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wavparse: port to 0.11

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This commit is contained in:
Mark Nauwelaerts 2011-07-06 16:10:34 +02:00
parent 9cf287ebcc
commit e9146e7aec
2 changed files with 138 additions and 135 deletions
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269
gst/wavparse/gstwavparse.c
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@ -89,38 +89,19 @@ GST_STATIC_PAD_TEMPLATE ("wavparse_sink",
GST_STATIC_CAPS ("audio/x-wav")
);
#define DEBUG_INIT(bla) \
#define DEBUG_INIT \
GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
GST_BOILERPLATE_FULL (GstWavParse, gst_wavparse, GstElement,
GST_TYPE_ELEMENT, DEBUG_INIT);
static void
gst_wavparse_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GstPadTemplate *src_template;
/* register pads */
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template_factory));
src_template = gst_pad_template_new ("wavparse_src", GST_PAD_SRC,
GST_PAD_SOMETIMES, gst_riff_create_audio_template_caps ());
gst_element_class_add_pad_template (element_class, src_template);
gst_object_unref (src_template);
gst_element_class_set_details_simple (element_class, "WAV audio demuxer",
"Codec/Demuxer/Audio",
"Parse a .wav file into raw audio",
"Erik Walthinsen <omega@cse.ogi.edu>");
}
#define gst_wavparse_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstWavParse, gst_wavparse, GST_TYPE_ELEMENT,
DEBUG_INIT);
static void
gst_wavparse_class_init (GstWavParseClass * klass)
{
GstElementClass *gstelement_class;
GObjectClass *object_class;
GstPadTemplate *src_template;
gstelement_class = (GstElementClass *) klass;
object_class = (GObjectClass *) klass;
@ -131,6 +112,19 @@ gst_wavparse_class_init (GstWavParseClass * klass)
gstelement_class->change_state = gst_wavparse_change_state;
gstelement_class->send_event = gst_wavparse_send_event;
/* register pads */
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&sink_template_factory));
src_template = gst_pad_template_new ("wavparse_src", GST_PAD_SRC,
GST_PAD_SOMETIMES, gst_riff_create_audio_template_caps ());
gst_element_class_add_pad_template (gstelement_class, src_template);
gst_element_class_set_details_simple (gstelement_class, "WAV audio demuxer",
"Codec/Demuxer/Audio",
"Parse a .wav file into raw audio",
"Erik Walthinsen <omega@cse.ogi.edu>");
}
static void
@ -172,9 +166,6 @@ gst_wavparse_reset (GstWavParse * wav)
if (wav->start_segment)
gst_event_unref (wav->start_segment);
wav->start_segment = NULL;
if (wav->close_segment)
gst_event_unref (wav->close_segment);
wav->close_segment = NULL;
}
static void
@ -189,7 +180,7 @@ gst_wavparse_dispose (GObject * object)
}
static void
gst_wavparse_init (GstWavParse * wavparse, GstWavParseClass * g_class)
gst_wavparse_init (GstWavParse * wavparse)
{
gst_wavparse_reset (wavparse);
@ -830,10 +821,11 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
if (!event || wav->state != GST_WAVPARSE_DATA) {
if (wav->start_segment)
gst_event_unref (wav->start_segment);
wav->start_segment =
// TODO
/* wav->start_segment =
gst_event_new_new_segment (FALSE, wav->segment.rate,
wav->segment.format, wav->segment.last_stop, wav->segment.duration,
wav->segment.last_stop);
wav->segment.last_stop);*/
res = TRUE;
} else {
/* convert seek positions to byte positions in data sections */
@ -887,7 +879,7 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
GST_PAD_STREAM_LOCK (wav->sinkpad);
/* save current position */
last_stop = wav->segment.last_stop;
last_stop = wav->segment.position;
GST_DEBUG_OBJECT (wav, "stopped streaming at %" G_GINT64_FORMAT, last_stop);
@ -899,7 +891,7 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
* right values in the segment to perform the seek */
if (event) {
GST_DEBUG_OBJECT (wav, "configuring seek");
gst_segment_set_seek (&seeksegment, rate, format, flags,
gst_segment_do_seek (&seeksegment, rate, format, flags,
cur_type, cur, stop_type, stop, &update);
}
@ -913,9 +905,9 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
/* bring offset to bytes, if the bps is 0, we have the segment in BYTES and
* we can just copy the last_stop. If not, we use the bps to convert TIME to
* bytes. */
if (!gst_wavparse_time_to_bytepos (wav, seeksegment.last_stop,
if (!gst_wavparse_time_to_bytepos (wav, seeksegment.position,
(gint64 *) & wav->offset))
wav->offset = seeksegment.last_stop;
wav->offset = seeksegment.position;
GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
wav->offset -= (wav->offset % wav->bytes_per_sample);
GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
@ -959,19 +951,7 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
if (flush) {
/* if we sent a FLUSH_START, we now send a FLUSH_STOP */
GST_DEBUG_OBJECT (wav, "sending flush stop");
gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop ());
} else if (wav->segment_running) {
/* we are running the current segment and doing a non-flushing seek,
* close the segment first based on the previous last_stop. */
GST_DEBUG_OBJECT (wav, "closing running segment %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, wav->segment.start, wav->segment.last_stop);
/* queue the segment for sending in the stream thread */
if (wav->close_segment)
gst_event_unref (wav->close_segment);
wav->close_segment = gst_event_new_new_segment (TRUE,
wav->segment.rate, wav->segment.format,
wav->segment.start, wav->segment.last_stop, wav->segment.start);
gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop (TRUE));
}
}
@ -982,29 +962,25 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT_CAST (wav),
gst_message_new_segment_start (GST_OBJECT_CAST (wav),
wav->segment.format, wav->segment.last_stop));
wav->segment.format, wav->segment.position));
}
/* now create the newsegment */
GST_DEBUG_OBJECT (wav, "Creating newsegment from %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, wav->segment.last_stop, stop);
" to %" G_GINT64_FORMAT, wav->segment.position, stop);
/* store the newsegment event so it can be sent from the streaming thread. */
if (wav->start_segment)
gst_event_unref (wav->start_segment);
wav->start_segment =
gst_event_new_new_segment (FALSE, wav->segment.rate,
wav->segment.format, wav->segment.last_stop, stop,
wav->segment.last_stop);
wav->start_segment = gst_event_new_segment (&wav->segment);
/* mark discont if we are going to stream from another position. */
if (last_stop != wav->segment.last_stop) {
if (last_stop != wav->segment.position) {
GST_DEBUG_OBJECT (wav, "mark DISCONT, we did a seek to another position");
wav->discont = TRUE;
}
/* and start the streaming task again */
wav->segment_running = TRUE;
if (!wav->streaming) {
gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,
wav->sinkpad);
@ -1051,9 +1027,10 @@ gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size)
if (gst_adapter_available (wav->adapter) < 8)
return FALSE;
data = gst_adapter_peek (wav->adapter, 8);
data = gst_adapter_map (wav->adapter, 8);
*tag = GST_READ_UINT32_LE (data);
*size = GST_READ_UINT32_LE (data + 4);
gst_adapter_unmap (wav->adapter, 0);
GST_DEBUG ("Next chunk size is %d bytes, type %" GST_FOURCC_FORMAT, *size,
GST_FOURCC_ARGS (*tag));
@ -1324,12 +1301,16 @@ gst_wavparse_stream_headers (GstWavParse * wav)
if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
goto exit;
} else {
guint8 *data;
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
&buf)) != GST_FLOW_OK)
goto header_read_error;
tag = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf));
size = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf) + 4);
data = gst_buffer_map (buf, NULL, NULL, -1);
tag = GST_READ_UINT32_LE (data);
size = GST_READ_UINT32_LE (data + 4);
gst_buffer_unmap (buf, data, -1);
}
GST_INFO_OBJECT (wav,
@ -1392,16 +1373,17 @@ gst_wavparse_stream_headers (GstWavParse * wav)
goto exit;
}
gst_adapter_flush (wav->adapter, 8);
data = gst_adapter_peek (wav->adapter, data_size);
data = gst_adapter_map (wav->adapter, data_size);
wav->fact = GST_READ_UINT32_LE (data);
gst_adapter_flush (wav->adapter, GST_ROUND_UP_2 (size));
gst_adapter_unmap (wav->adapter, GST_ROUND_UP_2 (size));
} else {
gst_buffer_unref (buf);
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
data_size, &buf)) != GST_FLOW_OK)
goto header_read_error;
wav->fact = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf));
gst_buffer_extract (buf, 0, &wav->fact, 4);
wav->fact = GUINT32_FROM_LE (wav->fact);
gst_buffer_unref (buf);
}
GST_DEBUG_OBJECT (wav, "have fact %u", wav->fact);
@ -1418,6 +1400,7 @@ gst_wavparse_stream_headers (GstWavParse * wav)
case GST_RIFF_TAG_acid:{
const gst_riff_acid *acid = NULL;
const guint data_size = sizeof (gst_riff_acid);
gfloat tempo;
GST_INFO_OBJECT (wav, "Have acid chunk");
if (size < data_size) {
@ -1434,21 +1417,26 @@ gst_wavparse_stream_headers (GstWavParse * wav)
goto exit;
}
gst_adapter_flush (wav->adapter, 8);
acid = (const gst_riff_acid *) gst_adapter_peek (wav->adapter,
acid = (const gst_riff_acid *) gst_adapter_map (wav->adapter,
data_size);
tempo = acid->tempo;
gst_adapter_unmap (wav->adapter, 0);
} else {
gst_buffer_unref (buf);
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
size, &buf)) != GST_FLOW_OK)
goto header_read_error;
acid = (const gst_riff_acid *) GST_BUFFER_DATA (buf);
acid = (const gst_riff_acid *) gst_buffer_map (buf, NULL, NULL,
GST_MAP_READ);
tempo = acid->tempo;
gst_buffer_unmap (buf, (guint8 *) acid, -1);
}
/* send data as tags */
if (!wav->tags)
wav->tags = gst_tag_list_new ();
gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
GST_TAG_BEATS_PER_MINUTE, acid->tempo, NULL);
GST_TAG_BEATS_PER_MINUTE, tempo, NULL);
size = GST_ROUND_UP_2 (size);
if (wav->streaming) {
@ -1469,15 +1457,17 @@ gst_wavparse_stream_headers (GstWavParse * wav)
if (gst_adapter_available (wav->adapter) < 12) {
goto exit;
}
data = gst_adapter_peek (wav->adapter, 12);
data = gst_adapter_map (wav->adapter, 12);
ltag = GST_READ_UINT32_LE (data + 8);
gst_adapter_unmap (wav->adapter, 0);
} else {
gst_buffer_unref (buf);
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset, 12,
&buf)) != GST_FLOW_OK)
goto header_read_error;
ltag = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf) + 8);
gst_buffer_extract (buf, 8, &ltag, 4);
ltag = GUINT32_FROM_LE (ltag);
}
switch (ltag) {
case GST_RIFF_LIST_INFO:{
@ -1560,11 +1550,11 @@ gst_wavparse_stream_headers (GstWavParse * wav)
if (gst_wavparse_calculate_duration (wav)) {
gst_segment_init (&wav->segment, GST_FORMAT_TIME);
gst_segment_set_duration (&wav->segment, GST_FORMAT_TIME, wav->duration);
wav->segment.duration = wav->duration;
} else {
/* no bitrate, let downstream peer do the math, we'll feed it bytes. */
gst_segment_init (&wav->segment, GST_FORMAT_BYTES);
gst_segment_set_duration (&wav->segment, GST_FORMAT_BYTES, wav->datasize);
wav->segment.duration = wav->datasize;
}
/* now we have all the info to perform a pending seek if any, if no
@ -1799,11 +1789,6 @@ gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
gst_element_add_pad (GST_ELEMENT_CAST (wav), wav->srcpad);
gst_element_no_more_pads (GST_ELEMENT_CAST (wav));
if (wav->close_segment) {
GST_DEBUG_OBJECT (wav, "Send close segment event on newpad");
gst_pad_push_event (wav->srcpad, wav->close_segment);
wav->close_segment = NULL;
}
if (wav->start_segment) {
GST_DEBUG_OBJECT (wav, "Send start segment event on newpad");
gst_pad_push_event (wav->srcpad, wav->start_segment);
@ -1839,7 +1824,7 @@ iterate_adapter:
* amounts of data regardless of the playback rate */
desired =
MIN (gst_guint64_to_gdouble (wav->dataleft),
wav->max_buf_size * wav->segment.abs_rate);
wav->max_buf_size * ABS (wav->segment.rate));
if (desired >= wav->blockalign && wav->blockalign > 0)
desired -= (desired % wav->blockalign);
@ -1888,11 +1873,13 @@ iterate_adapter:
goto pull_error;
/* we may get a short buffer at the end of the file */
if (GST_BUFFER_SIZE (buf) < desired) {
GST_LOG_OBJECT (wav, "Got only %u bytes of data", GST_BUFFER_SIZE (buf));
if (GST_BUFFER_SIZE (buf) >= wav->blockalign) {
buf = gst_buffer_make_metadata_writable (buf);
GST_BUFFER_SIZE (buf) -= (GST_BUFFER_SIZE (buf) % wav->blockalign);
if (gst_buffer_get_size (buf) < desired) {
gsize size = gst_buffer_get_size (buf);
GST_LOG_OBJECT (wav, "Got only %" G_GSIZE_FORMAT " bytes of data", size);
if (size >= wav->blockalign) {
buf = gst_buffer_make_writable (buf);
gst_buffer_resize (buf, 0, size - (size % wav->blockalign));
} else {
gst_buffer_unref (buf);
goto found_eos;
@ -1900,13 +1887,14 @@ iterate_adapter:
}
}
obtained = GST_BUFFER_SIZE (buf);
obtained = gst_buffer_get_size (buf);
/* our positions in bytes */
pos = wav->offset - wav->datastart;
nextpos = pos + obtained;
/* update offsets, does not overflow. */
buf = gst_buffer_make_writable (buf);
GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample;
GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample;
@ -1917,11 +1905,7 @@ iterate_adapter:
/* this will also push the segment events */
gst_wavparse_add_src_pad (wav, buf);
} else {
/* If we have a pending close/start segment, send it now. */
if (G_UNLIKELY (wav->close_segment != NULL)) {
gst_pad_push_event (wav->srcpad, wav->close_segment);
wav->close_segment = NULL;
}
/* If we have a pending start segment, send it now. */
if (G_UNLIKELY (wav->start_segment != NULL)) {
gst_pad_push_event (wav->srcpad, wav->start_segment);
wav->start_segment = NULL;
@ -1937,8 +1921,7 @@ iterate_adapter:
/* update current running segment position */
if (G_LIKELY (next_timestamp >= wav->segment.start))
gst_segment_set_last_stop (&wav->segment, GST_FORMAT_TIME,
next_timestamp);
wav->segment.position = next_timestamp;
} else if (wav->fact) {
guint64 bps =
gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
@ -1956,7 +1939,7 @@ iterate_adapter:
duration = GST_CLOCK_TIME_NONE;
/* update current running segment position with byte offset */
if (G_LIKELY (nextpos >= wav->segment.start))
gst_segment_set_last_stop (&wav->segment, GST_FORMAT_BYTES, nextpos);
wav->segment.position = nextpos;
}
if ((pos > 0) && wav->vbr) {
/* don't set timestamps for VBR files if it's not the first buffer */
@ -1972,13 +1955,10 @@ iterate_adapter:
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = duration;
/* don't forget to set the caps on the buffer */
gst_buffer_set_caps (buf, GST_PAD_CAPS (wav->srcpad));
GST_LOG_OBJECT (wav,
"Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
", size:%u", GST_TIME_ARGS (timestamp), GST_TIME_ARGS (duration),
GST_BUFFER_SIZE (buf));
gst_buffer_get_size (buf));
if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
goto push_error;
@ -2068,10 +2048,19 @@ pause:
const gchar *reason = gst_flow_get_name (ret);
GST_DEBUG_OBJECT (wav, "pausing task, reason %s", reason);
wav->segment_running = FALSE;
gst_pad_pause_task (pad);
if (ret == GST_FLOW_UNEXPECTED) {
/* handle end-of-stream/segment */
/* so align our position with the end of it, if there is one
* this ensures a subsequent will arrive at correct base/acc time */
if (wav->segment.format == GST_FORMAT_TIME) {
if (wav->segment.rate > 0.0 &&
GST_CLOCK_TIME_IS_VALID (wav->segment.stop))
wav->segment.position = wav->segment.stop;
else if (wav->segment.rate < 0.0)
wav->segment.position = wav->segment.start;
}
/* add pad before we perform EOS */
if (G_UNLIKELY (wav->first)) {
wav->first = FALSE;
@ -2115,7 +2104,7 @@ gst_wavparse_chain (GstPad * pad, GstBuffer * buf)
GstFlowReturn ret;
GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
GST_LOG_OBJECT (wav, "adapter_push %u bytes", GST_BUFFER_SIZE (buf));
GST_LOG_OBJECT (wav, "adapter_push %u bytes", gst_buffer_get_size (buf));
gst_adapter_push (wav->adapter, buf);
@ -2185,22 +2174,14 @@ gst_wavparse_sink_event (GstPad * pad, GstEvent * event)
GST_LOG_OBJECT (wav, "handling %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:
case GST_EVENT_SEGMENT:
{
GstFormat format;
gdouble rate, arate;
gint64 start, stop, time, offset = 0, end_offset = -1;
gboolean update;
gint64 start, stop, offset = 0, end_offset = -1;
GstSegment segment;
/* some debug output */
gst_segment_init (&segment, GST_FORMAT_UNDEFINED);
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
&start, &stop, &time);
gst_segment_set_newsegment_full (&segment, update, rate, arate, format,
start, stop, time);
GST_DEBUG_OBJECT (wav,
"received format %d newsegment %" GST_SEGMENT_FORMAT, format,
gst_event_copy_segment (event, &segment);
GST_DEBUG_OBJECT (wav, "received newsegment %" GST_SEGMENT_FORMAT,
&segment);
if (wav->state != GST_WAVPARSE_DATA) {
@ -2210,7 +2191,10 @@ gst_wavparse_sink_event (GstPad * pad, GstEvent * event)
/* now we are either committed to TIME or BYTE format,
* and we only expect a BYTE segment, e.g. following a seek */
if (format == GST_FORMAT_BYTES) {
if (segment.format == GST_FORMAT_BYTES) {
/* handle (un)signed issues */
start = segment.start;
stop = segment.stop;
if (start > 0) {
offset = start;
start -= wav->datastart;
@ -2218,8 +2202,8 @@ gst_wavparse_sink_event (GstPad * pad, GstEvent * event)
}
if (stop > 0) {
end_offset = stop;
stop -= wav->datastart;
stop = MAX (stop, 0);
segment.stop -= wav->datastart;
segment.stop = MAX (stop, 0);
}
if (wav->segment.format == GST_FORMAT_TIME) {
guint64 bps = wav->bps;
@ -2242,19 +2226,22 @@ gst_wavparse_sink_event (GstPad * pad, GstEvent * event)
goto exit;
}
segment.start = start;
segment.stop = stop;
/* accept upstream's notion of segment and distribute along */
gst_segment_set_newsegment_full (&wav->segment, update, rate, arate,
wav->segment.format, start, stop, start);
segment.time = segment.start = segment.position;
segment.duration = wav->segment.duration;
segment.base = gst_segment_to_running_time (&wav->segment,
GST_FORMAT_TIME, wav->segment.position);
gst_segment_copy_into (&segment, &wav->segment);
/* also store the newsegment event for the streaming thread */
if (wav->start_segment)
gst_event_unref (wav->start_segment);
wav->start_segment =
gst_event_new_new_segment_full (update, rate, arate,
wav->segment.format, start, stop, start);
GST_DEBUG_OBJECT (wav, "Pushing newseg update %d, rate %g, "
"applied rate %g, format %d, start %" G_GINT64_FORMAT ", "
"stop %" G_GINT64_FORMAT, update, rate, arate, wav->segment.format,
start, stop);
GST_DEBUG_OBJECT (wav, "Storing newseg %" GST_SEGMENT_FORMAT, &segment);
wav->start_segment = gst_event_new_segment (&segment);
/* stream leftover data in current segment */
gst_wavparse_flush_data (wav);
@ -2287,9 +2274,16 @@ gst_wavparse_sink_event (GstPad * pad, GstEvent * event)
/* fall-through */
case GST_EVENT_FLUSH_STOP:
{
GstClockTime dur;
gst_adapter_clear (wav->adapter);
wav->discont = TRUE;
dur = wav->segment.duration;
gst_segment_init (&wav->segment, wav->segment.format);
wav->segment.duration = dur;
/* fall-through */
}
default:
ret = gst_pad_event_default (wav->sinkpad, event);
break;
@ -2577,7 +2571,8 @@ static gboolean
gst_wavparse_sink_activate (GstPad * sinkpad)
{
GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (sinkpad));
gboolean res;
GstQuery *query;
gboolean pull_mode;
if (wav->adapter) {
gst_adapter_clear (wav->adapter);
@ -2585,33 +2580,43 @@ gst_wavparse_sink_activate (GstPad * sinkpad)
wav->adapter = NULL;
}
if (gst_pad_check_pull_range (sinkpad)) {
GST_DEBUG ("going to pull mode");
wav->streaming = FALSE;
res = gst_pad_activate_pull (sinkpad, TRUE);
} else {
GST_DEBUG ("going to push (streaming) mode");
query = gst_query_new_scheduling ();
if (!gst_pad_peer_query (sinkpad, query)) {
gst_query_unref (query);
goto activate_push;
}
gst_query_parse_scheduling (query, &pull_mode, NULL, NULL, NULL, NULL, NULL);
gst_query_unref (query);
if (!pull_mode)
goto activate_push;
GST_DEBUG_OBJECT (sinkpad, "activating pull");
wav->streaming = FALSE;
gst_object_unref (wav);
return gst_pad_activate_pull (sinkpad, TRUE);
activate_push:
{
GST_DEBUG_OBJECT (sinkpad, "activating push");
wav->streaming = TRUE;
wav->adapter = gst_adapter_new ();
res = gst_pad_activate_push (sinkpad, TRUE);
gst_object_unref (wav);
return gst_pad_activate_push (sinkpad, TRUE);
}
gst_object_unref (wav);
return res;
}
static gboolean
gst_wavparse_sink_activate_pull (GstPad * sinkpad, gboolean active)
{
GstWavParse *wav = GST_WAVPARSE (GST_OBJECT_PARENT (sinkpad));
if (active) {
/* if we have a scheduler we can start the task */
wav->segment_running = TRUE;
return gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop,
sinkpad);
} else {
wav->segment_running = FALSE;
return gst_pad_stop_task (sinkpad);
}
};

4
gst/wavparse/gstwavparse.h
View file

@ -66,7 +66,6 @@ struct _GstWavParse {
* the format for sure */
GstCaps *caps;
GstTagList *tags;
GstEvent *close_segment;
GstEvent *start_segment;
/* WAVE decoding state */
@ -110,9 +109,8 @@ struct _GstWavParse {
gboolean got_fmt;
gboolean streaming;
/* configured segment, start/stop expressed in time */
/* configured segment, start/stop expressed in time or bytes */
GstSegment segment;
gboolean segment_running;
/* for late pad configuration */
gboolean first;