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;
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
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
}
static void
-gst_wavparse_init (GstWavParse * wavparse, GstWavParseClass * g_class)
+gst_wavparse_init (GstWavParse * wavparse)
{
gst_wavparse_reset (wavparse);
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 */
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);
* 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);
}
/* 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);
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));
}
}
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);
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));
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,
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);
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) {
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) {
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, <ag, 4);
+ ltag = GUINT32_FROM_LE (ltag);
}
switch (ltag) {
case GST_RIFF_LIST_INFO:{
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
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);
* 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);
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;
}
}
- 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;
/* 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;
/* 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);
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 */
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;
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;
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);
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) {
/* 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;
}
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;
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);
/* 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;
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);
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);
}
};
projects / platform / upstream / gst-plugins-good.git / commitdiff