pad */
selpad->discont = TRUE;
GST_INPUT_SELECTOR_UNLOCK (sel);
gst_buffer_unref (buf);
/* figure out what to return upstream */
GST_OBJECT_LOCK (selpad);
if (selpad->always_ok)
res = GST_FLOW_OK;
else
res = GST_FLOW_NOT_LINKED;
GST_OBJECT_UNLOCK (selpad);
goto done;
}
flushing:
{
GST_DEBUG_OBJECT (pad, "We are flushing, discard buffer %p", buf);
GST_INPUT_SELECTOR_UNLOCK (sel);
gst_buffer_unref (buf);
res = GST_FLOW_WRONG_STATE;
goto done;
}
}
static GstFlowReturn
fs_funnel_chain (GstPad * pad, GstBuffer * buffer)
{
GstFlowReturn res;
FsFunnel *funnel = FS_FUNNEL (gst_pad_get_parent (pad));
FsFunnelPadPrivate *priv = gst_pad_get_element_private (pad);
GstEvent *event = NULL;
GstClockTime newts;
GstCaps *padcaps;
GST_DEBUG_OBJECT (funnel, "received buffer %p", buffer);
GST_OBJECT_LOCK (funnel);
if (priv->segment.format == GST_FORMAT_UNDEFINED) {
GST_WARNING_OBJECT (funnel, "Got buffer without segment,"
" setting segment [0,inf[");
gst_segment_set_newsegment_full (&priv->segment, FALSE, 1.0, 1.0,
GST_FORMAT_TIME, 0, -1, 0);
}
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buffer)))
gst_segment_set_last_stop (&priv->segment, priv->segment.format,
GST_BUFFER_TIMESTAMP (buffer));
newts = gst_segment_to_running_time (&priv->segment,
priv->segment.format, GST_BUFFER_TIMESTAMP (buffer));
if (newts != GST_BUFFER_TIMESTAMP (buffer)) {
buffer = gst_buffer_make_metadata_writable (buffer);
GST_BUFFER_TIMESTAMP (buffer) = newts;
}
if (!funnel->has_segment)
{
event = gst_event_new_new_segment_full (FALSE, 1.0, 1.0, GST_FORMAT_TIME,
0, -1, 0);
funnel->has_segment = TRUE;
}
GST_OBJECT_UNLOCK (funnel);
if (event) {
if (!gst_pad_push_event (funnel->srcpad, event))
GST_WARNING_OBJECT (funnel, "Could not push out newsegment event");
}
GST_OBJECT_LOCK (pad);
padcaps = GST_PAD_CAPS (funnel->srcpad);
GST_OBJECT_UNLOCK (pad);
if (GST_BUFFER_CAPS (buffer) && GST_BUFFER_CAPS (buffer) != padcaps) {
if (!gst_pad_set_caps (funnel->srcpad, GST_BUFFER_CAPS (buffer))) {
res = GST_FLOW_NOT_NEGOTIATED;
goto out;
}
}
res = gst_pad_push (funnel->srcpad, buffer);
GST_LOG_OBJECT (funnel, "handled buffer %s", gst_flow_get_name (res));
out:
gst_object_unref (funnel);
return res;
}
static GstFlowReturn
handle_buffer (GstSubParse * self, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
GstCaps *caps = NULL;
gchar *line, *subtitle;
feed_textbuf (self, buf);
/* make sure we know the format */
if (G_UNLIKELY (self->parser_type == GST_SUB_PARSE_FORMAT_UNKNOWN)) {
if (!(caps = gst_sub_parse_format_autodetect (self))) {
return GST_FLOW_UNEXPECTED;
}
if (!gst_pad_set_caps (self->srcpad, caps)) {
gst_caps_unref (caps);
return GST_FLOW_UNEXPECTED;
}
gst_caps_unref (caps);
}
while ((line = get_next_line (self)) && !self->flushing) {
/* Set segment on our parser state machine */
self->state.segment = &self->segment;
/* Now parse the line, out of segment lines will just return NULL */
GST_LOG_OBJECT (self, "Parsing line '%s'", line);
subtitle = self->parse_line (&self->state, line);
g_free (line);
if (subtitle) {
guint subtitle_len = strlen (subtitle);
/* +1 for terminating NUL character */
ret = gst_pad_alloc_buffer_and_set_caps (self->srcpad,
GST_BUFFER_OFFSET_NONE, subtitle_len + 1,
GST_PAD_CAPS (self->srcpad), &buf);
if (ret == GST_FLOW_OK) {
/* copy terminating NUL character as well */
memcpy (GST_BUFFER_DATA (buf), subtitle, subtitle_len + 1);
GST_BUFFER_SIZE (buf) = subtitle_len;
GST_BUFFER_TIMESTAMP (buf) = self->state.start_time;
GST_BUFFER_DURATION (buf) = self->state.duration;
/* in some cases (e.g. tmplayer) we can only determine the duration
* of a text chunk from the timestamp of the next text chunk; in those
* cases, we probably want to limit the duration to something
* reasonable, so we don't end up showing some text for e.g. 40 seconds
* just because nothing else is being said during that time */
if (self->state.max_duration > 0 && GST_BUFFER_DURATION_IS_VALID (buf)) {
if (GST_BUFFER_DURATION (buf) > self->state.max_duration)
GST_BUFFER_DURATION (buf) = self->state.max_duration;
}
gst_segment_set_last_stop (&self->segment, GST_FORMAT_TIME,
self->state.start_time);
GST_DEBUG_OBJECT (self, "Sending text '%s', %" GST_TIME_FORMAT " + %"
GST_TIME_FORMAT, subtitle, GST_TIME_ARGS (self->state.start_time),
GST_TIME_ARGS (self->state.duration));
ret = gst_pad_push (self->srcpad, buf);
}
/* move this forward (the tmplayer parser needs this) */
if (self->state.duration != GST_CLOCK_TIME_NONE)
self->state.start_time += self->state.duration;
g_free (subtitle);
subtitle = NULL;
if (ret != GST_FLOW_OK) {
GST_DEBUG_OBJECT (self, "flow: %s", gst_flow_get_name (ret));
break;
}
}
}
return ret;
}
/* For each buffer we receive we check if our collected condition is reached
* and if so we call the collected function. When this is done we check if
* data has been unqueued. If data is still queued we wait holding the stream
* lock to make sure no EOS event can happen while we are ready to be
* collected
*/
static GstFlowReturn
gst_collect_pads_chain (GstPad * pad, GstBuffer * buffer)
{
GstCollectData *data;
GstCollectPads *pads;
GstCollectPadsPrivate *priv;
GstFlowReturn ret;
GST_DEBUG ("Got buffer for pad %s:%s", GST_DEBUG_PAD_NAME (pad));
/* some magic to get the managing collect_pads */
GST_OBJECT_LOCK (pad);
data = (GstCollectData *) gst_pad_get_element_private (pad);
if (G_UNLIKELY (data == NULL))
goto no_data;
ref_data (data);
GST_OBJECT_UNLOCK (pad);
pads = data->collect;
priv = pads->abidata.ABI.priv;
GST_OBJECT_LOCK (pads);
/* if not started, bail out */
if (G_UNLIKELY (!pads->started))
goto not_started;
/* check if this pad is flushing */
if (G_UNLIKELY (data->abidata.ABI.flushing))
goto flushing;
/* pad was EOS, we can refuse this data */
if (G_UNLIKELY (data->abidata.ABI.eos))
goto unexpected;
/* see if we need to clip */
if (priv->clipfunc) {
buffer = priv->clipfunc (pads, data, buffer, priv->clipfunc_user_data);
if (G_UNLIKELY (buffer == NULL))
goto clipped;
}
GST_DEBUG ("Queuing buffer %p for pad %s:%s", buffer,
GST_DEBUG_PAD_NAME (pad));
/* One more pad has data queued */
pads->queuedpads++;
/* take ownership of the buffer */
if (data->buffer)
gst_buffer_unref (data->buffer);
data->buffer = buffer;
buffer = NULL;
/* update segment last position if in TIME */
if (G_LIKELY (data->segment.format == GST_FORMAT_TIME)) {
GstClockTime timestamp = GST_BUFFER_TIMESTAMP (data->buffer);
if (GST_CLOCK_TIME_IS_VALID (timestamp))
gst_segment_set_last_stop (&data->segment, GST_FORMAT_TIME, timestamp);
}
/* While we have data queued on this pad try to collect stuff */
do {
GST_DEBUG ("Pad %s:%s checking", GST_DEBUG_PAD_NAME (pad));
/* Check if our collected condition is matched and call the collected function
* if it is */
ret = gst_collect_pads_check_collected (pads);
/* when an error occurs, we want to report this back to the caller ASAP
* without having to block if the buffer was not popped */
if (G_UNLIKELY (ret != GST_FLOW_OK))
goto error;
/* data was consumed, we can exit and accept new data */
if (data->buffer == NULL)
break;
/* Check if we got removed in the mean time, FIXME, this is racy.
* Between this check and the _WAIT, the pad could be removed which will
* makes us hang in the _WAIT. */
GST_OBJECT_LOCK (pad);
if (G_UNLIKELY (gst_pad_get_element_private (pad) == NULL))
goto pad_removed;
GST_OBJECT_UNLOCK (pad);
GST_DEBUG ("Pad %s:%s has a buffer queued, waiting",
GST_DEBUG_PAD_NAME (pad));
/* wait to be collected, this must happen from another thread triggered
* by the _chain function of another pad. We release the lock so we
* can get stopped or flushed as well. We can however not get EOS
* because we still hold the STREAM_LOCK.
*/
GST_COLLECT_PADS_WAIT (pads);
GST_DEBUG ("Pad %s:%s resuming", GST_DEBUG_PAD_NAME (pad));
/* after a signal, we could be stopped */
if (G_UNLIKELY (!pads->started))
goto not_started;
/* check if this pad is flushing */
if (G_UNLIKELY (data->abidata.ABI.flushing))
goto flushing;
}
while (data->buffer != NULL);
unlock_done:
GST_DEBUG ("Pad %s:%s done", GST_DEBUG_PAD_NAME (pad));
GST_OBJECT_UNLOCK (pads);
unref_data (data);
if (buffer)
gst_buffer_unref (buffer);
return ret;
pad_removed:
{
GST_WARNING ("%s got removed from collectpads", GST_OBJECT_NAME (pad));
GST_OBJECT_UNLOCK (pad);
ret = GST_FLOW_NOT_LINKED;
goto unlock_done;
}
/* ERRORS */
no_data:
{
GST_DEBUG ("%s got removed from collectpads", GST_OBJECT_NAME (pad));
GST_OBJECT_UNLOCK (pad);
gst_buffer_unref (buffer);
return GST_FLOW_NOT_LINKED;
}
not_started:
{
GST_DEBUG ("not started");
gst_collect_pads_clear (pads, data);
ret = GST_FLOW_WRONG_STATE;
goto unlock_done;
}
flushing:
{
GST_DEBUG ("pad %s:%s is flushing", GST_DEBUG_PAD_NAME (pad));
gst_collect_pads_clear (pads, data);
ret = GST_FLOW_WRONG_STATE;
goto unlock_done;
}
unexpected:
{
/* we should not post an error for this, just inform upstream that
* we don't expect anything anymore */
GST_DEBUG ("pad %s:%s is eos", GST_DEBUG_PAD_NAME (pad));
ret = GST_FLOW_UNEXPECTED;
goto unlock_done;
}
clipped:
{
GST_DEBUG ("clipped buffer on pad %s:%s", GST_DEBUG_PAD_NAME (pad));
ret = GST_FLOW_OK;
goto unlock_done;
}
error:
{
/* we print the error, the element should post a reasonable error
* message for fatal errors */
GST_DEBUG ("collect failed, reason %d (%s)", ret, gst_flow_get_name (ret));
gst_collect_pads_clear (pads, data);
goto unlock_done;
}
}
