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gst/wavparse/gstwavparse.*: Update docs.

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Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_class_init),
(gst_wavparse_reset), (gst_wavparse_init),
(gst_wavparse_destroy_sourcepad), (gst_wavparse_fmt),
(gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
(gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
(gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
(gst_wavparse_loop), (gst_wavparse_chain),
(gst_wavparse_pad_convert), (gst_wavparse_pad_query),
(gst_wavparse_srcpad_event), (gst_wavparse_change_state),
(plugin_init):
* gst/wavparse/gstwavparse.h:
Update docs.
Use boilerplate.
Various code cleanups.
When the bitrate is not known (bps == 0 or compressed formats) let
downstream element guestimate the duration and position and don't
generate timestamps or durations. Fixes #405213.
Fix EOS and ERROR conditions in chain mode, we just need to forward the
error flowreturn upstream.
This commit is contained in:
Wim Taymans 2007-02-14 09:55:47 +00:00
parent b1aa8fef18
commit 2644d7178b
3 changed files with 281 additions and 240 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

23
ChangeLog
View file

@ -1,3 +1,26 @@
2007-02-14 Wim Taymans,,, <wim@fluendo.com>
* gst/wavparse/gstwavparse.c: (gst_wavparse_class_init),
(gst_wavparse_reset), (gst_wavparse_init),
(gst_wavparse_destroy_sourcepad), (gst_wavparse_fmt),
(gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
(gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
(gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
(gst_wavparse_loop), (gst_wavparse_chain),
(gst_wavparse_pad_convert), (gst_wavparse_pad_query),
(gst_wavparse_srcpad_event), (gst_wavparse_change_state),
(plugin_init):
* gst/wavparse/gstwavparse.h:
Update docs.
Use boilerplate.
Various code cleanups.
When the bitrate is not known (bps == 0 or compressed formats) let
downstream element guestimate the duration and position and don't
generate timestamps or durations. Fixes #405213.
Fix EOS and ERROR conditions in chain mode, we just need to forward the
error flowreturn upstream.
2007-02-13 Jan Schmidt <thaytan@mad.scientist.com>
* ext/gconf/Makefile.am:

491
gst/wavparse/gstwavparse.c
View file

@ -26,6 +26,10 @@
* <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>
@ -38,11 +42,11 @@
* <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)
*/
/*
@ -63,9 +67,6 @@
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);
@ -73,20 +74,18 @@ static gboolean gst_wavparse_sink_activate_pull (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",
@ -141,33 +140,11 @@ static GstStaticPadTemplate src_template_factory =
);
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)
@ -193,7 +170,6 @@ gst_wavparse_class_init (GstWavParseClass * klass)
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;
@ -217,7 +193,6 @@ gst_wavparse_dispose (GObject * object)
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_wavparse_reset (GstWavParse * wavparse)
{
@ -242,13 +217,12 @@ 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);
@ -261,7 +235,7 @@ gst_wavparse_init (GstWavParse * 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;
@ -271,7 +245,7 @@ static void
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;
}
}
@ -296,21 +270,6 @@ gst_wavparse_create_sourcepad (GstWavParse * wavparse)
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
@ -566,52 +525,68 @@ gst_wavparse_fmt (GstWavParse * wav)
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) {
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)
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 == 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;
}
wav->blockalign = header->blockalign;
wav->width = (header->blockalign * 8) / header->channels;
wav->depth = header->size;
wav->bps = header->av_bps;
if (wav->bps <= 0) {
no_bps:
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("Stream claims to bitrate of <= zero - invalid data"));
g_free (header);
return FALSE;
}
/* 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 {
no_caps:
{
GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL), (NULL));
return FALSE;
}
return TRUE;
}
static gboolean
@ -717,8 +692,6 @@ gst_wavparse_other (GstWavParse * wav)
}
#endif
static gboolean
gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
{
@ -736,7 +709,7 @@ 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;
}
@ -751,7 +724,7 @@ gst_wavparse_stream_init (GstWavParse * wav)
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;
@ -781,17 +754,22 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
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);
@ -809,34 +787,56 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
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);
@ -848,7 +848,10 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
}
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);
@ -865,6 +868,7 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
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;
@ -876,11 +880,12 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
/* 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);
@ -889,20 +894,23 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
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. */
@ -913,6 +921,7 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
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,
@ -924,6 +933,11 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
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.");
@ -946,9 +960,8 @@ gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size)
{
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);
@ -1027,14 +1040,12 @@ gst_wavparse_stream_headers (GstWavParse * wav)
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;
}
@ -1055,14 +1066,15 @@ gst_wavparse_stream_headers (GstWavParse * wav)
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
@ -1078,19 +1090,28 @@ gst_wavparse_stream_headers (GstWavParse * wav)
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;
@ -1105,11 +1126,17 @@ gst_wavparse_stream_headers (GstWavParse * wav)
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
@ -1149,9 +1176,8 @@ gst_wavparse_stream_headers (GstWavParse * wav)
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 */
@ -1235,13 +1261,18 @@ gst_wavparse_stream_headers (GstWavParse * wav)
(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
@ -1252,12 +1283,13 @@ gst_wavparse_stream_headers (GstWavParse * wav)
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);
@ -1279,14 +1311,6 @@ no_channels:
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),
@ -1311,7 +1335,6 @@ header_read_error:
}
}
/*
* Read WAV file tag when streaming
*/
@ -1323,10 +1346,10 @@ gst_wavparse_parse_stream_init (GstWavParse * wav)
/* _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 */
@ -1403,15 +1426,18 @@ gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
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;
}
}
@ -1424,7 +1450,7 @@ gst_wavparse_stream_data (GstWavParse * wav)
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:
@ -1478,7 +1504,7 @@ iterate_adapter:
obtained = GST_BUFFER_SIZE (buf);
/* our positions */
/* our positions in bytes */
pos = wav->offset - wav->datastart;
nextpos = pos + obtained;
@ -1486,23 +1512,33 @@ iterate_adapter:
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;
/* 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);
}
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);
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;
@ -1513,6 +1549,7 @@ iterate_adapter:
wav->dataleft = 0;
}
wav->offset += obtained;
/* Iterate until need more data, so adapter size won't grow */
if (wav->streaming) {
GST_LOG_OBJECT (wav,
@ -1520,32 +1557,13 @@ iterate_adapter:
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:
{
@ -1598,20 +1616,49 @@ gst_wavparse_loop (GstPad * pad)
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;
}
}
@ -1629,45 +1676,31 @@ gst_wavparse_chain (GstPad * pad, GstBuffer * buf)
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;
}
@ -1697,8 +1730,10 @@ gst_wavparse_pad_convert (GstPad * pad,
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;
@ -1708,6 +1743,8 @@ gst_wavparse_pad_convert (GstPad * pad,
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 =
@ -1717,7 +1754,6 @@ gst_wavparse_pad_convert (GstPad * pad,
res = FALSE;
goto done;
}
*dest_value -= *dest_value % wavparse->bytes_per_sample;
break;
case GST_FORMAT_DEFAULT:
@ -1764,14 +1800,6 @@ done:
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");
@ -1810,7 +1838,6 @@ gst_wavparse_pad_query (GstPad * pad, GstQuery * query)
gint64 curb;
gint64 cur;
GstFormat format;
gboolean res = TRUE;
curb = wav->offset - wav->datastart;
gst_query_parse_position (query, &format, NULL);
@ -1835,7 +1862,6 @@ gst_wavparse_pad_query (GstPad * pad, GstQuery * query)
gint64 endb;
gint64 end;
GstFormat format;
gboolean res = TRUE;
endb = wav->datasize;
gst_query_parse_duration (query, &format, NULL);
@ -1849,8 +1875,8 @@ gst_wavparse_pad_query (GstPad * pad, GstQuery * query)
gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, endb,
&format, &end);
}
}
break;
}
default:
format = GST_FORMAT_BYTES;
end = endb;
@ -1898,15 +1924,13 @@ gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event)
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;
}
@ -1974,17 +1998,10 @@ gst_wavparse_change_state (GstElement * element, GstStateChange transition)
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:
@ -1998,8 +2015,6 @@ plugin_init (GstPlugin * plugin)
{
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);
}

7
gst/wavparse/gstwavparse.h
View file

@ -76,13 +76,16 @@ struct _GstWavParse {
/* useful audio data */
guint16 depth;
gint rate;
gint rate;
guint16 channels;
guint16 blockalign;
guint16 width;
guint32 bps;
guint32 av_bps;
guint32 fact;
/* real bps used or 0 when no bitrate is known */
guint32 bps;
guint bytes_per_sample;
/* position in data part */