* <para>
* Parse a .wav file into raw or compressed audio.
* </para>
+ * <para>
+ * Wavparse supports both push and pull mode operations, making it possible to
+ * stream from a network source.
+ * </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* <programlisting>
* gst-launch gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink
* </programlisting>
- * Stream data from
+ * Stream data from a network url.
* </para>
* </refsect2>
*
- * Last reviewed on 2006-03-03 (0.10.3)
+ * Last reviewed on 2007-02-14 (0.10.6)
*/
/*
GST_DEBUG_CATEGORY_STATIC (wavparse_debug);
#define GST_CAT_DEFAULT (wavparse_debug)
-static void gst_wavparse_base_init (gpointer g_class);
-static void gst_wavparse_class_init (GstWavParseClass * klass);
-static void gst_wavparse_init (GstWavParse * wavparse);
static void gst_wavparse_dispose (GObject * object);
static gboolean gst_wavparse_sink_activate (GstPad * sinkpad);
gboolean active);
static gboolean gst_wavparse_send_event (GstElement * element,
GstEvent * event);
-static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstBuffer * buf);
static GstStateChangeReturn gst_wavparse_change_state (GstElement * element,
GstStateChange transition);
-static gboolean gst_wavparse_pad_query (GstPad * pad, GstQuery * query);
static const GstQueryType *gst_wavparse_get_query_types (GstPad * pad);
+static gboolean gst_wavparse_pad_query (GstPad * pad, GstQuery * query);
static gboolean gst_wavparse_pad_convert (GstPad * pad,
GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
+static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstBuffer * buf);
static void gst_wavparse_loop (GstPad * pad);
static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event);
-static void gst_wavparse_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec);
static const GstElementDetails gst_wavparse_details =
GST_ELEMENT_DETAILS ("WAV audio demuxer",
);
-static GstElementClass *parent_class = NULL;
-
-GType
-gst_wavparse_get_type (void)
-{
- static GType wavparse_type = 0;
-
- if (!wavparse_type) {
- static const GTypeInfo wavparse_info = {
- sizeof (GstWavParseClass),
- gst_wavparse_base_init,
- NULL,
- (GClassInitFunc) gst_wavparse_class_init,
- NULL,
- NULL,
- sizeof (GstWavParse),
- 0,
- (GInstanceInitFunc) gst_wavparse_init,
- };
-
- wavparse_type =
- g_type_register_static (GST_TYPE_ELEMENT, "GstWavParse",
- &wavparse_info, 0);
- }
- return wavparse_type;
-}
+#define DEBUG_INIT(bla) \
+ 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)
parent_class = g_type_class_peek_parent (klass);
- object_class->get_property = gst_wavparse_get_property;
object_class->dispose = gst_wavparse_dispose;
gstelement_class->change_state = gst_wavparse_change_state;
G_OBJECT_CLASS (parent_class)->dispose (object);
}
-
static void
gst_wavparse_reset (GstWavParse * wavparse)
{
if (wavparse->seek_event)
gst_event_unref (wavparse->seek_event);
wavparse->seek_event = NULL;
-
- /* we keep the segment info in time */
- gst_segment_init (&wavparse->segment, GST_FORMAT_TIME);
+ if (wavparse->adapter)
+ gst_adapter_clear (wavparse->adapter);
}
static void
-gst_wavparse_init (GstWavParse * wavparse)
+gst_wavparse_init (GstWavParse * wavparse, GstWavParseClass * g_class)
{
gst_wavparse_reset (wavparse);
GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_pull));
gst_pad_set_chain_function (wavparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavparse_chain));
- gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->sinkpad);
+ gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->sinkpad);
/* src, will be created later */
wavparse->srcpad = NULL;
gst_wavparse_destroy_sourcepad (GstWavParse * wavparse)
{
if (wavparse->srcpad) {
- gst_element_remove_pad (GST_ELEMENT (wavparse), wavparse->srcpad);
+ gst_element_remove_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad);
wavparse->srcpad = NULL;
}
}
GST_DEBUG_OBJECT (wavparse, "srcpad created");
}
-static void
-gst_wavparse_get_property (GObject * object,
- guint prop_id, GValue * value, GParamSpec * pspec)
-{
- GstWavParse *wavparse;
-
- wavparse = GST_WAVPARSE (object);
-
- switch (prop_id) {
- default:
- break;
- }
-}
-
-
#if 0
static void
gst_riff_strf_auds *header = NULL;
GstCaps *caps;
- if (!gst_riff_read_strf_auds (wav, &header)) {
- g_warning ("Not fmt");
- return FALSE;
- }
+ if (!gst_riff_read_strf_auds (wav, &header))
+ goto no_fmt;
wav->format = header->format;
wav->rate = header->rate;
wav->channels = header->channels;
- if (wav->channels == 0) {
- GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
- ("Stream claims to contain zero channels - invalid data"));
- g_free (header);
- return FALSE;
- }
+ if (wav->channels == 0)
+ goto no_channels;
+
wav->blockalign = header->blockalign;
wav->width = (header->blockalign * 8) / header->channels;
wav->depth = header->size;
wav->bps = header->av_bps;
- if (wav->bps <= 0) {
- GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
- ("Stream claims to bitrate of <= zero - invalid data"));
- g_free (header);
- return FALSE;
- }
+ if (wav->bps <= 0)
+ goto no_bps;
/* Note: gst_riff_create_audio_caps might need to fix values in
* the header header depending on the format, so call it first */
caps = gst_riff_create_audio_caps (header->format, NULL, header, NULL);
-
g_free (header);
- if (caps) {
- gst_wavparse_create_sourcepad (wav);
- gst_pad_use_fixed_caps (wav->srcpad);
- gst_pad_set_active (wav->srcpad, TRUE);
- gst_pad_set_caps (wav->srcpad, caps);
- gst_caps_free (caps);
- gst_element_add_pad (GST_ELEMENT (wav), wav->srcpad);
- gst_element_no_more_pads (GST_ELEMENT (wav));
- GST_DEBUG ("frequency %d, channels %d", wav->rate, wav->channels);
- } else {
+ if (caps == NULL)
+ goto no_caps;
+
+ gst_wavparse_create_sourcepad (wav);
+ gst_pad_use_fixed_caps (wav->srcpad);
+ gst_pad_set_active (wav->srcpad, TRUE);
+ gst_pad_set_caps (wav->srcpad, caps);
+ gst_caps_free (caps);
+ gst_element_add_pad (GST_ELEMENT_CAST (wav), wav->srcpad);
+ gst_element_no_more_pads (GST_ELEMENT_CAST (wav));
+
+ GST_DEBUG ("frequency %d, channels %d", wav->rate, wav->channels);
+
+ return TRUE;
+
+ /* ERRORS */
+no_fmt:
+ {
+ GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
+ ("No FMT tag found"));
+ return FALSE;
+ }
+no_channels:
+ {
+ GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
+ ("Stream claims to contain zero channels - invalid data"));
+ g_free (header);
+ return FALSE;
+ }
+no_bps:
+ {
+ GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
+ ("Stream claims to bitrate of <= zero - invalid data"));
+ g_free (header);
+ return FALSE;
+ }
+no_caps:
+ {
GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL), (NULL));
return FALSE;
}
-
- return TRUE;
}
static gboolean
}
#endif
-
-
static gboolean
gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
{
not_wav:
{
GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
- ("File is not an WAVE file: %" GST_FOURCC_FORMAT,
+ ("File is not a WAVE file: %" GST_FOURCC_FORMAT,
GST_FOURCC_ARGS (doctype)));
return FALSE;
}
if ((res = gst_pad_pull_range (wav->sinkpad,
wav->offset, 12, &buf)) != GST_FLOW_OK)
return res;
- else if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), buf))
+ else if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), buf))
return GST_FLOW_ERROR;
wav->offset += 12;
GstSegment seeksegment = { 0, };
if (event) {
+ GstFormat fmt;
+
GST_DEBUG_OBJECT (wav, "doing seek with event");
gst_event_parse_seek (event, &rate, &format, &flags,
&cur_type, &cur, &stop_type, &stop);
+ /* no negative rates yet */
+ if (rate < 0.0)
+ goto negative_rate;
+
+ fmt = wav->segment.format;
+
/* we have to have a format as the segment format. Try to convert
* if not. */
- if (format != GST_FORMAT_TIME) {
- GstFormat fmt;
-
- fmt = GST_FORMAT_TIME;
+ if (format != wav->segment.format) {
res = TRUE;
if (cur_type != GST_SEEK_TYPE_NONE)
res = gst_pad_query_convert (wav->srcpad, format, cur, &fmt, &cur);
stop_type = GST_SEEK_TYPE_SET;
}
+ /* get flush flag */
flush = flags & GST_SEEK_FLAG_FLUSH;
- if (flush && wav->srcpad) {
- GST_DEBUG_OBJECT (wav, "sending flush start");
- gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ());
+ /* now we need to make sure the streaming thread is stopped. We do this by
+ * either sending a FLUSH_START event downstream which will cause the
+ * streaming thread to stop with a WRONG_STATE.
+ * For a non-flushing seek we simply pause the task, which will happen as soon
+ * as it completes one iteration (and thus might block when the sink is
+ * blocking in preroll). */
+ if (flush) {
+ if (wav->srcpad) {
+ GST_DEBUG_OBJECT (wav, "sending flush start");
+ gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ());
+ }
} else {
gst_pad_pause_task (wav->sinkpad);
}
+ /* we should now be able to grab the streaming thread because we stopped it
+ * with the above flush/pause code */
GST_PAD_STREAM_LOCK (wav->sinkpad);
/* copy segment, we need this because we still need the old
* segment when we close the current segment. */
memcpy (&seeksegment, &wav->segment, sizeof (GstSegment));
+ /* configure the seek parameters in the seeksegment. We will then have the
+ * 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,
cur_type, cur, stop_type, stop, &update);
}
- if ((stop = seeksegment.stop) == GST_CLOCK_TIME_NONE)
+ /* figure out the last position we need to play. If it's configured (stop !=
+ * -1), use that, else we play until the total duration of the file */
+ if ((stop = seeksegment.stop) == -1)
stop = seeksegment.duration;
GST_DEBUG_OBJECT (wav, "cur_type =%d", cur_type);
if ((cur_type != GST_SEEK_TYPE_NONE)) {
- wav->offset =
- gst_util_uint64_scale_int (seeksegment.last_stop, wav->bps, GST_SECOND);
+ /* 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 (wav->bps)
+ wav->offset =
+ gst_util_uint64_scale_int (seeksegment.last_stop, wav->bps,
+ GST_SECOND);
+ else
+ wav->offset = seeksegment.last_stop;
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 (stop_type != GST_SEEK_TYPE_NONE) {
- wav->end_offset = gst_util_uint64_scale_int (stop, wav->bps, GST_SECOND);
+ if (wav->bps)
+ wav->end_offset = gst_util_uint64_scale_int (stop, wav->bps, GST_SECOND);
+ else
+ wav->end_offset = stop;
GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
wav->end_offset -= (wav->end_offset % wav->bytes_per_sample);
GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
if (gst_pad_query_peer_duration (wav->sinkpad, &bformat, &upstream_size))
wav->end_offset = MIN (wav->end_offset, upstream_size);
+ /* this is the range of bytes we will use for playback */
wav->offset = MIN (wav->offset, wav->end_offset);
wav->dataleft = wav->end_offset - wav->offset;
/* prepare for streaming again */
if (wav->srcpad) {
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 last_stop. */
+ * 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.accum, wav->segment.last_stop);
wav->segment.rate, wav->segment.format,
wav->segment.accum, wav->segment.last_stop, wav->segment.accum));
+ /* keep track of our last_stop */
seeksegment.accum = wav->segment.last_stop;
}
}
+ /* now we did the seek and can activate the new segment values */
memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
+ /* if we're doing a segment seek, post a SEGMENT_START message */
if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
- gst_element_post_message (GST_ELEMENT (wav),
- gst_message_new_segment_start (GST_OBJECT (wav),
+ gst_element_post_message (GST_ELEMENT_CAST (wav),
+ gst_message_new_segment_start (GST_OBJECT_CAST (wav),
wav->segment.format, wav->segment.last_stop));
}
- /* now send the newsegment */
- GST_DEBUG_OBJECT (wav, "Sending newsegment from %" G_GINT64_FORMAT
+ /* now create the newsegment */
+ GST_DEBUG_OBJECT (wav, "Creating newsegment from %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, wav->segment.last_stop, stop);
/* store the newsegment event so it can be sent from the streaming thread. */
wav->segment.format, wav->segment.last_stop, stop,
wav->segment.last_stop);
+ /* and start the streaming task again */
wav->segment_running = TRUE;
if (!wav->streaming) {
gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,
return TRUE;
/* ERRORS */
+negative_rate:
+ {
+ GST_DEBUG_OBJECT (wav, "negative playback rates are not supported yet.");
+ return FALSE;
+ }
no_format:
{
GST_DEBUG_OBJECT (wav, "unsupported format given, seek aborted.");
{
const guint8 *data = NULL;
- if (gst_adapter_available (wav->adapter) < 8) {
+ if (gst_adapter_available (wav->adapter) < 8)
return FALSE;
- }
data = gst_adapter_peek (wav->adapter, 8);
*tag = GST_READ_UINT32_LE (data);
if (!gst_wavparse_peek_chunk (wav, &tag, &size))
return GST_FLOW_OK;
- buf = gst_buffer_new ();
- gst_buffer_ref (buf);
gst_adapter_flush (wav->adapter, 8);
wav->offset += 8;
- GST_BUFFER_DATA (buf) = (guint8 *) gst_adapter_peek (wav->adapter, size);
- GST_BUFFER_SIZE (buf) = size;
+
+ buf = gst_adapter_take_buffer (wav->adapter, size);
} else {
- if ((res = gst_riff_read_chunk (GST_ELEMENT (wav), wav->sinkpad,
+ if ((res = gst_riff_read_chunk (GST_ELEMENT_CAST (wav), wav->sinkpad,
&wav->offset, &tag, &buf)) != GST_FLOW_OK)
return res;
}
if (tag != GST_RIFF_TAG_fmt)
goto invalid_wav;
- if (!(gst_riff_parse_strf_auds (GST_ELEMENT (wav), buf, &header, &extra)))
+ if (!(gst_riff_parse_strf_auds (GST_ELEMENT_CAST (wav), buf, &header,
+ &extra)))
goto parse_header_error;
if (wav->streaming) {
gst_adapter_flush (wav->adapter, size);
wav->offset += size;
- GST_BUFFER_DATA (buf) = NULL;
gst_buffer_unref (buf);
+ buf = NULL;
}
/* Note: gst_riff_create_audio_caps might need to fix values in
wav->channels = header->channels;
wav->blockalign = header->blockalign;
wav->depth = header->size;
- wav->bps = header->av_bps;
+ wav->av_bps = header->av_bps;
g_free (header);
if (wav->channels == 0)
goto no_channels;
- if (wav->bps == 0 && (wav->format == GST_RIFF_WAVE_FORMAT_MPEGL12 ||
- wav->format == GST_RIFF_WAVE_FORMAT_MPEGL3)) {
- /* Note: ugly workaround for mp2/mp3 embedded in wav, that relies on the
- * bitrate inside the mpeg stream */
- /* wav->bps = 1; */
- GST_INFO ("WAV file with bps==0 and format=mp2/3");
+ /* do format specific handling */
+ switch (wav->format) {
+ case GST_RIFF_WAVE_FORMAT_MPEGL12:
+ case GST_RIFF_WAVE_FORMAT_MPEGL3:
+ {
+ /* Note: workaround for mp2/mp3 embedded in wav, that relies on the
+ * bitrate inside the mpeg stream */
+ GST_INFO ("resetting bps from %d to 0 for mp2/3", wav->av_bps);
+ wav->bps = 0;
+ break;
+ }
+ default:
+ /* use the configured bps */
+ wav->bps = wav->av_bps;
+ break;
}
wav->width = (wav->blockalign * 8) / wav->channels;
GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign);
GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width);
GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth);
- GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps);
+ GST_DEBUG_OBJECT (wav, "av_bps = %u", (guint) wav->av_bps);
GST_DEBUG_OBJECT (wav, "frequency = %d", wav->rate);
GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels);
GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample);
+ /* bps can be 0 when we don't have a valid bitrate (mostly for compressed
+ * formats). This will make the element output a BYTE format segment and
+ * will not timestamp the outgoing buffers.
+ */
+ GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps);
+
GST_DEBUG_OBJECT (wav, "caps = %" GST_PTR_FORMAT, caps);
/* create pad later so we can sniff the first few bytes
gst_wavparse_get_upstream_size (wav, &upstream_size);
- /*
- wav is a st00pid format, we don't know for sure where data starts.
- So we have to go bit by bit until we find the 'data' header
+ /* wav is a st00pid format, we don't know for sure where data starts.
+ * So we have to go bit by bit until we find the 'data' header
*/
switch (tag) {
/* TODO : Implement the various cases */
(guint64) wav->fact);
GST_DEBUG_OBJECT (wav, "calculated bps : %d", wav->bps);
}
- if (wav->bps <= 0)
- goto no_bitrate;
- duration = gst_util_uint64_scale_int (wav->datasize, GST_SECOND, wav->bps);
- GST_DEBUG_OBJECT (wav, "Got duration %" GST_TIME_FORMAT,
- GST_TIME_ARGS (duration));
- gst_segment_set_duration (&wav->segment, GST_FORMAT_TIME, duration);
+ if (wav->bps > 0) {
+ duration = gst_util_uint64_scale_int (wav->datasize, GST_SECOND, wav->bps);
+ GST_DEBUG_OBJECT (wav, "Got duration %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (duration));
+ gst_segment_init (&wav->segment, GST_FORMAT_TIME);
+ gst_segment_set_duration (&wav->segment, GST_FORMAT_TIME, 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);
+ }
/* now we have all the info to perform a pending seek if any, if no
* event, this will still do the right thing and it will also send
gst_event_replace (event_p, NULL);
wav->state = GST_WAVPARSE_DATA;
+
return GST_FLOW_OK;
/* ERROR */
invalid_wav:
{
- GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
+ GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
("Invalid WAV header (no fmt at start): %"
GST_FOURCC_FORMAT, GST_FOURCC_ARGS (tag)));
g_free (codec_name);
g_free (codec_name);
return GST_FLOW_ERROR;
}
-no_bitrate:
- {
- GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
- ("Stream claims to have a bitrate of <= zero - invalid data"));
- g_free (header);
- g_free (codec_name);
- return GST_FLOW_ERROR;
- }
no_bytes_per_sample:
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
}
}
-
/*
* Read WAV file tag when streaming
*/
/* _take flushes the data */
tmp = gst_adapter_take_buffer (wav->adapter, 12);
+
GST_DEBUG ("Parsing wav header");
- if (!gst_wavparse_parse_file_header (GST_ELEMENT (wav), tmp)) {
+ if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), tmp))
return GST_FLOW_ERROR;
- }
wav->offset += 12;
/* Go to next state */
gst_pad_set_caps (wav->srcpad, wav->caps);
gst_caps_replace (&wav->caps, NULL);
- gst_element_add_pad (GST_ELEMENT (wav), wav->srcpad);
- gst_element_no_more_pads (GST_ELEMENT (wav));
+ gst_element_add_pad (GST_ELEMENT_CAST (wav), wav->srcpad);
+ gst_element_no_more_pads (GST_ELEMENT_CAST (wav));
GST_DEBUG_OBJECT (wav, "Send newsegment event on newpad");
- gst_pad_push_event (wav->srcpad, wav->newsegment);
- wav->newsegment = NULL;
+ if (wav->newsegment) {
+ gst_pad_push_event (wav->srcpad, wav->newsegment);
+ wav->newsegment = NULL;
+ }
if (wav->tags) {
- gst_element_found_tags_for_pad (GST_ELEMENT (wav), wav->srcpad, wav->tags);
+ gst_element_found_tags_for_pad (GST_ELEMENT_CAST (wav), wav->srcpad,
+ wav->tags);
wav->tags = NULL;
}
}
GstBuffer *buf = NULL;
GstFlowReturn res = GST_FLOW_OK;
guint64 desired, obtained;
- GstClockTime timestamp, next_timestamp;
+ GstClockTime timestamp, next_timestamp, duration;
guint64 pos, nextpos;
iterate_adapter:
obtained = GST_BUFFER_SIZE (buf);
- /* our positions */
+ /* our positions in bytes */
pos = wav->offset - wav->datastart;
nextpos = pos + obtained;
GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample;
GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample;
- /* and timestamps, be carefull for overflows */
- timestamp = gst_util_uint64_scale_int (pos, GST_SECOND, wav->bps);
- next_timestamp = gst_util_uint64_scale_int (nextpos, GST_SECOND, wav->bps);
+ if (wav->bps > 0) {
+ /* and timestamps if we have a bitrate, be carefull for overflows */
+ timestamp = gst_util_uint64_scale_int (pos, GST_SECOND, wav->bps);
+ next_timestamp = gst_util_uint64_scale_int (nextpos, GST_SECOND, wav->bps);
+ duration = next_timestamp - timestamp;
- GST_BUFFER_TIMESTAMP (buf) = timestamp;
- GST_BUFFER_DURATION (buf) = next_timestamp - timestamp;
+ /* update current running segment position */
+ gst_segment_set_last_stop (&wav->segment, GST_FORMAT_TIME, next_timestamp);
+ } else {
+ /* no bitrate, don't timestamp */
+ timestamp = GST_CLOCK_TIME_NONE;
+ next_timestamp = GST_CLOCK_TIME_NONE;
+ duration = GST_CLOCK_TIME_NONE;
+ /* update current running segment position with byte offset */
+ gst_segment_set_last_stop (&wav->segment, GST_FORMAT_BYTES, nextpos);
+ }
- /* update current running segment position */
- gst_segment_set_last_stop (&wav->segment, GST_FORMAT_TIME, next_timestamp);
+ 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 (GST_BUFFER_TIMESTAMP (buf)),
- GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), GST_BUFFER_SIZE (buf));
+ ", size:%u", GST_TIME_ARGS (timestamp), GST_TIME_ARGS (duration),
+ GST_BUFFER_SIZE (buf));
if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
goto push_error;
wav->dataleft = 0;
}
wav->offset += obtained;
+
/* Iterate until need more data, so adapter size won't grow */
if (wav->streaming) {
GST_LOG_OBJECT (wav,
wav->end_offset);
goto iterate_adapter;
}
-
return res;
/* ERROR */
found_eos:
{
GST_DEBUG_OBJECT (wav, "found EOS");
- /* we completed the segment */
- wav->segment_running = FALSE;
- if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
- GstClockTime stop;
-
- if ((stop = wav->segment.stop) == -1)
- stop = wav->segment.duration;
-
- gst_element_post_message (GST_ELEMENT (wav),
- gst_message_new_segment_done (GST_OBJECT (wav), GST_FORMAT_TIME,
- stop));
- } else {
- if (G_UNLIKELY (wav->first)) {
- wav->first = FALSE;
- gst_wavparse_add_src_pad (wav, NULL);
- }
- gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
- }
- return GST_FLOW_WRONG_STATE;
+ return GST_FLOW_UNEXPECTED;
}
pull_error:
{
default:
g_assert_not_reached ();
}
-
return;
/* ERRORS */
pause:
- GST_LOG_OBJECT (wav, "pausing task %d", ret);
- gst_pad_pause_task (wav->sinkpad);
- if (GST_FLOW_IS_FATAL (ret)) {
- /* for fatal errors we post an error message */
- GST_ELEMENT_ERROR (wav, STREAM, FAILED,
- (_("Internal data stream error.")),
- ("streaming stopped, reason %s", gst_flow_get_name (ret)));
- if (wav->srcpad != NULL)
- gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
+ {
+ 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 (GST_FLOW_IS_FATAL (ret) || ret == GST_FLOW_NOT_LINKED) {
+ if (ret == GST_FLOW_UNEXPECTED) {
+ /* add pad before we perform EOS */
+ if (G_UNLIKELY (wav->first)) {
+ wav->first = FALSE;
+ gst_wavparse_add_src_pad (wav, NULL);
+ }
+ /* perform EOS logic */
+ if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
+ GstClockTime stop;
+
+ if ((stop = wav->segment.stop) == -1)
+ stop = wav->segment.duration;
+
+ gst_element_post_message (GST_ELEMENT_CAST (wav),
+ gst_message_new_segment_done (GST_OBJECT_CAST (wav),
+ wav->segment.format, stop));
+ } else {
+ if (wav->srcpad != NULL)
+ gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
+ }
+ } else {
+ /* for fatal errors we post an error message, post the error
+ * first so the app knows about the error first. */
+ GST_ELEMENT_ERROR (wav, STREAM, FAILED,
+ (_("Internal data flow error.")),
+ ("streaming task paused, reason %s (%d)", reason, ret));
+ if (wav->srcpad != NULL)
+ gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
+ }
+ }
+ return;
}
}
case GST_WAVPARSE_START:
GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_START");
if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK)
- goto pause;
+ goto done;
if (wav->state != GST_WAVPARSE_HEADER)
break;
/* otherwise fall-through */
-
case GST_WAVPARSE_HEADER:
GST_DEBUG_OBJECT (wav, "GST_WAVPARSE_HEADER");
if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
- goto pause;
+ goto done;
if (!wav->got_fmt || wav->datastart == 0)
break;
wav->state = GST_WAVPARSE_DATA;
- /* fall-through */
+ /* fall-through */
case GST_WAVPARSE_DATA:
if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
- goto pause;
+ goto done;
break;
default:
g_assert_not_reached ();
}
-
- return ret;
-
-pause:
- GST_LOG_OBJECT (wav, "pausing task %d", ret);
- gst_pad_pause_task (wav->sinkpad);
- if (GST_FLOW_IS_FATAL (ret)) {
- /* for fatal errors we post an error message */
- GST_ELEMENT_ERROR (wav, STREAM, FAILED,
- (_("Internal data stream error.")),
- ("streaming stopped, reason %s", gst_flow_get_name (ret)));
- if (wav->srcpad != NULL)
- gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
- }
+done:
return ret;
}
wavparse = GST_WAVPARSE (gst_pad_get_parent (pad));
- if (wavparse->bytes_per_sample == 0)
- goto no_bytes_per_sample;
+ if (*dest_format == src_format) {
+ *dest_value = src_value;
+ return TRUE;
+ }
if (wavparse->bps == 0)
goto no_bps;
switch (*dest_format) {
case GST_FORMAT_DEFAULT:
*dest_value = src_value / wavparse->bytes_per_sample;
+ /* make sure we end up on a sample boundary */
+ *dest_value -= *dest_value % wavparse->bytes_per_sample;
break;
case GST_FORMAT_TIME:
*dest_value =
res = FALSE;
goto done;
}
- *dest_value -= *dest_value % wavparse->bytes_per_sample;
break;
case GST_FORMAT_DEFAULT:
return res;
/* ERRORS */
-no_bytes_per_sample:
- {
- GST_DEBUG_OBJECT (wavparse,
- "bytes_per_sample 0, probably an mp3 - channels %d, width %d",
- wavparse->channels, wavparse->width);
- res = FALSE;
- goto done;
- }
no_bps:
{
GST_DEBUG_OBJECT (wavparse, "bps 0, cannot convert");
gint64 curb;
gint64 cur;
GstFormat format;
- gboolean res = TRUE;
curb = wav->offset - wav->datastart;
gst_query_parse_position (query, &format, NULL);
gint64 endb;
gint64 end;
GstFormat format;
- gboolean res = TRUE;
endb = wav->datasize;
gst_query_parse_duration (query, &format, NULL);
gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, endb,
&format, &end);
}
- }
break;
+ }
default:
format = GST_FORMAT_BYTES;
end = endb;
case GST_EVENT_SEEK:
{
res = gst_wavparse_perform_seek (wavparse, event);
+ gst_event_unref (event);
break;
}
default:
- res = FALSE;
+ res = gst_pad_push_event (wavparse->sinkpad, event);
break;
}
-
- gst_event_unref (event);
-
return res;
}
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
- case GST_STATE_CHANGE_PAUSED_TO_READY:{
- GstEvent **event_p = &wav->seek_event;
-
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_wavparse_destroy_sourcepad (wav);
- gst_event_replace (event_p, NULL);
gst_wavparse_reset (wav);
- if (wav->adapter) {
- gst_adapter_clear (wav->adapter);
- }
break;
- }
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
{
gst_riff_init ();
- GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
-
return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY,
GST_TYPE_WAVPARSE);
}
projects / platform / upstream / gst-plugins-good.git / commitdiff